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Administrator Guide

Version 8.4 | June 2014 | 3725-74600-020

Polycom® RealPresence®
Collaboration Server
800s, Virtual Edition

Copyright© 2014, Polycom, Inc. All rights reserved. No part of this document may be reproduced, translated into another
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Contents

Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1
About the Polycom® RealPresence® Collaboration Server 800s and Polycom® RealPresence®
Collaboration Server, Virtual Edition Administrator’s Guide . . . . . . . . . . . . . . . . . . . . . . . . . . . 1
Who Should Read This Guide? . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2
Prerequisites . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2
How This Guide is Organized . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2
About the Polycom® RealPresence® Collaboration Server 800s and Polycom® RealPresence®
Collaboration Server, Virtual Edition Administrator’s Guide System . . . . . . . . . . . . . . . . . . . . . 3
Network Services Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4
IP Networks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4
Workstation Requirements . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4

Conferencing Modes Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 7
AVC Conferencing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 7
Continuous Presence (CP) Conferencing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 7
Video Protocol Support in CP Conferences . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 9
AVC Basic Conferencing Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 9
Supplemental Conferencing Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 10
SVC-based Conferencing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 11
SVC Conferencing Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 13
MCU Supported Resolutions for SVC Conferencing . . . . . . . . . . . . . . . . . . . . . . . . . 14
Mixed CP and SVC Conferencing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 16
MCU Resource Capacities for Mixed CP and SVC Conferences . . . . . . . . . . . . . . . . . . . 17

Using Conference Profiles . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 19
Conferencing Parameters Defined in a Profile . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 19
Conferencing Capabilities in the Various Conferencing Modes . . . . . . . . . . . . . . . . . . . . 20
Default Profile Settings in CP Conferencing Mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 21
Default Profile Settings in SVC Only Conferencing Mode . . . . . . . . . . . . . . . . . . . . . . . . . 22
Default Profile Settings in a Mixed CP and SVC Conferencing Mode . . . . . . . . . . . . . . . 23
Viewing the List of Conference Profiles . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 25
Profiles Toolbar . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 26
Modifying an Existing Profile . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 26
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Deleting a Conference Profile . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 27
Defining New Profiles . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 27
Exporting and Importing Conference Profiles . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 28
Guidelines for Exporting and Importing Conference Profiles . . . . . . . . . . . . . . . . . . . . . . 28
Exporting Conference Profiles . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 28
Exporting All Conference Profiles from an MCU . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 28
Exporting Selected Conference Profiles . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 29
Importing Conference Profiles . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 30

Defining AVC-Based Conference Profiles . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 32
Defining AVC CP Conferencing Profiles . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 32
Additional Information for Setting CP Profiles . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 63
Gathering Phase . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 63
Gathering Phase Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 64
Gathering Phase Duration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 65
Enabling the Gathering Phase Display . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 66
Overlay Layouts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 66
Guidelines for using the Overlay Layouts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 67
Selecting the Overlay Layouts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 68
Site Names Definition . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 69
Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 69
Shorten the Site Name Display . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 69
Sending Text Messages During a Conference Using Message Overlay . . . . . . . . . . . . . 71
Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 71
Selecting the Chinese Font for Text Display . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 72
Selecting the Chinese Font . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 72

Defining SVC and Mixed CP and SVC Conference Profiles . . . . . . . . . . . . . . . . 74
Defining SVC Conference Profiles . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 74
Defining Mixed CP and SVC Conferencing Profiles . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 85

Video Protocols and Resolution Configuration for CP Conferencing . . . . . . . . 87
Video Resolutions in AVC-based CP Conferencing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 87
Video Display with CIF, SD and HD Video Connections . . . . . . . . . . . . . . . . . . . . . . 87
H.264 High Profile Support in CP Conferences . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 88
Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 88
CP Conferencing with H.263 4CIF . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 88
H.263 4CIF Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 88
The CP Resolution Decision Matrix . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 89
H.264 Base Profile and High Profile Comparison . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 89

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Default Minimum Threshold Line Rates and Resource Usage Summary . . . . . . . . . . . . . 91
Resolution Configuration for CP Conferences . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 92
Modifying the Resolution Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 92
Resolution Configuration - Basic . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 93
Resolution Configuration - Detailed . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 94
Flag Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 96
Setting the Maximum CP Resolution for Conferencing . . . . . . . . . . . . . . . . . . . . . . . 96
Minimum Frame Rate Threshold for SD Resolution . . . . . . . . . . . . . . . . . . . . . . . . . . 96
Additional Video Resolutions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 96
w448p Resolution . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 96
Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 96
Content . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 98
Packet Loss Compensation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 98
Enabling Support of the w448p Resolution . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 98
Collaboration Server System Flag Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 99
Additional Intermediate Video Resolutions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 99

Sharing Content During Conferences . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 100
Content Control Protocols . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 100
Guidelines for Controlling Content Protocol . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 100
Supported Content Control Protocols . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 100
Content Sharing Using H.239 Protocol . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 101
Content Sharing Using BFCP Protocol . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 101
Guidelines for Using SIP BFCP Content . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 101
Content Sharing Using People+Content Protocol . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 102
Guidelines for Content Sharing Using People+Content Protocol . . . . . . . . . . . . . . . 102
Content Media Protocols . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 103
Content Transmission Methods . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 104
Content Video Switching . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 104
Highest Common . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 104
Fixed Rate . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 105
Multiple Content Resolutions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 105
Guidelines for Sharing Contents using Multiple Content Resolutions . . . . . . . . . . . 105
Content Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 106
Customized Content Rate in AVC CP Conferences . . . . . . . . . . . . . . . . . . . . . . . . . . . . 106
MCU Usage Modes of Content Protocols . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 107
H.263 (AVC CP Conferences) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 107
H.263 & H.264 Auto Selection (AVC Conferences) . . . . . . . . . . . . . . . . . . . . . . . . . . . . 107
H.264 Cascade and SVC Optimized . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 107
H.264 HD (AVC CP default) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 108

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H.264 Content Sharing Properties . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 108
Guidelines for Sharing Content Using H.264 HD . . . . . . . . . . . . . . . . . . . . . . . . . . . 108
Content Sharing Related Issues . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 109
Sharing Content in Cascaded Environments . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 109
Sending Content to Legacy Endpoints (CP Only) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 109
Guidelines for Sending Content to Legacy Endpoints . . . . . . . . . . . . . . . . . . . . . . . 109
Content Display on Legacy Endpoints . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 110
Sending Content to Legacy Endpoints in Telepresence Mode . . . . . . . . . . . . . . . . . 110
Exclusive Content Mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 111
Guidelines for Sharing Content in Exclusive Content Mode . . . . . . . . . . . . . . . . . . . 111
Forcing Other Content Capabilities . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 112
Managing Noisy Content Connections . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 112
Useful Procedures in Content Sharing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 113
Defining Content Sharing Parameters for a Conference . . . . . . . . . . . . . . . . . . . . . . . . . 113
H.264 Cascade and SVC Optimized Content Sharing in AVC CP Conferences . . . . . . 115
Selecting a Customized Content Rate in AVC CP Conferences . . . . . . . . . . . . . . . . . . . 117
Sharing Content in Multiple Content Resolutions Mode . . . . . . . . . . . . . . . . . . . . . . . . . 118
Changing the Default Layout for Content Display on Legacy Endpoints . . . . . . . . . . . . 119
Giving and Cancelling Token Ownership (AVC Participants) . . . . . . . . . . . . . . . . . . . . . 120
Stopping a Content Session . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 121
Content Sharing Reference Tables . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 122
Resolutions and Content Rate Reference Tables . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 122
H.263 Content Rate Table . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 122
H.264 Resolution per Content Rate Table . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 122
H.264 Highest Common Content Rates Table . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 123
H.264 Cascade and SVC Optimized (Fixed) Content Rates Table . . . . . . . . . . . . . 124
Legacy Content Endpoint Default Layouts Table . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 125

Implementing Media Encryption for Secured Conferencing . . . . . . . . . . . . . . 127
Media Encryption Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 127
Mixing Encrypted and Non-encrypted Endpoints in one Conference . . . . . . . . . . . . . . . . . . 128
Direct Connection to the Conference . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 129
Connection to the Entry Queue . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 130
Moving from the Entry Queue to Conferences or Between Conferences . . . . . . . . . . . . 130
Recording Link Encryption . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 131
Enabling Media Encryption for a Conference . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 131
Setting the Encryption Flags . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 132
Enabling Encryption in the Profile . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 132
Enabling Encryption at the Participant Level . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 133
Monitoring the Encryption Status . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 134

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Setting Conferences for Telepresence Mode (AVC CP) . . . . . . . . . . . . . . . . . . 136
Collaboration Server Telepresence Mode Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 136
System Level . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 136
Conference Level . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 136
Automatic Detection of Immersive Telepresence (ITP) Sites . . . . . . . . . . . . . . . . . . . . . 137
Horizontal Striping . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 137
Cropping . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 138
Gathering Phase with ITP Room Systems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 138
Aspect ratio for standard endpoints . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 138
Skins and Frames . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 138
RPX and OTX Video Layouts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 138
Room Switch Telepresence Layouts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 141
Telepresence Display Decision Matrix . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 141
Guidelines for Managing the Room Switch Telepresence Layouts by the MCU . . . . . . . 142
Sending Content to Legacy Endpoints in Telepresence Conferences . . . . . . . . . . . . . . . . . 143
Guidelines for Sending Content to Legacy Endpoints in Telepresence Conferences . . . 143
Content Display on Legacy Endpoints in Telepresence Conferences . . . . . . . . . . . 143
Enabling Telepresence Mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 144
Monitoring Telepresence Mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 146
Monitoring Ongoing Conferences . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 146
Monitoring Participant Properties . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 147
Creating Multiple Cascade Links Between Telepresence Conferences . . . . . . . . . . . . . . . . 148
Guidelines for Creating Multiple Cascading Links between Conferences . . . . . . . . . . . . 148
Enabling and Using Multiple Cascade Links . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 149
Creating a Link Participant . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 151
Link Participant in the Dial Out RMX . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 151
Participant Link in the Dial In RMX . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 153
Monitoring Multiple Cascade Links . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 154
Disconnection Causes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 154

Additional Conferencing Information . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 156
Video Preview (AVC Participants Only) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 156
Video Preview Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 156
Workstation Requirements to Display Video Preview . . . . . . . . . . . . . . . . . . . . . . . . . . . 157
Testing your Workstation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 157
Previewing the Participant Video . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 158
Auto Scan and Customized Polling in Video Layout (CP Conferences Only) . . . . . . . . . . . . 159
Guidelines for Using Auto Scan and Customized Polling . . . . . . . . . . . . . . . . . . . . . . . . 160
Enabling the Auto Scan and Customized Polling (CP Only Conferences) . . . . . . . . . . . 160
Enabling the Auto Scan . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 160

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Customized Polling . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 161
Packet Loss Compensation (LPR and DBA) AVC CP Conferences . . . . . . . . . . . . . . . . . . . 162
Packet Loss . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 163
Causes of Packet Loss . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 163
Effects of Packet Loss on Conferences . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 163
Lost Packet Recovery . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 163
Lost Packet Recovery Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 163
Enabling Lost Packet Recovery . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 163
Monitoring Lost Packet Recovery . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 164
Network Quality Indication (AVC Endpoints) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 166
Network Quality Levels . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 166
Indication Threshold Values . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 166
Guidelines for Displaying the Network Quality icons . . . . . . . . . . . . . . . . . . . . . . . . . . . . 167
Customizing Network Quality Icon Display . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 167
Lecture Mode (AVC CP Only) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 169
Enabling Lecture Mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 169
Selecting the Conference Lecturer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 169
Enabling the Automatic Switching . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 171
Lecture Mode Monitoring . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 172
Restricting Content Broadcast to Lecturer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 175
Muting Participants Except the Lecturer (AVC CP Only) . . . . . . . . . . . . . . . . . . . . . . . . 176
Guidelines for Muting all the Participants Except the Lecturer . . . . . . . . . . . . . . . . . 176
Enabling the Mute Participants Except Lecturer Option . . . . . . . . . . . . . . . . . . . . . . 177
Audio Algorithm Support . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 178
Audio Algorithm Support Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 178
SIP Encryption . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 178
Mono . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 178
Stereo . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 179
Monitoring Participant Audio Properties . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 180
Automatic Muting of Noisy Endpoints . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 181
Enabling or Disabling the Automatic Muting of Noisy Endpoints . . . . . . . . . . . . . . . . . . 182
Enabling or Disabling the Automatic Muting of Noisy Endpoints at the MCU Level . 183
Permanent Conference . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 184
Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 184
Enabling a Permanent Conference . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 184

Cascading Conferences . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 185
Video Layout in Cascading conferences (CP and mixed CP and SVC) . . . . . . . . . . . . . 185
Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 186
Flags controlling Cascade Layouts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 187

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Basic Cascading . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 187
Basic Cascading using IP Cascaded Link . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 187
Dialing Directly to a Conference . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 188
Dialing to an Entry Queue . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 188
Automatic Identification of the Cascading Link . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 189

Meeting Rooms . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 190
Meeting Rooms List . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 191
Use Time Out as DTMF Delimiter . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 192
Meeting Room Toolbar & Right-Click Menu . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 193
Creating a New Meeting Room . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 193

Entry Queues, Ad Hoc Conferences and SIP Factories . . . . . . . . . . . . . . . . . . 195
Entry Queues . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 195
Defining a New Entry Queue . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 196
Listing Entry Queues . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 200
Modifying the EQ Properties . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 200
Transit Entry Queue . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 200
Setting a Transit Entry Queue . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 200
IVR Provider Entry Queue (Shared Number Dialing) . . . . . . . . . . . . . . . . . . . . . . . . . . . 201
Call Flow . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 201
Guidelines for Setting the Entry Queue as IVR Provider . . . . . . . . . . . . . . . . . . . . . 201
Configuring the Collaboration Server as IVR Provider . . . . . . . . . . . . . . . . . . . . . . . 202
Configuring the MCU to Support External IVR Services via the MCCF-IVR . . . . . . 202
SIP Factories . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 202
Creating SIP Factories . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 203
SIP Registration & Presence for Entry Queues and SIP Factories with SIP Servers . . . . . . 205
Guidelines for registering Entry Queues and SIP Factories with SIP Servers . . . . . 205
Monitoring Registration Status . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 206
Ad Hoc Conferencing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 206
. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 206

Address Book . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 207
Viewing the Address Book . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 208
Displaying and Hiding the Group Members in the Navigation Pane . . . . . . . . . . . . . . . . 209
Participants List Pane Information . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 209
Displaying and Hiding the Address Book . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 210
Adding Participants from the Address Book to Conferences . . . . . . . . . . . . . . . . . . . . . . . . . 210
Adding Individual Participants from the Address Book to Conferences . . . . . . . . . . . . . 211
Adding a Group from the Address Book to Conferences . . . . . . . . . . . . . . . . . . . . . . . . 211

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Participant Groups . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 211
Managing Groups in the Address Book . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 211
Managing the Address Book . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 213
Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 213
Adding a Participant to the Address Book . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 213
Adding a New participant to the Address Book Directly . . . . . . . . . . . . . . . . . . . . . . . . . 213
Substituting E.164 Number in Dial String . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 218
Adding a Participant from an Ongoing Conference to the Address Book . . . . . . . . . . . . 219
Modifying Participants in the Address Book . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 220
Deleting Participants from the Address Book . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 221
Copying or Moving a Participant . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 222
Searching the Address Book . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 223
Filtering the Address Book . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 224
Filtering Address Book Data Using a Predefined Pattern . . . . . . . . . . . . . . . . . . . . . 224
Filtering Address Book Data Using a Custom Pattern . . . . . . . . . . . . . . . . . . . . . . . 225
Clearing the Filter . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 227
Obtaining the Display Name from the Address Book . . . . . . . . . . . . . . . . . . . . . . . . . . . 228
Guidelines for Obtaining the Display Name from the Address Book . . . . . . . . . . . . 228
Enabling and Disabling the Obtain Display Name from Address Book Feature . . . . 228
Importing and Exporting Address Books . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 229
Exporting an Address Book . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 229
Importing an Address Book . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 229
Integrating the Global Address Book (GAB) of Polycom RealPresence Resource Manager
or Polycom CMA with the Collaboration Server . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 230
Guidelines for integrating with the Global Address Book of Polycom RealPresence
Resource Manager or Polycom CMA . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 230

Scheduling Reservations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 233
Guidelines for Scheduling Reservations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 233
System . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 233
Resources . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 233
Reservations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 234
Using the Reservation Calendar . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 235
Toolbar Buttons . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 235
Reservations Views . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 236
Week View . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 236
Day View . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 236
Today View . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 237
List View . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 237
Changing the Calendar View . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 238
Scheduling Conferences Using the Reservation Calendar . . . . . . . . . . . . . . . . . . . . . . . . . . 240
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Creating a New Reservation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 240
Managing Reservations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 246
Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 246
Viewing and Modifying Reservations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 246
Using the Week and Day views of the Reservations Calendar . . . . . . . . . . . . . . . . 246
Adjusting the Start Times of all Reservations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 248
Deleting Reservations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 250
Searching for Reservations using Quick Search . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 250

Operator Assistance & Participant Move . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 252
Operator Conferences . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 252
Operator Conference Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 253
Defining the Components Enabling Operator Assistance . . . . . . . . . . . . . . . . . . . . . . . . 253
Defining a Conference IVR Service with Operator Assistance Options . . . . . . . . . . 253
Defining an Entry Queue IVR Service with Operator Assistance Options . . . . . . . . 256
Defining a Conference Profile for an Operator Conference . . . . . . . . . . . . . . . . . . . 257
Starting an Ongoing Operator Conference . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 259
Saving an Operator Conference to a Template . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 262
Starting an Operator Conference from a Template . . . . . . . . . . . . . . . . . . . . . . . . . 262
Monitoring Operator Conferences and Participants Requiring Assistance . . . . . . . . . . . 263
Requesting Help . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 263
Participant Alerts List . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 264
Audible Alarms . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 265
Using Audible Alarms . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 265
Moving Participants Between Conferences . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 265
Moving Participants Options . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 266

Conference Templates . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 268
Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 268
Using Conference Templates . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 269
Toolbar Buttons . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 269
Creating a New Conference Template . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 270
Creating a new Conference Template from Scratch . . . . . . . . . . . . . . . . . . . . . . . . . . . . 270
Saving an Ongoing or AVC-based CP Operator Conference as a Template . . . . . . . . . 277
Starting an Ongoing Conference From a Template . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 278
Starting an Operator Conference from a Template (AVC Conferencing) . . . . . . . . . 279
Scheduling a Reservation From a Conference Template . . . . . . . . . . . . . . . . . . . . . . . . . . . 280
Deleting a Conference Template . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 282
Exporting and Importing Conference Templates . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 282
Exporting Conference Templates . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 283

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Exporting All Conference Templates from an MCU . . . . . . . . . . . . . . . . . . . . . . . . . 283
Exporting Selected Conference Templates . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 285
Importing Conference Templates . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 286

Polycom Conferencing for Microsoft Outlook® . . . . . . . . . . . . . . . . . . . . . . . . . 289
Setting up the Calendaring Solution . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 290
Calendaring Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 296
Creating and Connecting to a Conference . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 298
Creating a Conference . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 298
Connecting to a Conference . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 299
Collaboration Server Standalone Deployment . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 300
Collaboration Server and Polycom RealPresence DMA System Deployment . . . . . 300
Polycom Solution Support . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 300

Conference and Participant Monitoring . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 301
General Monitoring . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 301
Conference Level Monitoring . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 301
Viewing the Properties of Ongoing CP and Mixed CP and SVC Conferences . . . . . . . . 302
Viewing the Properties of Ongoing SVC-based Conferences . . . . . . . . . . . . . . . . . . . . . 312
Monitoring of Operator Conferences and Participants Requiring Assistance
(CP and Mixed CP and SVC Conferences) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 318
Requesting Help . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 319
Request to Speak . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 320
Participant Alerts List . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 320
Participant Level Monitoring . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 320
Viewing the Properties of Participants . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 322
Monitoring IP Participants . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 322
Monitoring SIP BFCP Content . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 334
Detecting Endpoint Disconnection . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 335
H323 Endpoint Disconnection Detection . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 335
Monitoring Telepresence Participant Properties . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 337

Recording Conferences . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 338
Creating Multiple Virtual Recording Rooms on the RSS . . . . . . . . . . . . . . . . . . . . . . . . . . . . 338
Configuring the Collaboration Server to Enable Recording . . . . . . . . . . . . . . . . . . . . . . . . . . 339
Defining the Recording Link . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 339
Enabling the Recording Features in a Conference IVR Service . . . . . . . . . . . . . . . . . . . 341
Enabling the Recording in the Conference Profile . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 342
Recording Link Encryption . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 344
Recording Link Encryption Flag Setting . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 344
Recording Link Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 345
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Managing the Recording Process . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 345
Recording Layout . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 345
Using the Collaboration Server Web Client to Manage the Recording Process . . . . . . . 346
Using DTMF Codes to Manage the Recording Process . . . . . . . . . . . . . . . . . . . . . . . . . 348

Users, Connections, and Notes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 349
Collaboration Server Users . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 349
User Types . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 349
Administrator . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 349
Administrator Read-only . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 349
Operator . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 349
Chairperson . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 349
Auditor . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 350
Machine Account . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 350
Listing Users . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 350
Adding a New User . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 350
Deleting a User . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 351
Changing a User’s Password . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 352
Disabling a User . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 352
Enabling a User . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 353
Renaming a User . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 354
Machine Account . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 354
Guidelines for defining a machine account . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 355
Monitoring . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 355
Active Directory . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 356
Connections . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 356
Viewing the Connections List . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 356
Notes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 356
Using Notes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 357

IP Network Services . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 358
IP Network Services . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 358
Management Network (Primary) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 359
Default IP Service (Conferencing Service - Media and signaling) . . . . . . . . . . . . . . . . . 359
Modifying the Management Network in the RealPresence Collaboration Server 800s . . 360
Modifying the Default IP Network Service in the RealPresence Collaboration
Server 800s . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 365
Viewing the Management Network in the RealPresence Collaboration Server
Virtual Edition . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 379
IP Network Monitoring . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 383

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Using IPv6 Networking Addresses for Collaboration Server Internal and
External Entities . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 388
Collaboration Server Internal Addresses . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 388
External Entities . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 388
IPv6 Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 388
Ethernet Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 389
NAT (Network Address Translation) Traversal . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 390
Deployment Architectures . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 390
Remote Connection Using the Internet . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 390
Business to Business Connections . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 391
FW (Firewall) NAT Keep Alive . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 392
System Configuration in SBC environments . . . . . . . . . . . . . . . . . . . . . . . . . . . . 392
SIP Proxy Failover With Polycom® Distributed Media Application™ (DMA™) 7000 . . . . . . 393
Polycom® RealPresence® Collaboration Server 800s and Polycom® RealPresence®
Collaboration Server Virtual Edition Network Port Usage . . . . . . . . . . . . . . . . . . . . . . . . . . . 393
LAN Redundancy . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 395
Configuration Requirements . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 395
Signaling and Media Redundancy . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 395
Hardware Monitor Indications . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 396
Multiple Network Services . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 397
Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 397
Resource Allocation and Capacity . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 398
First Time Installation and Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 398
Connecting the Cables to the RealPresence Collaboration Server 800s . . . . . . . . . . . . 399
Collaboration Server Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 399
System Flags and License Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 399
IP Network Service Definition . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 400
Setting a Network Service as Default . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 404
Signaling Host IP Address and MCU Prefix in GK Indications . . . . . . . . . . . . . . . . . 405
Resolution Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 405
Conference Profile . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 405
Signaling Monitor . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 406
Conferencing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 406
Defining AVC Dial Out Participants . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 406
Monitoring Conferences . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 407
Resource Report . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 407
Port Gauge Indications . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 408

IVR Services . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 409
IVR Services List . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 409
IVR Services Toolbar . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 410
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Adding Languages . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 411
Uploading a Message File to the Collaboration Server . . . . . . . . . . . . . . . . . . . . . . . . . . 412
Defining a New Conference IVR Service . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 414
Defining a New Conference IVR Service . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 414
Change to Chairperson . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 428
Controlling the receipt of in-band and out-of-band DTMF Codes . . . . . . . . . . . . . . . 428
Entry Queue IVR Service . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 428
Defining a New Entry Queue IVR Service . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 429
Setting a Conference IVR Service or Entry Queue IVR Service as the Default Service . 434
Modifying the Conference or Entry Queue IVR Service Properties . . . . . . . . . . . . . . . . . . . . 436
Replacing the Music File . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 436
Adding a Music File . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 437
Creating Audio Prompts and Video Slides . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 437
Recording an Audio Message . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 438
Creating a Welcome Video Slide . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 442
Inviting Participants using DTMF . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 443
Invite Call Flow . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 443
Entering Additional DTMF Codes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 443
Error Handling . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 443
Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 444
Enabling the Invite Participants using DTMF Option . . . . . . . . . . . . . . . . . . . . . . . . . . . . 444
Disabling the Invite Participant Option . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 447
External IVR Service Control . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 448
IVR Services Support with TIP Protocol . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 448
Guidelines for TIP Support with IVR Services . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 448
Default IVR Prompts and Messages . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 449
Volume Control of IVR Messages, Roll Call and Music . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 453
IVR Services in TIP-Enabled Conferences . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 453
IVR Services in TIP-Enabled Conferences Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . 454
Entry Queue and Virtual Entry Queue Access . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 454
Configuring the Conference and Entry Queue IVR Services . . . . . . . . . . . . . . . . . . 454

Call Detail Record (CDR) Utility . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 455
The CDR File Properties . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 455
CDR File Formats . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 455
Multi-Part CDR Files . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 456
Enabling the Multi-Part CDR Option . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 457
CDR File Contents . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 457
Viewing, Retrieving and Archiving Conference Information . . . . . . . . . . . . . . . . . . . . . . . . . 458
Viewing the Conference Records . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 458

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Multi-part CDR File display . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 459
Refreshing the CDR List . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 460
Retrieving and Archiving Conference CDR Records . . . . . . . . . . . . . . . . . . . . . . . . . . . 460

RMX Manager Application . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 461
Installing the RMX Manager Application . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 461
Accessing or downloading the RMX Manager Installer . . . . . . . . . . . . . . . . . . . . . . . . . . 462
Accessing the RMX Manager Application Installer Directly from the MCU . . . . . . . . 462
Downloading the Installation files from Polycom Support Site . . . . . . . . . . . . . . . . . 463
Accessing the RMX Manager Installer from the Login screen . . . . . . . . . . . . . . . . . 464
Installing the RMX Manager on Your Workstation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 464
Installing the RMX Manager for Multi-User Capability . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 466
Starting the RMX Manager Application . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 468
Connecting to the MCU . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 470
RMX Manager Main Screen . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 472
MCUs Pane . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 472
Conferences Pane . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 474
Collaboration Server Management . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 475
List Pane . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 475
Status Bar . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 475
Address Book . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 476
Conference Templates . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 477
Adding MCUs to the MCUs List . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 477
Starting a Conference . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 479
Starting a Conference from the Conferences Pane . . . . . . . . . . . . . . . . . . . . . . . . . . . . 480
Starting a Reservation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 480
Starting an Ongoing Conference or Reservation From a Template . . . . . . . . . . . . . . . . 481
Monitoring Conferences . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 482
Grouping the Participants by MCU . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 483
Start Monitoring/Stop Monitoring . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 484
Modifying the MCU Properties . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 486
Disconnecting an MCU . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 487
Removing an MCU from the MCUs Pane . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 487
Changing the RMX Manager Language . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 488
Import/Export RMX Manager Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 488

Administration and Utilities . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 491
System and Participant Alerts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 491
System Alerts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 491
Participant Alerts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 493

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RMX Time . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 494
Altering the clock . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 494
Resource Management . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 496
Resource Capacity . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 496
MCU Capacities in CP Only Conferencing and SVC Only Conferencing . . . . . . . . . 496
MCU Capacities in Mixed CP and SVC Conferencing . . . . . . . . . . . . . . . . . . . . . . . 497
AVC Conferencing - Voice . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 497
Forcing Video Resource Allocation to CIF Resolution . . . . . . . . . . . . . . . . . . . . . . . 497
Resource Reports . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 498
Displaying the Resource Report . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 499
Resource Capacities in AVC CP, SVC and Mixed Mode Conferences . . . . . . . . . . . . . 501
MCU Resource Management by RealPresence Resource Manager, Polycom CMA
and Polycom RealPresence DMA System . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 501
Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 501
Port Usage Threshold . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 501
Setting the Port Usage Threshold . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 502
SIP Dial-in Busy Notification . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 502
Port Usage Gauge . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 503
System Information . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 504
SNMP (Simple Network Management Protocol) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 505
MIBs (Management Information Base) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 505
Traps . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 506
Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 506
MIB Files . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 506
Private MIBs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 506
Support for MIB-II Sections . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 506
The Alarm-MIB . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 506
H.341-MIB (H.341 – H.323) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 506
Standard MIBs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 507
Unified MIB . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 507
Traps . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 509
Status Trap . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 510
RMX MIB entities that do not generate traps. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 511
Defining the SNMP Parameters in the Collaboration Server . . . . . . . . . . . . . . . . . . . . . 512
Audible Alarms . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 520
Using Audible Alarms . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 520
Audible Alarm Permissions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 521
Stop Repeating Message . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 521
Configuring the Audible Alarms . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 521
User Customization . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 521

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Replacing the Audible Alarm File . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 522
Multilingual Setting . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 523
Customizing the Multilingual Setting . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 523
Banner Display and Customization . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 524
Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 525
Non-Modifiable Banner Text . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 525
Sample 1 Banner . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 525
Sample 2 Banner . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 526
Sample 3 Banner . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 526
Sample 4 Banner . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 526
Customizing Banners . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 527
Banner Display . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 528
Login Screen Banner . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 528
Main Screen Banner . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 528
Software Management . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 529
Backup and Restore Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 529
Using Software Management . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 530
Ping the Collaboration Server . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 531
Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 531
Using Ping . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 531
Notification Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 532
Logger Diagnostic Files . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 534
Information Collector . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 536
Standard Security Mode . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 536
Using the Information Collector . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 537
Step 1: Creating the Information Collector Compressed File . . . . . . . . . . . . . . . . . . 537
Step 2: Saving the Compressed File . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 538
Step 3: Viewing the Compressed File . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 538
Auditor . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 538
Auditor Files . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 539
Retrieving Auditor Files . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 539
Auditor File Viewer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 541
Audit Events . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 544
Alerts and Faults . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 544
Transactions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 545
ActiveX Bypass . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 547
Installing ActiveX . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 547
Resetting the Collaboration Server 800s . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 548
Resetting the RealPresence Collaboration Server Virtual Edition . . . . . . . . . . . . . . . . . . . . . 548
Upgrading and Downgrading . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 551

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Upgrading or Downgrading the Polycom® RealPresence® Collaboration
Server 800s and Polycom® RealPresence® Collaboration Server Virtual Edition . . . . . 572

System Configuration Flags . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 578
Modifying System Flags . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 578
Manually Adding and Deleting System Flags . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 590
Manually Adding Flags to the CS_MODULE_PARAMETERS Tab . . . . . . . . . . . . . 612
Deleting a Flag . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 613
Auto Layout Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 613
Customizing the Default Auto Layout . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 613
LEGACY_EP_CONTENT_DEFAULT_LAYOUT Flag Values . . . . . . . . . . . . . . . . . . . . . 616
CS_ENABLE_EPC Flag . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 616
Automatic Password Generation Flags . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 616
Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 617
Enabling the Automatic Generation of Passwords . . . . . . . . . . . . . . . . . . . . . . . . . . 617
. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 619

Hardware Monitoring . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 620
Viewing the Status of the Hardware Components . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 620
Hardware Monitor Toolbar . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 621
Viewing the Properties of Polycom® RealPresence® Collaboration Server 800s and
Polycom® RealPresence® Collaboration Server Virtual Edition Hardware Components . . . 621
FAN Properties: . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 623
Diagnostics . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 625

Appendix A - Disconnection Causes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 629
IP Disconnection Causes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 629

Appendix B - Active Alarms . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 635
Appendix C - CDR Fields, Unformatted File . . . . . . . . . . . . . . . . . . . . . . . . . . . . 646
The Conference Summary Record . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 647
Event Records . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 648
Standard Event Record Fields . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 648
Event Types . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 649
Event Specific Fields . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 654
Disconnection Cause Values . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 679
MGC Manager Events that are not Supported by the Collaboration Server . . . . . . . . . . . . . 682

Appendix D - Ad Hoc Conferencing and External Database Authentication . 684
Ad Hoc Conferencing without Authentication . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 684

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Ad Hoc Conferencing with Authentication . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 685
Entry Queue Level - Conference Initiation Validation with an External Database
Application . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 685
Conference Access with External Database Authentication . . . . . . . . . . . . . . . . . . . . . . . . . 687
Conference Access Validation - All Participants (Always) . . . . . . . . . . . . . . . . . . . . . . . 688
Conference Access Validation - Chairperson Only (Upon Request) . . . . . . . . . . . . . . . . 689
System Settings for Ad Hoc Conferencing and External Database Authentication . . . . . . . . 690
Ad Hoc Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 690
Authentication Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 690
MCU Configuration to Communicate with an External Database Application . . . . . 691
Enabling External Database Validation for Starting New Ongoing Conferences . . . 692
Enabling External Database Validation for Conferences Access . . . . . . . . . . . . . . . 693

Appendix E - Participant Properties Advanced Channel Information . . . . . . . 695
Appendix G - Configuring Direct Connections to the Collaboration Server . . 697
Management Network (Primary) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 697
Configuring the Workstation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 697
Connecting to the Management Network . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 701
Connecting to the Collaboration Server via Modem . . . . . . . . . . . . . . . . . . . . . . . . . . . . 702
Procedure 1: Install the RMX Manager . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 702
Procedure 2: Configure the Modem . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 702
Procedure 3: Create a Dial-up Connection . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 703
Procedure 4: Connect to the Collaboration Server . . . . . . . . . . . . . . . . . . . . . . . . . . 707

Appendix H - Integration Into Microsoft Environments . . . . . . . . . . . . . . . . . . 709
Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 709
Conferencing Entities Presence . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 710
Multiple Networks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 710
Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 710
Interactive Connectivity Establishment (ICE) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 710
ICE Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 710
Connecting to the Collaboration Server in ICE Environment . . . . . . . . . . . . . . . . . . 711
Dialing Methods . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 711
Integrating the Collaboration Server into the Microsoft Office Communications
Server Environment . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 713
Setting the Matched URI Dialing Method . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 713
Configuring the Office Communications Server for Collaboration Server Systems . 714
Setting the Trusted Host for Collaboration Server in the Office Communications
Server . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 714
Setting the Static Route for Collaboration Server in the OCS . . . . . . . . . . . . . . . . . 716

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Setting the Static Route & Trusted Host for Collaboration Server in the Load
Balancer Server (Optional) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 717
Configuring the Collaboration Server System . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 719
Dialing to an Entry Queue, Meeting Room or Conference Using the Matched URI
Method . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 719
Setting the Numerical Dialing Method . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 719
Setting the Numerical Dialing for Collaboration Server Meeting Rooms . . . . . . . . . 719
Optional. Removing the Collaboration Server from the Host Authorization List . . . . 720
Configuring the Collaboration Server as a Routable Gateway . . . . . . . . . . . . . . . . . 721
Establishing a Voice Route to the Collaboration Server “Voice” Gateway . . . . . . . . 722
Configuring Office Communicator Users for Enterprise Voice . . . . . . . . . . . . . . . . . 725
Starting a Conferencing Call from the MOC . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 729
Setting Simultaneous Numerical Dialing and Matched URI Routing . . . . . . . . . . . . . . . . 729
PFX Method - Creating the Security (TLS) Certificate in the OCS and Exporting the
Certificate to the Collaboration Server Workstation . . . . . . . . . . . . . . . . . . . . . . . . . . . . 730
Retrieving the Certificate from the OCS to be sent to the Collaboration Server
Workstation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 735
Optional. Creating the Certificate Password File (certPassword.txt) . . . . . . . . . . . . 737
Supporting Remote and Federated Users in Office Communications Server ICE
Environment . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 738
Creating an Active Directory Account for the Collaboration Server . . . . . . . . . . . . . 738
Enabling the Collaboration Server User Account for Office Communication Server

740

Configure the Collaboration Server for ICE dialing . . . . . . . . . . . . . . . . . . . . . . . . . . 741
Registering the Collaboration Server as a Trusted Application for Lync 2010/2013 . . . . . . . 742
Configure the Collaboration Server FQDN in the DNS . . . . . . . . . . . . . . . . . . . . . . . 742
Configure Collaboration Server Static Route and Trusted Application . . . . . . . . . . . 744
Configure the Collaboration Server for Lync 2010/2013 . . . . . . . . . . . . . . . . . . . . . . 747
Import and install the Certificate on the Collaboration Server . . . . . . . . . . . . . . . . . 752
Collaboration Server System Flag Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 755
Enabling the Microsoft Environment . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 755
Microsoft RTV Video Protocol Support in CP Conferences . . . . . . . . . . . . . . . . . . . . . . . . . . 757
Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 757
Participant Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 758
Monitoring RTV . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 759
Controlling Resource Allocations for Lync Clients Using RTV Video Protocol . . . . . . . . 759
Threshold HD Flag Settings using the RTV Video Protocol . . . . . . . . . . . . . . . . . . . 760
Sharing Content via the Polycom CSS Plug-in for Lync Clients . . . . . . . . . . . . . . . . . . . . . . 760
Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 761
Configuring the MCU for Content Sharing via the Polycom CSS Plug-in . . . . . . . . . . . . 761
Setting the System Flag . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 761
Conference Profile Settings . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 762

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Monitoring the Participant connection . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 762
Adding Presence to Conferencing Entities in the Buddy List . . . . . . . . . . . . . . . . . . . . . . . . 764
Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 764
Enabling the Registration of the Conferencing Entities . . . . . . . . . . . . . . . . . . . . . . . . . . 765
Creating an Active Directory Account for the Conferencing Entity . . . . . . . . . . . . . . 765
Enabling the Conferencing Entity User Account for Office Communication Server
or Lync Server . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 767
Defining the Microsoft SIP Server in the IP Network Service . . . . . . . . . . . . . . . . . . 768
Enabling Registration in the Conference Profile . . . . . . . . . . . . . . . . . . . . . . . . . . . . 768
Verifying the Collaboration Server Conferencing Entity Routing Name and Profile . 769
Monitoring the Registration Status of a Conferencing Entity in the Collaboration
Server Web Client or RMX Manager Application . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 770
Conferencing Entity List . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 770
Conferencing Entity Properties . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 771
Collaboration Server Configuration for CAC Implementation . . . . . . . . . . . . . . . . . . . . . . . . 773
Conferencing Behavior in CP Conferences . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 774
Monitoring Participant Connections . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 774
Connecting a Collaboration Server Meeting Room to a Microsoft AV-MCU Conference . . . 775
Configuring the Collaboration Server for Federated (ICE) Dialing . . . . . . . . . . . . . . . . . . . . 776
Monitoring the Connection to the STUN and Relay Servers in the ICE Environment . . . 777
Monitoring the Participant Connection in ICE Environment . . . . . . . . . . . . . . . . . . . . . . 778
Active Alarms and Troubleshooting . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 780
Active Alarms . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 780
ICE Active Alarms . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 781
Troubleshooting . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 783
Known Issues . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 783
Polycom Solution Support . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 783
Lync 2013 SVC Connectivity to Polycom MCU . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 785
Deployment Architectures . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 785
Backward compatibility to Lync 2010 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 786
Video Resource Requirements and Implications . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 786
Support for HD1080p Resolution . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 786
Limit Maximum Resolution for MS SVC Using a System flag . . . . . . . . . . . . . . . . . . 787
ICE Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 787
Federation Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 787
System Flags for Cropping Control . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 788
Sharing Content during a Conference . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 788
Cisco TIP Support . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 790
Lync 2013 Participant monitoring . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 790
Monitoring Participant Properties - Channel Status Tab . . . . . . . . . . . . . . . . . . . . . . 791

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Monitoring Participant Properties - Channel Status - Advanced Tab . . . . . . . . . . . . 792
Deployment Architecture 1 - Collaboration Server Hosted . . . . . . . . . . . . . . . . . . . . . . . . . . 793
Look and Feel . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 793
Deployment Architecture 2 - MS AV MCU Cascade . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 794
Look and Feel for Lync clients and Group Series Endpoints . . . . . . . . . . . . . . . . . . . . . 795
Look and Feel for Legacy Endpoints . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 796
Video Resource Requirement Selection in MS AV MCU Cascade . . . . . . . . . . . . . . . . . 796
Video Forcing and Changing Layout in MS AV MCU Cascade . . . . . . . . . . . . . . . . 797
Handle Low Bit Rate Calls From the AV MCU . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 797
Remove Empty Cells From the Video Layout . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 797
Configuring the Collaboration Server as a Trusted Application for Lync 2013 . . . . . . . . . . . 798
Registering the Collaboration Server as a Trusted Application for Lync 2010/2013 . . . . . . . 798
Configure the Collaboration Server FQDN in the DNS . . . . . . . . . . . . . . . . . . . . . . . 798
Configure Collaboration Server Static Route and Trusted Application . . . . . . . . . . . 801
Configure the Collaboration Server for Lync 2010/2013 . . . . . . . . . . . . . . . . . . . . . . 804
Import and install the Certificate on the Collaboration Server . . . . . . . . . . . . . . . . . 809
. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 811

Appendix I - Polycom Open Collaboration Network (POCN) . . . . . . . . . . . . . . 812
Collaboration With Cisco’s Telepresence Interoperability Protocol (TIP) . . . . . . . . . . . . . . . 812
Deployment Architectures . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 813
Single Company Model - Polycom and Cisco Infrastructure . . . . . . . . . . . . . . . . . . . . . . 813
Call Flows . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 817
Multipoint call with DMA . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 817
Multipoint call without DMA . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 818
Company to Company Models Using a Service Provider . . . . . . . . . . . . . . . . . . . . . . . . 819
Model 1 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 819
Call Flow . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 821
Multipoint call via Service Provider - Model 1 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 821
Multipoint call via Service Provider - Model 2 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 822
Call Flow . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 824
Multipoint call via Service Provider - Model 2 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 824
Administration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 825
Gatekeepers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 825
Standalone Polycom CMA/DMA System as a Gatekeeper . . . . . . . . . . . . . . . . . . . 825
Standalone Cisco IOS Gatekeeper . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 825
Neighbored Cisco IOS and Polycom CMA/DMA Gatekeepers . . . . . . . . . . . . . . . . . 825
DMA . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 825
CUCM . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 825
Configuring the Cisco and Polycom Equipment . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 826

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Cisco Equipment . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 826
CUCM . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 826
IOS Gatekeeper . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 826
IOS and CMA Gatekeepers (Neighbored) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 826
Polycom Equipment . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 826
Configuring the Collaboration Server . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 827
Configuring Entry Queues and IVR Services . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 828
Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 829
Content . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 829
Procedure 1: Set the MIN_TIP_COMPATIBILITY_LINE_RATE System Flag . . . . . 830
Procedure 2: Configuring Collaboration Server to statically route outbound SIP
calls to DMA or CUCM . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 830
Procedure 3: Configuring the Collaboration Server’s H.323 Network Service
to register with CMA gatekeeper . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 831
Procedure 4: Configuring a TIP Enabled Profile on the Collaboration Server . . . . . 832
Content Sharing Behavior . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 837
Procedure 5: Configuring an Ad Hoc Entry Queue on the Collaboration Server
if DMA is not used . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 838
Procedure 6: Configuring a Meeting Room on the Collaboration Server . . . . . . . . . 839
Procedure 7: Configuring Participant Properties for dial out calls . . . . . . . . . . . . . . 839
Collaboration with Microsoft and Cisco . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 840
Deployment Architecture . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 841
Call Flow . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 844
Multipoint Calls using DMA . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 844
Administration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 845
DMA . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 845
Microsoft Lync Server . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 845
CUCM . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 845
Solution Interoperability Table . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 845
TIP Layout Support & Resource Usage . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 847
Supported TIP Resolutions and Resource Allocation . . . . . . . . . . . . . . . . . . . . . . . . . . . 847
Supported Resolutions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 847
Resource Allocation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 847
Configuring the Microsoft, Cisco and Polycom Components . . . . . . . . . . . . . . . . . . . . . . . . 848
Content Sharing Behavior . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 853
Encryption . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 854
Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 855
Resolution Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 858
Endpoints . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 859
Content . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 859
Operations During Ongoing Conferences . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 860
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Monitoring . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 860
CTS Participants . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 860
Lync Participants (RTV) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 862
Known Limitations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 863

Restoring Defaults and System Recovery . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 865
Restore to Factory Security Defaults . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 865
Comprehensive Restore to Factory Defaults . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 868
Comprehensive Restore to Factory Defaults Procedure . . . . . . . . . . . . . . . . . . . . . . . . . 869
Procedure A: Backup Configuration Files . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 869
Procedure B: Restore to Factory Defaults . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 870
Procedure C: Restore the System Configuration From the Backup . . . . . . . . . . . . . 873
System Recovery Using the Recovery DVD . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 873
Preparation for System Recovery . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 873
Performing the Recovery Process . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 874
Completing the System Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 875

Appendix K - SIP RFC Support . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 876

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Overview
About the Polycom® RealPresence® Collaboration
Server 800s and Polycom® RealPresence®
Collaboration Server, Virtual Edition Administrator’s
Guide
The Polycom® RealPresence® Collaboration Server 800s and Polycom® RealPresence® Collaboration
Server, Virtual Edition Administrator’s Guide provides instructions for configuring, deploying, and
administering Polycom Multipoint Control Units (MCUs) for video conferencing. This guide will help you
understand the Polycom video conferencing components, and provides descriptions of all available
conferencing features.
This guide will help you perform the following tasks:
● Customize the Collaboration Server conferencing entities such as conference Profiles, IVR Services,
Meeting Rooms, Entry Queues, etc., to your organization’s needs (optional). In the CloudAxis
solution environment, these entities should be defined in the Polycom® RealPresence® Distributed
Media Application™ (DMA®).
● Define Collaboration Server Users.
● Advanced conference Management
● Define video protocols and resolution configuration for CP conferencing
● Optional. Configure Templates, the Address Book and schedule Reservations. In the CloudAxis
solution environment, these entities should be defined in the RealPresence DMA system.
● Record Conferences
● Configure the Collaboration Server to support special call flows and conferencing requirements, such
as Cascading Conferences.
● Configure the Collaboration Server to support Polycom third party and partner environments such as
Microsoft, IBM, Cisco, Avaya, Broadsoft and Siemens.
● Configure the Collaboration Server for special applications and needs by setting various system
flags.
● Manage and troubleshoot the Collaboration Server’s performance.
The Polycom® RealPresence Collaboration Server 800s / Virtual Edition Getting Started Guide provides
description of basic conferencing operations. It will help you perform the following tasks:
● Unpack the Collaboration Server system and install it on a rack (Collaboration Server 800s only).
● Connect the required cables to the Collaboration Server (Collaboration Server 800s only).
● Perform basic configuration procedures.
● Start a new conference and connect participants/endpoints to it.
● Monitor ongoing conferences
● Perform basic operations and monitoring tasks
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Overview

Who Should Read This Guide?
System administrators and network engineers should read this guide to learn how to properly set up
Polycom Collaboration Server systems. This guide describes administration-level tasks.
For detailed description of first time installation and configuration, description of the Collaboration Server
(RMX) Web Client, and basic operation of your Collaboration Server system, see the Polycom®
RealPresence Collaboration Server 800s / Virtual Edition Getting Started Guide.

Prerequisites
This guide assumes the user has the following knowledge:
● Familiarity with Windows® XP or Windows 7 operating systems and interface.
● Familiarity with Microsoft® Internet Explorer® Version 7, 8 or 9.
● Basic knowledge of video conferencing concepts and terminology.

How This Guide is Organized
The following typographic conventions are used in this guide to distinguish types of in-text information.
Typographic Conventions
Convention

Description

Bold

Highlights interface items such as menus, soft keys, flag names, and directories. Also used
to represent menu selections and text entry to the phone.

Italics

Used to emphasize text, to show example values or inputs, file names and to show titles of
reference documents available from the Polycom Support Web site and other reference
sites.

Underlined Blue

Used for URL links to external Web pages or documents. If you click on text in this style,
you will be linked to an external document or Web page.

Blue Text

Used for cross referenced page numbers in the same or other chapters or documents. If
you click on blue text, you will be taken to the referenced section.
Also used for cross references. If you click the italic cross reference text, you will be taken
to the referenced section.



Indicates a variable for which you must enter information specific to your installation,
endpoint, or network. For example, when you see , enter the IP address of
the described device.

>

Indicates that you need to select an item from a menu. For example, Administration >
System Information indicates that you need to select System Information from the
Administration menu.

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Overview

About the Polycom® RealPresence® Collaboration
Server 800s and Polycom® RealPresence®
Collaboration Server, Virtual Edition Administrator’s
Guide System
The Polycom® RealPresence® Collaboration Server 800s and Polycom® RealPresence® Collaboration
Server Virtual Edition are high performance, scalable, IP-network (H.323 and SIP) MCUs that provide
feature-rich and easy-to-use multipoint voice and video conferencing.
The MCU can be used as a standalone device to run voice and video conferences or it can be used as part
of a solution provided by Polycom. This solution may include the following components:
● Polycom® RSS™ 4000 - provides one-touch recording and secure playback on video conferencing
systems, tablets and smartphones, or from your Web browser.
● Polycom® Distributed Media Application™ (DMA™) system - provides call control and MCU
virtualization with carrier-grade redundancy, resiliency and scalability.
● Polycom® RealPresence® Resource Manager - centrally manages, monitors and delivers Cloud
based Video as a Service (VaaS) and enterprise video collaboration.
● Polycom® RealPresence® Access Director™ - removes communication barriers and enables
internal and external teams to collaborate more easily and effectively over video.
The following diagram describes the multipoint video conferencing configuration with the Collaboration
Server as a standalone system.
Multipoint Video Conferencing using a RealPresence Collaboration Server 800s/Virtual Edition

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Overview

The RealPresence Collaboration Server 800s / Virtual Edition provide multipoint voice and video
conferencing.
The RealPresence Collaboration Server 800s / Virtual Edition unit can be controlled via the LAN, by the
Collaboration Server Web Client application, using Internet Explorer installed on the user’s workstation or
the RMX Manager application. The RMX Manager can control several MCU units..
In the RealPresence Collaboration Server 800s unit, MCU management and IP conferencing are performed
via two different LAN ports. The networks can be separated in Maximum Security Environments.

Network Services Guidelines
IP Networks
In the RealPresence Collaboration Server 800s, system management and IP conferencing are performed
via a single LAN port.
Management uses LAN1 and IP network Services use LAN2. When enabling multiple services,
management and the IP network service (1) share LAN1, the second IP network service (2) uses LAN2.

Workstation Requirements
The RMX Web Client and RMX Manager applications can be installed in an environment that meets the
following requirements:
● Minimum Hardware – Intel® Pentium® III, 1 GHz or higher, 1024 MB RAM, 500 MB free disk space.
● Workstation Operating System – Microsoft® Windows® XP, Windows® 7, and Windows® 8.
● Network Card – 10/100/1000 Mbps.
● Web Browser - Microsoft® Internet Explorer® Version 7, 8, 9, and 10.
● Collaboration Server Web client and RMX Manager are optimized for display at a resolution of 1280
x 800 pixels and a magnification of 100%
The following table lists the environments (Web Browsers and Operating Systems) with which the
Collaboration Server Web Client and RMX Manager applications are supported.
Collaboration Server Wen Client/RMX Manager Environment Interoperability Table
Web Browser

Operating System

Internet Explorer 7

Windows Vista™
Windows 7

Internet Explorer 8

Windows 7

Internet Explorer 9

Windows 7 and Windows 8

Internet Explorer 10*

Windows 8

.

.Net Framework 2.0 is required and installed automatically.
If ActiveX installation is blocked, see the Polycom® RealPresence Collaboration Server 800s /
Virtual Edition Administrator’s Guide.

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Overview

.Net Framework 2.0 SP1 or above is required and installed automatically. Internet Explorer must be
enabled to allow running Signed ActiveX.
If ActiveX installation is blocked, see the ActiveX Bypass.

Collaboration Server Web Client does not support larger Windows text or font sizes. It is
recommended to set the text size to 100% (default) or Normal in the Display settings in Windows
Control Panel on all workstations. Otherwise, some dialog boxes might not appear properly aligned.
To change the text size, select Control Panel>Display. For Windows XP, click the Appearance tab,
select Normal for the Font size and click OK. For Windows 7, click the Smaller - 100% option and
click OK.

When installing the Collaboration Server Web Client, Windows Explorer >Internet Options>
Security Settings must be set to Medium or less.

It is not recommended to run RMX Web Client and Polycom CMAD applications simultaneously on
the same workstation.

If you have problems getting the Collaboration Server Web Client to work with Windows 8, it is
recommended to run Internet Explorer as an administrator by holding the shift key and right-clicking
on the IE icon, and then select Run as Administrator.

.Net Framework 2.0 SP1 or above is required and installed automatically. Internet Explorer must be
enabled to allow running Signed ActiveX.
If ActiveX installation is blocked, see the ActiveX Bypass.

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Overview

For Windows 7™ Security Settings, see the Polycom® RealPresence Collaboration Server 800s / Virtual
Edition Getting Started Guide, Windows 7™ Security Settings.
For Internet Explorer 8 configuration, see the Polycom® RealPresence Collaboration Server 800s / Virtual
Edition Getting Started Guide, Windows 7™ Security Settings.

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Conferencing Modes Overview
The MCU system offers the following types of conferences (Conferencing Modes), based on the video
protocol and the video display during the conference:
● AVC-based Conferencing - CP Only (Video Transcoding)
● SVC-based Conferencing (Media Relay) - SVC Only
● Mixed AVC and SVC Conferencing - CP and SVC

AVC Conferencing
AVC-based Conferences allow endpoints that support AVC video to connect to these conferences. AVC
(Advanced Video Coding) video refers to the H.264 video protocols used to send and receive video. On the
Collaboration Server system it also includes all the standard video protocols such as H.261, H.263, and
RTV.
All endpoints (including SVC-enabled endpoints) have AVC capabilities and can connect to AVC
conferences running on the MCU. AVC-based endpoints can connect using different signaling protocols and
different video protocols.
Based on the video processing required during the conference, the Collaboration Server offers the
Continuous Presence Conferencing Mode for AVC-based conferencing.
The Conferencing Mode determines the video display options (full screen or split screen with all participants
viewed simultaneously) and the method in which the video is processed by the MCU (with or without using
the MCU’s video resources).

Continuous Presence (CP) Conferencing
The dynamic Continuous Presence (CP) capability of the Collaboration Server system enables viewing
flexibility by offering multiple viewing options and window layouts for video conferencing. It enables several
participants to be viewed simultaneously and each connected endpoint uses its highest video, audio and
data capabilities up to the maximum line rate set for the conference.
AVC-based endpoints can connect to the conference using any:
● Signaling protocol: H.323, SIP, (and RTV line rate, up to a maximum line rate defined for the
conference
● Video Protocol: H.261, H.263, H.264 Base Profile and H.264 High Profile) and at any resolution and
frame rate, provided they meet the minimum requirements set for the conference:
 Video Resolutions: from QCIF, CIF and up to 1080p30
 Frame rates up to 30fps
In Continuous Presence conferences, the MCU receives the video stream from each endpoint at the video
rate, video resolution and frame rate that it is capable of sending, and it superimposes all the received
streams into one video stream that includes the input from the other endpoints arranges in the selected
video layout.

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Conferencing Modes Overview

Participants do not see themselves in the video layout. By Default, the speaker is shown in the top left layout
cell in symmetric layouts, in the larger cell in asymmetric layouts, or in full screen. The speaker sees the
previous speakers (their number depends on the number of cells on the speaker’s layout.
The Continuous Presence video session offers layouts to accommodate different numbers of participants
and conference settings including support of the VUI annex to the H.264 protocol for endpoints that transmit
wide video instead of 4CIF resolution. Each participant can select his/her layout for viewing during the
conference, as can be seen in AVC Continuous Presence (CP) video streams and built layouts.
For conferences with more participants than display squares, the Collaboration Server dynamic video mix
capability allows the viewed sites to be modified throughout the conference. The displayed layout can be
changed during an ongoing conference, allowing a participant to view different screen layouts of the other
conference participants. These layout options allow conferences to have greater flexibility when displaying
a large number of participants and maximizes the screen’s effectiveness.
AVC Continuous Presence (CP) video streams and built layouts

Video quality in Continuous Presence conferences is affected by the conference line rate (that determines
the maximum line rate to be used by the connecting endpoints), and the video capabilities of the endpoints
such as the video protocol, video resolution and frame rate. Content sharing is available in all CP
conferences.
This requires extensive processing of the video sent to each participant in the conference. The higher the
video rate and resolution, the more processing power is required.
By default every conference, Entry Queue and Meeting Room has the ability to declare the maximum CP
resolution as defined for the system. This includes conferences launched by the Collaboration Server Web
Client and conferences started via the API.

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Conferencing Modes Overview

CP conferencing is defined in the Conference profile by setting the following main
features:
● Setting the Conferencing Mode to CP only
● Conference Line Rate
● Video Layout

Video Protocol Support in CP Conferences
The video protocol selected by the system determines the video compression standard used by the
endpoints. In Continuous Presence conferences, the system selects the best video protocol for each of the
endpoint according to he endpoint’s capabilities.
The following Video protocols are supported in CP conferences:
● H.261 - the legacy video compression algorithm mandatory to all endpoints. It is used by endpoints
that do not support other protocols.
● H.263 - a video compression algorithm that provides a better video quality than H.261. This standard
is not supported by all endpoints.
● H.264 Base Profile - a video compression standard that offers improved video quality, especially at
line rates lower than 384 Kbps.
H.264 High Profile allows higher quality video to be transmitted at lower line rates.
● RTV - a video protocol that provides high quality video conferencing capability to Microsoft OCS
(Office Communicator Server) endpoints at resolutions up to HD720p30. (SIP only).

AVC Basic Conferencing Parameters
The main parameters that define the quality of an AVC-based video conference and its display are:
● Line (Bit) Rate - The transfer rate of video and audio streams. The higher the line (bit) rate, the better
the video quality. The MCU supports the following line rates:
 CP Conferences: 64kbps to 4096kbps
● Audio Algorithm - The audio compression algorithm determines the quality of the conference audio.
● Video protocol, video format, frame rate, annexes, and interlaced video mode - These
parameters define the quality of the video images. The Collaboration Server will send video at the
best possible resolution supported by endpoints regardless of the resolution received from the
endpoints.
 When Sharpness is selected as the Video Quality setting in the Conference Profile, the
Collaboration Server will send 4CIF (H.263) at 15fps instead of CIF (H.264) at 30fps.
 H.264 High Profile protocol provides better compression of video images in line rates lower than
384 Kbps and it will be automatically selected for the endpoint if it supports H.264 High Profile. If
the endpoint does not support H.264 High Profile, the Collaboration Server will try H.264 Base
Profile which provides good compression of video images in line rates lower than 384 Kbps (better
than H.263 and not as good as H.264 High Profile).
 When working with Collaboration Servers at low bit rates (128, 256, or 384Kbps), HDX endpoints
will transmit SD15 resolution instead of 2CIF resolution.
When using a full screen (1x1) conference layout, the Collaboration Server transmits the same
resolution it receives from the endpoint.
● Video resolution:

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Conferencing Modes Overview

 H.261 CIF/QCIF – Supported in Continuous Presence (CP) conferences at resolutions of 288 x
352 pixels (CIF) and 144 x 176 pixels (QCIF). Both resolutions are supported at frame rates of up
to 30 frames per second.
 H.263 4CIF - A high video resolution available to H.263 endpoints that do not support H.264. It is
only supported for conferences in which the video quality is set to sharpness and for lines rates
of 384kbps to 1920kbps.
 Standard Definition (SD) - A high quality video protocol which uses the H.264 and H.264 High
Profile video algorithms. It enables compliant endpoints to connect to Continuous Presence
conferences at resolutions of 720 x 576 pixels for PAL systems and 720 x 480 pixels for NTSC
systems.
 High Definition (HD) – HD is an ultra-high quality video resolution that uses the H.264 and H.264
High Profile video algorithms. Depending on the Collaboration Server’s type, compliant endpoints
are able to connect to conferences at the following resolutions:
 720p (1280 x 720 pixels) - all Collaboration Server types
 1080p (1920 x 1080 pixels) in Collaboration Server
● Lost Packet Recovery (LPR) - LPR creates additional packets that contain recovery information
used to reconstruct packets that are lost during transmission.

Supplemental Conferencing Features
In addition to basic parameters that determine the quality of the video, additional features can be enabled,
adding capabilities to the conference, or enabling special conferencing modes:
● Content Sharing (H.239) – Allows compliant endpoints to transmit and receive two simultaneous
streams of conference data to enable Content sharing. H.239 is also supported in cascading
conferences. Both H.263 and H.264 Content sharing protocols are supported. If all endpoints
connected to the conference have H.264 capability, Content is shared using H.264, otherwise
Content is shared using H.263.
For more information, see Sharing Content During Conferences.
● Encryption – Used to enhance media security at conference and participant levels. For more
information, see Implementing Media Encryption for Secured Conferencing..
● Conference Recording - The Collaboration Server enables audio and video recording of
conferences using Polycom RSS recording system.
● Lecture Mode (CP Conferences only) – The lecturer is seen by all participants in full screen while
the lecturer views all conference participants in the selected video layout.
For more information, see Lecture Mode (AVC CP Only).
● Presentation Mode (CP Conferences only) – When the current speaker’s speech exceeds a
predefined time (30 seconds), the conference layout automatically changes to full screen, displaying
the current speaker as the conference lecturer on all the participants’ endpoints. During this time the
speaker’s endpoint displays the previous conference layout. When another participant starts talking,
the Presentation Mode is cancelled and the conference returns to its predefined video layout.
Presentation mode is available with Auto Layout and Same Layout.
 If the speaker in a video conference is an Audio Only participant, the Presentation Mode is
disabled for that participant.
 Video forcing works in the same way as in Lecture Mode when Presentation Mode is activated,
that is, forcing is only enabled at the conference level, and it only applies to the video layout
viewed by the lecturer.

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Conferencing Modes Overview

● Telepresence Mode (CP Conferences only) - enables the connection of numerous high definition
telepresence rooms and of different models (such as TPX and RPX) into one conference maintaining
the telepresence experience. This mode is enabled by a special license.
● TIP Support (CP Conferences only) - TIP is a proprietary protocol created by Cisco for deployment
in Cisco TelePresence systems (CTS). Polycom’s solution is to allow the Collaboration Server to
natively inter-operate with Cisco TelePresence Systems, ensuring optimum quality multi-screen,
multipoint calls.

SVC-based Conferencing
The SVC-Based conferencing mode provides video without transcoding by the MCU, hence requiring less
video resources while providing better error resiliency and lower latency.
Using the SVC video protocol, SVC conferences provide video bit streams at different resolutions, frame
rates and line rates to SVC-enabled endpoints with various display capabilities and layout configurations.
In the SVC-based conference, each SVC-enabled endpoint transmits multiple bit streams, called
simulcasting, to the Collaboration Server. Simulcasting enables each endpoint to transmit at different
resolutions and frame rates such as 720p at 30fps, 15fps, and 7.5fps, 360p at 15fps and 7.5fps, and 180p
at 7.5fps.
The Polycom SVC-enabled endpoints (such as the Polycom® RealPresence® Desktop for Windows® and
Polycom® RealPresence® Mobile) compose the layout according to their layout settings and video
capabilities. This enables the MCU to send or relay the selected video streams to each endpoint without
processing the video streams and sending the composite video layout to the endpoints.

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Conferencing Modes Overview

SVC video streams and Layouts
-

The video streams displayed in the conference layout on each endpoint is obtained from the different
streams received from each of the endpoints displayed in the layout. Depending on the size of the video cell
in the configured layout, the endpoint requests the video stream in the required resolution from the
Collaboration Server. The higher the display quality and size, the higher the requested resolution will be sent
to the endpoint. The endpoint creates the displayed layout from the different video streams it receives.
For instance, an SVC endpoint might want to receive three video streams at different frame rates and
resolutions, and create a conference layout with the received video streams. Each SVC-enabled endpoint
sends encoded SVC bit streams to the MCU to relay to the other SVC-enabled endpoints in the conference.
The endpoints encode the video in multiple resolutions and decodes the multiple video input streams.
For example:
RealPresence mobile client (2) will transmit two resolutions; one that is suited for RealPresence Desktop
client (3) and a second that is suited for two other endpoints: RealPresence Desktop client (4) and (1).
RealPresence Desktop client (1) transmits two resolutions; one that is suited for RealPresence Mobile client
(2) and a second that is suited for RealPresence Desktop client (4).
The MCU determines which of the incoming resolutions to send to each endpoint. It does not perform any
SVC encoding and decoding, or any transcoding of the video streams. The Collaboration Server functions
as the multipoint media relay to the endpoints. For voice activated selection of the video streams, the
Collaboration Server determines which of the incoming bit streams to send to each endpoint.

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Advantages of SVC Conferencing
SVC increases the scalability of video networks and enables mass desktop video deployments. Some of
the advantages of SVC conferencing are:
● Offers high-resolution video conferencing with low end-to-end latency, improved error resiliency and
higher system capacities.
● Allows the SVC-enabled video endpoints to manage display layouts, supporting multiple line rates,
resolutions and frame rates.
● The Collaboration Server functions as a media relay server providing low cost production benefits.
The Collaboration Server reduces bandwidth usage by only selecting the necessary video stream to
be sent to the endpoints.

SVC Conferencing Guidelines
You can run SVC-based conferences when following the guidelines listed below.
● SVC conferences are supported only with the following:
 SVC Licensing
 SIP over UDP signaling
 SIP over TLS Signaling
 Polycom SVC-enabled endpoints (Polycom® RealPresence® Desktop, Polycom®
RealPresence® Mobile)
 Ad Hoc conferencing via Meeting Rooms and ongoing conferences
● SVC Only conferences can run on the same MCU as AVC Only conferences.
● End-to-end latency on a local network (same site), is around 200mSec to ensure AV sync (also known
as Lip-sync).
● Dial-out is not available in SVC Only conference.
● Dial-in is available as follows:
 AVC endpoints (participants) can only connect to an AVC conference or Mixed CP and SVC
conference. When dialing into SVC Only conferences they will be disconnected and the calls fail.
 SVC endpoints support both AVC and SVC video protocols.
When dialing into SVC Only conferences, they connect as SVC endpoints.
When dialing into AVC Only conferences, they connect as AVC endpoints. They cannot connect
to an AVC conference using the SVC capabilities.
● SVC endpoints can connect to conferences via Entry Queues, however:
 The Entry Queue and Conference Modes must match - both SVC Only or both Mixed.
 Both the Entry Queue and the Conference must have the same line rate.
● SVC endpoints cannot be moved between conferences.
● Content is supported in H.264 (AVC).
 Only the H.264 Cascade and SVC Optimized option is supported.
 LPR and DBA are not supported for SVC content sharing.
● In SVC Only conferences and Mixed CP and SVC conferences, Auto Layout is the default and the
layout display for SVC endpoints is controlled from the endpoint application.
● Site names display on SVC endpoints is controlled from the SVC endpoints.
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● When a RealPresence DMA system is part of the solution, the DMA is used as the SIP proxy and the
SVC endpoint subscribes to the RealPresence DMA system for call control. If a RealPresence DMA
system is not part of the solution, the SVC endpoint dial directly to the Collaboration Server using IP
addresses is the SIP dialing strings.
● When Hot backup is enabled, all the conferences are created on the Slave MCU.
● When Hot Backup is activated and the Slave MCU becomes the Master MCU:
 All AVC endpoints will be reconnected to the AVC conferences. SVC endpoints connected to AVC
conferences using their AVC capabilities will be reconnected to their AVC conferences.
 SVC endpoints cannot be reconnected to their SVC Only conferences as dial-out is not supported
for SVC endpoints. These endpoints will have to manually reconnect to their SVC conferences.
● Cascading between SVC Only conferences or between AVC and SVC Only conferences is not
supported.
● The following functionality and features are not supported during SVC Only conferences:
 FECC
 Skins. The video cells are displayed on the endpoint’s default background.
 IVR functionality
 Conference Gathering phase
 All DTMF enabled features during the conference
 Manual selection of video layout
 Chairperson functionality
 Media Encryption
 Recording of SVC Only conferences
 Text messaging using Message Overlay

MCU Supported Resolutions for SVC Conferencing
The MCU automatically selects the resolution and frame rate according to the conference line rate. The
table below details the maximum resolution and frame rates supported by the MCU for each conference line
rate. The actual video rate, resolution and frame rates displayed on each endpoints is determined by the
endpoint’s capabilities.
SVC Conferencing - Maximum Supported Resolutions per Simulcast Stream
Conference Line Rate
(kbps)

Profile

Maximum
Resolution

Max. Frame
Rate (fps)

Audio Rate
(kbps)

1472 - 2048

High Profile

720p

30fps

48

1024 - 1472

High Profile

720p

15fps

48

768 - 1024

High Profile

720p

15fps

48

512 - 768

High Profile

360p

30fps

48

256 - 512

Base Profile

180p

15fps

48

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SVC Conferencing - Maximum Supported Resolutions per Simulcast Stream
Conference Line Rate
(kbps)

Profile

Maximum
Resolution

Max. Frame
Rate (fps)

Audio Rate
(kbps)

192 - 256

Base Profile

180p

30fps

48

128 - 192

Base Profile

180p

15fps

48

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Mixed CP and SVC Conferencing
In a mixed CP (AVC) and SVC conference, AVC-based endpoints and SVC-enabled endpoints can be
supported in the same conference.
In a mixed CP (AVC) and SVC conference, SVC endpoints transmit multiple resolutions and temporal layers
to the Collaboration Server like the SVC-based conferences, while AVC endpoints, for example, send only
one AVC video stream to the Collaboration Server. Other endpoints (also referred to as AVC endpoints as
opposed to SVC endpoints) can send different video protocols, such as H.263. The Collaboration Server
relays SVC-encoded video bit streams to the SVC-enabled endpoints in the conference according to their
request. This enables the video conference layouts to be automatically assembled by the endpoint. AVC
endpoints connected to the conference send a single AVC video bit stream to the Collaboration Server,
which is then transcoded to SVC video streams. SVC-enabled endpoints receive the AVC converted video
bit streams through the Collaboration Server from the AVC endpoints as a single SVC video bit stream.
Alternatively, AVC endpoints receive a single video bit stream with the defined video conference layout from
the Collaboration Server. In this mixed mode conferencing, both SVC and AVC endpoints in the conference
receive the same CP layout.
The following diagram illustrates an example of a mixed CP and SVC conferencing mode:

In this example, an SVC endpoint (1) receives three video streams at different frame rates and resolutions,
and creates the conference layout with the received video streams. The video bit stream that the SVC
endpoint receives from the AVC endpoint (3) is decoded in the Collaboration Server and then encoded into
an SVC bit stream in the required resolution.
Alternatively, an AVC endpoint (4) sends a single resolution video stream to the Collaboration Server. The
Collaboration Server first decodes the SVC bit streams and AVC bit streams, then the Collaboration Server
composes the video layout for the AVC endpoint and sends a single resolution video stream with the video
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layout to the participant. In the displayed example, the Collaboration Server creates different video layouts
for each AVC endpoint.

MCU Resource Capacities for Mixed CP and SVC Conferences
In a mixed CP and SVC conference, video resources are allocated according to the MCU type and the
translation pools (AVC to SVC and SVC to AVC) used to convert video streams. Translation pools are
dynamically allocated, when the conference becomes a mixed CP and SVC conference; resources are not
released when the conference stops being a mixed CP and SVC conference. The translation pools send
one SVC to AVC stream with a resolution of 360p, two AVC to SVC streams with a resolution of 360p and
180p for AVC HD endpoints, and one video stream with a resolution of 180p for AVC SD endpoints. When
a video stream with a resolution of 360p is not available, a video stream with a resolution of 180p is sent
instead.
Translations between different endpoints can be done without using the highest resolution, thus saving
translation resources. CP video layouts in mixed CP and SVC conferences support the standard resolutions
as in normal CP conferences.
Taking these factors into consideration and the type of MCU deployed in the environment, the resource
capacities for a mixed CP and SVC conference can vary.
The following table describes an example of the resource capacity allocations for the RealPresence
Collaboration Server:
Resource Capacity Allocations
Resource Type

Number of Available Ports

Mixed CP and SVC (HD) (Example)

20 AVC
90 SVC

HD720p30

40

SD (@ 30 fps)

40

SVC Only

60

CIF (@ 30 fps)

60

The first four resource types in the resource capacity allocations table are endpoints in a CP only conference
or a mixed CP and SVC conference before the actual resource allocations occur.
In a mixed CP and SVC conference, video resources are used according to the amount of both AVC and
SVC participants in the conference and according to the actual type of the conference - mixed CP and SVC
conferences or CP only conferences. The ratio of resources in a mixed conference is one AVC HD (720p30)
video resource to three SVC video resources, meaning for each AVC HD video resource, three SVC video
resources can be allocated.
In this resource capacity allocations example, the mixed CP and SVC conference can allocate a
combination of AVC and SVC ports depending on the endpoints that are defined in the actual conference.
For example, a conference can be defined as a mixed CP and SVC conference but will only allocate
resources as a mixed conference when both AVC and SVC endpoints join the conference. When there are
only one resource type of endpoints participating in the conference, such as AVC or SVC, the resource
allocations are assigned according to the type of endpoint. For instance, a mixed CP and SVC conference
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with HD endpoints assigned, can have 60 or 120 ports allocated depending on the server configuration.
When an SVC endpoint joins the conference, the conference becomes an actual mixed conference and the
resource allocations are divided between the AVC and SVC endpoints. The Resource Report will reflect this
by showing an increase in the resource usage.
The following diagram illustrates the amount of AVC to SVC port resources that are used in an actual mixed
CP and SVC conference:

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Using Conference Profiles
In the Polycom® RealPresence® CloudAXIS™ Suite, the Conference Profiles are defined in the
RealPresence DMA system component and should not be defined directly in the Polycom®
RealPresence® Collaboration Server 800s and Polycom® RealPresence® Collaboration Server
Virtual Edition component.

Conference Profiles include conference parameters such as Conferencing Mode, conference line rate,
video and content sharing resolutions and settings, video layout, encryption, Lost Packet Recovery (LPR),
etc. Profiles eliminate the need to define these parameters for each new conference created on the MCU.
They are stored on the Collaboration Server and they enable you to define all types of conferences.
The maximum number of Conference Profiles that can be defined is 80.
Conference Profiles are assigned to Conferences, Meeting Rooms, Reservations and Entry Queues. The
same Profile can be assigned to different conferencing entities. When modifying the Profile parameters, the
changes will be applied to all the conferencing entities to which the profile is assigned.
Conference Profile options differ according to the selected Conferencing Mode. Profiles can be defined for
AVC (Advanced Video Codec) CP conferencing Mode, SVC (Scalable Video Codec) conferencing Mode or
Mixed CP and SVC Conferencing Mode.
Conference Profiles can be saved to Conference Templates along with all participant parameters, including
their Personal Layout and Video Forcing settings. It enables administrators and operators to create, save,
schedule and activate identical conferences quickly and easily.

Conferencing Parameters Defined in a Profile
When defining a new video Profile, you select the parameters that determine the video display on the
participant’s endpoint and the quality of the video, according to the selected Conferencing Mode. When
defining a new conference Profile, the system uses default values for the selected conferencing Mode.

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Conferencing Capabilities in the Various Conferencing Modes
The following table summarizes the conferencing capabilities and options available in the different
Conferencing Modes.
Conferencing Capabilities in the Different Conferencing Modes
Feature

CP Only

Mixed CP & SVC

SVC Only


*


*


*


**


**










Conference Type
Operator Conferences
Entry Queues
Permanent Conference
Cascading
Conferencing Feature
IVR

Reduced IVR set for SVC
endpoints
Dial Out







Auto Redial



LPR




***


***

Content



‡



All Content Settings, All
Content Protocols

Graphics Only, H.264
Cascade & SVC
Optimized (only)

Presentation Mode







Lecture Mode







Same Layout







Layout Selection





Layout set to Auto Layout
and defined on the endpoint

AVC endpoints only
Skins





Graphics Only, H.264
Cascade & SVC Optimized



AVC endpoints only
Encryption

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

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Conferencing Capabilities in the Different Conferencing Modes
Feature

CP Only

Recording



Mixed CP & SVC

SVC Only





AVC recording only
Site Names




AVC endpoints only

Managed by the endpoint
(not via MCU)

* Entry Queue & Destination Conference must have the same profile (i.e. SVC only to SVC only, Mixed CP and SVC
to Mixed CP and SVC)
** Only Basic Cascading is available
*** For AVC, the LPR error resiliency is used, however for SVC endpoints, new error resiliency methods are used.
‡. Content Line Rate is fixed at 128Kbps.

Default Profile Settings in CP Conferencing Mode
The Collaboration Server is shipped with a default Conference Profile for CP conferences which allows
users to immediately start standard ongoing CP conferences. These are also the default settings when
creating a new Profile. The default settings are as follows:
Default CP Only Conference Profile Settings
Setting

Value

Profile Name

Factory_Video_Profile

Line Rate

384Kbps

Operator Conference

Disabled

Encryption

Disabled

Packet Loss Compensation (LPR
and DBA)

Enabled for CP Conferences

Auto Terminate

•
•

Auto Redialing

Disabled

Exclusive Content Mode

Disabled

Enable FECC

Enabled

Video Quality

Sharpness

Maximum Resolution

Auto

Content Settings

HiResGraphics (High Res Graphics)

Content Protocol

H.264 HD

Presentation Mode

Disabled

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Default CP Only Conference Profile Settings
Setting

Value

Same Layout

Disabled

Lecturer View Switching

Disabled

Auto Scan Interval

Disabled (10)

Auto Layout

Enabled

Mute participants except the lecturer

Disabled

Skin

Polycom

IVR Name

Conference IVR Service

Recording

Disabled

Site Names display

Disabled

Network Services - SIP Registration

Disabled

Network Services - Accept Calls

Enabled

Default Profile Settings in SVC Only Conferencing Mode
The Collaboration Server is shipped with a default Conference Profile for SVC Only conferences which
allows users to immediately start standard ongoing SVC Only conferences. These are also the default
settings when creating a new Profile. The default settings are as follows:
Default SVC Only Conference Profile Settings
Setting

Value

Profile Name

Factory_SVC_Video_Profile

Line Rate

1920Kbps

Operator Conference

Not supported

Encryption

Disabled

Packet Loss Compensation (LPR and
DBA)

Not supported

Auto Terminate

•
•

Auto Redialing

Not supported

Exclusive Content Mode

Disabled

Enable FECC

Disabled

Video Quality

Sharpness

Maximum Resolution

Auto

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Default SVC Only Conference Profile Settings
Setting

Value

Content Settings

Graphics

Content Protocol

H.264 Cascading and SVC Optimized

Presentation Mode

Not applicable

Same Layout

Not applicable

Lecturer View Switching

Not applicable

Auto Scan Interval

Not applicable

Auto Layout

Enabled (Only available option)

Mute participants except the lecturer

Not applicable

IVR Name

Conference IVR Service

Network Services - SIP Registration

Disabled

Network Services - Accept Calls

Enabled

Default Profile Settings in a Mixed CP and SVC Conferencing Mode
The Collaboration Server is shipped with a default Conference Profile (CP and SVC) for mixed CP and SVC
conferences which enables users to immediately start a standard ongoing mixed CP and SVC conference.
These are also the default settings when creating a new Profile. (During mixed SVC & CP conferences,
PSTN (Audio Only) calls are supported.) Dial-out is not available in Mixed CP and SVC conferences.
The default settings are as follows:
Default Mixed CP and SVC Conference Profile Settings
Setting

Value

Profile Name

Factory_Mix_SVC_CP_Video_Profile

Line Rate

1920Kbps

Operator Conference

Disabled

Encryption

Enabled

Packet Loss Compensation (LPR
and DBA)

Enabled for AVC participants only

Auto Terminate

•
•

Auto Redialing

Disabled

Font for text over video

Enabled for AVC participants only

Exclusive Content Mode

Disabled

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Default Mixed CP and SVC Conference Profile Settings
Setting

Value

Enable FECC

Enabled

Video Quality

Sharpness

Maximum Resolution

Auto

Content Settings

Graphics

Content Protocol

H.264 Cascade and SVC Optimized (only)

Presentation Mode

Disabled

Same Layout

Enabled

Lecturer View Switching

Disabled

Auto Scan Interval

Disabled

Auto Layout

Enabled

Mute participants except the lecturer

Disabled

Skin

Classic (for AVC participants)

IVR Name

Conference IVR Service

Recording

Enabled

Site Names display

Enabled for AVC participants only

Network Services - SIP Registration

Disabled

Network Services - Accept Calls

Enabled

Network quality indication

Enabled for AVC participants only

This Profile is automatically assigned to the following conferencing entities:
Name

ID

Meeting Rooms
Maple_Room

1001

Oak_Room

1002

Juniper_Room

1003

Fig_Room

1004

Entry Queue
Default EQ

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Viewing the List of Conference Profiles
Existing Conference Profiles are listed in the Conference Profiles list pane.

To list Conference Profiles:
1 In the RMX Management pane, expand the Rarely Used list.
2 In the RMX Management pane, Click the Conference Profiles button.
The Conference Profiles are displayed in the Conference Profiles List pane.

The number of the currently defined Conference Profiles appears in the title of the list pane.
The following Conference Profile properties are displayed in the List pane:
Conference Profiles Pane Columns
Field

Description

Name

The name of the Conference Profile.

Layout

Displays either “Auto Layout” or an icon of the layout selected for the profile.
For information about video layouts, see .

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Conference Profiles Pane Columns
Field

Description

Line Rate

The maximum bit rate in kbps at which endpoints can connect to the conference.

Routing Name

Displays the Routing Name defined by the user or automatically generated by the
system.

Encryption

Displays if media encryption is enabled for the Profile.

Profiles Toolbar
The Profile toolbar provides quick access to the Profile functions:
Profile Toolbar buttons
Button

Button Name

Description

New Profile

To create a new Profile.

Delete Profile

To delete a Profile, click the Profile name and then click this
button.

Import Profile

To import Conference Profiles from another MCU in your
environment.

Export Profile

To export Conference Profiles to a single XML file that can be
used to import the Conference Profiles on multiple MCUs.

Modifying an Existing Profile
You can modify any of the Profile’s parameters but you cannot rename the Profile.

To modify the Profile properties:
1 In the Conference Profiles List, double -click the Profile icon or right-click the Profile icon, and then
click Profile Properties.

The Profile Properties - General dialog box opens.
2 Modify the required Profile parameter(s).
3 Click OK.

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Deleting a Conference Profile
You can delete Profiles from the Profiles list.
A Conference Profile cannot be deleted if it is being used by Meeting Rooms, Reservations, Entry
Queues, and SIP Factories. A Profile that is assigned to only one ongoing conference and no other
conferencing entity can be deleted.

To delete a Conference Profile:
1 List the Profile that are currently defined in the system. For details, see Viewing the List of
Conference Profiles.
2 In the Conference Profiles list, select the Conference Profile you want to delete.
3 Click the Delete Profile ( ) button.
or
Right-click the Conference Profile to be deleted and select Delete Profile from the menu.
4 In the confirmation dialog box, click OK.
The Conference Profile is deleted.

Defining New Profiles
In the Polycom® RealPresence® CloudAXIS™ Suite, the Conference Profiles are defined in the
Polycom® RealPresence® DMA® system component and should not be defined directly in the
Polycom® RealPresence® Collaboration Server 800s and Polycom® RealPresence® Collaboration
Server Virtual Edition component.

Profiles are the basis for the definition of all ongoing conferences, Reservations, Meeting Rooms, Entry
Queues, and Conference Templates and they contain only conference properties.
Profiles can be defined for the following Conferencing Modes: AVC (Advanced Video Codec) CP , SVC
(Scalable Video Codec) or Mixed CP and SVC. The Profile tabs and options change according to the
selected Conferencing Mode and only supported options are available for selection. Unsupported options
are disabled (grayed out).
CP Conferencing Mode also offers a special functional conference - Operator Conference.
To facilitate the definition process of a new Profile, the system displays default values for each parameter
so you need only to modify the required settings.

To define a new Profile:
1 In the RMX Management pane, expand the Rarely Used list.
2 In the RMX Management pane, click Conference Profiles.
3 In the Conference Profiles pane, click the New Profile button.
The New Profile – General dialog box opens.
4 In the Display Name field, enter the Profile name.

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5 Select the appropriate Conferencing Mode: CP, SVC Only or CP and SVC.
The New Profile tabs and options change according to the selected Conferencing Mode and only
supported options are available for selection.
6 Define the Profile parameters as described in:
 Defining AVC CP Conferencing Profiles
 Defining SVC Conference Profiles
 Defining Mixed CP and SVC Conferencing Profiles

Exporting and Importing Conference Profiles
Conference Profiles can be exported from one MCU and imported to multiple MCUs in your environment,
enabling you to copy the Conference Profiles definitions to other systems. This can save configuration time
and ensures that identical settings are used for conferences running on different MCUs. This is especially
important in environments using cascading conferences that are running on different MCUs.

Guidelines for Exporting and Importing Conference Profiles
● Only Collaboration Server system administrators can export and import Conference Profiles.
Operators are only allowed to export Conference Profiles.
● You can select a single, multiple, or all Conference Profiles to be exported.
● Conference Templates and their related Conference Profiles can be exported and imported
simultaneously using the Conference Templates export and import function. For more information,
see Exporting and Importing Conference Templates.

Exporting Conference Profiles
Conference Profiles are exported to a single XML file that can be used to import the Conference Profiles on
multiple MCUs. Using the Export Conference Profile feature, you can export all or selected Conference
Profiles from an MCU.

Exporting All Conference Profiles from an MCU
To export all Conference Profiles from an MCU:
1 List the Profile that are currently defined in the system. For details, see Viewing the List of
Conference Profiles.
2 In the Conference Profiles List toolbar, click the Export Conference Profiles
button or
right-click anywhere in the Conference Profiles pane, and then click Export Conference Profiles.

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The Conference Profile - Export dialog box is displayed.

3 In the Export Path field, click Browse to navigate to the location of the desired path where you want
to save the exported file.
4 In the Profiles file name field, type the file name prefix. The file name suffix (_confProfiles.xml) is
predefined by the system. For example, if you type Profiles01, the exported file name is defined
as Profiles01_confProfiles.xml.
5 Click OK to export the Conference Profiles to a file.
If the export file with the same file name already exists, a prompt is displayed.

6 Click Yes to replace the exported file or click No to cancel the export operation and return to the
Conference Profiles list. You can modify the export file name and restart the export operation.

Exporting Selected Conference Profiles
You can select a single Conference Profile or multiple Conference Profiles and export them to a file to be
imported to other MCUs in your environment.

To export selected Conference Profiles:
1 List the Profile that are currently defined in the system. For details, see Viewing the List of
Conference Profiles.
2 In the Conference Profiles pane, select the profiles you want to export.
3 In the Conference Profiles List toolbar, click the Export Conference Profiles
button or
right-click the selected Conference Profiles, and then click Export Selected Conference Profiles.

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The Conference Profile - Export dialog box is displayed.

4 In the Export Path field, click Browse to navigate to the location of the desired path where you want
to save the exported file.
5 In the Profiles file name field, type the file name prefix. The file name suffix (_confProfiles.xml) is
predefined by the system. For example, if you type Profiles01, the exported file name is defined
as Profiles01_confProfiles.xml.
6 Click OK to export the Conference Profiles to a file.
If the export file with the same file name already exists, a prompt is displayed.

7 Click Yes to replace the exported file or click No to cancel the export operation and return to the
Conference Profiles list. You can modify the export file name and restart the export operation.

Importing Conference Profiles
If your environment includes two or more MCUs, import previously exported Conference Profiles to your
MCU to save configuration time and ensure that all MCUs use the same conferencing parameters.
Conference Profiles are not imported when a Conference Profile with that name already exists or if
an IVR Service which is assigned to any of the imported Profile does not exist in the MCU.

Conference Profiles are not imported when a Conference Profile with that name already exists or if an IVR
Service which is assigned to any of the imported Profile does not exist in the MCU.

To import Conference Profiles:
1 Display the Conference Profiles List. For details, see Viewing the List of Conference Profiles.

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2 In the Conference Profiles List toolbar, click the Import Conference Profiles
button or
right-click the Conference Profiles pane, and then click Import Conference Profiles.

The Conference Profile - Import dialog box is displayed.

3 In the Import Path field, click Browse to navigate to the path and file name of the exported
Conference Profiles you want to import.
4 Click OK to import the Conference Profiles.
When Conference Profiles cannot be imported, a Message Alert window is displayed with the
profiles that were not imported.

Conference Profiles that are not problematic are imported.
5 Click Cancel to exit the Message Alerts window.
The imported Conference Profiles appear in the Conference Profiles list.

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Defining AVC-Based Conference Profiles
In the RealPresence CloudAxis Solution, the Conference Profiles are defined in the Polycom®
RealPresence® DMA® component and should not be defined directly in the Polycom®
RealPresence® Collaboration Server 800s and Polycom® RealPresence® Collaboration Server
Virtual Edition component.

AVC-based Conference Profile options differ according to the selected Conferencing Mode. To facilitate the
definition process of a new Profile, the system displays default values for each parameter so you need only
to modify the required settings.

Defining AVC CP Conferencing Profiles
When defining a new Profile, you select the parameters that determine the video display on the participant’s
endpoint, the quality of the video, content sharing parameters, whether the conference will be recorded,
encryption, Telepresence mode and other conferencing parameters.
The following parameters are defined:
● New AVC CP Profile - General Parameters
● New AVC CP Profile - Advanced Parameters
● New AVC CP Profile - Gathering Settings Parameters
● New AVC CP Profile - Video Quality Parameters
● New AVC CP Profile - Video Settings Parameters
● New AVC CP Profile - Audio Settings Parameters
● New AVC CP Profile - IVR Parameters
● New AVC CP Profile - Recording Parameters
● New AVC CP Profile - Site Names Parameters
● New AVC CP Profile - Message Overlay Parameters
● New AVC CP Profile - Network Services Parameters
● New AVC CP Profile - Layout Indications Parameters

To define a new CP Profile:
1 In the RMX Management pane, click Conference Profiles.

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2 In the Conference Profiles pane, click the New Profile button.
The New Profile – General dialog box opens.

3 Define the Profile name and, if required, the Profile General parameters:
New AVC CP Profile - General Parameters
Field/Option

Description

Display Name

Enter a unique Profile name, as follows:
• English text uses ASCII encoding and can contain the most characters (length
varies according to the field).
• European and Latin text length is approximately half the length of the maximum.
• Asian text length is approximately one third of the length of the maximum.
It is recommended to use a name that indicates the Profile type, such as CP or
Operator conference.
Note: This is the only parameter that must be defined when creating a new profile.
Note: This field is displayed in all tabs.

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New AVC CP Profile - General Parameters (Continued)
Field/Option

Description

Line Rate

Select the conference bit rate. The line rate represents the combined video, audio
and Content rate.
The default setting is 384 Kbps.
Note:
This field is displayed in all tabs.

Conferencing Mode

Select the required Conferencing Mode. The selection affects the available tabs
and their fields.
For CP conferencing, make sure that CP (Continuous Presence) is selected to
define a CP conference Profile (it is the default option).
Note:
This field is displayed in all tabs.

Routing Name

Enter the Profile name using ASCII characters set.
The Routing Name can be defined by the user or automatically generated by the
system if no Routing Name is entered as follows:
• If an all ASCII text is entered in Display Name, it is used also as the Routing
Name.
• If any combination of Unicode and ASCII text (or full Unicode text) is entered in
Display Name, the ID (such as Conference ID) is used as the Routing Name.

Operator Conference
(CP Only)

Select this option to define the profile of an Operator conference.
When defining an Operator Conference, the Send Content to Legacy Endpoints
option in the Video Quality tab is cleared and disabled.

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4 Click the Advanced tab.
The New Profile – Advanced dialog box opens.

5 Define the following parameters:
New AVC CP Profile - Advanced Parameters
Field/Option

Description

Encryption

Select the Encryption option for the conference:
• Encrypt All - Encryption is enabled for the conference and all conference
participants must be encrypted.
• No Encryption - Encryption is disabled for the conference.
• Encrypt when Possible - Enables the negotiation between the MCU and the
endpoints and let the MCU connect the participants according to their
capabilities, where encryption is the preferred setting.

LPR

When selected (default for CP conferences), Lost Packet Recovery creates
additional packets that contain recovery information used to reconstruct packets
that are lost during transmission.

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New AVC CP Profile - Advanced Parameters (Continued)
Field/Option

Description

Auto Terminate

When selected (default), the conference automatically ends when the
termination conditions are met:
• Before First Joins — No participant has connected to a conference during
the n minutes after it started. Default idle time is 10 minutes.
• At the End - After Last Quits — All the participants have disconnected from
the conference and the conference is idle (empty) for the predefined time
period. Default idle time is 1 minute.
• At the End - When Last Participant Remains — Only one participant is still
connected to the conference for the predefined time period (excluding the
recording link which is not considered a participant when this option is
selected). Default idle time is 1 minute.
Note: The selection of this option is automatically cleared and disabled when
the Operator Conference option is selected. The Operator conference cannot
automatically end unless it is terminated by the Collaboration Server User.

Auto Redialing

The Auto Redialing option instructs the Collaboration Server to automatically
redial H.323 and SIP participants that have been abnormally disconnected from
the conference.
• Auto Redialing is disabled by default.
• Auto Redialing can be enabled or disabled during an ongoing conference
using the Conference Properties – Advanced dialog box.
• The Collaboration Server will not redial an endpoint that has been
disconnected from the conference by the participant.
• The Collaboration Server will not redial an endpoint that has been
disconnected or deleted from the conference by an operator or administrator.

Exclusive Content Mode

Select this option to limit the Content broadcasting to one participant, preventing
other participants from interrupting the Content broadcasting while it is active.

Enable FECC

This option is enabled by default, allowing participants in the conference to
control the zoom and PAN of other endpoints in the conference via the FECC
channel. Clear this check box to disable this option for all conference
participants.

FW NAT Keep Alive

The MCU can be configured to send a FW NAT Keep Alive message at specific
Intervals for the RTP, UDP and BFCP channels.
For more information see FW (Firewall) NAT Keep Alive.

Interval

If needed, modify the NAT Keep Alive Interval field within the range of 1 - 86400
seconds.For more information see FW (Firewall) NAT Keep Alive.

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New AVC CP Profile - Advanced Parameters (Continued)
Field/Option

Description

MS AV MCU Cascade
Mode

This enables you to set Cascade Mode as either Resource Optimized or Video
Optimized.
• Resource Optimized
System resource usage is optimized by allowing high resolution connections
only at high line rates and may result in lower video resolutions for some line
rates. This option allows youto save MCU resources and increase the
number of participant connections.
• Video Quality Optimized
Video is optimized through higher resolution connections at lower line rates
increasing the resource usage at lower line rates. This may decrease the
number of participant connections.
For more information, see H.264 Base Profile and High Profile Comparison.

6 For CP Conferences only: Click the Gathering Settings tab.

7 Optional. Define the following fields if the conference is not launched by the Polycom Conferencing
Add-in for Microsoft Outlook:

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•

•

If the conference is launched by the Polycom Conferencing Add-in for Microsoft Outlook the field
information is received from the meeting invitation and existing field value are overridden. For
more information see Polycom Conferencing for Microsoft Outlook® .
Gathering is not supported in Cascading Conferences.

For more information see Auto Scan and Customized Polling in Video Layout (CP Conferences Only).
8 Click the Video Quality tab.
The New Profile – Video Quality dialog box opens.

9 Define the following parameters:

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New AVC CP Profile - Video Quality Parameters
Field/Option

Description

People Video Definition
Video Quality

Sharpness is the only supported content format that supports higher video
resolutions.
Depending on the amount of movement contained in the conference video, select
either:
• Motion – For a higher frame rate without increased resolution. When selected,
Video Clarity is disabled.
• Sharpness – For higher video resolution and requires more system resources.
Note: When Sharpness is selected as the Video Quality setting in the conference
Profile, the Collaboration Server will send 4CIF (H.263) at 15fps instead of CIF
(H.264) at 30fps.

Maximum Resolution

This setting overrides the Maximum Resolution setting of the Resolution
Configuration dialog box.
The administrator can select one of the following Maximum Resolution options:
• Auto (default) - The Maximum Resolution remains as selected in the
Resolution Configuration dialog box.
• CIF
• SD
• HD720
• HD1080
Maximum Resolution settings can be monitored in the Profile Properties - Video
Quality and Participant Properties - Advanced dialog boxes.
Notes:
• The Resolution field in the New Participant - Advanced dialog box allows
Maximum Resolution to be further limited per participant endpoint.
• The Maximum Resolution settings for conferences and participants cannot be
changed during an ongoing conference.

Content Video Definition
Content Settings

Select the transmission mode for the Content channel:
• Graphics — basic mode, intended for normal graphics
• Hi-res Graphics (AVC CP Only) — a higher bit rate intended for high resolution
graphic display
• Live Video (AVC CP Only) — Content channel displays live video
• Customized Content Rate (AVC CP Only) — manual definition of the
Conference Content Rate, mainly for cascading conferences.
Selection of a higher bit rate for the Content results in a lower bit rate for the people
channel.

AS SIP Content

AS-SIP is an implementation of SIP that utilizes SIP’s built in security features.
When selected, content is shared using the Multiple Resolutions mode and is not
supported in any other Content sharing mode.
For more information, see Enabling AS-SIP Content.

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New AVC CP Profile - Video Quality Parameters (Continued)
Field/Option

Description

Multiple Content
Resolutions

Click this check box to enable the Multiple Content Resolutions mode, in which
content is shared in multiple streams, one for each video protocol: H.263 and
H.264. This allows endpoints with different protocols to connect and disconnect
without having to restart Content sharing in the middle of a conference. For more
information, see Sharing Content Using Multiple Content Resolutions Mode.
When enabled, the H.264 is always selected and can not be deselected.
Note: If Multiple Content Resolutions is selected in a Cascading environment, the
Content Protocol must be set to H.264 Cascade and SVC Optimized and H.264
Cascade must be checked as the Transcode to: setting.
Optional. Select additional protocols:
• H.263 - if the conference will include H.263-capable endpoints that do not support
H.264 protocol for content sharing.
• H.264 Cascade - if the conference will include cascading links that should use a
fixed video format for content sharing.
Optional. If H.264 Cascade is selected, select the desired Content Resolution.

Content Protocol

Select the Content Protocol to be used for content sharing in Highest Common
Content Sharing Mode.
• H.263 (AVC CP only)
Content is shared using the H.263 protocol.
Use this option when most of the endpoints support H.263 and some endpoints
support H.264.
• H.263 & H.264 Auto Selection (AVC CP only)
When selected, content is shared using H.263 if a mix of H.263-supporting and
H.264-supporting endpoints are connected, or H.264 if all connected endpoints
have H.264 capability.
• H.264 Cascade and SVC Optimized
All Content is shared using the H.264 content protocol and is optimized for use
in cascaded conferences.
• H.264 HD (AVC CP only, default)
Ensures high quality Content when most endpoints support H.264 and HD
resolutions.
Note: When Multiple Content Resolutions is selected, the Content Protocol field
is hidden.
For more information, see Content Protocols and Defining Content Sharing
Parameters for a Conference.

H.264 High Profile
(Check Box)

The H.264 High Profile check box is un-checked by default and is displayed next to
the Content Protocol drop-down menu if all the following conditions are met:
• The selected Conferencing Mode is AVC-CP.
• Multiple Resolutions (Content Transcoding) is not selected.
• The selected Content Protocol is Cascade and SVC Optimized.
If H.264 HD, H.264 Cascade and SVC Optimized is selected, the Content
Resolution is set according to the line rate.
• TIP Compatibility (in the Profile - Advanced dialog box) is selected as None or
Video Only.

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New AVC CP Profile - Video Quality Parameters (Continued)
Field/Option

Description

Content Resolution

Select the Content Resolution and frame rate according to the selected Content
Sharing Mode (Highest common Content or Multiple Resolution Contents) and the
video protocol. For more information, see Defining Content Sharing Parameters for
a Conference.

Content Rate
drop-down menu

The Content Rate drop-down menu is displayed next to the Content Resolution
drop-down menu when:
• H.264 Cascade and SVC Optimized is the selected Content Protocol
and
• CustomizedContentRate is the selected Content Setting.

Send Content to
Legacy Endpoints
(CP only)

When enabled (default), Content can be sent to H.323/SIP endpoints that do not
support H.239 Content (legacy endpoints) over the video (people) channel. For
more information see Sending Content to Legacy Endpoints (AVC Only).
Notes:
• When enabled, an additional HD video resource is allocated to the conference.
• This option is valid when sending Content as a separate stream is enabled by the
System Flag ENABLE_H239 set to YES.
• Select this option when Avaya IP Softphone will be connecting to the conference.
• If the Same Layout option is selected, the Send Content to Legacy Endpoints
selection is cleared and is disabled.
• Once an endpoint is categorized as Legacy, it will not be able to restore its
content to the Content channel and will receive content only in the video channel.

10 Click the Video Settings tab.
The New Profile - Video Settings dialog box opens.

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11 Define the video display mode and layout using the following parameters:
New AVC CP Profile - Video Settings Parameters
Field/Option

Description

Presentation Mode

Select this option to activate the Presentation Mode.
In this mode, when the current speaker speaks for a predefined time (30 seconds),
the conference changes to Lecture Mode. When another participant starts talking,
the Presentation Mode is cancelled and the conference returns to the previous
video layout.

(CP only)

Same Layout
(CP only)
Lecture View
Switching

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Select this option to force the selected layout on all participants in a conference.
Displays the same video stream to all participants and personal selection of the
video layout is disabled. In addition, if participants are forced to a video layout
window, they can see themselves.
Select this option to enable automatic switching of participants on the Lecturer’s
screen when Lecture Mode is enabled for the conference.
The automatic switching is enabled when the number of participants exceeds the
number of video windows displayed on the Lecturer’s screen.
Note: Lecture Mode is enabled in the Conference Properties – Participants tab.
For more information, see Lecture Mode (AVC CP Only).

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New AVC CP Profile - Video Settings Parameters (Continued)
Field/Option

Description

Auto Scan Interval(s)

Select the time interval, 5 - 300 seconds, that Auto Scan uses to cycle the display
of participants that are not in the conference layout in the selected cell.
Auto Scan is often used in conjunction with Customized Polling which allows the
cyclic display to be set to a predefined order for a predefined time period.

(CP only)
Auto Layout
(CP only)

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When selected (default), the system automatically selects the conference layout
based on the number of participants currently connected to the conference. When a
new video participant connects or disconnects, the conference layout automatically
changes to reflect the new number of video participants.
For more information, see Auto Layout – Default Layouts in CP Conferences.
Clear this selection to manually select a layout for the conference.
The default Auto Layout settings can be customized by modifying default Auto
Layout system flags in the System Configuration file. For more information see,
Auto Layout Configuration.
Note: In some cases, the default layout automatically selected for the conference
contains more cells than the number of connected participants, resulting in an
empty cell. For example, if the number of connected participants is 4, the default
layout is 2x2, but as only 3 participants are displayed in the layout (the participants
do not see themselves), one cell is empty.

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Auto Layout – Default Layouts in CP Conferences
Number of Video Participants

Auto Layout Default Settings

0–2

3

4–5

6–7

8-10

11

12+

In layout 2+8, the two central windows display the last two speakers in the conference: the current
speaker and the “previous” speaker. To minimize the changes in the layout, when a new speaker is
identified the “previous” speaker is replaced by the new speaker while the current speaker remains
in his/her window.

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The Collaboration Server supports the VUI addition to the H.264 protocol for endpoints that transmit
wide video (16:9) in standard 4SIF resolution.

When there is a change of speaker in a Continuous Presence conference, the transition is set by
default to fade in the current speaker while fading out the previous speaker.
To make this transition visually pleasant, fading in the current speaker while fading out the previous
speaker is done over a period of 500 milliseconds.
The Fade In / Fade Out feature can be disabled by adding a new flag to the System Configuration.
The Value of the new flag must be: FADE_IN_FADE_OUT=NO.
For more information about System Flags, see the Modifying System Flags.

12 To select the Video Layout for the conference, click the required number of windows from the
layouts bar and then select the windows array. The selected layout is displayed in the Video Layout
pane.
Video Layout Options
Number of Video Windows

Available Video Layouts

1

2

3

4

5+

9

10+

13 Click the Audio Settings tab.

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The New Profile - Audio Settings dialog box opens.

14 Define the following parameters:

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New AVC CP Profile - Audio Settings Parameters
Field/Option

Description

Mute participant
except lecturer

When the Mute Participants Except Lecturer option is enabled, the audio of all
participants in the conference except for the lecturer can be automatically muted
upon connection to the conference. This prevents other conference participants
from accidentally interrupting the lecture, or from a noisy participant affecting the
audio quality of the entire conference. Muted participants cannot unmute
themselves unless they are unmuted from the Collaboration Server Web
Client/RMX Manager.
You can enable or disable this option during the ongoing conference.
Notes:
• When enabled, the mute indicator on the participant endpoints are not visible
because the mute participants was initiated by the MCU. Therefore, it is
recommended to inform the participants that their audio is muted by using the
Message Overlay function.
In the Collaboration Server Web Client/RMX Manager the mute by MCU
indicator is listed for each muted participant in the Audio column in the
Participants pane.
• This option can be disabled during an ongoing conference, thereby unmuting all
the participants in the conference.
• If the endpoint of the designated lecturer is muted when the lecturer connects to
the conference, the lecturer remains muted until the endpoint has been unmuted.
• When you replace a lecturer, the MCU automatically mutes the previous lecturer
and unmutes the new lecturer.
• When you disconnect a lecturer from the conference or the lecturer leaves the
conference, all participants remain muted but are able to view participants in
regular video layout until the you disable the Mute Participants Except Lecturer
option.
• A participant can override the Mute Participants Except Lecturer option by
activating the Mute All Except Me option using the appropriate DTMF code,
provided the participant has authorization for this operation in the IVR Services.
The lecturer audio is muted and the participant audio is unmuted. You can
reactivate the Mute Participants Except Lecturer option after a participant has
previously activated the Mute All Except Me option. The participant is muted and
the lecturer, if designated, is unmuted.
• In cascaded conferences, all participants (including the link participant) are
muted. Only the lecturer is not muted.

Speaker Change
Threshold

Indicates the amount of time a participant must speak continuously before
becoming the speaker.
Select the desired threshold:
• Auto (Default, 3 seconds)
• 1.5 seconds
• 3 seconds
• 5 seconds

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15 Click the Skins tab to modify the background and frames.
The New Profile - Skins dialog box opens.

16 Select one of the Skin options.

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17 Click the IVR tab.
The New Profile - IVR dialog box opens.

18 If required, set the following parameters:
New AVC CP Profile - IVR Parameters
Field/Option

Description

Conference IVR
Service

The default conference IVR Service is selected. You can select another conference
IVR Service if required.

Conference Requires
Chairperson

Select this option to allow the conference to start only when the chairperson
connects to the conference and to automatically terminate the conference when the
chairperson exits. Participants who connect to the conference before the
chairperson are placed on Hold and hear background music (and see the Welcome
video slide). Once the conference is activated, the participants are automatically
connected to the conference.
When the check box is cleared, the conference starts when the first participant
connects to it and ends at the predefined time or according to the Auto Terminate
rules when enabled.

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New AVC CP Profile - IVR Parameters (Continued)
Field/Option

Description

Terminate conference
after chairperson
leaves

Select this check box to automatically terminate the conference after the
chairperson leaves. When the chairperson leaves, the “Chairperson Has Left” IVR
message is played to all participants, at which point the conference terminates. This
way an operator does not need to monitor a conference to know when to terminate
it manually.
If there is a single chairperson in the conference who is changed to a regular
participant the conference will be terminated as if the chairperson left. If there is
more than one chairperson, then changing one chairperson to a regular participant
will not terminate the conference. It is therefore recommended that before changing
a single chairperson to regular participant, another participant first be changed to
chairperson.
Terminate Conference After Chairperson Leaves is not supported in cascaded
environments.

19 Optional. Click the Recording tab to enable conference recording with Polycom RSS 2000/4000.
The New Profile - Recording dialog box opens.

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20 Define the following parameters:
New AVC CP Profile - Recording Parameters
Parameter

Description

Enable Recording

Select this check box to enable the Recording of the conference. If no
Recording Links are found, an error message is displayed.

Recording Link

Select the Recording Link to be used for conference recording.
Recording Links defined on the Collaboration Server can be given a
descriptive name and can be associated with a Virtual Recording
Room (VRR) saved on the Polycom® RSS™ 4000 (Recording and
Streaming Server). For more information see Recording Conferences

Start Recording

Select when to start the recording:
• Immediately – conference recording is automatically started upon
connection of the first participant.
• Upon Request – the operator or chairperson must initiate the
recording (manual).

Display Recording
Icon

This option is automatically selected to display a Recording Indication
to all conference participants informing them that the conference is
being recorded.
Clear the selection to prevent the display of the recording icon.

The Recording link (which is listed as a participant in the conference) does not support H.264 High
Profile. If recording a conference that is set to H.264 High Profile, the Recording participant connects
as Audio Only and records only the conference Audio.

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21 Click the Site Names tab.
The New Profile - Site Names dialog box opens.

Using the Site Name dialog box, you can control the display of the site names by defining the font,
size, color, background color and transparency and position within the Video Window. For a detailed
description of the site names options see Site Names Definition.
22 Define the following parameters:
New AVC CP Profile - Site Names Parameters
Field

Description

Display Mode

Select the display mode for the site names:
• Auto - Display the Site Names for 10 seconds whenever the Video Layout
changes.
• On - Display the Site Names for the duration of the conference.
• Off (default) - Do not display the Site Names and all other fields in this tab are
grayed and disabled

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New AVC CP Profile - Site Names Parameters
Field

Description

Font Size

Click the arrows to adjust the font size (in points) for the display of Site Names.
Choose a Font Size that is suitable for viewing at the conference’s video resolution.
For example, if the resolution is CIF, a larger Font Size should be selected for
easier viewing.
Range: 9 - 32 points
Default: 12 points

Background Color

Select the color of the Site Names display text.
The color and background for Site Names display text is dependent on whether a
Plain Skin or a Picture Skin was selected for the conference in the Profile - Skins
tab. The choices are:

Note: Choose a Background Color combination that is suitable for viewing at the
conference’s video resolution. At low resolutions, it is recommended to select
brighter colors as dark colors may not provide for optimal viewing.

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New AVC CP Profile - Site Names Parameters
Field

Description

Display Position

Select the pre-set position for the display of the Site (endpoint) Names.
Selection

Site Names Position

LeftTop (Default)

Top

RightTop

LeftMiddle

RightMiddle

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New AVC CP Profile - Site Names Parameters
Field

Description

Display Position
(cont.)

LeftBottom

Bottom

RightBottom

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23 Click the Message Overlay tab.
The New Profile - Message Overlay dialog box opens.

Message Overlay enables you to send text messages to all participants during ongoing Continuous
Presence conferences.
The text message is seen as part of the in the participant’s video layout on the endpoint screen or
desktop display.
For more details, see Sending Text Messages During a Conference Using Message Overlay.

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24 Define the following fields:
New AVC CP Profile - Message Overlay Parameters
Field

Description

Enable

This option is disabled by default. Select this check box to enable Message Overlay
or clear it to disable it.

Content

Enter the message text. The message text can be up to 50 Chinese characters.

Font Size

Click the arrows to adjust the font size (points) for the display of the message text.
Font size range: 9 - 32 points, default: 24 points
Note: In some languages, for example Russian, when a large font size is selected,
both rolling and static messages may be truncated if the message length exceeds
the resolution width.

Color

From the drop-down menu select the color and background of the displayed text.
The choices are:

Default: White text on red background.
Vertical Position

Move the slider to the right to move the vertical position of the displayed text
downward within the Video Layout.
Move the slider to the left to move the vertical position of the displayed text upward
within the Video Layout.
Default: Top Left (10)

Background
Transparency

Move the slider to the left to decrease the transparency of the background of the
message text. 0 = No transparency (solid background color).
Move the slider to the right to increase the transparency of the background of the
message text. 100 = Full transparency (no background color).
Default: 50

Display Repetition

Click the arrows to increase or decrease the number of times that the text message
display is to be repeated.
Default: 3

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New AVC CP Profile - Message Overlay Parameters
Field

Description

Display Speed

Select whether the text message display is static or moving across the screen, the
speed in which the text message moves: Static, Slow, Fast
Default: Slow

As the fields are modified the Preview changes to show the effect of the changes.
For example:

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25 Click the Network Services tab.
The New Profile - Network Services dialog box opens.

Registration of conferencing entities such as ongoing conferences, Meeting Rooms, Entry
Queuesand SIP Factories with SIP servers is done per conferencing entity. This allows better control
on the number of entities that register with each SIP server. Selective registration is enabled by
assigning a conference Profile in which registration is configured to the required conferencing
entities. Assigning a conference Profile in which registration is not configure to conferencing entities
will prevent them from registering. By default, Registration is disabled in the Conference Profile, and
must be enabled in Profiles assigned to conferencing entities that require registration.
26 Define the following parameters:
New AVC CP Profile - Network Services Parameters
Parameter

Description

IP Network Services
Service Name

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This column lists all the defined Network Services, one or several depending on the
system configuration.

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New AVC CP Profile - Network Services Parameters
Parameter

Description

SIP Registration

To register the conferencing entity to which this profile is assigned with the SIP
Server of the selected Network Service, click the check box of that Network Service
in this column.
When SIP registration is not enabled in the conference profile, the Collaboration
Server's registering to SIP Servers will each register with an URL derived from its
own signaling address.

Accept Calls

To prevent dial in participants from connecting to a conferencing entity when
connecting via a Network Service, clear the check box of the Network Service from
which calls cannot connect to the conference.

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27 Click the Layout Indications tab.
The Layout Indications dialog is displayed.

The Layout Indications are displayed in the conference video layout as an icon group that includes:
 Recording
 Audio Participants
 Network Quality

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The Layout Indications tab is only displayed if either AVC-CP or AVC-CP and SVC (mixed mode) is
selected.
Beginning with version 8.4, the configuration options of the Layout Indications tab have replaced the
following system flags:
• DISABLE_SELF_NETWORK_IND
• SELF_IND_LOCATION
For further information on layout indications see Layout Indications (AVC Endpoints).

28 Modify the following parameters to configure the display, position, and duration of the indication
icons.
New AVC CP Profile - Layout Indications Parameters
Field

Description

Position

Use the Position drop-down menu to configure the display position of the indication
icons group. Icons can be displayed in the following positions:
• Top-left
• Bottom-left
• Top center (default)
• Bottom
• Top-right
• Bottom-right

Recording

Select the check box to display the Recording icon.
The Recording indication icon is a duplicate of the Display Recording Icon field in the
Recording tab of the Profile dialog.
For more information, see Recording Conferences.

Audio Participants

Select the check box to display the Audio Participants icon.
• Permanent - Sets the Audio Participant Indication to display permanently. In this
setting it is only displayed when audio participants are connected.
• On audio participants change - Sets the Audio Participant Indication to display for
a short period only when the number of audio participants changes.
 Duration - Sets the amount of time the Audio Participant icon displays.
For more information see Audio Participants Indication.

Network Quality

To display the Network Quality icon, select the check box.
For more information see Network Quality Indication (AVC Endpoints).

29 Click OK to complete the Profile definition.
A new Profile is created and added to the Conference Profiles list.

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Additional Information for Setting CP Profiles
This section includes detailed explanation of various CP Profile settings:
● Gathering Phase
● Overlay Layouts
● Site Names Definition
● Sending Text Messages During a Conference Using Message Overlay
● Selecting the Chinese Font for Text Display

Gathering Phase
The Gathering Phase of an AVC (CP only) conference is the time period during which participants are
connecting to a conference. During the Gathering Phase, a mix of live video from connected endpoints is
combined with both static and variable textual information about the conference into a slide which is
displayed on all connected endpoints.

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During the Gathering Phase, the audio of all participants can be heard, and the video of active speakers is
displayed in the video windows as they begin talking.
All connected participants are kept informed about the current conference status including names of
connected participants, participant count, participant type (video/audio) etc.

Gathering Phase Guidelines
● Gathering Phase is only available in AVC only (CP only) conferences. It is not supported in SVC Only
conferences.
● The Gathering Phase slide can be displayed at any time during the conference by entering the Show
Participants DTMF code, *88.
Note: When the display of the Gathering Phase slide is removed, the message overlay text is also
removed.
● The names of the first eight participants to connect are displayed. If eight or more participants
connect, the 8th row displays “…”.
● Static text in the Gathering Phase slide such as the field headings: Organizer, Duration, Video/Audio
Participants, Access Number, IP are always displayed in the language as configured in the Polycom
Virtual Meeting Rooms Add-in for Microsoft Outlook. The following languages are supported:
 English
 French
 German
 International Spanish
 Korean
 Japanese
 Simplified Chinese
● Dynamic text in the Gathering Phase slide such as the meeting name, participants’ names, access
numbers and the additional information entered in the Info1/2/3 fields of the Gathering Settings tab
of the conference Profile are displayed in the language of the meeting invitation.
● The language of a Gathering Phase slide of a conference configured to include a Gathering Phase
that is not launched by the Polycom Conferencing Add-in for Microsoft Outlook is configured by the
administrator. Using the Collaboration Server Web Client, the administrator selects the language for
the Gathering Phase slide. The language selected can be different to that of the Collaboration Server
Web Client used by the administrator to perform the configuration.

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●

Content can be sent during the Gathering Phase. The content is displayed in the large video window
of the participant’s layout while the Gathering slide is displayed in a smaller video window in the
layout.

● Gathering is not supported in Cascading Conferences.

Gathering Phase Duration
The duration of the Gathering Phase can be customized by the administrator so that it is long enough to be
viewed by most connected participants yet short enough so as not to over extend into the scheduled
conferencing time.
The Gathering Phase duration is configured for the Collaboration Server, by the following System Flags in
system.cfg in the Setup >System Configuration:
● CONF_GATHERING_DURATION_SECONDS
Range: 0 - 3600 seconds
Default: 180 seconds
The Gathering Phase duration of the conference is measured from the scheduled start time of the
conference.
Example: If the value of the flag is set to 180, the Gathering slide is displayed for three minutes to all
participants starting at the conference Start Time, and ending three minutes after the conference Start
Time.
For participants who connect before Start Time, the Gathering slide is displayed from the time of
connection until the end of the Gathering duration period.
● PARTY_GATHERING_DURATION_SECONDS
Range: 0 - 3600 seconds
Default: 15 seconds
The value of this flag determines the duration of the display of the Gathering slide for participants that
connect to the conference after the conference Start Time.
Participants connecting to the conference very close to of the end of the Gathering Phase (when there
are fewer seconds left to the end of the Gathering Phase than specified by the value of the flag) have
the Gathering slide displayed for the time specified by the value of the flag.
Example: If the value of the flag is set to 15, the Gathering Phase slide is displayed to the participant
for 15 seconds.

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Enabling the Gathering Phase Display
The Gathering Phase is enabled for per conference in the Conference Profile. The profile also includes the
dial-in numbers and the optional additional information to display on the slide.
Conferences that are configured to include a Gathering Phase that are not launched by the Polycom
Conferencing Add-in for Microsoft Outlook need the following information to be entered via the New Profile
or Profile Properties — Gathering Settings dialog box:
● Display Name (Optional, the Meeting Name is used if left blank.)
● Displayed Language
● Access Number 1 / 2 (Optional.)
● Additional Information (Optional free text)
 Info 1
 Info 2
 Info 3
Conferences launched by the Polycom Conferencing Add-in for Microsoft Outlook receive this information
from the meeting invitation.
For more information see Defining New Profiles .

Overlay Layouts
In Overlay Layouts additional participant endpoints can be displayed over the full screen display of the
conference speaker.
The following Overlay Layouts are available for use in CP Conferences:

Although the following Overlay Layout is included in the Profiles - Video Settings dialog box, it is not
available for use in any Conferencing Mode and is only available when included in the Polycom® Multipoint
Layout (MLA) application:

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These Overlay Layouts will only be available in ITP (Telepresence) conferences when support for
Overlay Layouts is included in the Polycom® Multipoint Layout (MLA) application.

Guidelines for using the Overlay Layouts
● The Overlay Layouts are supported:
 In CP Conferencing Mode only.
 With ITP, non-ITP and CTS endpoints used only as standard endpoints.
 With both new and classic Skins in Collaboration Server CP mode. For more information see the
Polycom® RealPresence Collaboration Server 800s/Virtual Edition Administrator’s Guide, “Skins”
on Click the Skins tab to modify the background and frames.
● Overlay Layouts are not supported in ITP conferences as they are not supported by the MLA
application.
● The Overlay Layouts are 20% of the height of the endpoint display and are supported on endpoints
of both 16:9 and 4:3 aspect ratios.
● Overlay Layouts are recommended for use with high resolution endpoints.
● Overlay Layouts are not selected as defaults by the system and are not included in the Auto Layout
settings.
● Message Overlay is not affected by the use of Overlay Layouts and is displayed on top of the video
layouts.
● Site Names are displayed for all cells. Because the smaller cells are located at the bottom of the large
cell, when enabling Site Names it is advisable not to locate the Site Name at the bottom of the cells.
● Standalone Endpoint Cells are displayed each with a border. For all Overlay Layouts, border color is
dependent on the selected Skin.
● System behavior for Video Forcing and Personal Layout Control when using the Overlay Layouts
during an ongoing conference is the same as for other video layouts.
● Overlay Layouts are only available for selection for the Conference Layout and are not available for
selection for Personal Layout.
● During an ongoing conference you cannot select the Overlay Layouts via PCM or Click&View.
● PCM menus can be used when the Overlay Layouts are active, and they are displayed as the top
layer.

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Selecting the Overlay Layouts
The Overlay Layouts are selected in the New Profile - Video Settings dialog box, in the Overlay tab of the
Video Layout tree.

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Site Names Definition
You can control the display of the site names by defining the font, size, color, background color and
transparency and position within the video window in the Profile - Site Name dialog box.

Guidelines
● Site Names display is Off by default in a new profile.
● Site Names can be enabled to function in one of two modes:
 Auto – Site names are displayed for 10 seconds whenever the conference layout changes.
 On – Site names are displayed for the duration of the conference.
● During the display of the site names, the video frame rate is slightly reduced
● Site Names display characteristics (position, size, color) can by modified during an ongoing
conference using the Conference Properties - Site Names dialog box. Changes are immediately
visible to all participants.
● Site Names display text and background color is dependent on the Skin selected for the conference:
 Plain Skins - Site Names text is displayed without a background.
 Picture Skins - Site Names text is displayed with a background.

Shorten the Site Name Display
The SIP_OMIT_DOMAIN_FROM_PARTY_NAME System Flag can be used to remove Domain Names
from SIP dial-in participants’ Site Names. This prevents long domain names being appended to SIP
participant names, as frequently happens when the Collaboration Server is used with a DMA.
The flag must be manually added to the System Configuration and its value modified as follows:
● YES (Default) - The domain name is omitted from SIP dial-in participant names.
● NO - The domain name remains as part of SIP dial-in participant names.
For more information, see Modifying System Flags.

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Sending Text Messages During a Conference Using Message Overlay
The Message Overlay option in the Conference Profile allows the operator or administrator to send text
messages to all participants during an ongoing conference.
The text message is seen as part of the participant’s video layout on the endpoint screen or desktop display.

Guidelines
● Text messaging using Message Overlay is supported in:
 Continuous Presence (CP) conferences
 Same Layout mode
 Encrypted conferences
 With Unicode or ASCII characters
● Text messages using Message Overlay cannot be displayed:
 In Lecture Mode
 When the PCM menu is active
 On endpoints that have their video suspended
● Text messaging using Messages Overlay can be enabled, disabled or modified (content and display
parameters) during the ongoing conference.
● The number of characters for each language can vary due to the type of font used, for example, the
available number of characters for Chinese is 18, while for English and Russian it is 48.
 In some languages, for example Russian, when large font size is selected, both rolling and static
messages may be truncated if the message length exceeds the resolution width.
● Changes to the Message Overlay Content or display characteristics (position, size, color and speed)
are immediately visible to all participants. When there is a current Message Overlay:
 The current message is stopped immediately, even it has not completed all of its repetitions.
 The Display Repetition count is reset to 1.
 The new message content is displayed  times or until it is stopped and
replaced by another content change.
● If during the ongoing conference the Show Number of Participants DTMF option (default DTMF
*88) is used, when the displayed number of participants is removed, the message overlay text is also
removed.
● The text messages cannot be sent via the Content channel.

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● Message Overlay text settings are not saved in the Conference Template when saving an ongoing
conference as a Conference Template.
● Text messages can also be sent to individual or several participants during the ongoing conferences.
For more details, see thePolycom® RealPresence Collaboration Server 800s / Virtual Edition Getting
Started Guide Sending Text Messages During a Conference Using Message Overlay.
For a detailed description of all the Message Overlay parameters, see New AVC CP Profile - Message
Overlay Parameters.

Selecting the Chinese Font for Text Display
When using the RMX Web Client or the RMX Manager in Chinese (either Simplified Chinese or Traditional
Chinese is selected as an available language in the Setup > Customize Display Settings > Multilingual
Setting, you can select one of several Chinese fonts for use when sending text over video. The font is used
to display text for the following:
● Display of Site Names
● Test messages sent using Message Overlay
● Text displayed on the Gathering slide when Chinese is selected as the display language

Selecting the Chinese Font
The Chinese fonts can be selected in the CP Conference Profile - Advanced dialog box only.

The following Chinese fonts are available for selection:
● Heiti (Default)
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● Songti
● Kaiti
● Weibei
The Chinese font cannot be changed during an existing conference. It can only be modified in the
conference profile.
A participant moved to another conference will be shown the font used by the new conference.

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Defining SVC and Mixed CP and SVC
Conference Profiles
In the Polycom® RealPresence® CloudAXIS™ Suite, the Conference Profiles are defined in the
RealPresence DMA system component and should not be defined directly in the Polycom®
RealPresence® Collaboration Server 800s and Polycom® RealPresence® Collaboration Server
Virtual Edition component.

Defining SVC Conference Profiles
The SVC conference Profile definition is started by selecting SVC as the Conferencing Mode. The dialog
boxes and their options change as the conference behavior and the MCU video processing change. For
example, site name display is performed and controlled by the SVC endpoint and not by the MCU as in CP
conferences.

To define SVC Only Profile:
1 In the RMX Management pane, click Conference Profiles.
2 In the Conference Profiles pane, click the New Profile button.
The New Profile – General dialog box opens.
By default, the Conferencing Mode is set to CP.

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3 Select SVC Only to define the SVC Profile.

The profile tabs and options change accordingly and only supported options are available for
selection. Unsupported options are disabled (grayed out).
4 Define the Profile name and, if required, the Profile - General parameters:
New SVC Profile - General Parameters
Field/Option

Description

Display Name

Enter a unique Profile name, as follows:
• English text uses ASCII encoding and can contain the most characters (length
varies according to the field).
• European and Latin text length is approximately half the length of the maximum.
• Asian text length is approximately one third of the length of the maximum.
This is the only parameter that must be defined when creating a new profile.
Note: This field is displayed in all tabs.

Line Rate

Select the conference bit rate. The line rate represents the combined video, audio
and Content rate.
The default setting for SVC Only conference is 1920kbps.
Note: This field is displayed in all tabs.

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New SVC Profile - General Parameters
Field/Option

Description

Routing Name

Enter the Profile name using ASCII characters set.
You can define the Routing Name or it can be automatically generated by the
system if no Routing Name is entered as follows:
• If an all ASCII text is entered in Display Name, it is used also as the Routing
Name.
• If any combination of Unicode and ASCII text (or full Unicode text) is entered in
Display Name, the ID (such as Conference ID) is used as the Routing Name.

5 Click the Advanced tab.
The New Profile – Advanced dialog box opens.

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6 Define the following supported parameters:
New SVC Profile - Advanced Parameters
Field/Option

Description

Encryption

Select the Encryption option for the conference:
• Encrypt All - Encryption is enabled for the conference and all conference
participants must be encrypted.
• No Encryption - Encryption is disabled for the conference.
• Encrypt when Possible - enables the negotiation between the MCU and the
endpoints and let the MCU connect the participants according to their
capabilities, where encryption is the preferred setting. For connection guidelines
see Mixing Encrypted and Non-encrypted Endpoints in one Conference.
For more information, see Packet Loss Compensation (LPR and DBA) AVC CP
Conferences.

Auto Terminate

When selected (default), the conference automatically ends when the termination
conditions are met:
• Before First Joins — No participant has connected to a conference during the
n minutes after it started. Default idle time is 10 minutes.
• At the End - After Last participant Quits — All the participants have
disconnected from the conference and the conference is idle (empty) for the
predefined time period. Default idle time is 1 minute.
• At the End - When Last Participant Remains — Only one participant is still
connected to the conference for the predefined time period (excluding the
recording link which is not considered a participant when this option is selected).
It is not recommended to select this option for SVC Conferences. Default idle
time is 1 minute.

Exclusive Content
Mode

When selected, Content broadcasting is limited to one participant preventing other
participants from interrupting the Content broadcasting while it is active. For more
details, see

FW NAT Keep Alive

When selected, a FW NAT Keep Alive message is sent at an interval defined in the
field below the check box.

Interval

The time in seconds between FW NAT Keep Alive messages.

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7 Click the Video Quality tab.
The New Profile – Video Quality dialog box opens.

8 In SVC Conferencing Mode, the video and Content sharing parameters cannot be modified and they
are set to the following parameters:
New SVC Profile - Video Quality Parameters
Field/Option

Description

People Video Definition
Video Quality

Only Sharpness is available in SVC Conferencing Mode. The MCU
sends the video stream in the resolution required by the endpoint.

Maximum Resolution

Only Auto is available in SVC Conferencing Mode. The MCU sends
the video stream in the resolution required by the endpoint.

Content Video Definition
Content Settings

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Only Graphics is available in SVC Conferencing Mode for
transmission of Content. It offers the basic mode, intended for normal
graphics. For more information, see Video Preview (AVC Participants
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New SVC Profile - Video Quality Parameters
Field/Option

Description

Content Protocol

H.264 Cascade and SVC Optimized is the only available Content
Protocol for content sharing during SVC-based conferences.
In this mode, all Content is shared using the H.264 content protocol
and all endpoints must use the set video resolution and frame rate
(720p 5fps). Endpoints that do not support these settings cannot
share content.

9 Click the Video Settings tab.

In SVC conferences, each endpoint determines its own video layout and there is no conference level
layout selected. Therefore, all the Video Settings parameters are disabled.
10 Click the Audio Settings tab.

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11 If required, define the Speaker Change Threshold: Auto (Default, 3 seconds), 1.5.3.5.
It indicates the amount of time a participant must speak continuously before becoming the speaker.

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12 Click the IVR tab.

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13 If required, set the following parameters:
New SVC Profile - IVR Parameters
Field/Option

Description

Conference IVR
Service

The default conference IVR Service is selected. You can select
another conference IVR Service if required.

Conference Requires
Chairperson

Select this option to allow the conference to start only when the
chairperson connects to the conference and to automatically
terminate the conference when the chairperson exits. Participants
who connect to the conference before the chairperson are placed on
Hold and hear background music (and see the Welcome video slide).
Once the conference is activated, the participants are automatically
connected to the conference.
When the check box is cleared, the conference starts when the first
participant connects to it and ends at the predefined time or according
to the Auto Terminate rules when enabled.

Terminate conference
after chairperson
leaves

Select this check box to automatically terminate the conference after
the chairperson leaves. When the chairperson leaves, the
“Chairperson Has Left” IVR message is played to all participants, at
which point the conference terminates. This way an operator does not
need to monitor a conference to know when to terminate it manually.
If there is a single chairperson in the conference who is changed to a
regular participant the conference will be terminated as if the
chairperson left. If there is more than one chairperson, then changing
one chairperson to a regular participant will not terminate the
conference. It is therefore recommended that before changing a
single chairperson to regular participant, another participant first be
changed to chairperson.
Terminate Conference After Chairperson Leaves is not supported in
cascaded environments.

The following IVR features are not supported during SVC conferences:
 Roll Call
 Invite Participants
 Entry and Exit tones
 Click & View
 PCM

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14 Click the Network Services tab.
The New Profile - Network Services tab opens.

Registration of conferencing entities such as ongoing conferences, Meeting Rooms, and SIP
Factories with SIP servers is done per conferencing entity. This allows better control of the number
of entities that register with each SIP server. Selective registration is enabled by assigning a
conference Profile in which registration is configured for the required conferencing entities. Assigning
a conference Profile in which registration is not configure for conferencing entities will prevent them
from registering. By default, Registration is disabled in the Conference Profile, and must be enabled
in Profiles assigned to conferencing entities that require registration.
15 Define the following parameters:
New SVC Profile - Network Services Parameters
Parameter

Description

IP Network Services
Service Name

This column lists all the defined Network Services, one or several
depending on the system configuration.

SIP Registration

To register the conferencing entity to which this profile is assigned
with the SIP Server of the selected Network Service, click the check
box of that Network Service in this column.
When SIP registration is not enabled in the conference profile, the
Collaboration Server's registering to SIP Servers will each register
with an URL derived from its own signaling address.

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New SVC Profile - Network Services Parameters
Parameter

Description

Accept Calls

To prevent dial in participants from connecting to a conferencing
entity when connecting via a Network Service, clear the check box of
the Network Service from which calls cannot connect to the
conference.

16 Click OK to complete the Profile definition.
A new Profile is created and added to the Conference Profiles list.

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Defining Mixed CP and SVC Conferencing Profiles
In the Polycom® RealPresence® CloudAXIS™ Suite, the Conference Profiles are defined in the
RealPresence DMA system component and should not be defined directly in the Polycom®
RealPresence® Collaboration Server 800s and Polycom® RealPresence® Collaboration Server
Virtual Edition component.

The mixed CP and SVC Profile is based on the CP Profile with a few of the CP options disabled for
compatibility between AVC and SVC protocols and to enable the media conversion between these two
modes. The Gathering Settings and the Message Overlay options are unavailable in this Conferencing
Mode.
In a mixed CP and SVC conference, the Chairperson can be either an AVC-enabled or SVC-enabled
endpoint.

To configure a mixed AVC and SVC conference:
1 In the Management pane, click Conference Profiles.
2 In the Conference Profiles pane, click the New Profile button.
The New Profile - General dialog box is displayed.
3 In the Conferencing Mode list, select CP and SVC to define a mixed AVC and SVC conference.

Using the various Profile tabs, you can define the following profile parameters:

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● CP and SVC Profile - Advanced parameters - these parameters are the same as for CP
conferences. For details, see New AVC CP Profile - Advanced Parameters.
● CP and SVC Profile - Video Quality parameters - to enable the sharing of video between SVC and
AVC, the common denominator parameters (in this conference, the SVC parameters) are selected
for the conference. For more details, see New SVC Profile - Video Quality Parameters.
● CP and SVC Profile - Video Settings parameters - the video layout parameters apply only to the
AVC-enabled endpoints and do not apply to SVC-enabled endpoints as the SVC endpoints generate
their own layout. Options that are not supported in SVC conferencing are disabled in this dialog box,
for example, Telepresence Mode. For more details, see New AVC CP Profile - Video Settings
Parameters.
● CP and SVC Profile - Audio Settings parameters - options that are not supported in SVC
conferencing are disabled in this dialog box. For more details, see New AVC CP Profile - Audio
Settings Parameters.
● CP and SVC Profile - Skins parameters - the display of a video skin applies only to the AVC-enabled
endpoints and do not apply to SVC-enabled endpoints as the SVC endpoints generate their own
layout.
● CP and SVC Profile - IVR parameters - to enable the same IVR behavior and DTMF usage for SVC
and AVC, the common denominator parameters (in this conference, the SVC parameters) are
selected for the conference. For more details, see New SVC Profile - IVR Parameters.
● CP and SVC Profile - Recording parameters - these parameters are the same as for CP
conferences as the recording is done in AVC format. For details, see New AVC CP Profile - Recording
Parameters.
● CP and SVC Profile - Site Names parameters - these parameters are the same as for CP
conferences as they apply the AVC-enabled endpoints. SVC-enabled endpoints generate the site
name display independent of the MCU. For details, see New AVC CP Profile - Site Names
Parameters.
● CP and SVC Profile - Network Services parameters - these parameters are the same as for CP and
SVC conferences . For details, see New AVC CP Profile - Network Services Parameters.

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Configuration for CP Conferencing
Video Resolutions in AVC-based CP Conferencing
The following video resolution information applies to AVC Conferencing Mode.
For a description of resolutions for SVC Conferencing Mode see Defining SVC and Mixed CP and
SVC Conference Profiles on page 74.

The Collaboration Server always attempts to connect to endpoints at the highest line rate defined for the
conference. If the connection cannot be established using the conference line rate, the Collaboration Server
attempts to connect at the next highest line rate at its highest supported resolution.
Depending on the line rate, the Collaboration Server sends video at the best possible resolution supported
by the endpoint regardless of the resolution received from the endpoint.
The video resolution is also defined by the Video Quality settings in the Profile.
The combination of frame rate and resolution affects the number of video resources required on the MCU
to support the call.
The following resolutions are supported:
● CIF 352 x 288 pixels.
● SD 720 x 576 pixels
● HD 720p 1280 x 720 pixels.

Video Display with CIF, SD and HD Video Connections
Although any combination of CIF, SD and HD connections is supported in all CP conferences, the following
rules apply:
● In a 1X1 Video Layout:
 SD: If the speaker transmits CIF, the MCU will send CIF to all participants, including the SD
participants. In any other layout the MCU will transmit to each participant at the participant’s
sending resolution.
 HD: The MCU transmits speaker resolution (including input from HD participants) at up to SD
resolution. If 1x1 is the requested layout for the entire duration of the conference.
● In asymmetrical Video Layouts:
 SD: A participant in the large frame that sends CIF is displayed in CIF.
 HD: Where participants’ video windows are different sizes, the Collaboration Server transmits HD
and receives SD or lower resolutions.
● In panoramic Video Layouts:
 SD: Participants that send CIF also receive CIF.
 HD: the Collaboration Server transmits HD and receives SD or lower resolutions, the
Collaboration Server scales images from SD to HD resolution.

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H.264 High Profile Support in CP Conferences
The H.264 High Profile is a new addition to the H.264 video protocol suite. It uses the most efficient video
data compression algorithms to even further reduce bandwidth requirements for video data streams.
Video quality is maintained at bit rates that are up to 50% lower than previously required. For example, a
512Kbps call will have the video quality of a 1Mbps HD call while a 1Mbps HD call has higher video quality
at the same (1Mbps) bit rate.
H.264 High-Profile should be used when all or most endpoints support it.

Guidelines
● H.264 High Profile is supported in H.323 and SIP networking environments.
● H.264 High Profile is supported in Continuous Presence conferences at all bit rates, video resolutions
and layouts.
● H.264 High Profile is the first protocol declared by the Collaboration Server, to ensure that endpoints
that support the protocol will connect using it.
Setting minimum bit rate thresholds that are lower than the default may affect the video quality of
endpoints that do not support the H.264 High Profile.
● For monitoring purposes, the Collaboration Server and endpoint H.264 High Profile capability is listed
in the Participant Properties - H.245 and SDP tabs for H.323 participants and SIP participants
respectively.
For more information see "IP Participant Properties” on page 12-20.
● H.264 High Profile is not supported:
 For Content Sharing
 As an RSS Recording link
 With Video Preview

CP Conferencing with H.263 4CIF
The video resolution of 4CIF in H.263 endpoints is only supported for line rates of 384 Kbps to 1920 Kbps
as shown in the table below.
Video Quality vs. Line Rate

The Collaboration Server Web Client supports monitoring of H.263 4CIF information. The H.245 or SDP tab
includes the additional information.
The creation of a new H.263 4CIF slide is supported in the IVR Service in addition to the current H.263 IVR
slide. If users utilize the default Polycom slides that are delivered with the Collaboration Server, the slide’s
resolution will be as defined in the profile, i.e. SD, HD, CIF, etc.
For more information see “High Resolution Slides” on page 571.

H.263 4CIF Guidelines
● H.263 4CIF is supported with H.323 and SIP connection endpoints.

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● H.263 4CIF is supported in CP mode only.
● Click & View is supported in H.263 4CIF.
● AES encryption is supported with H.263 4CIF.
● H.263 4CIF is supported in recording by the RSS2000 and other recording devices.
● All video layouts are supported in H.263 4CIF, except 1x1 layout. In a 1x1 layout, the resolution will
be CIF.
● For information about Resource Usage see Table 19-7 on page 19-8.
● H.239 is supported in H.263 4CIF and is based on the same bandwidth decision matrix as for HD.

The CP Resolution Decision Matrix
All the CP resolution options and settings are based on a decision matrix which matches video resolutions
to connection line rates, with the aim of providing the best balance between resource usage and video
quality at any given line rate.
The following factors affect the decision matrices:
● The video protocol used: H.264 base Profile or H.264 High Profile. The H.264 High Profile maintains
the Video quality at bit rates that are up to 50% lower than previously required. For example, a 512
kbps call will have the video quality of a 1Mbps HD call while a 1Mbps HD call has higher video quality
at the same (1Mbps) bit rate.
By default, the system shipped with three pre-defined settings of the decision matrix for H.264 Base Profile
and three pre-defined settings of the decision matrix for H.264 High Profile:
● Resource-Quality Balanced (default)
A balance between video quality and resource usage.
● Resource Optimized
System resource usage is optimized by allowing high resolution connections only at high line rates
and may result in lower video resolutions (in comparison to other resolution configurations) for some
line rates. This option allows to save MCU resources and increase the number of participant
connections.
● Video Quality Optimized
Video is optimized through higher resolution connections at lower line rates increasing the resource
usage at lower line rates. This may decrease the number of participant connections.

Video Resource Usage
Video resource usage is dependent on the participant’s line rate, resolution and Video Quality settings.

H.264 Base Profile and High Profile Comparison
The following illustrations show a comparison between the resolutions used at various line rates for H.264
Baseline and the H.264 High Profile Video Quality setting.

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Resolution usage for H.264 High Profile and H.264 Base Profile at various line rates when Resolution
Configuration is set to Resource-Quality Balanced

Resolution usage for H.264 High Profile and H.264 Base Profile for Motion at various line rates when
Resolution Configuration is set to Video Quality Optimized

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Resolution usage for H.264 High Profile and H.264 Base Profile at various line rates when Resolution
Configuration is set to Resource Optimized

Default Minimum Threshold Line Rates and Resource Usage Summary
The following Table summarizes the Default Minimum Threshold Line Rates and Video Resource usage for
each of the pre-defined optimization settings for each Resolution, H.264 Profile, Video Quality setting.
Default Minimum Threshold Line Rates and Video Resource Usage
The table above lists resource consumption for H.264:
• CIF resolution consumes 1 resources.
• 4CIF resolution consumes 1 resources.
• HD720p resolution consumes 2 resources.

Optimization Mode
Resolution

Profile

Balanced
Sharpness

Motion

Resource
Sharpness

Motion

Video Quality
Sharpness

Motion

HD1080p30

Default
kbps

High

1536

4096

1024

Base

4096

4096

1728

HD720p60

Default
kbps

High

1280

1920

832

Base

1920

1920

1280

HD720p30

Default
kbps

High

832

1920

512

Base

1024

1920

832

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Optimization Mode
Resolution

Profile

Balanced
Sharpness

Resource

Motion

Sharpness

Video Quality

Motion

Sharpness

Motion

SD 60

Default
kbps

High

768

1024

512

Base

1024

1024

768

SD 30

Default
kbps

High

256

384

256

Base

256

384

256

CIF 60

Default
kbps

High

256

384

256

Base

384

384

256

CIF 30

Default
kbps

High

64

64

64

64

64

64

Base

64

64

64

64

64

64

Resolution Configuration for CP Conferences
The Resolution Configuration dialog box enables Collaboration Server administrators to override the default
video resolution decision matrix, effectively creating their own decision matrix. The minimum threshold line
rates at which endpoints are connected at the various video resolutions can be optimized by adjusting the
resolution sliders.
System resource usage is also affected by the Resolution Configuration settings. For more information see
Video Resource Usage on page 89 and Default Minimum Threshold Line Rates and Resource Usage
Summary on page 91.

Guidelines
● Resolution Slider settings affect all Continuous Presence (CP) conferences running on the
Collaboration Server. Video Switched conferences are not affected.
● A system restart is not needed after changing the Resolution Slider settings.
● Resolution Slider settings cannot be changed if there are ongoing conferences running on the
Collaboration Server.

Modifying the Resolution Configuration
The Resolution Configuration dialog box is accessed by clicking Setup > Resolution Configuration in the
Collaboration Server Setup menu.
Clicking the Detailed Configuration button toggles the display of the Detailed Configuration pane, which
displays sliders for modifying minimum connection threshold line rates for endpoints that support H.264
Base Profile or High Profile.

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The Detailed Configuration pane can also be opened by clicking the Manual radio button in the Resolution
Configuration pane.

Resolution Configuration - Basic
The Resolution Configuration -Basic dialog box contains the following panes:
● Max CP Resolution Pane

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● Resolution Configuration Pane

Maximum CP Resolution Pane
● The Collaboration Server can be set to one of the following Maximum CP Resolutions:HD 1080p30
● HD 720p30
● SD 30
● CIF 30

Limiting Maximum Resolution
Before a selection is made in this pane, the Maximum CP Resolution of the system is determined by the
MAX_CP_RESOLUTION System Flag.

Resolution Configuration - Detailed
H.264 High Profile allows higher quality video to be transmitted at lower bit rates.
However, setting minimum bit rate thresholds that are lower than the default may affect the video quality of
endpoints that do not support the H.264 High Profile. The Collaboration Server uses two decision matrices
(Base Profile, High Profile) to enable endpoints to connect according to their capabilities.

Resolution Configuration Sliders
The Detailed Configuration dialog box allows the administrator to configure minimum connection threshold
bit rates for endpoints that support H.264 High Profile and those that do not support H.264 High Profile by
using the following slider panes:

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● Base Profile - Endpoints that do not support H.264 High Profile connect at these minimum threshold
bit rates.

● High Profile - Endpoints that support H.264 High Profile connect at these minimum threshold bit rates.

Although the default minimum threshold bit rates provide acceptable video quality, the use of higher
bit rates usually results in better video quality.
These Video Quality settings are selected per conference and are defined in the conference Profile and they
determine the resolution matrix that will be applied globally to all conferences. The resolution matrix is
determined by the resolution configuration and can be viewed in the Resolution Configuration sliders.
System Resource usage is affected by the Resolution Configuration settings.
Example
As shown in following diagram:

● Moving the HD720p30 resolution slider from 1024kbps to 1920kbps increases the minimum
connection threshold line rate for that resolution. Endpoints connecting at line rates between
1024kbps and 1920kbps that would have connected at HD 720p30 resolution will instead connect at
SD 30 resolution. Each of the affected endpoints will connect at lower resolution but will use 1 system
resource instead of 2 system resources.

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Flag Settings
Setting the Maximum CP Resolution for Conferencing
The MAX_CP_RESOLUTION flag value is applied to the system during First-time Power-up and after a
system upgrade. The default value is HD720p30.
All subsequent changes to the Maximum CP Resolution of the system are made by selections in the Max
Resolution pane of the Resolution Configuration dialog box.
The Collaboration Server can be set to one of the following resolutions:
● HD1060p30
● HD720p30
● SD 30
● CIF 30

Minimum Frame Rate Threshold for SD Resolution
The MINIMUM_FRAME_RATE_THRESHOLD_FOR_SD System Flag can be added and set to prevent low
quality, low frame rate video from being sent to endpoints by ensuring that an SD channel is not opened at
frame rates below the specified value. For more information see Modifying System Flags on page 754.

Additional Video Resolutions
The following higher video quality resolutions are available:
● CIF 352 x 288 pixels at 50 fps.
● WCIF 512 x 288 pixels at 50 fps.
● WSD 848 x 480 pixels at 50 fps.
● W4CIF

1024 x 576 pixels at 30 fps.

● HD 720p 1280 x 720 pixels at 30fps.
● HD 1080p 1920 x 1080 pixels at 30 fps.
The video resolution transmitted to any endpoint is determined by the endpoint’s capabilities, the
conference line rate and the Conference Profile’s Motion and Sharpness settings.

w448p Resolution
For improved interoperability with Tandberg MXP 990/3000 endpoints, the appropriate System Flag settings
will force the Collaboration Server to send w448p (768x448 pixels) at 25fps as a replacement resolution for
WSD15 (848x480) and SD15 (720x576 pixels).

Guidelines
● The w448p resolution is supported:
 In CP mode.

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 At conference line rates of 384kbps and 512kbps.
 With H.323 and SIP.
H.323 endpoints must identify themselves as Tandberg MXP during capabilities exchange.
 In all Video Layouts.
 In 1x1 Layout:
 When Video Clarity is Off, the Collaboration Server transmits the same resolution as it
receives.
 When Video Clarity is On, the Collaboration Server changes the transmitted resolution to
w448p.
For more information see the Polycom® RealPresence Collaboration Server 800s/Virtual Edition
Administrator’s Guide, Video Clarity on page 2-9 "Video Clarity” on page 2-7.
● Resource consumption for the w448p resolution is the same as for SD and WSD resolutions.
The following table lists the video outputs from the Collaboration Server to the Tandberg Endpoints for both
16:9 Aspect Ratio when the w448p resolution is enabled.
Video Output to Tandberg Endpoints- Aspect Ratio 16:9

Video Quality
Tandberg

Collaboration
Server

Line
Rate
Kbps

H.323
SIP

Motion

Sharpness

H.323
SIP

Sharpness*

Network
Environment

Sharpness

Resolution

Frame
Rate
fps

Resolution

Frame
Rate fps

Tandberg to
Collaboration Server

Collaboration Server to
Tandberg

384

512x288

30

768x448

25

512

768x448

30

768x448

25

384

1024x576

15

768x448

25

512

1024x576

15

768x448

25

* It is recommend to set the endpoint to Motion to ensure the transmission of the higher frame rates of 25fps/30fps to
the Collaboration Server.
The following table list the video outputs from the Collaboration Server to the Tandberg Endpoints for 4:3
Aspect Ratio when the w448p resolution is enabled.
Video Output to Tandberg Endpoints - Aspect Ratio 4:3

Video Quality
Network
Environment
H.323
SIP
ISDN

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Tandberg

Collaboration
Server

Line
Rate
Kbps

Motion

Sharpness

Resolution

Frame
Rate
fps

Resolution

Frame
Rate
fps

Tandberg to
Collaboration Server

Collaboration Server
to Tandberg

384

576x448 ‡

25

768x448

25

512

576x448 ‡

25

768x448

25

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Video Quality
Network
Environment
H.323
SIP
ISDN

Tandberg

Collaboration
Server

Line
Rate
Kbps

Sharpness*

Sharpness

Resolution

Frame
Rate
fps

Resolution

Frame
Rate
fps

Tandberg to
Collaboration Server

Collaboration Server
to Tandberg

384

4CIF

15

768x448

25

512

4CIF

15

768x448

25

* It is recommend to set the endpoint to Motion to ensure the transmission of the higher frame rates of 25fps/30fps to
the Collaboration Server.
‡ MXP 990/3000 endpoints transmit 576x448 pixels. Other MXP endpoints may transmit other resolutions
eg. CIF.

Content
Sharing and receiving Content is supported.
Bandwidth allocated to the Content channel during Content sharing may cause the video resolution to be
decreased as from w448p to w288p.
When Content sharing stops and the full bandwidth becomes available, video resumes at the previous
w448p resolution.
For more information see the Polycom® RealPresence Collaboration Server 800s/Virtual Edition
Administrator’s Guide, "H.239 Protocol” on page 4-2"H.239 Protocol” on page 4-2.

Packet Loss Compensation
If there is Packet Loss in the network and Dynamic Bandwidth Allocation (DBA) is activated, allocating
bandwidth for Lost Packet Recovery, video resolution decreases from w448p
to w288p.
When Packet Loss ceases and DBA no longer needs to allocate bandwidth for Lost Packet Recovery, the
full bandwidth becomes available and video resumes at the previous w448p resolution.
For more information see the Polycom® RealPresence Collaboration Server 800s/Virtual Edition
Administrator’s Guide, Packet Loss Compensation (LPR and DBA) AVC CP Conferences.

Enabling Support of the w448p Resolution
w448p resolution support for Tandberg endpoints requires setting of the following entities:
● Tandberg endpoint
● Collaboration Server flags
● Collaboration Server Conference Profile

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Collaboration Server System Flag Settings
● On the Collaboration Server, the Video Quality field in the New Profile - Video Quality dialog box must
be set to Sharpness.
For more information see Polycom® RealPresence Collaboration Server 800s/Virtual Edition Administrator’s
Guide, "Defining New Profiles” on page 2-19"Defining a CP Conference Profile” on page 2-11.

Additional Intermediate Video Resolutions
Two higher quality, intermediate video resolutions replace the transmission of CIF (352 x 288 pixels) or SIF
(352 x 240 pixels) resolutions to endpoints that have capabilities between:
● CIF (352 x 288 pixels) and 4CIF (704 x 576 pixels) – the resolution transmitted to these endpoints is
432 x 336 pixels.
● SIF (352 x 240 pixels) and 4SIF (704 x 480 pixels) – the resolution transmitted to these endpoints is
480 x 352 pixels.
The frame rates (depending on the endpoint’s capability) for both intermediate resolutions are 25 or 30 fps.

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Sharing Content During Conferences
Content such as graphics, presentations, documents, or live video can be shared with conference
participants.
Content sharing architecture is comprised of various aspects:
● Content Control Protocols - H.239 (for H.323), BFCP (for SIP), or People+Content (Polycom’s
protocol used for CP conferences prior to H.239 creation)
● Content Media Protocols - H.263 (AVC only), H.264 (all conferencing modes), or TIP (Cicso’s
proprietary protocol for TelePresence endpoints).
● Content Transmission Methods - Content Video Switching and Multiple Content Resolutions.
● Sharing Content in Cascading Environments

Content Control Protocols
Endpoints wishing to share content, request the Content Token from the MCU (in cascaded environments,
the Master MCU). The MCU uses the control protocol to grant the Content Token to the requesting endpoint
(unless in Exclusive Content Mode).
Endpoints incapable of sharing content using one of the supported control protocols, can view content via
the people video layout (see Sending Content to Legacy Endpoints (CP Only)).

Guidelines for Controlling Content Protocol
● Only the Content Token owner may send content.
● Content Token ownership is valid until:
 A new endpoint requests token ownership (unless in Exclusive Content Mode).
 The owner relinquishes it.
 The Content Token owner endpoint disconnects from the conference.
 It is cancelled by the MCU user

Supported Content Control Protocols
Polycom supports content sharing using one of the following content control protocols:
● H.239 - For H.323 participants
● BFCP - For SIP participants (over TCP or UDP)
● People+Content - Polycom’s proprietary content control protocol; for H.323 participants
● TIP Auto-Collaboration (CISCO TIP participants). (see Prefer TIP) - TIP Auto-Collaboration for H.264
content sharing
Endpoints supporting the content control protocols above can share content within the same conference.
Video endpoints not supporting one of the content control protocols above, can view content on the people
video layout if the conference is set to Send Content to Legacy Endpoints (see Sending Content to Legacy
Endpoints (CP Only)).
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Note that since TIP content implies using H.264 content media protocol, endpoint supporting only
H.263 are considered Legacy content endpoints (see Sending Content to Legacy Endpoints (CP
Only)) when TIP content is used.

Content Sharing Using H.239 Protocol
This protocol is used by H.323 endpoints.
The H.239 protocol allows compliant endpoints to share content stream simultaneously with video.
Cascaded links declare H.239 capabilities, and are supported in Star and MIH cascading topologies. For
more details, see Cascading Conferences - H.239-enabled MIH Topology.
Endpoints may not send content while connecting to an Entry Queue.

Content Sharing Using BFCP Protocol
This protocol is used by SIP endpoints.
The MCU supports BFCP over either TCP or UDP, which enables the MCU to share content with both SIP
client types.

Guidelines for Using SIP BFCP Content
For SIP clients supporting BFCP/TCP or BFCP/UDP:
● BFCP content is not supported over SIP links (in Gateway and cascading scenarios). Therefore, in
cascading environment, the cascaded link must be defined as H.323 for content to be shared.
● BFCP/UDP is supported in both IPv4 and IPv6 addressing modes. BFCP/TCP is supported only in
IPv4 addressing mode.
● Note that Lync endpoints, though considered as SIP endpoints, do not use BFCP to share content,
and use a Microsoft proprietary protocol for that purpose. There are two options to bypass that
limitation:
 Polycom CSS plug-in (see Content Sharing via Polycom CSS Plug-in for Microsoft Lync Clients).
This is the preferred option, and is used whenever the CSS plug-in is currently used.
 Treating Lync endpoints as Legacy, thus viewing content via the people video layout (see Sending
Content to Legacy Endpoints (CP Only)).
● BFCP utilizes an unsecured channel (port 60002/TCP) even when SIP TLS is enabled. If security is
of higher priority than SIP content sharing, SIP/BFCP can be disabled by manually adding the
ENABLE_SIP_PEOPLE_ PLUS_CONTENT system flag, and setting its value to NO.
● SIP and BFCP capabilities are by default declared to all endpoints.
Capabilities declaration is controlled by the ENABLE_SIP_PPC_FOR_ALL_USER_AGENT system
flag, whose default value is YES, meaning BFCP capability is declared to all vendors’ endpoints.
When set to NO, the MCU declares SIP over BFCP capabilities only to Polycom and Avaya
endpoints. Note that a SIP proxy might remove the agent information, thus preventing the capability
declaration to Polycom and Avaya endpoints as well.
● Set ENABLE_FLOW_CONTROL_REINVITE system flag to NO when SIP BFCP is enabled.
● If the system flags mentioned above do not exist in the system, they must be manually added (see
Modifying System Flags).
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● Due to UC-APL requirement, BFCP over TCP is not supported in Ultra Secure Mode (Collaboration
Server 1500/1800/2000/4000).

Content Sharing via Polycom CSS Plug-in for Microsoft Lync Clients
From version 8.1, Polycom CSS (Content Sharing Suite) plug-in for Lync clients allows Lync clients to
receive and send content via SIP BFCP, without having to use the people video layout.
The CSS plug-in invokes a separate call for content with or without video. This call may be invoked per a
Lync client.

BFCP support in dial-out Connections
For SIP dial-out clients supporting both TCP and UDP, the preferred protocol is BFCP/UDP. However, this
preference can be modified, by adding the SIP_BFCP_DIAL_OUT_MODE system flag and modifying its
value to TCP (see Manually Adding and Deleting System Flags).
The Collaboration Server’s content sharing, as determined by the system flags settings and SIP client
capabilities, is summarized in the following table:
System Flag - SIP_BFCP_DIAL_OUT_MODE
Flag Value
AUTO (Default)

SIP Client: BFCP Support
UDP
BFCP/UDP selected as
content sharing protocol

UDP
TCP

TCP
BFCP/TCP selected as
content sharing protocol

UDP and TCP
BFCP/UDP selected as
content sharing protocol

Cannot share content
Cannot share content

BFCP/TCP selected as content sharing protocol

BFCP support in dial-in Connections
● The MCU shares content with dial-in SIP clients according to their preferred BFCP protocol.
● SIP clients connected as audio-only cannot share content.

Content Sharing Using People+Content Protocol
People+Content utilizes a different content control protocol, and is Polycom’s proprietary protocol used prior
to H.239.
This protocol is supported in CP conferences, and is applicable for H.323 endpoints.

Guidelines for Content Sharing Using People+Content Protocol
● If an endpoint supports both H.239 and People+Content protocols, H.239 is selected as the preferred
communication protocol.
● H.263 and H.263 and H.264 Auto-selection are the only supported content media protocols usage
modes (see MCU Usage Modes of Content Protocols).
● People+Content is enabled by default. It can be disabled for all conferences and endpoints by
manually adding the ENABLE_EPC System Flag, and setting its value to NO (default value is YES).

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● Endpoints supporting People+Content (for example, FX endpoints) may require a different signaling
protocol. For these endpoints, manually add the System Flag CS_ENABLE_EPC, and set its value
to YES (default value is NO).

Content Media Protocols
The RealPresence Collaboration Server transmits content using the following content media protocols:
● H.263 (Annex T) - Base profile.
● H.264 - Base profile.
For single MCU conferences, the MCU determines the media protocol by applying the Highest
Common principle (see Highest Common).
.

H.264 Supported Resolutions for AVC (non-TIP) Conferences
Conference
Resolution

Soft MCU

Multiple Content
Resolution

720p5





720p30





1080p15





● TIP - Cisco’s proprietary protocol for TelePresence endpoints. The supported TIP content rate is
XGA, 5 fps, 512 Kbps, base profile (see Appendix I - Polycom Open Collaboration Network (POCN)).
LPR has no effect on content rate, though due to the increased number of frames, video rate is
decreased.

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Content Transmission Methods
There are two methods used for content transmission:
● Content Video Switching
● Multiple Content Resolution

Content Video Switching
When using this method, content sent by endpoints is transmitted as-is to all content-capable participants.
The content rate is determined using one of the following methods:
● Highest Common
● Fixed Rate

Highest Common
This method is applied for single MCU conferences.

The Highest Common method is applied in a few levels:
● Within the same protocol - The highest common content rate
● Between protocols - H.263 and H.264
In this method, the content rate is negotiated to highest common capabilities supported by the endpoints
connected to the conference. Therefore, if the conference includes participants supporting both lower and
higher content capabilities, the content is sent to all endpoints using the lowest endpoint capabilities.
Note, that when content is currently shared using H.263, the MCU determines the content rate, and the
endpoint sharing the content determines the resolution and frame rate. However, when content is shared
using H.264, the MCU determines content rate, resolution and frame rate (depending on endpoints
capabilities and conference settings).
When a new endpoint with lower content capabilities joins while content is shared, content sharing
parameters are downgraded to this endpoint capabilities:
● If downgrade affects the protocol (from H.264 to H.263), content sharing is terminated, protocol
automatically downgrades to H.263, or in H.264, and content sharing should be manually resumed.
● If downgrade affects only the content rate, resolution or frame rate (in H.264), the relevant content
parameters automatically downgrade during content sharing.
Once this endpoint leaves the conference during content sharing, no change occurs in the protocol or
content rate. In H.264 content, resolution or frame rate may automatically upgrade.

Content Setting Highest Common Calculation
For Graphics, Hi-res Graphics and Live Video (see Content Settings), the highest common content is
calculated for the conference each time an endpoint connects. Highest Common content bit rate is
calculated using line rate (the only factor for H.263), resolution, and frame rate (additional factors for H.264).
Therefore, if an endpoint connects to an ongoing conference at lower values of these parameters, content
parameters are re-calculated and decreased accordingly.

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During content sharing, the MCU does not permit endpoints to increase their content bit rate, only their
content resolution.
For more information, see H.263 Content Rate Table and H.264 Highest Common Content Rates Table. For
information on minimum content rates as determined by system flags, see H.264 HD System Flags.

Fixed Rate
This method must be used for cascaded and SVC-enabled conferences.

In this method the content rate is fixed, and endpoints not supporting this rate do not receive content over
the content channel. Endpoints whose capabilities are too low, can only view content over the people video
layout (see Sending Content to Legacy Endpoints (CP Only)).
For SVC-enabled conferences, the MCU uses H.264, and 128 Kbps as the fixed content rate.
For H.263 cascaded conferences, the MCU automatically uses a fixed rate (once a cascaded link is
detected) according to the content rates described in Highest Common and Fixed Content Bit Rate
Allocation for H.263.
For H.264 cascaded conferences, the MCU uses a fixed rate according to the content rates described in
H.264 Cascade and SVC Optimized (Fixed) Content Rates Table.

Multiple Content Resolutions
This option is available only in AVC CP conferencing mode.

In this method, the content is shared in multiple streams, one for each video protocol: H.263 (optional) and
H.264HD (mandatory). Separate video resources are used for processing the content for each of the
required content streams. The MCU then applies the Highest-Common principle for sharing content with
each group of endpoints. In cascading conferences, an additional resource is dedicated for the cascading
link(s), with a fixed (H.264 Cascade and SVC Optimized) rate shared over the cascaded link(s).
This allows endpoints with different capabilities to connect/disconnect without having to repeatedly restart
content sharing process.
Endpoints not supporting the content capabilities set for the conference, receive the content over the people
video layout (see Sending Content to Legacy Endpoints (CP Only)).
Multiple Content Resolutions option is not supported in Ultra Secure Mode.

Guidelines for Sharing Contents using Multiple Content Resolutions
● Multiple Content Resolutions is supported only in CP conferences.
● Content is always provided to H.264 HD endpoints.

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● The Send Content to Legacy Endpoints option is set, and cannot be modified.
● Additional resources are allocated to the conference (in addition to resources for the conference
participants) for content processing:
 Resources are allocated only upon content sharing beginning, but once allocated, they remain
unchanged until conference ends.
 If the conference configuration (line rate and content settings) yields HD720p30 as the highest
content resolution, 1.5 HD video resources are allocated to the conference, whereas a highest
resolution of HD1080p15, increases the allocated video resources to 2 HD.
● An additional content stream can be sent to H.263 endpoints, in which case a separate resource is
allocated. H.263 supported resolutions are CIF, 4CIF, XGA.
● Content can be sent to cascaded links, in which case an additional video resource is allocated. The
links in both MCUs must use the same content parameters (meaning content rate, resolution, and
frame rate).
● When resources are insufficient, Multiple Content Resolutions mode is disabled, and content sharing
reverts to Content Video Switching mode.
 If H.264 Cascade was selected for the conference, the conference uses H.264 Cascade and
SVC Optimized as the content protocol.
 If H.264 Cascade was not selected for the conference, H.264 HD protocol is used.
● H.264 endpoints with no HD or H.263 capabilities, view content over the video layout (see Sending
Content to Legacy Endpoints (CP Only)).
● TIP endpoints cannot share content, only view it over the people video layout.
● When AS SIP is enabled for the conference, Multiple Resolutions is selected and cannot be
modified, H.264 is enabled, and both H.263 and H.264 Cascade are disabled. Resources are
allocated on conference beginning. All this enables AS SIP performance in high-traffic environments.

Content Settings
The Content channel can transmit in one of the following modes:
● Graphics – For standard graphics. This is the default mode in AVC conferences and the only
supported mode for SVC enabled conferences.
● Hi-res Graphics (AVC Only) – Requiring a higher bit rate to increase display quality or highly detailed
graphics.
● Live Video (AVC Only) – Highest bit rate, for video clips or live video display.
● Customized Content Rate (AVC Only) - Allowing manual definition of the Conference Content Rate.
Content sharing consumes a quota of the video rate, depending on the content required quality; the better
the content required quality (such as in Live Video) the less bit rate remains for the video.
For each of the settings, the MCU allocates an approximate percentage (determined by the table in Content
Sharing Reference Tables) of the conference video capability towards content sharing: for Graphics - 33%,
for Hi-res Graphics - 50%, and for Live Video - 66%. However, in actuality this percentage might be lower,
since the endpoint capabilities are also taken into account when making this calculation.

Customized Content Rate in AVC CP Conferences
Customized Content Rate functionality may be implemented (see Selecting a Customized Content Rate in
AVC CP Conferences) when the content rate automatically calculated by the MCU is unsuitable.
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For example, in a cascaded environment, the conference rate must be identical for all links. Yet, capabilities
may vary widely between the cascading conferences (such as one conference supporting 2 Mbps, while the
other - 512kbps).
The rates listed by the MCU are up to 66% of the conference maximum line rate.
Customized Content rate list comprised of a single value of 0, means the conference bit rate capability is
too low to share content.
Selecting a line rate too low for the selected Customized Content Rate, results in an error message display.

MCU Usage Modes of Content Protocols
Depending on the endpoints capabilities, you can determine the content sharing experience by selecting the
appropriate protocol and system behavior from the Content Protocol list:
● H.263 - AVC Only
● H.263 & H.264 Auto Selection - AVC Only
● H.264 Cascade and SVC Optimized - All conferencing modes
● H.264 HD - AVC Only
Endpoints without the content capabilities matching the conference content sharing requirements, can
connect to the conference but cannot share or view content in the content channel. Depending on Legacy
definitions, they can view content in the people video layout (see Sending Content to Legacy Endpoints (CP
Only)).

H.263 (AVC CP Conferences)
In this mode, all endpoints share content using H.263 protocol. Select this option when either most
endpoints support H.2634, or to share content over a cascading conference, in which case, the cascading
link should be created prior to participants joining the conference.
For single MCU scenario, the Highest Common principle (see Highest Common) is applied to determine
content parameters. In cascading environments, a fixed rate is used (see Fixed Rate).

H.263 & H.264 Auto Selection (AVC Conferences)
Select this option to share content using a mix of H.263 and H.264 capable endpoints. Until version 7.6
(including), this option is named Up to H.264.
For single MCU scenario, the Highest Common principle (see Highest Common) is applied to determine
content parameters. In cascading environments, an H.263 fixed rate is used (see Fixed Rate) regardless of
endpoint capabilities.

H.264 Cascade and SVC Optimized
This content sharing option applies for H.264 content media protocol, and provides fixed content rate (see
Fixed Rate) and resolution according to the conference line rate. It must be used for Cascading or
SVC-enabled conferences.
The H.264 Cascade and SVC Optimized option maintains content quality, and prevents content refreshes
upon participants connect/disconnect from the conference.
In cascading environments using this option, the cascade link signaling must be H.323.
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● In AVC conferences - The H.264 Cascade and SVC Optimized option must be used for AVC
cascaded conferences sharing H.264 content. The selected content is defined by the conference
parameters:
 Line rate
 Content settings (Graphics, Hi-res Graphics, or Live Video)
 Resolution configuration
● In SVC enabled conferences - The H.264 Cascade and SVC Optimized option must be used for
SVC enabled conferences sharing content over H.264.The used content rate is 128Kbps (fixed) and
720p5 resolution. Endpoints not supporting the required content parameters (content line rate and
resolution) cannot share or view content.
For more information see H.264 Cascade and SVC Optimized (Fixed) Content Rates Table.

H.264 HD (AVC CP default)
H.264 HD may be used for content sharing only for a single MCU, but not in cascading conferences.

H.264 Content Sharing Properties
The H.264 HD option should be selected if most endpoints in the conference support H.264 to ensure high
quality content.
When this protocol option is selected, content minimal bit rate is determined. For more information, see
Setting the Minimum Content Rate for Each Content Quality Setting for H.264 HD.
Content parameters are determined by the Collaboration Server applying the Highest Common principle
(see Highest Common), for H.264 media content protocol only.

Guidelines for Sharing Content Using H.264 HD
● Only endpoints supporting HD H.264 content (at least HD720p5) can share content.
● This option is not available in SVC-enabled conferencing modes.

Setting the Minimum Content Rate for Each Content Quality Setting for H.264 HD
System flags determine the minimum line rate required for endpoints to share H.264 high quality content for
each of the Content Settings: Graphics, Hi Resolution Graphics and Live Video.
H.264 HD System Flags
Content Settings

Flag Name

Range

Default

Graphics

H264_HD_GRAPHICS_MIN_CONTENT_RATE

0-1536

128

Hi Resolution Graphics

H264_HD_HIGHRES_MIN_CONTENT_RATE

0-1536

256

Live Video

H264_HD_LIVEVIDEO_MIN_CONTENT_RATE

0-1536

384

To change the system flag value, the flag must be first manually added to the System Configuration (see
Modifying System Flags.
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Content Sharing Related Issues
Sharing Content in Cascaded Environments
In cascaded environments, content must be shared using the same content rate, and in H.264 conferences,
the same resolution and frame rate as well.
Sharing content has different constraints and guidelines over each of the signaling protocols:
● Over H.323 cascaded links  Fixed content rate is used.
 In cascading environments with non-Polycom MCUs, the Polycom MCU must be defined as
Master
 For H.263 cascading links:
 The cascading link must be created before connecting the participants.
 The cascaded link’s Master/Slave role must be determined, in topologies including more than
two MCUs.
 For H.264 cascading links, the following should be determined in advance:
 The H.264 Cascade and SVC Optimized content protocol must be selected.
 The cascaded link’s Master/Slave role must be configured for the link’s participants.
● Over SIP - No content sharing is supported over SIP cascaded links.
Gateway calls are implemented as cascaded links, typically SIP, in which case, content
cannot be shared via the content channel. It may be viewed over the people video layout,
depending on Legacy configuration (see Sending Content to Legacy Endpoints (CP Only)).

Sending Content to Legacy Endpoints (CP Only)
The Collaboration Server can be configured to send content to endpoints not supporting the conference
content parameters (legacy endpoints) over the people video layout, thus allowing the participants to view
content. However, these endpoints cannot share content.

Guidelines for Sending Content to Legacy Endpoints
● A separate HD video resource is allocated to the conference for content sending to legacy endpoints.
Allocation is performed only once a legacy content endpoint is connected to the conference, and a
content session is initiated and transmitted via the people video layout. Once the resource is
allocated, it remains allocated to the conference until its end.
If the system cannot allocate the resource required for sending the content, the conference status
changes to Content Resource Deficiency, and content cannot be sent to the legacy endpoints.
● Endpoints receiving content via the people video layout, use the same video protocol and resolution
they use for receiving video.
● Content cannot be sent to Legacy endpoints when Same Layout mode is selected for the conference.
● When content is transmitted, endpoints’ Site Names cannot be viewed.

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● Content can be sent to Legacy endpoints in gateway calls, depending on content configuration on
Master/Slave MCUs (for example, if the gateway conference is using H.263 content media protocol,
and the conference MCU is using H.264 content media protocol).
● Content becomes unavailable to Legacy endpoints moved to an Operator conference.
● A Polycom FX endpoint dialing in to a Collaboration Server receives content using People+Content.
However, an FX endpoint dialed out from a Collaboration Server with receives content via the people
video layout using People+Content only if Send Content to Legacy Endpoints is enabled in the
Conference Profile.

Content Display on Legacy Endpoints
When content is sent to content legacy endpoints, their video layout automatically changes to the content
layout defined by the system flag LEGACY_EP_CONTENT_DEFAULT_LAYOUT, and the content is shown
in the larger/top-left cell. The video layouts of the other conference participants do not change.
The switch to the content layout occurs in Auto Layout, Presentation Mode, Lecture Mode, and when a
layout is selected for the conference.
In Lecture Mode, when content is sent to legacy endpoints, switching to the content layout, results in the
content shown in the lecturer/speaker window, while the lecturer is shown in the second window. If the layout
contains more than two windows, all other windows are empty. The non-lecturer legacy content participants
see the lecturer in full screen.
The LEGACY_EP_CONTENT_DEFAULT_LAYOUT flag’s default is a layout of 1+4, where the content is
shown in the larger window, and the conference participants are shown in the smaller windows. This default
can be changed (see Changing the Default Layout for Content Display on Legacy Endpoints).
When content is stopped, the layout of the legacy participants terminates as well.
Legacy participants can change their layout using Click&View. In such a case, the content is forced to the
larger/top-left cell.
The Collaboration Server user can also change the layout for the legacy content endpoints participants
(selecting personal layout).
When forcing a video participant to the Content window (instead of Content), the Content display can be
restored only by selecting any other video layout.

Sending Content to Legacy Endpoints in Telepresence Mode
To ignore personal layouts during Telepresence conferences (while working with MLA), set the value of the
flag FORCE_LEGACY_EP_CONTENT_LAYOUT_ON_TELEPRESENCE to YES.
If the layout for displaying content in Legacy endpoints include multiples cells, MCU may populate
Telepresence room streams sources in remote cells.

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Exclusive Content Mode
In this mode, the MCU prevents participants other than the current content sharer, from sharing content.
Exclusive content may be limited to the lecturer, as described below.
To modify exclusive content mode by granting (or cancelling) of token ownership, see Giving and Cancelling
Token Ownership (AVC Participants)).

Guidelines for Sharing Content in Exclusive Content Mode
● Exclusive Content Mode is available in all conferencing modes.
● Exclusive Content Mode is enabled or disabled (system default) by a check box in the Conference
Profile - Advanced tab, or during an ongoing conference using the Conference Properties Advanced tab.

● In Exclusive Content Mode, when the RESTRICT_CONTENT_BROADCAST_TO_LECTURER
system flag is set to:
 NO - The first participant to send content becomes the Content Token holder, and releasing the
Content Token allows other participants to acquire the token, and begin transmitting content.
 YES - Only the designated Lecturer can be the Content Token holder.
● The Exclusive Content Mode check box replaces the EXCLUSIVE_CONTENT_ MODE system flag
used in previous versions to control exclusive content mode for the system.
● In Exclusive Content Mode, an endpoint attempting to send content, immediately after another
endpoint starts sending content, results with a momentary interruption (slide), before resuming
normal content stream.

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Forcing Other Content Capabilities
The H239_FORCE_CAPABILITIES system flag allows additional control of content sharing:
● When set to NO (default) - The MCU merely verifies the endpoint supports the content protocols:
H.263 or H.264.
● When set to YES - The MCU verifies frame rate, bit rate, resolution, annexes, and all other
parameters of content as declared by an endpoint during the capabilities negotiation phase. If the
endpoint does not support the content capabilities of the MCU, the participant cannot share content
over a dedicated content channel.

Managing Noisy Content Connections
The system can identify participants sending frequent content display refresh requests (usually as a result
of a problematic network connection), which cause frequent refreshing of content display, and degrading of
viewing quality.
When the system identifies such “noisy” participants, it automatically suspends these requests to avoid
affecting the quality of the content viewed by the other conference participants. This process is controlled
by the following system flags:
● MAX_INTRA_REQUESTS_PER_INTERVAL_CONTENT The maximum number of refresh (intra) requests per 10-second intervals allowed for an endpoint.
Beyond that number, content sent by this participant is identified as “noisy”, and its refresh requests
are suspended.
Default setting: 3
● MAX_INTRA_SUPPRESSION_DURATION_IN_SECONDS_CONTENT The duration, in seconds, for ignoring the participant’s content display refresh requests.
Default setting: 10
● CONTENT_SPEAKER_INTRA_SUPPRESSION_IN_SECONDS The interval, in seconds, between content refresh (intra) requests sent from the MCU to the content
sender due to refresh requests initiated by other conference participants. Additional refresh requests
received within that interval are deferred to the next interval.
Default setting: 5

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Useful Procedures in Content Sharing
For all MCUs, content sharing parameters are defined in the conference profile Video Quality dialog box.

Defining Content Sharing Parameters for a Conference
For RealPresence Collaboration Server VE, the available content options change according to the selected
conferencing mode.

To set the content sharing parameters:
» In the Content Video Definition section, set the values for the Content Settings and Protocol as
follows:
Content sharing Options
Field

Description

Content Settings

Select the transmission mode for the Content channel:
• Graphics — Basic mode, intended for normal graphics
• Hi-res Graphics (AVC CP Only) — A higher bit rate intended for high resolution
graphic display
• Live Video (AVC CP Only) — Content channel displays live video
• Customized Content Rate (AVC CP Only) - Manual definition of the Conference
Content Rate, mainly for cascading conferences.
For a description of each of these options, see Content Settings.

AS-SIP Content

AS-SIP is a SIP implementation utilizing SIP’s built-in security features.
When selected, content is shared using the Multiple Resolutions mode, and is not
supported in any other content sharing mode.

Multiple Resolutions

Select this check box to enable Multiple Content Resolutions mode, for both H.263
and H.264 content protocols.
When enabled, H.264 is always selected and cannot be modified.

(CP conferencing
mode only)

Optional. Select additional protocols:
• H.263 - If the conference includes endpoints with H.263 capabilities.
• H.264 Cascade - If the conference includes cascading links and you want to
define the video settings for content sharing.
For more information, see Sharing Content Using Multiple Content Resolutions
Mode.
Content Protocol

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•

H.263 (AVC Only) - Content is shared using the H.263 protocol.
H.263 & H.264 Auto Selection (AVC Only) - Content is shared according to
conference participants capabilities.
• H.264 HD (AVC Only, default) - Content is shared using the H.264 HD protocol.
• H.264 Cascade and SVC Optimized - Content is shared using the H.264 content
protocol (fixed), and optimized for use in SVC only and cascaded conferences.
For a detailed description of each of these settings, see Content Media Protocols.

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Content sharing Options
Field

Description

Content Resolution

Select a Content Resolution from the pull-down menu.
The Content Resolutions available for selection are dependent on the content
sharing mode (Highest Common or Multiple Content Resolutions), Line Rate and
Content Settings selected for the conference.
For a full list of Content Resolutions see H.264 Supported Resolutions for AVC
(non-TIP) Conferences.
Note: This field is displayed only when H.264 Cascade and SVC Optimized is
selected, and is enabled for selection in CP conferences (AVC CP) Only. This option
is disabled in SVC conferences.

Send Content to
Legacy Endpoints

When enabled (default), content is sent to H.323/SIP (Collaboration Server VE)
endpoints not supporting MCU content control protocol (legacy endpoints) over the
people video layout (see Sending Content to Legacy Endpoints (CP Only)).

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H.264 Cascade and SVC Optimized Content Sharing in AVC CP
Conferences
When H.264 Cascade and SVC Optimized is selected in AVC CP conference as the Content Protocol,
an additional field, Content Resolution is displayed in the Content Video Definition pane.
The Content Resolution value determines the fixed resolution and frame rate to be used for content
sharing in cascaded conferences.
The Content Resolutions available for selection are dependent on the Line Rate and Content Settings
selected for the conference.

The Content Rate drop-down menu list depends on Customized Content Rate being selected as the
Content Setting, and cannot exceed 66% of the conference line rate capability.

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Selecting a Customized Content Rate in AVC CP Conferences
To Select the Customized Content Rate:
1 In the Content Settings list, select Customized Content Rate.
When selected, a drop-down menu of the available conference content rates is displayed. These
content rates vary according to the selected conference Line Rate.

2 Select the required content rate.
If Customized Content Rate is already selected along with the content rate, and you attempt to modify
the conference line rate to a value that does not support the selected customized content rate, an
error message is displayed.
Modify the Content Line Rate or Conference Line Rate, or modify the Content Setting.

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Sharing Content in Multiple Content Resolutions Mode
For information on this content sharing mode see Multiple Content Resolutions.
The Multiple Content Resolutions mode can be modified in the conference Profile, in the Video Quality Tab.

To enable Multiple Content Resolutions:
1 If the Conference Mode is not CP (Continuous Presence), select CP.

2 Select the Multiple Resolutions check box.
By default, H.264 is always selected, and cannot be modified.
3 Select additional protocols:
 H.263 - If the conference includes H.263 (only) capable endpoints.
 H.264 Cascade - If the conference includes cascading links.
If H.264 Cascade is selected, select the Content Resolution.
4 Click OK.

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Changing the Default Layout for Content Display on Legacy Endpoints
The default layout used for displaying content for legacy endpoints, is defined by the system flag
LEGACY_EP_CONTENT_DEFAULT_LAYOUT.
The configured default layout is 1+4 (
CP_LAYOUT_1P4VER). You can change the default layout
configuration by modifying this flag value.

To modify System Flags:
1 On the Collaboration Server menu, click Setup > System Configuration.
The System Flags dialog box opens.
2 In the MCMS_PARAMETERS tab, double-click the LEGACY_EP_CONTENT_DEFAULT_LAYOUT
flag.
The Edit Flag dialog box is displayed.
3 In the Value field, modify the flag value to the desired layout (see Legacy Content Endpoint Default
Layouts Table).
4 Click OK.
The flag is updated in the MCMS_PARAMETERS list.
5 Click OK.
6 For flag changes (including deletion) to take effect, reset the MCU (see Resetting the RMX).

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Giving and Cancelling Token Ownership (AVC Participants)
For information on exclusive content ownership, see Exclusive Content Mode.

To give token ownership:
1 In the Participants list, right-click the AVC-enabled endpoint you wish to define as the exclusive
Content Token owner.

2 Select Change To Content Token Owner in the drop-down menu.
The endpoint receives exclusive ownership of the Content Token, and an indication icon is displayed
in the Role column of the participant’s entry in the Participants list.

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To cancel token ownership:
1 In the Participants list, right-click the endpoint that currently has Content Token ownership.

2 Select Cancel Content Token Owner in the drop-down menu.
Content Token ownership is cancelled for the endpoint.

Stopping a Content Session
In some cases, when a participant ends the content sharing session by his/her endpoint, the content token
is not released, preventing other participants from sharing content.
The Collaboration Server user can withdraw the content token from the current holder, and return it to the
MCU for assignment to other endpoints.

To end the current Content session:
» In the Conferences list pane, right-click the conference icon, and select Abort H.239 Session.

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Content Sharing Reference Tables
Resolutions and Content Rate Reference Tables
The values in the tables below indicate the maximum negotiated content rate (for both H.263 and
H.264) and resolution/frame rate (only for H.264).
When using Highest Common, endpoints may lower the content parameters, whereas when using
fixed rate, endpoints must comply with the content parameters as determined by the MCU.
The actual content parameters used by the content sharing endpoint are determined by that endpoint,
and may be lower.

H.263 Content Rate Table
The table below describes the content rates for both Highest Common and Fixed content bit rates.
Highest Common and Fixed Content Bit Rate Allocation for H.263
1152

Content
Settings
/ MCU

64

128

256

768

1280

1536

96

192

320

384

512

832

1024

1472

1728

1920

64

64

128

128

256

256

384

512

512

64

128

192

256

384

512

512

768

1024

64

128

256

384

512

512

768

1024

1280

Graphics 33%
RPCS VE

0

Hi-res Graphics 50%
RPCS VE

0

Live Video 66%
RPCS VE

0

H.264 Resolution per Content Rate Table
The table below describes the resolution as negotiated by the MCU according to the content rate.
Maximum Negotiated Resolution and Frame Rate per Content Rate for H.264 Base Profile
Bit Rate Allocated to Content
Channel (Kbps)

Maximum Negotiated Content
Resolution

Frames/Second

64-512

H.264 HD720

5

512-768

H.264 HD720

30

768-1280

H.264 HD1080

15

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H.264 Highest Common Content Rates Table
The table below summarizes the Highest Common maximum content rates as negotiated by the MCU. This
table applies only to single MCU (non-cascading) and non-SVC enabled conferences.
The values in the table below are inapplicable if lower than those of the respective system flags (see
Setting the Minimum Content Rate for Each Content Quality Setting for H.264 HD).

Highest Common Content Bit Rate for H.264 Base Profile
1152

Content
Settings
/ MCU

64

128

256

768

1280

1536

96

192

320

384

512

832

1024

1472

1728

1920

64

64

128

128

256

256

384

512

512

64

128

192

256

384

512

512

768

1024

64

128

256

384

512

512

768

1024

1280

Graphics 33%
RPCS VE

Hi-res Graphics 50%
RPCS VE

Live Video 66%
RPCS VE

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H.264 Cascade and SVC Optimized (Fixed) Content Rates Table
The table below summarizes the content rates as they are determined according to resolutions and
conference line rates.
For information on the supported resolutions for each of the MCU types see H.264 Resolution per Content
Rate Table.
Highest Common Content Bit Rate for H.264 Base Profile
1152

Content
Settings
/ MCU

64

128

256

768

1280

1536

96

192

320

384

512

832

1024

1472

1728

1920

64

64

128

128

256

256

384

512

512

64

128

192

256

384

512

512

768

768

64

128

256

384

512

512

768

1024

1280

Graphics 33%
RPCS VE

Hi-res Graphics 50%
RPCS VE

Live Video 66%
RPCS VE

Bit Rate of Content Channel per Line Rate for H.264 Cascade and SVC Optimized for Base Profile
1472

Cascade
Resolution

64

128

256

832

1536

96

192

320

384

512

768

1024

1152

1280

1728

1920

64

64

128

128

256

256

256

256

256

256

512

512

Graphics 33%
720p5
720p30
1080p15

768

Hi-res Graphics 50%
720p5

64

128

192

256

384

384

720p30

384

512

512

512

512

512

512

512

768

768

1080p15

Live Video 66%
720p5
720p30
1080p15

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128

256

384

512

512

768

768

768

768

512

768

768

768

768

768

768

768

768

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Legacy Content Endpoint Default Layouts Table
The table below describes the supported Legacy content layouts, and their corresponding
LEGACY_EP_CONTENT_DEFAULT_LAYOUT flag values.
LEGACY_EP_CONTENT_DEFAULT_LAYOUT Flag Values
Layout

Flag Value
CP_LAYOUT_1X1
CP_LAYOUT_1X2
CP_LAYOUT_1X2HOR
CP_LAYOUT_1X2VER
CP_LAYOUT_2X1
CP_LAYOUT_1P2HOR
CP_LAYOUT_1P2HOR_UP
CP_LAYOUT_1P2VER
CP_LAYOUT_2X2
CP_LAYOUT_1P3HOR_UP
CP_LAYOUT_1P3VER
CP_LAYOUT_1P4HOR_UP
CP_LAYOUT_1P4HOR
CP_LAYOUT_1P4VER
CP_LAYOUT_1P5
CP_LAYOUT_1P7
CP_LAYOUT_1P8UP

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LEGACY_EP_CONTENT_DEFAULT_LAYOUT Flag Values
Layout

Flag Value
CP_LAYOUT_1P8CENT
CP_LAYOUT_1P8HOR_UP
CP_LAYOUT_3X3
CP_LAYOUT_2P8
CP_LAYOUT_1P12
CP_LAYOUT_4X4

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Implementing Media Encryption for
Secured Conferencing
Encryption is available at the conference and participant levels, based on AES 128 (Advanced Encryption
Standard) and is fully H.233/H.234 compliant and the Encryption Key exchange DH 1024-bit
(Diffie-Hellman) standards.

Media Encryption Guidelines
● Encryption is not available in all countries and it is enabled in the MCU license. Contact Polycom
Support to enable it.
● Media encryption is supported in CP, SVC Only and mixed CP and SVC Conferencing Modes.
● Endpoints must support both AES 128 encryption and DH 1024 key exchange standards which are
compliant with H.235 (H.323) to encrypt and to join an encrypted conference.
● The encryption mode of the endpoints is not automatically recognized, therefore the encryption mode
must be set for the conference or the participants (when defined).
● Conference level encryption must be set in the Profile, and cannot be changed once the conference
is running.
● If an endpoint connected to an encrypted conference stops encrypting its media, it is disconnected
from the conference.
● In Cascaded conferences, the link between the cascaded conferences must be encrypted in order to
encrypt the conferences.
● The recording link can be encrypted when recording from an encrypted conference to the RSS that
is set to encryption. For more information, see Dial Out Recording Link Encryption.
● Encryption of SIP Media is supported using SRTP (Secured Real-time Transport Protocol) and the
AES key exchange method.
● Encryption of SIP Media requires the encryption of SIP signaling - TLS Transport Layer must be used.
● Encryption of SIP Media is supported in conferences as follows:
 All media channels are encrypted: video, audio and FECC.
 Collaboration Server SRTP implementation complies with Microsoft SRTP implementation.
 LPR is not supported with SRTP.
 The ENABLE_SIRENLPR_SIP_ENCRYPTION System Flag enables the SirenLPR audio
algorithm when using encryption with the SIP protocol. The default value of this flag is NO
meaning SirenLPR is disabled by default for SIP participants in an encrypted conference. To
enable SirenLPR the System Flag must be added to system.cfg and its value set to YES.
 The SEND_SRTP_MKI System Flag enables or disables the inclusion of the MKI field in SRTP
packets sent by the Collaboration Server. The default value of the flag is YES.
Add the flag to system.cfg and set its value set to NO to disable the inclusion of the MKI field in SRTP
packets sent by the Collaboration Server when using endpoints that cannot decrypt SRTP-based
audio and video streams if the MKI (Master Key Identifier) field is included in SRTP packets sent by

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the Collaboration Server. When all conferences on the RMX will not have MS-Lync clients
participating and will have 3rd party endpoints participating. This setting is recommended for
Maximum Security Environments.
Add the flag to system.cfg and set its value set to YES when Microsoft Office Communicator and Lync
Clients. When any conferences on the RMX will have both MS-Lync clients and Polycom endpoints
participating. Some 3rd party endpoints may be unsuccessful in participating in conferences with this
setting.
Polycom endpoints function normally regardless of the setting of this flag.
For more information, see Modifying System Flags.

Mixing Encrypted and Non-encrypted Endpoints in one
Conference
Mixing encrypted and non-encrypted endpoints in one conference is possible, based on the Encryption
option “Encrypt When Possible” in the Conference Profile - Advance dialog box.
The option “Encrypt When Possible” enables the negotiation between the MCU and the endpoints and let
the MCU connect the participants according to their capabilities, where encryption is the preferred setting.
Defined participants that cannot connect encrypted are connected non-encrypted, with the exception of
dial-out SIP participants.
When the conference encryption is set to "Encrypt when possible", SIP dial out participants whose
encryption is set to AUTO can only connect with encryption, otherwise they are disconnected from
the conference.In CISCO TIP environments, dial in endpoints that are registered to CUCM can only
connect as non-encrypted when the conference encryption is set to “Encrypt when possible” as the
CUCM server sends the Invite command without SDP.

When the conference encryption is set to "Encrypt when possible", SIP dial out participants whose
encryption is set to AUTO can only connect with encryption, otherwise they are disconnected from
the conference.

The same system behavior can be applied to undefined participants, depending on the setting of the System
Flag FORCE_ENCRYPTION_FOR_UNDEFINED_PARTICIPANT_IN_WHEN_AVAILABLE_MODE:
● When set to NO and the conference encryption in the Profile is set to “Encrypt When Possible”, both
Encrypted and Non-encrypted undefined participants can connect to the same conferences, where
encryption is the preferred setting.
● When set to YES (default), Undefined participants must connect encrypted, otherwise they are
disconnected.
For defined participants, connection to the conference is decided according to the encryption settings in the
conference Profile, the Defined Participant’s encryption settings.
For undefined participants, connection to the conference is decided according to the encryption settings in
the conference Profile, the System Flag setting and the connecting endpoint’s Media Encryption
capabilities.

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Direct Connection to the Conference
The following table summarizes the connection status of participants, based on the encryption settings in
the conference Profile, the Defined Participant’s encryption settings or the System Flag setting for undefined
participants and the connecting endpoint’s Media Encryption capabilities.
Connection of Participants to the Conference based on Encryption Settings
Conference
Encryption
Setting
No Encryption

Encrypt All

Encrypt When
Possible

Defined Participant
Encryption
Setting

Connection status

Auto

Connected,
non-encrypted

No

Connected,
non-encrypted

Yes

Connected only if encrypted.
Non-encrypted endpoints are
disconnected as encryption is
forced for the participant.

Auto

Connected, encrypted.
Non-encrypted endpoints are
disconnected

No

Disconnected (cannot be
added to the conference)

Yes

Connected, encrypted

Auto

All defined participants except
dial-out SIP participants:
Connect encrypted - Endpoints
with encryption capabilities.
Connect non-encrypted endpoints without encryption
capabilities.
Defined dial-out SIP
participant: Connect only if
encrypted.
Non-encrypted endpoints are
disconnected.

No

Connected, non-encrypted

Yes

Connected, encrypted

Undefined Participant
Connection Status
*Flag = No

Connection Status
*Flag = YES

Connected
non-encrypted
(Encryption is not
declared by the
Collaboration
Server, therefore the
endpoint does not
use encryption)

Connected
non-encrypted
(Encryption is not
declared by the
Collaboration
Server, therefore the
endpoint does not
use encryption)

Connect only if
encrypted.
Non-encrypted
endpoints are
disconnected

Connect only if
encrypted.
Non-encrypted
endpoints are
disconnected

Connect encrypted Endpoints with
encryption
capabilities.
Connect
non-encrypted endpoints without
encryption
capabilities

Connect only if
encrypted.
Non-encrypted
endpoints are
disconnected.

* System Flag =
FORCE_ENCRYPTION_FOR_UNDEFINED_PARTICIPANT_IN_WHEN_AVAILABLE_MODE

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Connection to the Entry Queue
An undefined participant connecting to an Entry Queue inherits the encryption characteristics of the Entry
Queue as defined in the Entry Queue’s profile.
Participants can be moved from the Entry Queue and the destination conference if both conferencing
entities have the same Profile settings, i.e. from SVC Only Entry Queue to SVC Only conference and from
mixed CP and SVC Entry Queue to a mixed CP and SVC conference, etc.
The following table summarizes the connection possibilities for a participant that is to be moved from an
Entry Queue to a destination conference for each of the conference Profile and Entry Queue encryption
options.
Connection of Undefined Participants to the Entry Queue Based on Encryption Settings
Entry Queue Encryption
Setting

Undefined Participant Connection to the Entry Queue
*Flag = No

*Flag = YES

No Encryption

Connected, non-encrypted
(Encryption is not declared by the
Collaboration Server, therefore endpoint
does not use encryption)

Connected, non-encrypted
(Encryption is not declared by the
Collaboration Server, therefore endpoint
does not use encryption)

Encrypt All

Connected only if encrypted.
Non-encrypted endpoints are
disconnected

Connected only if encrypted.
Non-encrypted endpoints are
disconnected

Encrypt When Possible

Connected encrypted - Endpoints with
encryption capabilities.
Connected non-encrypted - endpoints
without encryption capabilities

Connected only if encrypted.
Non-encrypted endpoints are
disconnected.

* System Flag =
FORCE_ENCRYPTION_FOR_UNDEFINED_PARTICIPANT_IN_WHEN_AVAILABLE_MODE

Moving from the Entry Queue to Conferences or Between Conferences
Participants can be moved from the Entry Queue and the destination conference if both conferencing
entities have the same Profile settings, i.e. from SVC Only Entry Queue to SVC Only conference and from
mixed CP and SVC Entry Queue to a mixed CP and SVC conference, etc.
When moving participants from the Entry Queue to the destination conference, or when the Collaboration
Server user moves AVC participants from one conference to another (SVC participants cannot be moved
between conferences), the connection rules are similar and they are summarized in Table 5-53:

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Moving Participants from the Entry Queue to the Destination conference or between conferences Based on
the Encryption Settings
Destination
Conference
Encryption Setting

Current Participant Encryption Status
Encrypted
*Flag = NO

*Flag = YES

Non-Encrypted
*Flag = NO

*Flag = YES

No Encryption

Move succeeds, connected encrypted

Move succeeds, connected
non-encrypted

Encrypt All

Move succeeds, connected encrypted.

Move fails, disconnected.

Encrypt When Possible

Move succeeds,
connected
encrypted

Move succeeds,
connected
non-encrypted

Move succeeds,
connected
encrypted

Connected only if
endpoint was a
defined participant
in the source
conference.
Otherwise, move
fails.

* System Flag =
FORCE_ENCRYPTION_FOR_UNDEFINED_PARTICIPANT_IN_WHEN_AVAILABLE_MODE

Recording Link Encryption
Recording Links are treated as regular participants, however the
ALLOW_NON_ENCRYPT_RECORDING_LINK_IN_ENCRYPT_CONF system flag must be set to YES if
a non-encrypted Recording Link is to be allowed to connect to an encrypted conference.
The following table summarizes the connection possibilities for a Recording Link that is to be connected to
a conference for each of the conference profile and Entry Queue encryption options.
Connections by Recording Link and Conference Encryption Settings
Conference
Profile Setting

Recording Link Connection Status according to flag:
ALLOW_NON_ENCRYPT_RECORDING_ LINK_IN_ENCRYPT_CONF
YES

NO

Encrypt All

Connected encrypted if possible,
otherwise connected non-encrypted.

Connected only if encrypted, otherwise
disconnected

No Encryption

Connected non-encrypted

Connected non-encrypted

Encrypt when possible

Connected encrypted if possible,
otherwise connected non-encrypted.

Connected encrypted if possible,
otherwise connected non-encrypted.

Enabling Media Encryption for a Conference
Media encryption is enabled at three levels:
● MCU level - Setting the Encryption Flags
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● Conference level - Enabling Encryption in the Profile
● Participant level - Enabling Encryption at the Participant Level
You must first set the system flags for the MCU before media encryption can be enabled for the conference
and participants.

Setting the Encryption Flags
Enabling the media encryption for the MCU is usually performed once an it is applicable to all conferences
running on the MCU.

To modify the Encryption flags:
1 Click Setup>System Configuration.
The System Flags dialog box opens.
2 Set the
FORCE_ENCRYPTION_FOR_UNDEFINED_PARTICIPANT_IN_WHEN_AVAILABLE_MODE flag
to YES or NO.
3 If recording will be used in encrypted conferences, set the
ALLOW_NON_ENCRYPT_RECORDING_LINK_IN_ENCRYPT_CONF flag to YES or NO.
4 Click OK.
For more information, see Modifying System Flags.
5 Reset the MCU for flag changes to take effect.

Enabling Encryption in the Profile
Encryption for the conference is in the Profile and cannot be changed once the conference is running.

To enable encryption at the conference level:
» In the Conference Profile Properties – Advanced dialog box, select one of the following Encryption
options:

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 Encrypt All - Encryption is enabled for the conference and all conference participants must be
encrypted.
 No Encryption - Encryption is disabled for the conference.
 Encrypt when possible - enables the negotiation between the MCU and the endpoints and let
the MCU connect the participants according to their capabilities, where encryption is the preferred
setting. For connection guidelines see Mixing Encrypted and Non-encrypted Endpoints in one
Conference.
For more information about recording encrypted conferences, see Dial Out Recording Link
Encryption.

Enabling Encryption at the Participant Level
You can select the encryption mode for each of the defined participants. Encryption options are affected by
the settings of the flag in the system configuration. Undefined participants are connected with the Participant
Encryption option set to Auto, inheriting the conference/Entry Queue encryption setting.

To enable encryption at the participant level:
» In the Participant Properties – Advanced dialog box, in the Encryption list, select one of the
following options: Auto, On, or Off.

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 Auto - The participant inherits the conference/Entry Queue encryption setting. The participant
connects as encrypted only if the conference is defined as encrypted.
 Yes - The participant joins the conference/Entry Queue as encrypted.
 No - The participant joins the conference/Entry Queue as non-encrypted.

Monitoring the Encryption Status
The conference encryption status is indicated in the Conference Properties - General dialog box.
The participant encryption status is indicated by a check mark in the Encryption column in the Participants
list pane.
The participant encryption status is also indicated in the Participant Properties – SDP tab, where SRTP
indication is listed for each encrypted channel (for example, audio and video).

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An encrypted participant who is unable to join a conference is disconnected from the conference. The
disconnection cause is displayed in the Participant Properties – Connection Status dialog box, Security
Failure indication, and the Cause box identifies the encryption related situation.
For more information about monitoring, see Conference and Participant Monitoring .

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Mode (AVC CP)
Collaboration Server supports the Telepresence Mode in AVC CP conferences allowing multiple participants
to join a telepresence conference from RPX and OTX high definition rooms as well as traditional, standard
definition video conferencing systems.
OTX (Telepresence) and RPX (Realpresence) room systems are configured with high definition cameras
and displays that are set up to ensure that all participants share a sense of being in the same room.
Participants using two RealPresence RPX HD 400 Room Systems

The following are examples of situations where an Collaboration Server is needed for Telepresence
configurations:
● RPX to OTX
● RPX 2-cameras/screens to RPX 4-cameras/screens
● 3 or more RPXs
● 3 or more OTXs

Collaboration Server Telepresence Mode Guidelines
System Level
● The Collaboration Server system must be licensed for Telepresence Mode.
● The system must be activated with a Telepresence enabled license key.

Conference Level
● The Telepresence Mode and Telepresence Layout Mode fields are only displayed in the
Conference Profile dialog box if the Collaboration Server has a Telepresence license installed.
● A Telepresence conference must have Telepresence Mode enabled in its profile.
● In Telepresence Mode, ITP sites are automatically detected.
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● When Telepresence mode is selected in a conference profile, the following options are disabled:
 Borders
 Site Names
 Speaker Indication
 Skins
 Same Layout
 Presentation Mode
 Auto Layout
 Lecture Mode
● The master (center) camera is used for video, audio and content.
● Conference Templates can be used to simplify the setting up Telepresence conferences where
precise participant layout and video forcing settings are crucial. Conference Templates:
 Save the conference Profile.
 Save all participant parameters including their Personal Layout and Video Forcing settings.
● An ongoing Telepresence conference can be saved to a Conference Template for later re-use.
For more information see Using Conference Templates.

Automatic Detection of Immersive Telepresence (ITP) Sites
When the conference Telepresence Mode is set to Auto (Default) ITP endpoints are automatically detected.
If an ITP endpoint is detected in such conference, ITP features are applied to all endpoints and the
Collaboration Server sends conference video with the following options disabled:
● Borders
● Site names
● Speaker indication
● Skins
● Same Layout
● Presentation Mode
● Auto Layout
● Lecture Mode
The ITP features are dynamic, and if all ITP endpoints disconnect from the conference, normal conference
video is resumed for the remaining all participants. ITP features are re-applied to all participants should an
ITP endpoint re-connects to that conference.

Horizontal Striping
Horizontal Striping is used by the Collaboration Server in order to prevent cropping and preserve the aspect
ratio of video for all Telepresence Modes.

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Cropping
Cropping is used by the Collaboration Server in order to preserve the aspect ratio of video for all
Telepresence Modes.
Cropping is controlled by the ITP_CROPPING system flag in the system configuration, providing different
cropping options according to the endpoints participating in the Telepresence conference.
By default, the flag is set to ITP. In this mode, the area to be stripped is cropped equally from the top and
the bottom (as shown in the example below). For more details, see Modifying System Flags.

Gathering Phase with ITP Room Systems
When a conference is configured to include a Gathering Phase, only one endpoint name is displayed for the
ITP room in the connected participant list of the Gathering slide. The ITP room endpoint with the suffix “1”
in its name receives the Gathering slide.

Aspect ratio for standard endpoints
Standard endpoints (non-ITP) receive video from the Collaboration Server with the same aspect ratio as that
which they transmitted to the Collaboration Server.

Skins and Frames
When Telepresence Mode is enabled, no Skin is displayed and the system uses a black background.
Frames around individual layout windows and the speaker indication are disabled.

RPX and OTX Video Layouts
Additional video layouts have been created to give Telepresence operators more video layout options when
configuring OTX and RPX room systems. These additional video layout options are available to all
endpoints on both conference layout and Personal Layout levels.

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OTX / RPX – Additional Video Layouts
Number of Endpoints

Layouts

1

2
4:3

3

4

4:3

4:3

4:3

4:3

4:3

4:3

4:3

4:3

4:3

4:3

4:3

4:3

4:3

4:3

4:3

4:3

4:3

4:3

4:3

4:3

4:3

4:3

4:3

4:3

4:3

5

9

10+

The following example illustrates the use of standard and additional Collaboration Server Telepresence
layouts when connecting four Room Systems as follows:
● Two OTX Room Systems

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 2 active cameras
 6 screens
● Two RPX Room Systems
 8 cameras
 8 screens
RPX and OTX Room System connected using the RealPresence Collaboration Server

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Room Switch Telepresence Layouts
The Room Switch Telepresence layouts normally controlled by the MLA can be managed by the MCU to
speed updating the conference layouts in large conferences with many endpoints.
Whether the MLA or the MCU controls the Room Switch Telepresence layouts is determined by the
MANAGE_TELEPRESENCE_ROOM_SWITCH_LAYOUTS flag. This flag must be manually added before
changing its value. No system reset is required.
The values are:
● NO (Default) - The MCU does not manage Telepresence Room Switch Layouts and they continue to
be managed by the MLA.
● YES - The MCU manages Telepresence Room Switch Layouts.
When the MCU controls the Telepresence Room Switch layouts
(MANAGE_TELEPRESENCE_ROOM_SWITCH_LAYOUTS = YES) the display is affected according to
the Telepresence Mode Settings in the Conference Profile as follows:
● If the Telepresence Mode = ON
 If no ITP endpoints are connected to the conference, the RMX Room Switch layout applies, in
which case only the speaker is seen.
 When a single participant using an ITP endpoint with either single or multiple screens connects to
the conference, the participant will see black screens.
● If the Telepresence Mode = AUTO
 If no ITP endpoints are connected to the conference, the RMX CP layout applies (unless the
conference layout is defined).
 When a single participant using an ITP endpoint with multiple screens connects to the conference,
the participant will see black screens.
 When a single participant using an ITP endpoint with a single screen connects to the conference,
the MCU will display a self-view of the participant.
● When a TIP system with 3 screens joins a conference, the layout is updated on all screen
simultaneously.
● When a Polycom ITP system with 2, 3, or 4 screens joins the conference, the layout is updated on all
screens simultaneously.

Telepresence Display Decision Matrix
How the speaker video is displayed on the screens of the conference participants is dependent on the
relationship between the number of screens the speaker endpoint contains and the number of screens of
the endpoints of the other conference participants.

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The following Telepresence Display Decision Matrix” table shows how the speaker video will be displayed
on the various participant endpoints when the MCU is managing Telepresence Room Switch conference
layouts.

For example, if the speaker’s endpoints has two screens and the participant’s endpoint only one, the
participant’s display is divided into two video layout cells with each video layout cell showing the input of one
of the speaker’s screens (endpoint).
If the participant endpoint has two screens, and the speaker endpoint only one, the speaker’s video will be
displayed on one of the participant’s screens, while the second screen remains black.

Guidelines for Managing the Room Switch Telepresence Layouts by
the MCU
● Only Room Switch layouts can be managed by the MCU. CP (Continuous Presence) layouts continue
to be managed by the MLA.
● Only CP-AVC conferences are supported.
● Lync Clients (with CSS add-in) are supported.
● SVC endpoints are not supported.
● It is recommended that the Speaker Change Threshold be set to 3 seconds.
● Telepresence endpoints are named using a text name followed by a number. For example, if an OTX
Telepresence room is named Oak, the three endpoint names would be Oak1, Oak2, and Oak3.
● Lecture mode is not supported in Telepresence Room Switch conferences managed by the MCU.
(This is because in Lecture mode, unlike Room Switch mode, the lecturer receives the CP layout of
conference participants.)
● Personal layouts are disabled. Therefore, any features that use personal layouts like Click&View can
not be used to change the layout, and Click&View DTMF digits will be ignored.
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● Changing the flag affects only future conferences. Conferences currently running are not affected.
● The Send Content To Legacy Endpoints feature is enabled by default when Telepresence mode is
enabled.
● Layout attributes (no skins, no site names and no borders) should continue for Telepresence layouts
managed by the RMX.

Sending Content to Legacy Endpoints in Telepresence
Conferences
The Collaboration Server can be configured to manage the layouts of to H.323/SIP/ISDN endpoints that do
not support H.239 Content (legacy endpoints) over the video (people) channel in Telepresence conferences
when Content is being sent. This feature is controlled using the FORCE_LEGACY_EP_CONTENT
_LAYOUT_ON_TELEPRESENCE flag. This flag must be added to change the value.
The values of the flag are:
● NO (Default) - The MCU does not manage the layouts while Content is sent. Personal layout
changes, for example, by MLA, override the default MCU layout. Legacy endpoints may not display
Content in Telepresence conferences due to layout changes.
● YES - The MCU manages the layouts while Content is sent. Personal layout changes, for example,
by MLA, are ignored. The layouts for legacy endpoints are managed by the MCU.

Guidelines for Sending Content to Legacy Endpoints in Telepresence
Conferences
● MLA layout change requests for legacy endpoints will be ignored until Content is stopped. At that
point, MLA can be used again.
● Click&View can not be used to change the layout while Content is being sent.
● The Polycom Touch Control can not be used to change the layout while Content is being sent.

Content Display on Legacy Endpoints in Telepresence Conferences
When Content is sent to legacy endpoints in Telepresence conferences, their video layout automatically
changes to a “Content layout” which is defined by the system flag
LEGACY_EP_CONTENT_DEFAULT_LAYOUT. If MLA is managing the Telepresnce layout prior to Content
being sent, the MCU takes over managing the layout of Legacy endpoints once Content is started. The video
layouts of the other conference participants continue to be managed by MLA.
If MLA was managing the Telepresence layouts, when Content ends, control of the layouts for legacy
endpoints goes back to the MLA after a short time.

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Enabling Telepresence Mode
Telepresence Mode must be configured in a new or existing Conference Profile.

To enable Telepresence in a new or existing Conference Profile:
1 In the RMX Management pane, click Conference Profiles.
2 Click the New Profiles button or open an existing Conference Profile.
3 Define the various profile General, Advanced, Gathering Settings and Video Quality parameters.
For more information on defining Profiles, see Defining New Profiles .
4 Click the Video Settings tab.

5 In the Telepresence Mode field, select one of the following options:
 OFF - When OFF is selected, normal conference video is sent by the Collaboration Server.
 AUTO (Default) - The ITP features are dynamic. When AUTO is selected and an ITP endpoint is
detected, ITP features are applied to the conference video for all participants. If all ITP endpoints
disconnect from the conference, normal conference video is resumed for all remaining
participants. ITP features are re-applied for all participants should an ITP endpoint re-connect to
the conference.
When Telepresence Mode is set to Auto and a one-screen Telepresence unit is in use, the
Collaboration Server controls layouts instead of the MLA. For more information see Polycom
Multipoint Layout Application (MLA) User’s Guide for Use with Polycom Telepresence Solutions.
 ON - ITP features are always applied to the conference video for all participants regardless of
whether there are ITP endpoints connected or not.
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6 In the Telepresence Layout Mode field, select the Telepresence Layout Mode to be used in the
conference. This field is used by VNOC operators and Polycom Multi Layout Applications to retrieve
Telepresence Layout Mode information from the Collaboration Server.
The following modes can be selected (as required by the VNOC and Polycom Multi Layout
Applications):
 Manual
 Continuous presence - Room Continuous Presence (Default)
 Room Switch - Voice Activated Room Switching
7 Select the required video layout.
8 Click OK.
When Telepresence Mode is enabled, the Skins options are disabled as the system uses a black
background and the frames and speaker indication are disabled.

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Monitoring Telepresence Mode
Monitoring Ongoing Conferences
An additional status indicator, Telepresence Mode Enabled, is displayed in the Conference Properties - Video
Settings tab when monitoring ongoing conferences.

The Telepresence Mode Enabled, Telepresence Mode and Telepresence Layout Mode fields are
only enabled if the Collaboration Server has a Telepresence license installed.

If Telepresence Mode is enabled, a check mark is displayed in the check box. This option is grayed as this
is a status indicator and cannot be used to enable or disable Telepresence Mode.

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Monitoring Participant Properties
An additional status indicator, Telepresence, is displayed in the Participant Properties - Advanced tab
when monitoring conference participants.

The Telepresence mode of the participant is indicated:
● RPX - the participant’s endpoint is transmitting 4:3 video format.
● OTX - the participant’s endpoint is transmitting 16:9 video format.
● None.

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Creating Multiple Cascade Links Between
Telepresence Conferences
You can create multiple Cascading links between Collaboration Servers hosting conferences that include
Immersive Telepresence Rooms (ITP) such as Polycom’s OTX and RPX Room Systems.

Guidelines for Creating Multiple Cascading Links between
Conferences
● Basic Cascading topology is used. For more information see the RealPresence Collaboration
Server 800s Administrator’s Guide, Basic Cascading Using IP Cascaded Link.
● Multiple Cascade Links between conferences are implemented by creating a Link Participant which
consists of a main link and sub-links which are automatically generated and sequentially numbered.
For more information see Creating a Link Participant, Creating a Link Participant.
● All cascaded links must use H.323 protocol.
● Multiple Cascade Links are supported in CP conferencing mode.
● The number of cascading links is defined manually according to the maximum number of Room
System cameras in the cascaded conference.
● When the active speaker is in an Immersive Telepresence Room, Multiple Cascade Links are used,
one link for each of the Room System’s cameras.
 An RPX 4xx Room System requires 4 Cascaded Links to carry the video of its 4 cameras.
 An RPX 2xx Room System requires 2 Cascaded Links to carry the video of its 2 cameras.
 An OTX 3xx Room System requires 3 Cascaded Links to carry the video of its 3 cameras. The
OTX Room System must be configured as Room Switch in order to send multiple streams. When
configured in CP Mode, its cameras zoom out and all 3 screens are sent as one stream.
● The number of links is defined when creating the Link Participant. Each conference in the cascade
must have a Link Participant with the same number of Multiple Cascade Links defined. Calls from
Link Participants not defined with the same number of links are rejected. Number of cascading links
is not identical for all conferences is listed as the Call Disconnection Cause. For more information see
Creating a Link ParticipantCreating a Link Participant and Monitoring Multiple Cascade
LinksMonitoring Multiple Cascade Links.
● Although it is possible to disconnect and reconnect specific Multiple Cascade Links using the
RealPresence Collaboration Server Web Client / RealPresence Collaboration Server Manager it not
advisable to do so.
 If the main link is disconnected all sub-links are disconnected and deleted.
Reconnecting the main link reconnects all sub-links.
 If a sub-link is disconnected it remains disconnected until it is manually reconnected.
 The number of Multiple Cascade Links cannot be modified while any of the links are in a
disconnected state. All previous links must be deleted before modification is possible.
For more information see Monitoring Multiple Cascade LinksMonitoring Multiple Cascade Links.
●

A Link Participant can be dragged from the address book into a conference.
 If it is the first Link Participant in the conference, the number of Multiple Cascade Links defined
for the participant are created and connected.

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 If it is not the first Link Participant in the conference, the number of Multiple Cascade Links defined
for the participant is ignored.
● If there are insufficient resources to connect all Multiple Cascade Links in either of the RMXs, none
of the links are connected and resources deficiency -0 is listed as the Call Disconnection Cause. For
more information see Monitoring Multiple Cascade LinksMonitoring Multiple Cascade Links.
● Multiple Cascade Links that are not used by MLA are inactive but continue to consume resources.
● All RMXs participating in the cascade must have the same Telepresence Mode definitions, either all
defined as CP or all defined as Room Switch.
● When Multiple Cascade Links are defined in the Conference Profile, the Layout Type field of the Link
Participant’s Participant Properties - Media Sources dialog box is set to Conference and cannot be
modified.
● TIP Telepresence Rooms (CTS) are supported without Content. For more information see the
Collaboration With Cisco’s Telepresence Interoperability Protocol (TIP).

Enabling and Using Multiple Cascade Links
The settings required to enable Multiple Cascade Links on the RMX are minimal and are described in
Creating a Link ParticipantCreating a Link Participant.
Most of the layout configuration is performed using Polycom’s Multipoint Layout Application (MLA).
The figures, RMX Telepresence Layout Mode - Room Switch and RMX Telepresence Layout Mode Continuous Presence show example layouts and media flows when MLA is configured for a cascading
conference between two RMXs.
In the figure RMX Telepresence Layout Mode - Room Switch:
● The OTX Room System connects to RMX A.
● The RPX Room System connects to RMX B.
● This layout requires that the Telepresence Layout Mode to be set to Room Switch in the Conference
Profiles of the Cascading Conferences in each RMX.
● The current speaker is a participant in the RPX ITP Room.

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● Directional media flows, A  B, are shown separately for readability purposes.
RMX Telepresence Layout Mode - Room Switch

In this figure:
 An HDX endpoint and an OTX Room System connects to RMX A.
 An HDX endpoint and an RPX Room System connects to RMX B.
 This layout requires that the Telepresence Layout Mode to be set to Continuous Presence in
the Conference Profiles of the Cascading Conferences in each RMX.

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● The current speaker is the HDX endpoint connected to RMX A.
RMX Telepresence Layout Mode - Continuous Presence

For more information see:
● Telepresence Layout Mode .
● Polycom® Multipoint Layout Application (MLA) User’s Guide for Use with Polycom Telepresence
Solutions
● Polycom® Immersive Telepresence (ITP) Deployment Guide

Creating a Link Participant
Link Participant in the Dial Out RMX
The Link Participant is defined in the New Participant dialog box.
In the General tab:

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● Dialing Direction must be selected as Dial out.
● Type must be selected as H.323.
For more information see the Creating a Cascade Enabled Dial-out/Dial-in Participant Link .
In the Advanced tab:
(This field is only enabled if the RMX system is licensed for Telepresence Mode.)
● In the Cascade drop-down menu, select either Master or Slave.

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● In the Number of cascaded links (for ITP) drop-down menu, select the maximum number of
Multiple Cascade Links required according to the number of Room System endpoints in the cascaded
conference.
This field enables the administrator to select the maximum number of Multiple Cascade Links
required according to the number of Room System endpoints in the cascaded conference.
For example if an RPX 4xx is included, the number of links required is 4.

The RMX automatically adds a number suffix to the name of the Link Participant, for example if the
Participant Link Name is CascadeLink and the Number of cascaded links (for ITP) field is set to 4, the
following Multiple Cascade Links are created:
● CascadeLink-1
● CascadeLink-2
● CascadeLink-3
● CascadeLink-4

Participant Link in the Dial In RMX
The call from Participant Link defined in the Dial-out RMX is identified by the Dial-in RMX as having been
initiated by a Participant Link.
Suffixes are appended the Multiple Cascade Links according to the Number of cascaded links (for ITP)
field depending on whether the Dial -In Participant Link is defined or un-defined:

Participant Link is un-defined
The Multiple Cascade Link names are automatically assigned by the RMX.
For example on a RMX 1500 the names of the links are:

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● POLYCOM RMX 1500-1
● POLYCOM RMX 1500-2
● POLYCOM RMX 1500-3, etc.

Participant Link is a defined
The Multiple Cascade Link names are assigned according to the name of the defined participant that is to
function as the cascade link and the Number of cascaded links (for ITP) information sent by the calling
Dial-Out Participant Link.
For example if the defined participant that is to function as the cascade link is named
Cascade_Link_From_B the names of the links are:
● Cascade_Link_From_B-1
● Cascade_Link_From_B-2
● Cascade_Link_From_B-3, etc.

Monitoring Multiple Cascade Links
Multiple Cascade Links connections can be monitored in the Participants list of the RMX Web Client / RMX
Manager main screen:

Disconnection Causes
● If there are insufficient resources to connect all the required links:
 None of the links are connected.
 The first link is listed as Disconnected in the Participants list of the RMX Web Client / RMX
Manager main screen.

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▪

Resource deficiency is listed as the Call Disconnection Cause in the Participant Properties Connection Status dialog box.

● If a calling Link Participant is not defined with same number of links as all the other Link Participants
in the cascaded conferences:
 The call is rejected.
 The Call Disconnection Cause is: Number of cascading links is not identical for all conferences.

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In the RealPresence CloudAxis Solution, the conferencing parameters are defined in the
RealPresence DMA system component and should not be defined directly in the Polycom®
RealPresence® Collaboration Server 800s and Polycom® RealPresence® Collaboration Server
Virtual Edition component.

Various conferencing modes and video features require additional settings, such as system flag settings,
conference parameters and other settings. In depth explanations of these additional settings are described
in the following sections:
● Video Preview (AVC Participants Only)
● Auto Scan and Customized Polling in Video Layout (CP Conferences Only)
● Packet Loss Compensation (LPR and DBA) AVC CP Conferences
● Network Quality Indication (AVC Endpoints)
● Lecture Mode (AVC CP Only)
● Audio Algorithm Support

Video Preview (AVC Participants Only)
Collaboration Server users can preview the video sent from the participant to the conference (MCU) and the
video sent from the conference to the participant. It enables the Collaboration Server users to monitor the
quality of the video sent and received by the participant and identify possible quality degradation.
The video preview is displayed in a separate window independent to the Collaboration Server Web Client.
All Web Client functionality is enabled and conference and participant monitoring as well as all other user
actions can be performed while the video preview window is open and active. Live video is shown in the
preview window as long as the window is open. The preview window closes automatically when the
conference ends or when participant disconnects from the conference. It can also be closed manually by
the Collaboration Server user.

Video Preview Guidelines
● Video Preview is supported in CP Conferencing Mode only.
● Video preview is available for AVC participants. It is not available for SVC participants.
● Video preview window size and resolution are adjusted to the resolution of the PC that displays the
preview.
● Video Preview of the video sent from the conference to the participant is shown according to the line
rate and video parameters of the level threshold to which the participant is connected.
● All users can view a video preview.
● Only one preview window can be displayed for each Collaboration Server Web Client connection
(workstation).

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● Only one preview window can be displayed for a single conference and up to four preview windows
can be displayed for each system.
● Live video that is shown in the preview window does not include the Content when it is sent by the
participant.
● Video Preview is supported in cascaded conferences.
● If the video preview window is opened when the IVR slide is displayed to the participant, it will also
be displayed in the video preview window.
● Video Preview is supported with H.264 High Profile.
● Video Preview is not supported for endpoints using the RTV protocol.
● Video Preview is disabled in encrypted conferences.
● Video preview cannot be displayed when the participant’s video is suspended.
● Participant’s video preview and the Polycom Desktop application (such as CMAD) window cannot be
open and running simultaneously on the same PC as both require the same DirectDraw resource.

Workstation Requirements to Display Video Preview
To be able to display the video preview window, the following minimum requirements must be met:
● Windows XP, Windows Vista and Windows 7
● Internet Explorer 7 and later
● DirectX is installed
● DirectDraw Acceleration must be enabled and no other application is using the video resource
● Hardware acceleration must be enabled

Testing your Workstation
To ensure that your workstation can display the video preview window:
1 In Windows, click Start > Run.
The Run dialog box opens.
2 In the Open field, type dxdiag and press the Enter key or click OK.

A confirmation message is displayed.
3 Click Yes to run the diagnostics.
The DirectX Diagnostic Tool dialog box opens.
4 Click the Display tab.
To be able to display the video preview window, the DirectDraw Acceleration and Direct3D
Acceleration options must be Enabled.
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If the video card installed in the PC does not support DirectDraw Acceleration, a black window may
be viewed in the Video Preview window.
5 Click the Exit button.

Previewing the Participant Video
You can preview the video sent from the participant to the conference (MCU) and the video sent from the
conference to the participant by selecting the appropriate option from the Participant’s pop-up menu.

To preview the participant video:
1 List the conference participants in the Participants pane.
2 Right-click the participant whose video you want to preview and then click one of the following
options:

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 View Participant Sent Video - to display the video sent from the participant to the conference.
 View Participant Received Video - to display the video sent from the conference to the
participant.
The Video Preview window opens.

If the video card installed in the PC does not support DirectDraw Acceleration, a black window may
be viewed.

Auto Scan and Customized Polling in Video Layout
(CP Conferences Only)
Auto Scan enables you to define a single cell in the conference layout to cycle the display of participants
that are not in the conference layout.
Customized Polling allows the cyclic display to be set to a predefined order for a predefined time period.
The cyclic display only occurs when the number of participants is larger than the number of cells in the
layout.
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Guidelines for Using Auto Scan and Customized Polling
● Auto Scan and Customized Polling are supported in AVC CP conferences only.
● Participants that are in the conference layout will not appear in the Auto Scan enabled cell.
● If Customized Polling is not used to define the order of the Auto Scan it will proceed according to order
in which the participants connected to the conference.
● If the user changes the conference layout, the Auto Scan settings are not exported to the new layout.
If the user changes the conference layout back to the layout in which Auto Scan was enabled, Auto
Scan with the previous settings will be resumed.

Enabling the Auto Scan and Customized Polling (CP Only
Conferences)
Auto Scan and Customized Polling are enabled during the ongoing conference, in the Conference
Properties - Video Settings dialog box.

Enabling the Auto Scan
You enable the Auto Scan feature by selecting it in the Video Layout cell.

To enable Auto Scan:
1 In the Collaboration Server Web Client Main Screen - Conference list pane, double-click the
conference or right-click the conference and then click Conference Properties.
2 In the Conference Properties - General dialog box, click Video Settings.
The Video Settings dialog box is displayed.

3 If Auto Layout check box is selected, clear it.

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4 In the video layout cell to be designated for Auto Scan, click the drop-down menu button and select
Auto Scan.
5 Select from the Auto Scan Interval(s) list the scanning interval in seconds.
6 Click the Apply button to confirm and keep the dialog box open, or Click OK.

Customized Polling
The order in which the Auto Scanned participants are displayed in the Auto Scan enabled cell of the video
layout can be customized.

To define the scanning order in the Customized Polling tab:
1 Open the Conference Properties dialog box.
2 Click the Customized Polling tab.
The Customized Polling dialog box is displayed.

All conference participants are listed in the left pane (All Participants) while the participants that are
to be displayed in the Auto Scan enabled cell of the video layout are listed in the right pane (Scanning
Order).
The dialog box buttons are summarized in the following table:

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Customized Polling Dialog Box Command Buttons
Button

Description

Add

Select a participant and click this button to Add a the participant to the list of
participants to be Auto Scanned.
The participants name is removed from the All Participants pane.

Delete

Select a participant and click this button to Delete the participant from the list
of participants to be Auto Scanned.
The participants name is moved back to the All Participants pane.

Add All

Add all participants to the list of participants to be Auto Scanned.
All participants’ names are removed from the All Participants pane.

Delete All

Delete all participant from the list of participants to be Auto Scanned.
All participants’ names are moved back to the All Participants pane.

Up

Select a participant and click this button to move the participant Up in the
Scanning Order.

Down

Select a participant and click this button to move the participant Down in the
Scanning Order.

3 Optional. Add a participant to the list of participants to be Auto Scanned:
 Click on the participant’s name in the All Participants list and then click the Add button to move
the participant to the Scanning Order pane.
4 Optional. Delete a participant from the list of participants to be Auto Scanned:
 Click on a participant’s name in the Scanning Order list and then click the Delete button to move
the participant back to the All Participants pane.
5 Optional. Add all participants to the list of participants to be Auto Scanned by clicking the Add All
button.
6 Optional. Delete all participant from the list of participants to be Auto Scanned by clicking the Delete
All button.
7 Optional. Move the participant up in the Scanning Order by clicking the Up button.
8 Optional. Move the participant down in the Scanning Order by clicking the Down button.
9 Click the Apply button to confirm and keep the dialog box open, or click OK.

Packet Loss Compensation (LPR and DBA) AVC CP
Conferences
Lost Packet Recovery (LPR) and Dynamic Bandwidth Allocation (DBA) help minimize media quality
degradation that can result from packet loss in the network. Packet loss Compensation is available in AVC
CP Conferencing Mode only and is not supported in SVC Conferencing Mode or CP and SVC Conferencing
Mode.

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Packet Loss
Packet Loss refers to the failure of data packets, transmitted over an IP network, to arrive at their
destination. Packet Loss is described as a percentage of the total packets transmitted.

Causes of Packet Loss
Network congestion within a LAN or WAN, faulty or incorrectly configured network equipment or faulty
cabling are among the many causes of Packet Loss.

Effects of Packet Loss on Conferences
Packet Loss affects the quality of:
● Video – frozen images, decreased frame rate, flickering, tiling, distortion, smearing, loss of lip sync
● Audio – drop-outs, chirping, audio distortion
● Content – frozen images, blurring, distortion, slow screen refresh rate

Lost Packet Recovery
The Lost Packet Recovery (LPR) algorithm uses Forward Error Correction (FEC) to create additional
packets that contain recovery information. These additional packets are used to reconstruct packets that are
lost, for whatever reason, during transmission. Dynamic Bandwidth Allocation (DBA) is used to allocate the
bandwidth needed to transmit the additional packets.

Lost Packet Recovery Guidelines
● If packet loss is detected in the packet transmissions of either the video or Content streams:
 LPR is applied to both the video and Content streams.
 DBA allocates bandwidth from the video stream for the insertion of additional packets containing
recovery information.
● LPR is supported in H.323 and SIP networking environments only.
● In LPR-enabled Continuous Presence conferences:
 Both LPR-enabled and non-LPR-enabled endpoints are supported.
 The LPR process is not applied to packet transmissions from non-LPR-enabled IP (H.323 and
SIP) endpoints.
 Non-LPR-enabled endpoints can be moved to LPR-enabled conferences.
 LPR-enabled endpoints cannot be moved to non-LPR-enabled conferences.
● When connecting via an Entry Queue:
 A participant using an LPR-enabled endpoint can be moved to a non-LPR-enabled conference.
The participant is connected with LPR enabled.

Enabling Lost Packet Recovery
LPR is enabled or disabled in the Conference Profile dialog box.
● CP Conferences – LPR is enabled by default in the New Profile – Advanced dialog box.
For more information, see Defining New Profiles .

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Monitoring Lost Packet Recovery
In the Participant Properties – H.245 tab, LPR activity is displayed in all three panes.

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In the Participant Properties – Channel Status tab, check box indicators show LPR activation in the local
and remote (transmit and receive) channels.

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Network Quality Indication (AVC Endpoints)
If network quality issues occur, the Network Quality icon provide information to participants about their own
network quality and that of other participants displayed in the cells of the conference Video Layout.

Network Quality Levels
Network quality is determined by the percentage of packet loss according to the following default threshold
values:
● Packet loss less than 1% is considered Normal
● Packet loss in the range of 1% - 5% is considered Major
● Packet loss above 5% is considered Critical.

When network quality improves from Critical to Major remaining stable for 5 seconds, the Network Quality
Indicator is changed accordingly and when network quality improves from Major to Normal, remaining stable
for 5 seconds, the Network Quality Indicator is no longer displayed.

Indication Threshold Values
The default Major and Critical indication threshold values can be modified by manually adding the following
System Flags and modifying their values as required.

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Network Quality Icon - Indication Threshold Flags
Flag

Description

NETWORK_IND_MAJOR_PERCENTAGE

The percentage degradation due to packet loss required to
change the indicator from Normal to Major.
Default: 1

NETWORK_IND_CRITICAL_PERCENTAGE

The percentage degradation due to packet loss required to
change the indicator from Major to Critical.
Default: 5

For more information see the Polycom® RealPresence Collaboration Server 800s/Virtual Edition
Administrator’s Guide, Manually Adding and Deleting System Flags.

Guidelines for Displaying the Network Quality icons
Network Quality icons are not supported in SVC Conferencing Mode.
Network Quality icons are displayed for:
● The video channel only in AVC Conferencing Mode.
Content, audio and FECC channel quality issues are not indicated.
● The participant’s own endpoint:
 Network Quality icons are displayed by default and can be disabled
 For media transmitted to and received from the Collaboration Server (Video in / Video out).
● Participants displayed in the cells of the conference video layout:
 Network Quality icons are not displayed by default and can be enabled
 The media transmitted from the endpoint to the Collaboration Server (Video in).

Customizing Network Quality Icon Display
Display of the Network Quality icon can be customized for the participant’s own endpoint or for the
Participants displayed in the cells of the conference Video Layout.
The display of Network Quality icon (showing or hiding the icon) and the position of the icon in the video
layout cell can be customized by manually adding the following System Flags and modifying their values as
required.

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Network Quality Icon - Display Customization Flags
Flag

Description

DISABLE_SELF_NETWORK_IND

Disable the display of the Network Quality icon of the
participant’s own endpoint.
Default: NO
Range: YES / NO

DISABLE_CELLS_NETWORK_IND

Disable the display of Network Quality icons displayed in
the cells of the conference Video Layout.
Default: YES
Range: YES / NO

SELF_IND_LOCATION

Change the location of the display of the Network Quality
icon of the participant’s own endpoint.
Default: BOTTOM_RIGHT
Range:
• TOP_ LEFT
• TOP
• TOP_RIGHT
• BOTTOM_ LEFT
• BOTTOM
• BOTTOM_RIGHT

CELL_IND_LOCATION

Change the location of the display of Network Quality icons
displayed in the cells of the conference Video Layout.
Default: TOP_RIGHT
Range:
• BOTTOM_LEFT
• BOTTOM_RIGHT
• TOP_LEFT
• TOP_RIGHT

For more information see the Polycom® RealPresence Collaboration Server 800s/Virtual Edition
Administrator’s Guide, Manually Adding and Deleting System Flags.

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Lecture Mode (AVC CP Only)
Lecture Mode enables all participants to view the lecturer in full screen while the conference lecturer sees
all the other conference participants in the selected layout while he/she is speaking. When the number of
sites/endpoints exceeds the number of video windows in the layout, switching between participants occurs
every 15 seconds. Conference participants cannot change their Personal Layouts while Lecture Mode is
enabled.
Automatic switching is suspended when one of the participants begins talking, and it is resumed
automatically when the lecturer resumes talking.
Lecture Mode is available only in AVC CP Conferencing Mode.

Enabling Lecture Mode
Lecture Mode is enabled at the conference level by selecting the lecturer. Conference participants cannot
change their Personal Layouts while Lecture Mode is enabled.
Automatic switching between participants viewed on the lecturer’s screen is enabled in the conference
Profile.

Selecting the Conference Lecturer
Selecting a lecturer for the ongoing conference, enables the Lecture Mode. You can select the lecturer:
● during the definition of the ongoing conference
● after the conference has started and the participants have connected to the conference.

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To select the lecturer and enable the Lecture Mode while starting the conference:
● In the Conference Properties - Participant dialog box, enable the Lecture Mode in one of the
following methods:

Selecting a defined participant:
a Add participants to the conference either from the Address book or by defining new participants.
b In the Lecturer field, select the lecturer from the list of the defined participants.
Automatic selection of the lecturer:
 In the Lecturer field, select [Auto].
In this mode, the conference speaker becomes the lecturer.

To select the lecturer and enable the Lecture Mode during the ongoing conference:
1 Make sure that the participant you want to designate as the lecturer has connected to the
conference.
2 In the Conference Properties - Video Settings dialog box, in the Lecturer field, select the lecturer
from the list of the connected participants.

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Defined dial out participants and dial in participants are considered to be two separate participants
even if they have the same IP address/number. Therefore, if a defined dial-out participant is added to
the conference and the same participant then dials in (before the system dialed out to that
participant) the system creates a second participant in the Participants list and tries to call the
dial-out participant. If the dial-out participant was designated as the conference lecturer, the system
will not be able to replace that participant with the dial-in participant that is connected to the
conference.

Enabling the Automatic Switching
Automatic switching between participants viewed on the lecturer’s screen is enabled in the conference
Profile, or during the ongoing conference, in the Conference Properties.

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● In the Profile Properties - Video Settings dialog box, select the Lecturer View Switching check
box.

This option is activated when the conference includes more sites than windows in the selected layout.
If this option is disabled, the participants will be displayed in the selected video layout without
switching.
For more information about Profile definition, see Defining AVC CP Conferencing Profiles .
● Once the conference is running, in the Conference Properties - Video Settings dialog box, select the
Lecturer View Switching check box.

Lecture Mode Monitoring
A conference in which the Lecture Mode is enabled is started as any other conference. The conference runs
as an audio activated Continuous Presence conference until the lecturer connects to the conference. The
selected video layout is the one that is activated when the conference starts. Once the lecturer is connected,
the conference switches to the Lecture Mode.
When Lecturer View Switching is activated, it enables automatic switching between the conference
participants in the lecturer’s video window. The switching in this mode is not determined by voice activation
and is initiated when the number of participants exceeds the number of windows in the selected video
layout. In this case, when the switching is performed, the system refreshes the display and replaces the last
active speaker with the current speaker.
When one of the participants is talking, the automatic switching is suspended, showing the current speaker,
and it is resumed when the lecturer resumes talking.
If the lecturer is disconnected during an Ongoing Conference, the conference resumes standard
conferencing.
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Forcing is enabled at the Conference level only. It applies only to the video layout viewed by the lecturer as
all the other conference participants see only the lecturer in full screen.
If an asymmetrical video layout is selected for the lecturer (i.e. 3+1, 4+1, 8+1), each video window contains
a different participant (i.e. one cannot be forced to a large frame and to a small frame simultaneously).
When Lecture Mode is enabled for the conference, the lecturer is indicated by an icon in the Role column
of the Participants list.

To control the Lecture Mode during an Ongoing Conference:
During the Ongoing Conference, in the Conference Properties - Video Settings dialog box you can:
● Enable or disable the Lecture Mode and designate the conference lecturer in the Lecturer list; select
None to disable the Lecture Mode or select a participant to become the lecturer to enable it.

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● Designate a new lecturer.

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● Enable or disable the Lecturer View Switching between participants displayed on the lecturer monitor
by selecting or clearing the Lecturer View Switching check box.

● Change the video layout for the lecturer by selecting another video layout.

Restricting Content Broadcast to Lecturer
Content broadcasting can be restricted to the conference lecturer only, when one of the conference
participants is set as the lecturer (and not automatically selected by the system). Restricting the Content
Broadcast prevents the accidental interruption or termination of H.239 Content that is being shared in a
conference.
Content Broadcast restriction is enabled by setting the
RESTRICT_CONTENT_BROADCAST_TO_LECTURER system flag to ON. When set to OFF (default) it
enables all users to send Content.

When enabled, the following rules apply:
● Content can only be sent by the designated lecturer. When any other participant tries to send Content,
the request is rejected.
● If the Collaboration Server user changes the designated lecturer (in the Conference Properties Video Settings dialog box), the Content of the current lecturer is stopped immediately and cannot
be renewed.
● The Collaboration Server User can abort the H.239 Session of the lecturer.
● Content Broadcasting is not implemented in conferences that do not include a designated lecturer
and the lecturer is automatically selected by the system (for example, in Presentation Mode).

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Muting Participants Except the Lecturer (AVC CP Only)
When the Mute Participants Except Lecturer option in the Conference Profile is enabled, the audio of all
participants in the conference except for the lecturer can be automatically muted upon connection to the
conference. This prevents other conference participants from accidentally interrupting the lecture, or from a
noisy participant affecting the audio quality of the entire conference. Muted participants cannot unmute
themselves unless they are unmuted from the Collaboration Server Web Client/RMX Manager.

Guidelines for Muting all the Participants Except the Lecturer
● Both administrators and operators (users) are allowed to set the Mute Participants Except Lecturer
option.
● When the Mute Participants Except Lecturer option is enabled, the mute indicator on the participant
endpoints are not visible because the mute participants was initiated by the MCU. Therefore, it is
recommended to inform the participants that their audio is muted by using the Message Overlay
functions.
● When the Mute Participants Except Lecturer option is enabled in the Conference Profile settings, all
conferences to which this profile is assigned will start with this option enabled. All participants, except
for the designated lecturer, are muted.
● The Mute Participants Except Lecturer option can be enabled or disabled at any time after the start
of the conference. When enabled, it allows all the conference participants to converse before the
lecturer joins the conference or before they are muted. When disabled, it unmutes all the participants
in the conference.
● If the endpoint of the designated lecturer is muted when the lecturer connects to the conference, the
lecturer remains muted until the endpoint has been unmuted.
● When you replace a lecturer, the MCU automatically mutes the previous lecturer and unmutes the
new lecturer.
● When you disconnect a lecturer from the conference or the lecturer leaves the conference, all
participants remain muted but are able to view participants in regular video layout until the you disable
the Mute Participants Except Lecturer option.
● A participant can override the Mute Participants Except Lecturer option by activating the Mute All
Except Me option using the appropriate DTMF code, provided the participant has authorization for
this operation in the IVR Services properties. The lecturer audio is muted and the participant audio is
unmuted. You can reactivate the Mute Participants Except Lecturer option after a participant has
previously activated the Mute All Except Me option. The participant is muted and the lecturer, if
designated, is unmuted.
● In cascaded conferences, all participants (including the link participants) except the lecturer are
muted. Only the lecturer is not muted.

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Enabling the Mute Participants Except Lecturer Option
The Mute Participants Except Lecturer option is enabled or disabled (default) in the Conference Profile or
in an ongoing conference in the Profile Properties - Audio Settings tab.

When the Mute Participants Except Lecturer option is enabled and a conference has started, the Mute by
MCU icon is displayed in the Audio column in the Participants pane of each participant that is muted.

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Audio Algorithm Support
The Collaboration Server supports the following audio algorithms in AVC conferences: G.711, G.722,
G.722.1, G.722.1C, G.729A, Polycom Siren 7 (in mono), Siren14, Siren 22 (in mono or stereo) and
SirenLPR.
Polycom’s proprietary Siren 22 and industry standard G.719 audio algorithms are is supported for
participants connecting with Polycom endpoints.
The Siren 22 audio algorithm provides CD-quality audio for better clarity and less listener fatigue with audio
and visual communication applications. Siren 22 requires less computing power and has much lower
latency than alternative wideband audio technologies.
The SirenLPR audio algorithm provides CD-quality audio for better clarity and less listener fatigue with audio
and visual communication applications.
In SVC conferences, the system supports SAC (Scalable Audio Coding) audio algorithm.

Audio Algorithm Support Guidelines
● Siren 22 is supported in both mono and stereo.
● Stereo is supported in H.323 calls only.
● Siren 22 is supported by Polycom HDX and Group series endpoints, version 2.0 and later.
● SirenLPR is enabled by default and can be disabled by setting the system flag,
ENABLE_SIRENLPR, to NO.
● SirenLPR is supported:
 In IP (H.323, SIP) calls only.
 In CP conferences.
 With Polycom CMAD and HDX 3.0.1 and later and Group series endpoints.
 For mono audio at audio line rates of 32Kbps, 48Kbps and 64Kbps.
 For stereo audio at audio line rates of 64Kbps, 96Kbps and 128Kbps.

SIP Encryption
The ENABLE_SIRENLPR_SIP_ENCRYPTION System Flag enables the SirenLPR audio algorithm when
using encryption with the SIP protocol.
The default value of this flag is NO meaning SirenLPR is disabled by default for SIP participants in an
encrypted conference. To enable SirenLPR the System Flag must be added to system.cfg and its value set
to YES.

Mono
The Siren 22, and SirenLPR mono audio algorithms are supported at the following bit rates
:

Siren22, and SirenLPR Mono vs Bitrate
Audio Algorithm

Minimum Bitrate (kbps)

Siren22 64k
Siren22 48K
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Siren22, and SirenLPR Mono vs Bitrate
Audio Algorithm

Minimum Bitrate (kbps)

Siren22_32k
G.719_64k

384

G.719_48k
G.719_32k
G.728 16K
G.719_64k

384

SirenLPR_48k

256

Siren22_48K
G.719_48k
G.7221C_48k
Siren14_48k
SirenLPR_32k
Siren22_32k
G.719_32k

128

G.7221C_32k
Siren14_32k
SirenLPR

64

SirenLPR

48

SirenLPR

32

Stereo
The Siren 22Stereo, and SirenLPR audio algorithms are supported at the following bit rates.
Siren22Stereo, and SirenLPR vs Bitrate
Audio Algorithm

Minimum Bitrate (kbps)

Siren22Stereo_128k
SirenLPRStereo_128k

1024

G.719Stereo_128k

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Siren22Stereo, and SirenLPR vs Bitrate
Audio Algorithm

Minimum Bitrate (kbps)

Siren22Stereo_96k
SirenLPRStereo_96k
G.719Stero_96k

512

Siren14Stero_96k
SirenLPRStereo_64k
G.719Stereo_64k
Siren22Stereo_64k

384

Siren14Stereo_64k

Monitoring Participant Audio Properties
The audio algorithm used by the participant’s endpoint can be verified in the Participant Properties Channel Status dialog box.

To view the participant’s properties during a conference:
1 In the Participants list, right click the desired participant and select Participant Properties.
2 Click the Channel Status - Advanced tab.
The Participant Properties - Channel Status - Advanced dialog box is displayed.

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3 In the Channel Info field, select Audio In or Audio Out to display the audio parameters.

4 Click the OK button.

Automatic Muting of Noisy Endpoints
The automatic muting of noisy AVC-enabled endpoints can be used according to the following guidelines:
● The Auto mute noisy endpoints check box in the Profile Properties - Audio Settings dialog box is
enabled only when the ENABLE_SELECTIVE_MIXING flag is set to YES (default).
● It affects only AVC-based and audio only endpoints (non-SAC endpoints)
● It does not affect SVC-based endpoints
● It is supported in CP conferences and in Mixed CP and SVC conferences.
● In a mixed CP and SVC conferences, only the AVC-based endpoints can be automatically muted. If
the noisy endpoint is SVC-based, its audio channel will not be sent to the AVC-based endpoints, but
it will be sent to the other SVC-based endpoints.
● MCU reset is not required when changing the ENABLE_SELECTIVE_MIXING flag setting.
● When upgrading from a version prior to 8.1, the Auto mute noisy endpoints option is not automatically
enabled in the existing Profiles and it has to be manually enabled, if required.

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● In new Profiles that are created after the upgrade, the Auto mute noisy endpoints option is
automatically enabled.
If your conferencing environment includes the Polycom DMA, the conferences that are started from the
DMA will not include the Auto mute noisy endpoints parameter as it is not part of the DMA Profiles. In such
a case, when the parameter setting is unknown, the system will enable or disable the automatic muting of
noisy endpoints according to the flag setting - if the flag is set to YES, it will be enabled in the conference.
The following table summarizes the state (enabled or disabled) of the Automatic muting of noisy endpoints
feature depending on the ENABLE_SELECTIVE_MIXING flag setting and the Auto mute noisy endpoints
setting in the Profile Properties - Audio Settings:
ENABLE_SELECTIVE_MIXING flag
Setting

Auto mute noisy endpoints
setting

Automatic muting of noisy
endpoints State

YES

Yes (check box selected)

Enabled

YES

No (check box cleared)

Disabled

YES

Unknown (for example, the
conference is started from the
DMA)

Enabled

NO

Yes (check box selected)

Disabled

NO

No (check box cleared)

Disabled

NO

Unknown (for example, the
conference is started from the
DMA)

Disabled

Enabling or Disabling the Automatic Muting of Noisy Endpoints
The automatic muting of noisy endpoints can be enabled or disabled at the conference level (in the
Conference Profile) or at the system level, by changing the ENABLE_SELECTIVE_MIXING flag setting.
In new MCU installations, the automatic muting of noisy endpoints is automatically enabled on the MCU as
the ENABLE_SELECTIVE_MIXING flag is set to YES and the Auto mute noisy endpoints check box in the
Profile Properties - Audio Settings tab is selected.
You can disable the automatic muting of noisy endpoints by either setting the system flag to NO or clearing
the Auto mute noisy endpoints check box in the Profile Properties - Audio Settings tab.
If required, it is recommended to disabled the automatic muting of noisy endpoints at the conference level,
in the conference Profile without changing the flag settings.
In existing MCU sites, following the software upgrade the automatic muting of noisy endpoints is disabled
at the conference level in the existing conference Profile and has to be manually enabled in these profiles.
This option is automatically enabled when creating a new Profile.
Enabling or Disabling the Automatic Muting of Noisy Endpoints at the Conference Level
If the ENABLE_SELECTIVE_MIXING flag is set to YES, the automatic muting of noisy endpoints can be
enabled or disabled at the conference level in the Conference Profile - Audio Settings dialog box.
If the ENABLE_SELECTIVE_MIXING flag is set to NO, the automatic muting of noisy endpoints is disabled
at the conference level and cannot be enabled in the Conference Profile - Audio Settings dialog box.
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To disable/enable the automatic muting of noisy endpoints in the Conference Profile:
1 In a new or existing Conference Profile, click the Audio Settings tab.

 In new Profiles, the Auto mute noisy endpoints check box is selected by default.
 In existing profiles (after software upgrade from a version prior to 8.1), the Auto mute noisy
endpoints check box is cleared.
2 To enable the automatic muting of noisy endpoints, click the Auto mute noisy endpoints check
box.
3 Click OK.

Enabling or Disabling the Automatic Muting of Noisy Endpoints at the MCU Level
You can disable the automatic muting of noisy endpoints at the MCU level by changing the
ENABLE_SELECTIVE_MIXING flag setting to NO.
In such a case, the automatic muting of noisy endpoints at the conference level (in the Conference Profile
- Audio Settings dialog box) is disabled.

To modify the system flag setting:
● To modify NABLE_SELECTIVE_MIXING flag setting to NO, manually add it to system.cfg file and set
its value to NO.
For more details, see Modifying System Flags.

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Permanent Conference
A Permanent Conference is any ongoing conference with no pre-determined End Time continuing until it is
terminated by an administrator, operator or chairperson.

Guidelines
● Auto Terminate is disabled in Permanent Conferences.
● If participants disconnect from the Permanent Conference, resources are released.
● Entry Queues, Conference Reservations and SIP Factories cannot be defined as Permanent
Conferences.
● Additional participants can connect to the conference, or be added by the operator, if sufficient
resources are available.
● The maximum size of the Call Detail Record (CDR) for a Permanent Conference is 1MB.

Enabling a Permanent Conference
The Permanent Conference option is selected in the New Conference, New Meeting Room or New
Conference Templates dialog boxes.

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Cascading Conferences
Cascading information applies to AVC Conferencing Mode (CP and mixed CP and SVC) only.
Cascading is not supported with SVC Conferencing Mode.

Cascading enables administrators to connect one conference directly to one or several conferences,
depending on the topology, creating one large conference. The conferences can run on the same MCU or
different MCUs.
There are many reasons for cascading conferences, the most common are:
● Connecting two conferences on different MCUs at different sites.
● Utilizing the connection abilities of different MCUs, for example, different communication protocols.
The following cascading topologies are available for cascading:
● Basic Cascading - only two conferences are connected (usually running on two different
Collaboration Servers). The cascaded MCUs reside on the same network.
● Star Cascading - one or several conferences are connected to one master conference. Conferences
are usually running on separate MCUs. The cascaded MCUs reside on the same network.
System configuration and feature availability change according to the selected cascading topology.

Video Layout in Cascading conferences (CP and mixed CP and SVC)
Cascade links are treated as endpoints in CP conferences and are allocated resources according to
Resolution Configuration for CP Conferences on page 6. Cascaded links in 1x1 video layout are in SD
resolution.
When cascading two conferences, the video layout displayed in the cascaded conference is determined by
the selected layout in each of the two conferences. Each of the two conferences will inherit the video layout
of the other conference in one of their windows.
In order to avoid cluttering in the cascaded window, it is advised to select appropriate video layouts in each
conference before cascading them.

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Video Layouts in Cascaded Conferences

Guidelines
To ensure that conferences can be cascaded and video can be viewed in all conferences the following
guidelines are recommended:
● The same version installed on all MCUs participating the cascading topology
● The same license installed on all MCUs participating the cascading topology
● Same Conference Parameters are defined in the Profile of the conferences participating in the
cascading topology
 Conference line rates should be identical
 Content rate should be identical
 Same encryption settings
● DTMF codes should be defined with the same numeric codes in the IVR services assigned to the
cascading conferences
● DTMF forwarding is suppressed
● The video layout of the link is set to 1x1 by the appropriate system flag.
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● When the Mute Participants Except Lecturer option is enabled in the Conference Profile, all
participants (including the link participants) except the lecturer are muted. Only the lecturer is not
muted.

Flags controlling Cascade Layouts
● Setting the FORCE_1X1_LAYOUT_ON_CASCADED_LINK
_CONNECTION System Flag to YES (default) automatically forces the cascading link to Full Screen
(1x1) in CP conferences, hence displaying the speaker of one conference to a full window in the video
layout of the other conference.
Set this flag to NO when cascading between an Collaboration Server and an MGC that is functioning
as a Gateway, if the participant layouts on the MGC are not to be forced to 1X1.
● Setting the AVOID_VIDEO_LOOP_BACK_IN_CASCADE System Flag to YES (default) prevents
the speaker’s image from being sent back through the participant link from the cascaded conference.
This can occur in cascaded conferences with conference layouts other than 1x1. It results in the
speaker’s own video image being displayed in the speaker’s video layout.
This option is supported with Basic Cascading. If a Master MCU has two slave MCUs, participants
connected to the slave MCUs will not receive video from each other.
For more details on defining system flags, see Modifying System Flags.

Basic Cascading
In this topology, a link is created between two conferences, usually running on two different MCUs. The
MCUs are usually installed at different locations (states/countries) to save long distance charges by
connecting each participant to their local MCU, while only the link between the two conferences is billed as
long distance call.
● This is the only topology that enables IP cascading links:
 When linking two conferences using an IP connection, the destination MCU can be indicated by:
 IP address
 H.323 Alias
 If IP cascading link is used to connect the two conferences, both MCUs must be located in the
same network.
● One MCU can be used as a gateway.
● The configuration can include two Collaboration Servers.

Basic Cascading using IP Cascaded Link
In this topology, both MCUs can be registered with the same gatekeeper or the IP addresses of both MCUs
can be used for the cascading link. Content can be sent across the Cascading Link.

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Basic Cascading Topology - IP Cascading Link

For example, MCU B is registered with the gatekeeper using 76 as the MCU prefix.
The connection between the two conferences is created when a dial out IP participant is defined (added) to
conference A whose dial out number is the dial-in number of the conference or Entry Queue running on
MCU B.

Dialing Directly to a Conference
Dial out IP participant in conference A dials out to the conference running on MCU B entering the number
in the format:
[MCU B Prefix/IP address][conference B ID].
For example, if MCU B prefix is 76 and the conference ID is 12345, the dial number is 7612345.

Dialing to an Entry Queue
When dialing to an Entry Queue, the dial out participant dials the MCU B prefix or IP address of MCU B and
the Entry Queue ID in the format:
[MCU B Prefix/IP address][EQ B ID].
For example, if MCU B prefix is 76 and the Entry Queue ID is 22558, the dial number is 7622558.
When the participant from conference A connects to the Entry Queue, the system plays to all the participants
in Conference A the IVR message requesting the participant to enter the destination conference ID.
At this point, the Conference A organizer or any other participant in the conference can enter the required
information for the IVR session using DTMF codes. For example, the meeting organizer enters the
destination conference ID - 12345.
Any DTMF input from conference A is forwarded to the Entry Queue on MCU B to complete the IVR session
and enable the move of the participant to the destination conference B.
Once the DTMF codes are entered and forwarded to the Entry Queue on MCU B, the IVR session is
completed, the participant moved to the destination conference and the connection between the two
conferences is established.

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Automatic Identification of the Cascading Link
The system automatically identifies that the dial in participant is an MCU and creates a Cascading Link and
displays the link icon for the participant (
). The master-slave relationship is randomly defined by the
MCUs during the negotiation process of the connection phase.

System Flag Settings
The DTMF_FORWARD_ANY_DIGIT_TIMER_SECONDS flag determines the time period (in seconds) that
MCU A will forward DTMF inputs from conference A participants to MCU B.
Once the timer expires, most of the DTMF codes (excluding five operations as for IP links) entered in
conference A will not be forwarded to conference B. This is done to prevent an operation requested by a
participant individually (for example, mute my line) to be applied to all the participants in conference B.
Flag range (in seconds): 0 - 360000
This flag is defined on MCU A (the calling MCU).
If a flag is not listed in the System Flags list it must be added before it can be modified. For more details on
defining system flags, see Modifying System Flags.

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Meeting Rooms
In the RealPresence CloudAxis Solution, the virtual Meeting Rooms are defined in the RealPresence
DMA system component and should not be defined directly in the Polycom® RealPresence®
Collaboration Server 800s and Polycom® RealPresence® Collaboration Server Virtual Edition
component.

A Meeting Room is a conference saved on the MCU in passive mode, without using any of the system
resources. A Meeting Room is automatically activated when the first participant dials into it. Meeting Rooms
can be activated as many times as required. Once activated, a Meeting Room functions as any ongoing
conference.
The conferencing Mode of the Meeting Room is determined by the Profile assigned to it.
In SVC Conferencing Mode, dial-in is available as follows:
● AVC-capable endpoints (participants) can only connect to an AVC CP Meeting Room. When dialing
into SVC Only Meeting Room the calls fail.
● SVC-capable endpoints support both AVC and SVC video protocols. When dialing into SVC Only
conferences, they connect as SVC endpoints. When dialing into AVC CP Only conferences, they
connect as AVC endpoints.
● Both AVC and SVC endpoints can connect to a mixed CP and SVC conference.
In AVC CP Conferences, dial-out participants can be connected to the conference automatically, or
manually. In the automatic mode the system calls all the participants one after the other. In the manual
mode, the Collaboration Server user or meeting organizer instructs the conferencing system to call the
participant. Dial-out participants must be defined (mainly their name) and added to the conference. This
mode can only be selected at the conference/Meeting Room definition stage and cannot be changed once
the conference is ongoing.
A Meeting Room can be designated as a Permanent Conference.
For more information see Audio Algorithm Support.
The maximum of number of Meeting Rooms that can be defined is: 1000.
The system is shipped with four default Meeting Rooms:
Default Meeting Rooms List
Meeting Room Name

ID

Default Line Rate

Maple_Room

1001

1920 Kbps

Oak_Room

1002

1920 Kbps

Juniper_Room

1003

1920 Kbps

Fig_Room

1004

1920 Kbps

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Meeting Rooms List
Meeting Rooms are listed in the Meeting Room list pane.

To list Meeting Rooms:
» In the RMX Management pane, in the Frequently Used list, click the Meeting Rooms button

.

The Meeting Rooms list is displayed.

An active Meeting Room becomes an ongoing conference and is monitored in the same way as any
other conference.
The Meeting Room List columns include:
Meeting Rooms List Columns
Field

Description

Display Name

Displays the name and the icon of the Meeting Room in the Collaboration Server Web
Client.

(green)

An active video Meeting Room that was activated when the first
participant connected to it.
A passive video Meeting Room that is waiting to be activated.

(gray)
Routing Name

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The ASCII name that registers conferences, Meeting Rooms, Entry Queues and SIP
Factories in the various gatekeepers and SIP Servers. In addition, the Routing Name
is also:
• The name that endpoints use to connect to conferences.
• The name used by all conferencing devices to connect to conferences that must be
registered with the gatekeeper and SIP Servers.

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Meeting Rooms List Columns (Continued)
Field

Description

ID

Displays the Meeting Room ID. This number must be communicated to H.323
conference participants to enable them to dial in.

Duration

Displays the duration of the Meeting Room in hours using the format HH:MM
(default 01:00).

Conference Password

The password to be used by participants to access the
Meeting Room. If blank, no password is assigned to the
conference. This password is valid only in conferences
that are configured to prompt for a conference password
in the IVR Service.

Chairperson Password

Displays the password to be used by the users to identify
themselves as Chairpersons. They are granted additional
privileges. If left blank, no chairperson password is
assigned to the conference. This password is valid only in
conferences that are configured to prompt for a
chairperson password.

Profile

Displays the name of the Profile assigned to the Meeting Room. For more information,
see Defining New Profiles.

SIP Registration

The status of registration with the SIP server:
• Not configured - Registration with the SIP Server was not enabled in the
Conference Profile assigned to this conferencing Entity. In Multiple Networks
configuration, If one service is not configured while others are configured and
registered, the status reflects the registration with the configured Network Services.
The registration status with each SIP Server can be viewed in the Properties Network Services dialog box of each conferencing entity.
When SIP registration is not enabled in the conference profile, the Collaboration
Server's registering to SIP Servers will each register with a URL derived from its
own signaling address.
• Failed - Registration with the SIP Server failed. This may be due to incorrect
definition of the SIP server in the IP Network Service, or the SIP server may be
down, or any other reason the affects the connection between the Collaboration
Server or the SIP Server to the network.
• Registered - the conferencing entity is registered with the SIP Server.
• Partially Registered - This status is available only in Multiple Networks
configuration, when the conferencing entity failed to register to all the required
Network Services if more than one Network Service was selected.

The Collaboration Server
can be configured to
automatically generate
conference and
chairperson passwords
when these fields are left
blank. For more
information, see the
Automatic Password
Generation Flags.

Use Time Out as DTMF Delimiter
Users are able to change the behavior of the policy of number collection for VMR Entry Queues,
Conference-IDs and Passwords, allowing a time-out to be used as a stop indicator for the input string.
Note: This feature is not supported in the RealPresence Collaboration Server Virtual Edition.

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In previous versions, a # input at the end of the input string indicated completion of the input.
The administrator can configure the system, using the ENABLE_DTMF_NUMBER_WO_DELIMITER
system flag to change the previous system behavior, allowing a time-out to be used as a stop indicator for
the string input for the local IVR, when the MCU collects the Conference-ID in the local Entry Queue or the
Password (chairperson or participant) while routed to the conference.
The flag must be manually added to the System Configuration and its value modified as follows:
Value / Description

Flag Name

YES

ENABLE_DTMF_NUM
BER_WO_DELIMITER

If the timer expires, the received digits
validated even if there is no delimiter.
If the received number is not valid, the
system will prompt again for the number
according to number of retries. that are
configured.

NO
This is the default setting for
backward compatibility.
If the timer expires because no delimiter
is received, the number input is not valid.
The system will prompt again for the
number according to number of retries.
that are configured.

A System Reset in not required for the flag setting to take effect.
For more information see, Modifying System Flags in the Collaboration Server (RMX) Administrator’s Guide.

Meeting Room Toolbar & Right-Click Menu
The Meeting Room toolbar and right-click menus provide the following functionality:
Meeting Room Toolbar and Right-click Menus
Toolbar
button

Right-click menu

Description

New Meeting Room

Select this button to create a new Meeting Room.

Delete Meeting Room

Select any Meeting Room and then click this button to delete the
Meeting Room.

Dial out to AVC participants assigned to a Meeting Room will only start when the dial in participant
who has activated it has completed the connection process and the Meeting Room has become an
ongoing conference.

Creating a New Meeting Room
In the RealPresence CloudAxis Solution, virtual Meeting Rooms are defined in the RealPresence
DMA system component and should not be defined directly in the Polycom® RealPresence®
Collaboration Server 800s and Polycom® RealPresence® Collaboration Server Virtual Edition
component.

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To create a new meeting room:
● In the Meeting Rooms pane, click the New Meeting Room
in the pane and then click New Meeting Room.

button or right-click an empty area

The New Meeting Room dialog box is displayed.

The definition procedure is the same as for the new conference.
If SIP Factories are being used do not assign a Meeting Room the ID 7001. This ID is reserved for
the default SIP Factory.

For more information, see the Polycom® RealPresence Collaboration Server 800s / Virtual Edition
Getting Started Guide, Starting an AVC CP Conference from the Conferences Pane. Microsoft Lync
users can connect a Collaboration Server Meeting Room to a conference running on the Microsoft
A/V MCU. This allows Collaboration Server Lync users to connect with a conference in progress on
the A/V MCU and be an active participant in the conference.
For more information, see Connecting a Collaboration Server Meeting Room to a Microsoft AV-MCU
Conference.

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Entry Queues, Ad Hoc Conferences and
SIP Factories
Entry Queues
An Entry Queue (EQ) is a special routing lobby to access conferences. Participants connect to a single-dial
lobby and are routed to their destination conference according to the Conference ID they enter. The Entry
Queue remains in a passive state when there are no callers in the queue (in between connections) and is
automatically activated once a caller dials its dial-in number.
Participants can be moved from the Entry Queue and the destination conference if both conferencing
entities are set to the same conferencing parameters: Conferencing Mode, Line rate and video parameters.
For example, participants can be moved from SVC Only Entry Queue to SVC Only conference, or from a
mixed CP and SVC Entry Queue to a mix CP and SVC conference, from CP only Entry Queue to CP only
conference.
The maximum of number of Entry Queues that can be defined is 40.
The parameters (bit rate and video properties) with which the participants connect to the Entry Queue and
later to their destination conference are defined in the Conference Profile that is assigned to the Entry
Queue. For example, if the Profile Bit Rate is set to 384kbps, all endpoints connect to the Entry Queue and
later to their destination conference using this bit rate even if they are capable of connecting at higher bit
rates.
An Entry Queue IVR Service must be assigned to the Entry Queue to enable the voice prompts guiding the
participants through the connection process. The Entry Queue IVR Service also includes a video slide that
is displayed to the participants while staying in the Entry Queue (during their connection process).
Different Entry Queues can be created to accommodate different conferencing modes, conferencing
parameters (by assigning different Profiles) and prompts in different languages (by assigning different Entry
Queue IVR Services).
For more information, see IVR Services List.
The Entry Queue can also be used for Ad Hoc conferencing. If the Ad Hoc option is enabled for the Entry
Queue, when the participant enters the target conference ID the system checks whether a conference with
that ID is already running on the MCU. If not, the system automatically creates a new ongoing conference
with that ID. For more information about Ad Hoc conferencing, see Ad Hoc Conferencing.
An Entry Queue can be designated as Transit Entry Queue to which calls with dial strings containing
incomplete or incorrect conference routing information are transferred. For more information, see Transit
Entry Queue.

Default Entry Queue properties
The system is shipped with a default Entry Queue whose properties are shown in the following table.

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Default Entry Queue Properties
Parameter

Value

Display Name

DefaultEQ
The user can change the name if required.

Routing Name

DefaultEQ
The default Routing Name cannot be changed.

ID

1000

Profile name

Factory_Mixd_CP_SVC_Video_Profile. Profile Bit Rate is set to 1920Kbps.

Entry Queue Service

Entry Queue IVR Service. This is default Entry Queue IVR Service shipped with the
system and includes default voice messages and prompts in English.

Ad Hoc

Enabled

Defining a New Entry Queue
In the RealPresence CloudAxis Solution, virtual Entry Queues and ad-hoc conferences are defined
in the RealPresence DMA system component and should not be defined directly in the Polycom®
RealPresence® Collaboration Server 800s and Polycom® RealPresence® Collaboration Server
Virtual Edition component.

You can modify the properties of the default Entry Queue and define additional Entry Queues to suit different
conferencing requirements.

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To define a new Entry Queue:
1 In the RMX Management pane, In the Rarely Used menu, click Entry Queues.

2 In the Entry Queues list pane, click the New Entry Queue button.
The New Entry Queue dialog box opens.

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3 Define the following parameters:
Entry Queue Definitions Parameters
Option

Description

Display Name

The Display Name is the conferencing entity name in native language character
sets to be displayed in the Collaboration Server Web Client.
In conferences, Meeting Rooms, Entry Queues and SIP factories the system
automatically generates an ASCII name for the Display Name field that can be
modified using Unicode encoding.
• English text uses ASCII encoding and can contain the most characters
(length varies according to the field).
• European and Latin text length is approximately half the length of the
maximum.
• Asian text length is approximately one third of the length of the maximum.
The maximum length of text fields also varies according to the mixture of
character sets (Unicode and ASCII).
Maximum field length in ASCII is 80 characters. If the same name is already
used by another conference, Meeting Room or Entry Queue, the Collaboration
Server displays an error message requesting you to enter a different name.

Routing Name

Enter a name using ASCII text only. If no Routing Name is entered, the system
automatically assigns a new name as follows:
• If an all ASCII text is entered in Display Name, it is used also as the Routing
Name.
• If any combination of Unicode and ASCII text (or full Unicode text) is entered
in Display Name, the ID (such as Conference ID) is used as the Routing
Name.

Profile

Select the Profile to be used by the Entry Queue.
The default Profile is selected by default. This Profile determines the Bit Rate
and the video properties with which participants connect to the Entry Queue and
destination conference.
In Ad Hoc conferencing, it is used to define the new conference properties.

ID

Enter a unique number identifying this conferencing entity for dial in. Default
string length is 4 digits.
If you do not manually assign the ID, the MCU assigns one after the completion
of the definition. The ID String Length is defined by the flag
NUMERIC_CONF_ID_LEN in the System Configuration.

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Entry Queue Definitions Parameters (Continued)
Option

Description

Entry Queue Mode

Select the mode for the Entry Queue
Standard Lobby (default) - When selected, the Entry Queue is used as a
routing lobby to access conferences. Participants connect to a single-dial lobby
and are routed to their destination conference according to the Conference ID
they enter.
Ad Hoc - Select this option to enable the Ad Hoc option for this Entry Queue. In
this mode, when the participant enters the target conference ID the system
checks whether a conference with that ID is already running on the MCU. If not,
the system automatically creates a new ongoing conference with that ID.
IVR Only Service Provider - When selected, designates this Entry Queue as a
special Entry Queue that provides IVR Services to SIP calls on behalf of the
RealPresence DMA system. The IVR Only Service Provider Entry Queue does
not route the SIP calls to a target conference. Instead the RealPresence DMA
system handles the call. For more details, see IVR Provider Entry Queue
(Shared Number Dialing).
External IVR Control - IVR Services can be controlled externally from an
application server (such as the DMA) supporting the MCCF-IVR (Media Control
Channel Framework-Interactive Voice Response) package.
When selected, the connection process of the participant to the conference via
the Virtual Entry Queue is controlled and managed by an external IVR service of
an application server (for example, DMA).

Entry Queue IVR Service

The default Entry Queue IVR Service is selected. If required, select an alternate
Entry Queue IVR Service, which includes the required voice prompts, to guide
participants during their connection to the Entry Queue.

4 Click OK.
The new Entry Queue is added to the Entry Queues list.

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Listing Entry Queues
To view the list of Entry Queues:
● In the RMX Management pane- Rarely Used menu, click Entry Queues.
The Entry Queues are listed in the Entry Queues pane.

You can double-click an Entry Queue to view its properties.

Modifying the EQ Properties
To modify the EQ:
● In the Entry Queues pane, either double-click or right-click, and select Entry Queue Properties of
the selected Entry Queue in the list.
The Entry Queue Properties dialog box is displayed. All the fields may be modified except Routing
Name.

Transit Entry Queue
A Transit Entry Queue is an Entry Queue to which calls with dial strings containing incomplete or incorrect
conference routing information are transferred.
IP Calls are routed to the Transit Entry Queue when:
● A gatekeeper is not used, or where calls are made directly to the Collaboration Server’s Signaling IP
Address, with incorrect or without a Conference ID.
● When a gatekeeper is used and only the prefix of the Collaboration Server is dialed, with incorrect or
without a Conference ID.
● When the dialed prefix is followed by an incorrect conference ID.
When no Transit Entry Queue is defined, all calls containing incomplete or incorrect conference routing
information are rejected by the Collaboration Server.
In the Transit Entry Queue, the Entry Queue IVR Service prompts the participant for a destination
conference ID. Once the correct information is entered, the participant is transferred to the destination
conference.

Setting a Transit Entry Queue
The Collaboration Server factory default settings define the Default Entry Queue also as the Transit Entry
Queue. You can designate another Entry Queue as the Transit Entry Queue.

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Only one Transit Entry Queue may be defined per Collaboration Server and selecting another Entry Queue
as the Transit Entry Queue automatically cancels the previous selection.

To designate an Entry Queue as Transit Entry Queue:
1 In the RMX Management pane, Rarely Used list, click Entry Queues.
2 In the Entry Queues list, right-click the Entry Queue entry, and then click Set Transit Entry Queue.
The Entry Queue selected as Transit Entry Queue is displayed in bold.

To cancel the Transit Entry Queue setting:
1 In the RMX Management pane, Rarely Used list, click Entry Queues.
2 In the Entry Queues list, right-click the Transit Entry Queue entry, and then click Cancel Transit
Entry Queue.

IVR Provider Entry Queue (Shared Number Dialing)
In an environment that includes a RealPresence DMA system, the Collaboration Server Entry Queue can
be configured to provide the IVR Services on behalf of the RealPresence DMA system to SIP endpoints. It
displays the Welcome Slide, plays the welcome message and retrieves the destination conference ID that
is entered by the participant using DTMF codes.
To enable this feature, a special Entry Queue that is defined as IVR Only Service Provider is created. This
Entry Queue does not forward calls to conferences running on the Collaboration Server and its main
functionality is to provide IVR services.

Call Flow
The SIP participant dials the DMA Virtual Entry Queue number, for example 1000@dma.polycom.com.
The DMA forwards the SIP call to the Collaboration Server, to a special Entry Queue that is configured as
IVR Only Service Provider. The participant is prompted to enter the conference ID using DTMF codes.
Once the participant enters the conference ID, the conference ID is forwarded to the DMA, enabling the
DMA to connect the SIP endpoint to the destination conference or create a new conference and connect
the participant to that conference.

Guidelines for Setting the Entry Queue as IVR Provider
● An Entry Queue defined as IVR Only Service Provider does not route the SIP call to a target
conference and it cannot be used to route calls on the Collaboration Server. In such a configuration,
the DMA handles the calls. Therefore, normal Entry Queues must be defined separately.
● Operator Assistance must be disabled in the IVR Service assigned to this Entry Queue.
● Only the conference ID prompts should be configured. Other prompts are not supported in IVR Only
Service Provider configuration.
● ISDN and H.323 calls to this Entry Queue are rejected.
● The DMA must be configured to locate the IVR Only Service Provider Entry Queue on the
Collaboration Server. To locate the Entry Queue the DMA requires the Entry Queue’s ID number and
the Collaboration Server Signaling IP address (xxx.xx.xxx.xx).

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Configuring the Collaboration Server as IVR Provider
Entry Queue IVR Service
If required, create a special Entry Queue IVR Service in which the Operator Assistance option is disabled,
and only the Conference ID prompts are enabled.

Entry Queue
» In the New Entry Queue dialog box, Entry Queue Mode list, select IVR Only Service Provider.

 Enter the Entry Queue ID that will be used by the DMA to forward the SIP calls to this Entry
Queue.
 Select the special Entry Queue IVR Service if one was created.

Configuring the MCU to Support External IVR Services via the MCCF-IVR
The support of External IVR Services via the MCCF-IVR package is enabled by default in the Collaboration
Server (RMX) systems, by the flag ENABLE_MCCF which is set to YES.
However, in secured environments where the External IVR Services via the MCCF-IVR package is not
required and unused ports should be closed, this flag should be set to NO.
To change this flag value from YES to No, you must first add it to the System Configuration. For more details,
see Manually Adding and Deleting System Flags.

SIP Factories
A SIP Factory is a conferencing entity that enables SIP endpoints to create Ad Hoc conferences. The
system is shipped with a default SIP Factory, named DefaultFactory.
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.

The default SIP Factory uses the conferencing ID 7001. If a SIP Factory is being used do not assign
this ID to any conferencing entity, including conferences, reservations, and meeting rooms.

When a SIP endpoint calls the SIP Factory URI, a new conference is automatically created based on the
Profile parameters, and the endpoint joins the conference.
The SIP Factory URI must be registered with the SIP server to enable routing of calls to the SIP Factory. To
ensure that the SIP factory is registered, the option to register Factories must be selected in the Default IP
Network Service.
The maximum of number of SIP Factories that can be defined is 40.

Creating SIP Factories
To create a new SIP Factory:
1 In the RMX Management pane, Rarely Used list, click SIP Factories.
2 In the SIP Factories list pane, click the New SIP Factory button.
The New Factory dialog box opens.

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3 Define the following parameters:
:

New Factory Properties
Option

Description

Display Name

Enter the SIP Factory name that will be displayed.
The Display Name is the conferencing entity name in native language character
sets to be displayed in the Collaboration Server Web Client.
In conferences, Meeting Rooms, Entry Queues and SIP factories the system
automatically generates an ASCII name for the Display Name field that can be
modified using Unicode encoding.
• English text uses ASCII encoding and can contain the most characters
(length varies according to the field).
• European and Latin text length is approximately half the length of the
maximum.
• Asian text length is approximately one third of the length of the maximum.
The maximum length of text fields also varies according to the mixture of
character sets (Unicode and ASCII).
Maximum field length in ASCII is 80 characters. If the same name is already
used by another conference, Meeting Room or Entry Queue, the Collaboration
Server displays an error message requesting you to enter a different name.

Routing Name

The Routing Name is defined by the user, however if no Routing Name is
entered, the system will automatically assign a new name when the Profile is
saved as follows:
• If an all ASCII text is entered in Display Name, it is used also as the Routing
Name.
• If any combination of Unicode and ASCII text (or full Unicode text) is entered
in Display Name, the ID (such as Conference ID) is used as the Routing
Name.

Profile

The default Profile is selected by default. If required, select the conference
Profile from the list of Profiles defined in the MCU.
A new conference is created using the parameters defined in the Profile.

Automatic Connection

Select this check box to immediately accept the conference creator endpoint to
the conference. If the check box is cleared, the endpoint is redirected to the
conference and then connected.

4 Click OK.
The new SIP Factory is added to the list.

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SIP Registration & Presence for Entry Queues and SIP
Factories with SIP Servers
Entry Queues and SIP Factories can be registered with SIP servers. This enables Office Communication
Server or Lync server client users to see the availability status (Available, Offline, or Busy) of these
conferencing entities, and to connect to them directly from the Buddy List.

Guidelines for registering Entry Queues and SIP Factories with SIP Servers
● The Entry Queue or SIP Factory must be added to the Active Directory as a User.
● SIP Registration must be enabled in the Profile assigned to the Entry Queue or SIP Factory. For more
information see Defining New Profiles.

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Monitoring Registration Status
The SIP registration status can be viewed in the Entry Queue or SIP Factory list panes.

The following statuses are displayed:
● Not configured - Registration with the SIP Server was not enabled in the Conference Profile
assigned to the Entry Queue or SIP Factory.
When SIP registration is not enabled in the conference profile, the Collaboration Server's registering
to SIP Servers will each register with an URL derived from its own signaling address.
● Failed - Registration with the SIP Server failed.
This may be due to incorrect definition of the SIP server in the IP Network Service, or the SIP Server
may be down, or any other reason the affects the connection between the Collaboration Server or the
SIP Server to the network.
● Registered - The conferencing entity is registered with the SIP Server.
● Partially Registered - This status is available only in Multiple Networks configuration, when the
conferencing entity failed to register to all the required Network Services, if more than one Network
Service was selected for Registration.

Ad Hoc Conferencing
The Entry Queue can also be used for Ad Hoc conferencing. If the Ad Hoc option is enabled for the Entry
Queue, when the participant enters the target conference ID the system checks whether a conference with
that ID is already running on the MCU. If not, the system automatically creates a new ongoing conference
with that ID. The conference parameters are based on the Profile linked to the Entry Queue. As opposed to
Meeting Rooms, that are predefined conferences saved on the MCU, Ad Hoc conferences are not stored
on the MCU. Once an Ad Hoc conference is started, it becomes an ongoing conference, and is monitored
and controlled as any standard ongoing conference.
For more information about Ad Hoc conferencing, see Appendix D, Appendix D - Ad Hoc Conferencing and
External Database Authentication.

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Address Book
The Address Book stores information about the people and businesses you communicate with. The
Address Book stores, among many other fields, IP addresses, phone numbers and network communication
protocols used by the participant’s endpoint. By utilizing the Address Book you can quickly and efficiently
assign or designate participants to conferences. Groups defined in the Address Book help facilitate the
creation of conferences. Participants can be added to the Address Book individually or in Groups.
The maximum of number of Address Book entries that can be defined on the Collaboration Server is 4000.
When using the Polycom® CMA® and Polycom® RealPresence® Resource Manager® Global Address
Book, all entries are listed.
The Address Book can be organized into a multi-level hierarchical structure. It can be used to mirror the
organizational layout of the enterprises and it is especially suitable for large-scale enterprises with a
considerable number of conference participants and organizational departments and divisions. Groups in
the Address Book can contain sub-groups or sub-trees, and individual address book participant entities.
The Address Book provides flexibility in arranging conference participants into groups in multiple levels and
the capabilities to add groups or participants, move or copy participants to multiple groups within the address
book, and use the address book to add groups and participants to a conference or Conference Template.
Importing and exporting of Address Books enables organizations to seamlessly distribute up-to-date
Address Books to multiple Collaboration Server units. It is not possible to distribute Address Books to
external databases running on applications such as the RealPresence Resource Manager or Polycom
CMA. External databases can run in conjunction with Collaboration Server units, but must be managed from
the external application. For example, new participants cannot be added to the external database from the
Collaboration Server Web Client. To enable the Collaboration Server to run with an external database such
as Polycom RealPresence Resource Manager or Polycom CMA, the appropriate system configuration
flags must be set.
For more information, see Modifying System Flags.
Integration with the Global Address Book of the Polycom® RealPresence® Resource Manager®
or Polycom CMA® is supported. For more information, see Integrating the Global Address Book
(GAB) of Polycom RealPresence Resource Manager or Polycom CMA with the Collaboration
Server. Integration with the SE200 GAB (Global Address Book) is not supported.

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Viewing the Address Book
You can view the participants currently defined in the Address Book. The first time the Collaboration Server
Web Client is accessed, the Address Book pane is displayed.

The Address Book contains two panes:
● Navigation pane - contains the hierarchical tree and All Participants list
● List pane - displays the list of all the members of the selected group and sub-groups.

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The Navigation pane of the Address Book contains the following types of lists:
● Hierarchical — Displays a multi-level hierarchical tree of groups and participants. Double-clicking a
group on the navigation pane displays the group participants and sub-groups in the List pane.
● All Participants — Double-clicking this selection displays the single unique entity of all the
participants in a single level. When adding a participant to a group, the system adds a link to the
participant’s unique entity that is stored in the All Participants list. The same participant may be added
to many groups at different levels, and all these participant links are associated with the same
definition of the participant in the All Participants list. If the participant properties are changed in one
group, they will be changed in all the groups accordingly.

Displaying and Hiding the Group Members in the Navigation Pane
The currently selected group, whose group members are displayed in the Address Book List pane is
identified by a special icon
.

To expand the group to view the group members:
» Double-click the group name or click the Expand

button.

The address book entities and sub-groups of the group is displayed in the right group list pane. You
can drill down the sub-group to view address book entities in the sub-group.

To move up to the next level and view the members in the upper level:
» Double-click the navigation arrow

button in the group members pane.

To collapse a group:
» Double-click the group name or click the Collapse

button.

Participants List Pane Information
The Participants List pane displays the following information for each participant:

Participants List Pane
Field/Option

Description

Type

Indicates whether the participant is a video (

Name

Displays the name of the participant.

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Participants List Pane (Continued)
Field/Option

Description

IP Address/Phone

Enter the IP address of the participant’s endpoint.
• For H.323 participant define either the endpoint IP address or alias.
• For SIP participant define either the endpoint IP address or the SIP address.

Network

The network communication protocol used by the endpoint to connect to the conference:
H.323 or SIP.

Dialing Direction

Dial-in – The participant dials in to the conference.
Dial-out – The Collaboration Server dials out to the participant.

Encryption

Displays whether the endpoint uses encryption for its media.
The default setting is Auto, indicating that the endpoint must connect according to the
conference encryption setting.

For information on adding and modifying participants in the Address Book, see Managing the Address Book.

Displaying and Hiding the Address Book
The Address Book can be hidden it by clicking the anchor pin ( ) button in the pane header. The Address
Book pane closes and a tab is displayed at the right edge of the screen.
» Click the tab to re-open the Address Book.

Adding Participants from the Address Book to
Conferences
In the RealPresence CloudAxis Solution, Participant Address Book is defined in the Polycom®
RealPresence® Resource Manager® component and should not be defined directly in the
Polycom® RealPresence® Collaboration Server 800s and Polycom® RealPresence® Collaboration
Server Virtual Edition component.

You can add individual participants or a group of participants from the Address Book to a conference.

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Adding Individual Participants from the Address Book to Conferences
You can add a participant or multiple participants to a new conference, ongoing conferences, or to
Conference Templates by using the drag-and-drop operation.
Multiple selection of group levels is not available.

To add a participant to a new conference or an ongoing conference:
1 In the Address Book Navigation pane, select the group from which to add participants.
2 In the Address Book List pane, select the participant or participants you want to add to the
conference.
3 Click and hold the left mouse button and drag the selection to the Participants pane of the
conference.
The participants are added to the conference.

Adding a Group from the Address Book to Conferences
You can add a group of participants to a new conference, ongoing conferences, or to Conference
Templates by using the drag-and-drop operation.

To add a group to a new conference or an ongoing conference:
1 In the Address Book Navigation pane, select the group you want to add to the conference.
2 Click and hold the left mouse button and drag the selection to the Participants pane of the
conference.
The participants in the group level and all sub-levels are added to the conference.

Participant Groups
A group is a predefined collection of participants. A group provides an easy way to manage clusters of
participants that are in the same organizational structure and to connect a combination of endpoints to a
conference. For example, if you frequently conduct conferences with the marketing department, you can
create a group called “Marketing Team” that contains the endpoints of all members of the marketing team.
Groups can contain participants and sub-groups. You can define up to ten levels in the “Main” group.

Managing Groups in the Address Book
To manage the groups in the Address Book:
1 In the Address Book Navigation pane, right-click the group you want to manage.
The Groups menu is displayed.

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2 Select one of the following actions:
:

Address Book Navigation
Action

Description

New Group

Creates a new group within the current group.

New Participant

Adds a new participant within the current group.

Copy Group

Copies the current group to be pasted as an additional group.

Paste Group

Places the copied group into the current group. The group name of the
copied group is defined with “Copy” at the end of the group name. This
action is only available after a Copy Group action has been implemented.

Paste Participant

Places the copied participant into the current selected group. This action
is available after a Copy or Cut action was activated when selecting a
single participant or multiple participants.

Paste Participant as New

Pastes as a new participant into the selected group. This paste action
adds “Copy” at the end of the participant name. This action is only
available after a Copy action was activated for a single participant.

Rename Group

Renames the group name.

Delete Group

Deletes the group and all of its members. This action displays a message
requesting confirmation to delete the group and all members connected
with the group.

Additionally, you can drag a group from one location in the Address Book to another location, moving the
group and all its members, including sub-groups, to its new location using the drag-and-drop operation.
Moving a group to a new location can be done in the navigation pane or the list pane.

To drag a group from a location in the address book to another location:
1 Select the group you want to move.
2 Click and hold the left mouse button and drag the selection to the new location. The new location
can be either the “Main” root level or another group level.
The group and all its members (participants and groups) are moved to the new address book location.

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Managing the Address Book
Guidelines
● The multi-level Address Book can only be used in a local configuration on the Collaboration Server.
The hierarchical structure cannot be implemented with the Global Address Book (GAB).
● Up to ten levels can be defined in the hierarchical structure of the Address Book.
● The default name of the root level is “Main”. The “Main” root level cannot be deleted but the root level
name can be modified.
● Address Book names support multilingual characters.
● Participants in the Address Book can be copied to multiple groups. However, only one participant
exists in the Address Book. Groups that contain the same participants refer to the same definition
of the participant entity.

Adding a Participant to the Address Book
Adding participants to the Address Book can be performed by the following methods:
● Directly in the Address Book.
● Moving or saving a participant from an ongoing conference to the Address Book.
When adding dial-out participants to the ongoing conference, the system automatically dials out to the
participants using the Network Service (IP) defined for the connection in the participant properties.

Adding a New participant to the Address Book Directly
You can add a new participant to the “Main” group or to a group in the Address Book. Additionally, you can
add a participant from a new conference, ongoing conference, or Conference Template.

To add a new participant to the Address Book:
1 In the Address Book - Navigation pane, select the group to where you want to add the new
participant.
2 Click the New Participant button ( ) or right-click the group to where you want to add the
participant and select the New Participant option.

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 Alternatively, click anywhere in the List pane and select the New Participant option.
The New Participant - General dialog box opens.

3 Define the following fields:
New Participant - General
Field

Description

Name

Enter the name of the participant or the endpoint as it will be displayed in the
Collaboration Server Web Client.
The Name field can be modified using Unicode encoding.
• English text uses ASCII encoding and can contain the most characters (length
varies according to the field).
• European and Latin text length is approximately half the length of the maximum.
• Asian text length is approximately one third of the length of the maximum.
Maximum field length in ASCII is 80 characters.
The maximum length of text fields varies according to the mixture of character sets
used (Unicode and ASCII).
This field may not be left blank. Duplicate participant names, comma, and semi-colon
characters may not be used in this field.
This name can also become the endpoint name that is displayed in the video layout.
For more details about endpoint (site) names, see the Polycom® RealPresence
Collaboration Server 800s / Virtual Edition Getting Started Guide, Audio and Visual
Indications (AVC CP Conferencing)”.
Note: This field is displayed in all tabs.

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New Participant - General (Continued)
Field

Description

Endpoint Website

Click the Endpoint Website hyperlink to connect to the internal website of the
participant’s endpoint. It enables you to perform administrative, configuration and
troubleshooting activities on the endpoint.
The connection is available only if the IP address of the endpoint’s internal site is
defined in the Website IP Address field.
i

Dialing Direction

Select the dialing direction:
• Dial-in – The participant dials in to the conference. This field applies to IP
participants only.
• Dial-out – The MCU dials out to the participant.

Type

The network communication protocol used by the endpoint to connect to the
conference: H.323, or SIP or ISDN/PSTN.
The fields in the dialog box change according to the selected network type.

IP Address
(H.323 and SIP)

Enter the IP address of the participant’s endpoint.
• For H.323 participant define either the endpoint IP address or alias.
• For SIP participant define either the endpoint IP address or the SIP address.
For Collaboration Servers registered to a gatekeeper, the Collaboration Server can be
configured to dial and receive calls to and from H.323 endpoints using the IP address
in the event that the Gatekeeper is not functioning.

Alias Name/Type
(H.323 Only)

If you are using the endpoint’s alias and not the IP address, first select the type of
alias and then enter the endpoint’s alias:
• H.323 ID (alphanumeric ID)
• E.164 (digits 0-9, * and #)
• Email ID (email address format, e.g. abc@example.com)
• Participant Number (digits 0-9, * and #)
Notes:
• Although all types are supported, the type of alias is dependent on the gatekeeper’s
capabilities. The most commonly supported alias types are H.323 ID and E.164.
• This field is used to enter the Entry Queue ID, target Conference ID and
Conference Password when defining a cascaded lin.
• Use of the E.164 Number is dependent on the setting of the
REMOVE_IP_IF_NUMBER_EXISTS System Flag. For more information see
Substituting E.164 Number in Dial String.

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New Participant - General (Continued)
Field

Description

SIP Address/Type
(SIP Only)

Select the format in which the SIP address is written:
• SIP URI - Uses the format of an E-mail address, typically containing a user name
and a host name: sip:[user]@[host]. For example, sip:dan@polycom.com.
Note: If the SIP Address field contains an IPv6 address, it must be surrounded by
square brackets, for example, [::1].
• TEL URI - Used when the endpoint does not specify the domain that should
interpret a telephone number that has been input by the user. Rather, each domain
through which the request passes would be given that opportunity.
For example, a user in an airport might log in and send requests through an outbound
proxy in the airport. If the users enters “411” (this is the phone number for local
directory assistance in the United States), this number needs to be interpreted and
processed by the outbound proxy in the airport, and not by the user's home domain.
In this case, tel: 411 is the correct choice.

Endpoint Website
IP Address
(IP only)

Enter the IP address of the endpoint’s internal site to enable connection to it for
management and configuration purposes.
This field is automatically completed the first time that the endpoint connects to the
Collaboration Server. If the field is blank it can be manually completed by the system
administrator. The field can be modified while the endpoint is connected

Audio Only

Select this check box to define the participant as a voice participant, with no video
capabilities.

Extension/Identifier
String

Dial-out participants that connect to an external device such as Cascaded Links or
Recording Links may be required to enter a conference password or an identifying
string to connect. Enter the required string as follows:
[p]…[p][string]
For example: pp4566#
p - optional - indicates a pause of one second before sending the DTMF string. Enter
several concatenated [p]s to increase the delay before sending the string. The
required delay depends on the configuration of the external device or conference IVR
system.
String - enter the required string using the digits 0-9 and the characters * and #. The
maximum number of characters that can be entered is identical to the H.323 alias
length.
If the information required to access the device/conference is composed of several
strings, for example, the conference ID and the conference password, this information
can be entered as one string, where pauses [p] are added between the strings for the
required delays, as follows:
[p]…[p][string][p]…[p] [string]...
For example: p23pp*34p4566#

Extension/Identifier
String

The Collaboration Server automatically sends this information upon connection to the
destination device/conference. The information is sent by the Collaboration Server as
DTMF code to the destination device/conference, simulating the standard IVR
procedure.

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4 Usually, additional definitions are not required and you can use the system defaults for the
remaining parameters. In such a case, click OK.
To modify the default settings for advanced parameters, click the Advanced tab.
5 Define the following Advanced parameters:

New Participant - Advanced
Field

Description

Video Bit Rate / Auto
(IP Only)

The Auto check box is automatically selected to use the Line Rate defined for
the conference.
Note: This check box cannot be cleared when defining a new participant during
an ongoing conference.
To specify the video rate for the endpoint, clear this check box, and then select
the required video rate.

Video Protocol

Select the video compression standard that will be forced by the MCU on the
endpoint when connecting to the conference: H.261, H.263, H.264 or RTV.
Select Auto to let the MCU select the video protocol according to the endpoint’s
capabilities.

Resolution

The Auto check box is automatically selected to use the Resolution defined for
the conference.
To specify the Resolution for the participant, select the required resolution from
the drop-down menu.

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New Participant - Advanced (Continued)
Field

Description

Encryption

Select whether the endpoint uses encryption for its connection to the
conference.
Auto (default setting) indicates that the endpoint will connect according to the
conference encryption setting.

Cascaded (IP Only)

If this participant is used as a link between conferences select:
• Slave, if the participant is defined in a conference running on a Slave MCU.
• Master, if the participant is defined in a conference running on the Master
MCU.
It enables the connection of one conference directly to another conference using
an H.323 connection only. The conferences can run on the same MCU or
different MCU’s. For more information, see Basic Cascading using IP Cascaded
Link.

Precedence Domain
Name
(Dial-out SIP Only)

When Multi Level Precedence and Preemption is used, this is the Precedence
Domain Name for the participant.
For more information see MLPP (Multi Level Precedence and Preemption).

Precedence Level
(Dial-out SIP Only)

When Multi Level Precedence and Preemption is used, this is the Precedence
Level for the participant
For more information see MLPP (Multi Level Precedence and Preemption).

AGC

The Audio Gain Control (AGC) protocol that reduces noises is enabled by
default for the participants.
Clear this check box to disable the AGC feature.

6 To add general information about the participant, such as e-mail, company name, and so on, click
the Information tab and type the necessary details in the Info 1-4 fields. Text in the info fields can
be added in Unicode format (length: 31 characters).
7 Click OK.
The new participant is added to the selected group in the address book.

Substituting E.164 Number in Dial String
Between the time a conference is scheduled and when it becomes active, the IP of an endpoint may change,
especially in an environment that uses DHCP. The MCU can be set to ignore the IP address of a participant
when the conference starts. Instead, the alternative E.164 number will be used.
The flag, REMOVE_IP_IF_NUMBER_EXISTS controls this option. This flag must be manually added to
change its value. The values of this flag are:
● YES (default) - The IP address of an endpoint will be ignored if an E.164 number (or other) exists.
● NO - The IP address of an endpoint will be used.

Guidelines for Substituting E.164 Number in Dial String
● When this feature is enabled, the IP address field of participants in scheduled conferences and
conference templates will be empty.
● In order for the MCU to ignore the IP of H.323 participants, the following requirements must be met:
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 A gatekeeper must be defined.
 The alias of the participant must be defined.
 The alias type must be defined (not set to None).
● If an H.323 gatekeeper is defined but is not connected, the MCU will fail to connect to H.323 dial-out
participants.
● In order for the MCU to ignore the IP of SIP participants, the following requirements must be met:
 A SIP proxy must be defined.
 The SIP address must be defined.
● If a SIP proxy is defined but is not connected, the MCU will fail to connect to SIP dial-out participants.

Adding a Participant from an Ongoing Conference to the Address
Book
You can add a participant to the Address Book directly from an ongoing conference.
When adding a participant to the address book from a new conference, Participants list of an
ongoing conference or Conference Template, the participant is always added to the “Main” group.

To add a participant from the conference to the Address Book:
1 During an ongoing conference, select the participant in the Participant pane, and either click the
Add Participant to Address Book button ( ), or right-click and select Add Participant to
Address Book.
The participant is added to the Address Book.
Alternatively, you could:
a Double-click the participant’s icon, or right-click the participant icon and click Participant
Properties.

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The Participant Properties window opens.

b Click the Add to Address Book button.
If the participant name is already listed in the All Participants list, an error message is displayed. In
such a case, change the name of the participant before adding the participant to the address book.

Modifying Participants in the Address Book
When required, you can modify the participant’s properties.

To modify participant properties in the Address Book:
1 In the Address Book - Navigation pane, select the group to where the participant to modify is
listed.

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2 In the Address Book - List pane, double-click the participant’s icon.
The Participant’s Properties window is displayed.

3 Modify the necessary properties in the window, such as dialing direction, communication protocol
type, and so on. You can modify any property in any of the three tabs: General, Advanced and Info.
4 Click OK.
The changes to the participant’s properties are updated.

Deleting Participants from the Address Book
To delete participants from the Address Book:
1 In the Address Book - Navigation pane, select the group where the participant to delete is listed.
2 In the Address Book - List pane, either select the participant to delete, and then click the Delete
Participant ( ) button, or right-click the participant icon and then click the Delete Participant
option.

3 A confirmation message is displayed depending on the participant’s assignment to groups in the
address book:
a When the participant belongs to only one group: click Yes to permanently delete the participant
from the address book.
b When the participant belongs to multiple groups, a message is displayed requesting whether to
delete the participant from the Address Book or from the current selected group. Select:
 Current group to delete the participant from the selected group

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 Address Book to permanently delete the participant from the address book (all groups).
Click OK to perform the delete operation, or Cancel to exit the delete operation.

Copying or Moving a Participant
You can copy or move a participant from one group to another group using the Copy, Cut, and Paste
options. A participant can belong to multiple groups. However, there is only one entity per participant.
Groups that contain the same participants refer to the same definition of the participant entity. Alternatively,
you can drag a participant from one location in the Address Book to another location, moving the
participant to its new location using the drag-and-drop operation.
The cut and copy actions are not available when selecting multiple participants.

To copy or move a participant to another group:
1 In the Address Book - Navigation pane, select the group from where to copy the participant.
2 In the Address Book - List pane, select the participant you want to copy.
3 Right-click the selected participant, and select one of the following functions from the drop-down
menu:

Copy / Move Participant
Function

Description

Copy Participant

Copies the participant to be pasted into an additional group.

Cut Participant

Moves the participant from the current group to a different group. Alternatively,
you can move a participant to another location by dragging the participant to the
new location.

4 In the Address Book - Navigation pane, navigate and select the group in which you want to paste
the participant.
5 Right-click the selected group, and click one of the following Paste functions from the drop-down
menu:

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Paste Participant
Function

Description

Paste Participant

Creates a link to the participant entity in the pasted location.

Paste Participant as New

Pastes as a new participant into the selected group. This paste action adds
“Copy” to the end of the participant name.

The Paste functions are only available after a Copy or Cut action has been implemented.

To drag a participant from an address book group to another group:
1 Select the participant or participants you want to move.
2 Click and hold the left mouse button and drag the selection to the new group.
The participants are moved to the new address book group.

Searching the Address Book
You can search the Address Book for a participant’s name or a group name only on the currently selected
group/level.

To search for participants or groups in the current selected level:
1 In the Address Book - Navigation pane, select the group/level within to run the search.
2 In the Address Book toolbar, activate the search option by clicking the Find field.
The field clears and a cursor appears indicating that the field is active.
3 Type all or part of the participant’s name or group name and click the search button.

The closest matching participant entries are displayed and the Active Filter indicator turns on.

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Filtering the Address Book
The entries in an address book group can be filtered to display only the entries (participants or groups) that
meet criteria that you specify and hides entries that you do not want displayed. It enables you to select and
work with a subset of Address Book entries.
You can filter by more than one column, by adding additional filters (columns).
The filter applies to the displayed group. If All Participants option is selected, it applies to all the listed
participants.
Filtering can be done using:
● A predefined pattern
● Customized pattern
When you use the Find dialog box to search filtered data, only the data that is displayed is searched; data
that is not displayed is not searched. To search all the data, clear all filters.

Filtering Address Book Data Using a Predefined Pattern
To filter the data in an address book group:
1 In the Address Book - Navigation pane, select the group to filter.
2 In the Address Book - List pane, in the column that you want to use for filtering, click the filter ( )
button.
A drop-down menu is displayed containing all the matching patterns that can be applied to the
selected field.

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3 Click the matching pattern to be applied.
The filtered list is displayed with a filter indicator ( ) displayed in the selected column heading.
Example: If the user selects 172.21.41.104 as the matching pattern, the filtered group in the Address
Book is displayed as follows:

Filtering Address Book Data Using a Custom Pattern
To filter the data in an address book group:
1 In the Address Book - Navigation pane, select the group to filter.
2 In the Address Book - List pane, in the column that you want to use for filtering, click the filter ( )
button.

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3 Select the (Custom) option from the drop-down list.

The Custom Filtering dialog box opens.

4 In the Condition - Column text matches field, enter the filtering pattern.
For example, to list only endpoints that include the numerals 41 in their name, enter 41.

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5 Optional. Click the Add Condition button to define additional filtering patterns to further filter the list
and fine tune your search.
To clear a filtering pattern, click the Clear Condition button.
The filtered list is displayed with an active filter (blue) indicator ( ) displayed in the selected column
heading.
For example, if the filtering pattern is 41, the participants list includes all the endpoints that contain
the numerals 41 in their name.

Clearing the Filter
To clear the filter and display all entries:
1 In the filtered Address Book column heading, click the Active Filter indicator.
The pattern matching options menu is displayed.
2 Click (All).

The filter is deactivated and all the group/level entries are displayed.

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Obtaining the Display Name from the Address Book
The MCU can be configured to replace the name of the dial-in participant as defined in the endpoint (site
name) with the name defined in the Address Book.
In this process, the system retrieves the data (name, alias, number or IP address) of the dial-in participant
and compares it first with the conference defined dial-in participants and if the endpoint is not found, it then
searches for the endpoint with entries in the address book. After a match is found, the system displays the
participant name as defined in the address book instead of the site name, in both the video layout and the
Collaboration Server Web Client/Manager.
The system compares the following endpoint data with the address book entries:
● For H.323 participants, the system compares the IP address, Alias, or H.323 number.
● For SIP participants, the system compares the IP address or the SIP URI.

Guidelines for Obtaining the Display Name from the Address Book
● Only Users with Administrator and Operator Authorization Levels are allowed to enable and disable
the Obtain Display Name from Address Book feature.
● This feature is supported for IPv4 participants only .

Enabling and Disabling the Obtain Display Name from Address Book Feature
The Obtain Display Name from Address Book option can be enabled for all participants connecting to the
MCU if the name of the participants are defined in the Address Book.

To enable or disable the Obtain Display Name from Address Book option:
1 On the Collaboration Server main menu bar, click Setup > Customize Display Settings >
Ongoing Conferences.
The Ongoing Conferences dialog box is displayed.

2 Select the Obtain display name from address book check box to enable the feature or clear the
check box to disable the feature.
3 Click OK.

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Importing and Exporting Address Books
Address Books are proprietary Polycom data files that can only be distributed among Collaboration Server
units. The Address Books are exported in XML format, which are editable offline. If no name is assigned to
the exported Address Book, the default file name is:
EMA.DataObjects.OfflineTemplates.AddressbookContent_.xml

Exporting an Address Book
To Export an Address Book:
1 In the Address Book pane, click the Export Address Book (
in the pane, and click Export Address Book.

) button, or right-click an empty area

The Export Address Book dialog box is displayed.

2 Enter the desired path, or click the Browse button.
3 In the Save Address Book dialog box, select the directory to save the file. You may also rename
the file in the File Name field.
4 Click Save.
You will return to the Export File dialog box.
5 Click OK.
The exported Address Book is saved in the selected folder in XML format.

Importing an Address Book
To Import and Address Book:
1 In the Address Book pane, click the Import Address Book (
in the pane, and then click Import Address Book.

) button, or right-click an empty area

The Import Address Book dialog box is displayed.

2 Enter the path from which to import the Address Book, or click the Browse button.
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3 In the Open dialog box navigate to the desired Address Book file (in XML format) to import.
When importing an Address Book, participants with exact names in the current Address Book will be
overwritten by participants defined in the imported Address Book.

4 Click Open.
You will return to the Import File dialog box.
5 Click OK.
The Address Book is imported and a confirmation message is displayed at the end of the process.
6 Click Close.

Integrating the Global Address Book (GAB) of
Polycom RealPresence Resource Manager or Polycom
CMA with the Collaboration Server
The RealPresence Resource Manager or Polycom CMA application includes a Global Address Book (GAB)
with all registered endpoints. This address book can be used by the Collaboration Server users to add
participants to conferences.

Guidelines for integrating with the Global Address Book of Polycom
RealPresence Resource Manager or Polycom CMA
● The Collaboration Server can use only one address book at a time. After you integrate the Polycom
RealPresence Resource Manager or Polycom CMA with the Polycom Collaboration Server, the CMA
address book replaces the Collaboration Server internal address book.
● The Collaboration Server uses the RealPresence Resource Manager or Polycom CMA address book
in read-only mode. You can only add or modify CMA address book entries from the RealPresence
Resource Manager or Polycom CMA.
The Collaboration Server acts as a proxy to all address book requests between the Collaboration
Server Web Client and the CMA. Ensure that firewall and other network settings allow the
Collaboration Server access to the CMA server.

To Integrate the RealPresence Resource Manager or Polycom CMA Global Address Book
(GAB) with the Collaboration Server:
RealPresence Resource Manager or Polycom CMA Side
1 In the RealPresence or Polycom application, manually add the Polycom Collaboration Server
system to the RealPresence Resource Manager or Polycom CMA system as directed in the
RealPresence Resource Manager or Polycom CMA Operations Guide.

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2 In the RealPresence Resource Manager or Polycom CMA application, add a user or use an existing
user for Collaboration Server login as directed in the RealPresence Resource Manager or Polycom
CMA Operations Guide.
Write down the User Name and Password as they will be used later to define the Collaboration Server
connection to the RealPresence Resource Manager or Polycom CMA Global Address Book.

Collaboration Server Side
1 On the Collaboration Server menu, click Setup > System Configuration.
The System Flags - MCMS_PARAMETERS_USER dialog box opens.

2 Modify the values of the flags in the table below.
For more information, see Modifying System Flags.
System Flags for CMA Address Book Integration
Flag

Description

EXTERNAL_CONTENT_
DIRECTORY

The Web Server folder name. Change this name if you have changed the
default names used by the RealPresence Resource Manager or Polycom CMA
application.
Default: /PlcmWebServices

EXTERNAL_CONTENT_
IP

Enter the IP address of the RealPresence Resource Manager or Polycom CMA
server. For example: 172.22.185.89.
This flag is also the trigger for replacing the internal Collaboration Server
address book with the RealPresence Resource Manager or Polycom CMA
Global Address Book (GAB).
Leave this flag blank to disable address book integration with the RealPresence
Resource Manager or Polycom CMA server.

EXTERNAL_CONTENT_
PASSWORD

The password associated with the user name defined for the Collaboration
Server in the RealPresence Resource Manager or Polycom CMA server.

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System Flags for CMA Address Book Integration
Flag

Description

EXTERNAL_CONTENT_
USER

The login name defined for the Collaboration Server in the RealPresence
Resource Manager or Polycom CMA server defined in the format:
domain name/user name.

3 Click OK to complete the definitions.
4 When prompted, click Yes to reset the MCU and implement the changes to the system
configuration.

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In the RealPresence CloudAxis Solution, Reservations are scheduled in the RealPresence Resource
Manager component and should not be scheduled directly in the Polycom® RealPresence®
Collaboration Server 800s and Polycom® RealPresence® Collaboration Server Virtual Edition
component.

The Reservations option enables users to schedule conferences. These conferences can be launched
immediately or become ongoing, at a specified time on a specified date.
Scheduling a conference reservation requires definition of conference parameters such as the date and time
at which the conference is to start, the participants and the duration of the conference.
Scheduled conferences (Reservations) can occur once or repeatedly, and the recurrence pattern can vary.
The maximum number of reservations per Collaboration Server is 2000.

Guidelines for Scheduling Reservations
System
● By default, the Scheduler is enabled by a System Flag. The flag prevents potential scheduling
conflicts from occurring as a result of system calls from external scheduling applications such as the
Polycom® RealPresence® Resource Manager®, ReadiManager®, SE200, Polycom CMA™ and
others via the API.
If an external scheduling application is used, the flag INTERNAL_SCHEDULER must be manually
added to the System Configuration and its value must be set to NO.
For more information see Modifying System Flags.

Resources
● System resources are calculated according to the Collaboration Server’s license. For more
information, see Forcing Video Resource Allocation to CIF Resolution.
● System resource availability is partially checked when reservations are created:
 If a conference duration extension request is received from an ongoing conference, the request
is rejected if it would cause a resource conflict.
 If several reservations are scheduled to be activated at the same time and there are not enough
resources for all participants to be connected:
 The conferences are activated.
 Participants are connected to all the ongoing conferences until all system resources are used
up.
● If sufficient resources are not available in the system and a scheduled Reservation cannot be
activated, the Reservation is deleted from the schedule.
● Resources are reserved for participants at the highest video resolution supported by the Line Rate
specified in the conference Profile and up to the maximum system video resolution specified by the
Resolution Configuration dialog box.
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● When a new Reservation is created in the Reservation Calendar, the effect of the new Reservation
(including its recurrences) on available resources is checked. If resource deficiencies are found an
error message is displayed.
Defined dial-in or dial-out participants, Meeting Rooms, Entry Queues and new connections to
Ongoing conferences are not included in the resources calculation.

Reservations
● A Reservation that has been activated and becomes an ongoing conference is deleted from the
Reservation Calendar list.
● The maximum number of concurrent reservations is 80. Reservations with durations that overlap (for
any amount of time) are considered to be concurrent.
● System resource availability is partially checked when reservations are created:
 If a conference duration extension request is received from an ongoing conference, the request
is rejected if it would cause a resource conflict.
 If several reservations are scheduled to be activated at the same time and there are not enough
resources for all participants to be connected:
 The conferences are activated.
 Participants are connected to all the ongoing conferences until all system resources are used
up.
● A scheduled Reservation cannot be activated and is deleted from the schedule if:
 An Ongoing conference has the same Numeric ID.
 Sufficient resources are not available in the system.
● If a problem prevents a Reservation from being activated at its schedule time, the Reservation will
not be activated at all. This applies even if the problem is resolved during the Reservation’s
scheduled time slot.
● A Profile that is assigned to a Reservation cannot be deleted.
● Reservations are backed up and restored during Setup > Software Management >Backup
/Restore Configuration operations. For more information see Software Management.
● All existing reservations are erased by the Standard Restore option of the Administration > Tools >
Restore Factory Defaults procedure.
● Reservations can also be scheduled from Conference Templates. For more information see
Scheduling a Reservation From a Conference Template.

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Using the Reservation Calendar
To open the Reservation Calendar:
● In the RMX Management pane, click the Reservation Calendar button (

).

Toolbar Buttons
The toolbar buttons functions are described in the table below.
Reservations – Toolbar Buttons
Button

Description

New Reservation

Create a new reservation. The date and time of the new reservation is
set according to the highlighted blocks on the Reservation Calendar.
Click to delete the selected reservation.

Delete Reservation

Back

Next

Today

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Click to show the previous day or week, depending on whether Show
Day or Show Week is the selected.
Click to show the next day or week, depending on whether Show Day
or Show Week is the selected.
Click to show the current date in the Reservation Calendar in either
Show Day or Show Week view.

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Button

Show Week

Description
Change the calendar view to weekly display, showing a calendar week:
Sunday through Saturday
Click this button to show the day containing the selected time slot.

Show Day
Click to change to List View and display a list of all reservations.
Reservations List
Used to search for reservations by Display Name. (Available in
Reservations List view only).

Reservations Views
The Reservation Calendar list has the following views available:
● Week
● Day
● Today
● List
In all views the Main Window List Pane header displays the total number of reservations in the system.

Week View
By default the Reservation Calendar is displayed in Week view with the current date highlighted in orange.

Day View
A single day is displayed.

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Today View
The current date (Today), highlighted in orange, can be viewed in both Week View and Day View.

List View
List View does not have a calendar based format.

All Reservations are listed by:
•

Display Name

•

End Time

•

ID

•

Status

•

Internal ID

•

Conference Password

•

Start Time

•

Profile

The Reservations can be sorted, searched and browsed by any of the listed fields.

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Changing the Calendar View
To change between Week and Day views:
● In Week View: In the Reservation Calendar toolbar, click Show Day (

) to change to Day View.

or
In Day View: In the Reservation Calendar toolbar, click Show Week (

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To view Today (the current date):
● In Week View or Day View, in the Reservation Calendar toolbar, click the Today (
the current date displayed within the selected view.

) button to have

To change to List View:
1 In the Reservation Calendar toolbar, click, the Reservations List (

) button.

The Reservations List is displayed.

2 Optional. Sort the data by any field (column heading) by clicking on the column heading.
A
or
symbol is displayed in the column heading indicating that the list is sorted by this field, as
well as the sort order.
3 Optional. Click on the column heading to toggle the column’s sort order.

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To return to Calendar View:
● In the Reservation Calendar toolbar, click any of the buttons (Show Week/Show Day/Today) to
return to the required Reservation Calendar view.

Scheduling Conferences Using the Reservation
Calendar
In the RealPresence CloudAxis Solution, Reservations are scheduled in the RealPresence Resource
Manager component and should not be scheduled directly in the RealPresence Collabporatio Server
Virtual Edition component.

Creating a New Reservation
There are three methods of creating a new reservation:
● Method I – Creating a reservation with default duration of 1 hour
● Method II – Creating a reservation with default duration of ½ hour
● Method III – Interactively define the reservation duration
Each method requires the selection of a starting time slot in the Reservation Calendar. The default time slot
is the current half-hour period of local time.
In all views, if the New Reservation (
) button is clicked without selecting a starting time slot or if a time
slot is selected that is in the past, the Reservation becomes an Ongoing conference immediately and is not
added to the Reservations calendar.

After selecting a starting time slot in the Reservation Calendar you can create a reservation with a default
duration derived from the creation method used or by interactively defining the duration of the reservation.

Method I – To create a reservation with default duration of 1 hour:
● In the Reservation Calendar toolbar, click the New Reservation (
of 1 hour duration.

) button to create a reservation

Method II – To create a reservation with default duration of ½ hour:
● Right-click and select New Reservation to create a reservation of ½ hour default duration.

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Method III – To interactively define the duration:
1 In the calendar, click & drag to expand the time slot to select the required Date, Start Time and
Duration for the reservation.
2 In the Reservation Calendar toolbar, click the New Reservation (
New Reservation.

) button or right-click and select

Example: The following click & drag sequence would select a reservation for Tuesday, August 12,
2008, starting at 01:00 with a duration of 4 hours.

The duration of reservations created by any of the above methods can be modified in the Scheduler
tab of the New Reservation dialog box.

To create a new reservation:
1 Open the Reservation Calendar.
2 Select a starting time slot.

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3 Create the reservation using one of the three methods described above.
The New Reservation – General tab dialog box opens.

All the fields are the same as for the New Conference – General dialog box, described in the
Polycom® RealPresence Collaboration Server 800s / Virtual Edition Getting Started Guide, General
Tab.
New Reservation – Reserved Resources

When a Conference Profile is assigned to a Meeting Room or a Reservation, the Profile’s parameters
are not embedded in the Reservation, and are taken from the Profile when the reservation becomes an
ongoing conference. Therefore, any changes to the Profile parameters between the time the
Reservation or Meeting Room was created and the time that it is activated (and becomes an ongoing
conference) will be applied to the conference.
If the user wants to save the current parameters, a different Profile with these parameters must be
assigned, or a different Profile with the new parameters must be created.

4

Click the Schedule tab.

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5 Adjust the new reservation’s schedule by modifying the fields as described in the table below.
New Reservation – Schedule Tab
Field

Description

Start Time

Select the Start Time
of the Reservation.

End Time

Select the End Time of
the Reservation.

•

•

•

•

•

•

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The Start/End Times of the Reservation are
initially taken from the time slot selected in the
Reservation Calendar.
The Start/End Times can be adjusted by typing in
the hours and minutes fields or by clicking the
arrow buttons.
The Start/End dates can be adjusted by typing in
the date field or by clicking the arrow buttons or
using the calendar.
The start time of all the reservations can be
manually adjusted in one operation. For more
information see Adjusting the Start Times of all
Reservations.
End Time settings are initially calculated as Start
Time + Duration. End Time settings are
recalculated if Start Time settings are changed.
Changes to End Time settings do not affect Start
Time settings. However, the Duration of the
Reservation is recalculated.

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Field

Description

Recurring
Meeting

Select this option to set up a Recurring Reservation - a series of Reservations
to be repeated on a regular basis.
To create a recurring reservation, you must define a time period and a
recurrence pattern of how often the Reservation should occur: Daily, Weekly or
Monthly.

Recurrence
Pattern

Daily

If Daily is selected, the system automatically selects all the days of
the week. To de-select days (for example, weekends) clear their
check boxes.

Weekly

If Weekly is selected, the system automatically selects the day of
the week for the Reservation from the day selected in the
Reservation Calendar.
You can also define the recurrence interval in weeks. For example,
if you want the reservation to occur every second week, enter 2 in
the Recur every _ week(s) field.
To define a twice-weekly recurring Reservation, select the check
box of the additional day of the week on which the Reservation is to
be scheduled and set the recurrence interval to 1.

Monthly

If Monthly is selected, the system automatically selects the day of
the month as selected in the Reservation Calendar. You are
required to choose a recurrence pattern:
• Day (1-31) of every (1-12) month(s) - Repeats a conference on
a specified day of the month at a specified monthly interval. For
example, if the first Reservation is scheduled for the 6th day of
the current month and the monthly interval is set to 1, the
monthly Reservation will occur on the 6th day of each of the
following months.
• The (first, second,...,last) (Sun-Sat) of x month(s) - Repeats
a Reservation in a particular week, on a specified day of the
week at the specified monthly interval. For example, a recurrent
meeting on the third Monday every second month.

A series of Reservations can be set to end after a specified number of occurrences or by a
specific date. Select one of the following methods of terminating the series of Reservations:
End After

End After: x Occurrences - Ends a recurring series of Reservations after a
specific number (x) of occurrences.
Default: 1
(Leaving the field blank defaults to 1 occurrence.)

End by Date

End By Date: mm/dd/yyyy - Specifies a date for the last occurrence of the
recurring series of Reservations. The End By Date value can be adjusted by
typing in the date field or by clicking the arrow button and using the calendar
utility.
Default: Current date.

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6 Click the Participants tab.

The fields are the same as for the New Conference – Participants dialog box, described in the
Polycom® RealPresence Collaboration Server 800s / Virtual Edition Getting Started Guide,
Participants Tab.
Participant properties are embedded in the conferencing entity and therefore, if the participant
properties are modified in the Address Book (or Meeting Rooms) after the Reservation has been
created they are not applied to the participant when the Reservation is activated.

7 Optional. Add participants from the Participants Address Book.
For more information see Meeting Rooms and the Polycom® RealPresence Collaboration Server
800s / Virtual Edition Getting Started Guidee, To add participants from the Address Book: .
Between the time a conference is scheduled and when it becomes active, the IP of an endpoint may
change, especially in an environment that uses DHCP. The MCU can be set to ignore the IP address of
a participant when the conference starts. Instead, the alternative E.164 number will be used instead of
the IP address. For more information see Substituting E.164 Number in Dial String .

8 Optional. Add information to the reservation.
Information entered in the Information tab is written to the Call Detail Record (CDR) when the
reservation is activated. Changes made to this information before it becomes an ongoing conference
will be saved to the CDR.
For more information see the Polycom® RealPresence Collaboration Server 800s / Virtual Edition
Getting Started Guide, Information Tab.

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9 Click OK.
The New Reservation is created and is displayed in the Reservation Calendar.
If you create a recurring reservation all occurrences have the same ID.The series number (_0000n)
of each reservation is appended to its Display Name.
Example:
Conference Template name: Sales
Display Name for single scheduled occurrence:Sales
If 3 recurrences of the reservation are created:
Display Name for occurrence 1: Sales_00001
Display Name for occurrence 2: Sales_00002
Display Name for occurrence 3: Sales_00003

Managing Reservations
Reservations can be accessed and managed via all the views of the Reservations List.

Guidelines
● The Recurrence Pattern fields in the Schedule tab that are used to create multiple occurrences of a
Reservation are only displayed when the Reservation and its multiple occurrences are initially
created.
● As with single occurrence Reservations, only the Duration, Start Time and End Time parameters of
multiple occurrence reservations can be modified after the Reservation has been created.
● A single occurrence Reservation cannot be modified to become a multiple occurrence reservation.
● Reservations can only be modified one at a time and not as a group.
● If Reservations were created as a recurring series, the system gives the option to delete them
individually, or all as series.

Viewing and Modifying Reservations
Reservations can be viewed and modified by using the Week and Day views of the Reservations Calendar
or by using the Reservation Properties dialog box.

Using the Week and Day views of the Reservations Calendar
In the Week and Day views each Reservation is represented by a shaded square on the Reservation
Calendar. Clicking on a Reservation selects the Reservation. A dark blue border is displayed around the
edges of the Reservation indicating that it has been selected.
The Start Time of the Reservation is represented by the top edge of the square while the End Time is
represented by the bottom edge.

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The cursor changes to a vertical double arrow (
square.

) when it is moved over the top and bottom sides of the

To move the Reservation to another time slot:
1 Select the Reservation.
2 Hold the mouse button down and drag the Reservation to the desired time slot.
3 Release the mouse button.

To change the Reservation’s Start time:
1 Select the Reservation.
2 Move the mouse over the top edge of the Reservation’s square.
3 When the cursor changes to a vertical double arrow (
edge to the desired Start Time.

) hold the mouse button down and drag the

4 Release the mouse button.

To change the Reservation’s End time:
1 Select the Reservation.
2 Move the mouse over the bottom edge of the Reservation’s square.
3 When the cursor changes to a vertical double arrow (
edge to the desired End Time.

) hold the mouse button down and drag the

4 Release the mouse button.

To View or Modify Reservations using the Reservation Properties dialog box:
1 In the Reservations List, navigate to the reservation (or its recurrences) you want to view, using the
Show Day, Show Week, Today, Back, Next or List buttons.
2 Double-click, or right-click and select Reservation Properties, to select the reservation to be
viewed or modified.
The Reservation Properties – General dialog box opens.
3 Select the tab(s) of the properties you want to view or modify.
4 Optional. Modify the Reservation Properties.

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5 Click OK.
The dialog box closes and modifications (if any) are saved.

Adjusting the Start Times of all Reservations
When utilizing GMT offset (for example, Daylight Saving Time change), the start time of the reoccurring
reservations scheduled before the Collaboration Server time change are not updated accordingly (although
their start times appear correctly in the Reservations list, when checking the reservation properties the start
time is incorrect).
Following the Collaboration Server time change, the start time of all reoccurring reservations must be
manually adjusted in one operation.
Using this option, the start times of all reservations currently scheduled on the Collaboration Server are
adjusted with the same offset.

To adjust the reoccurring reservations start time after the GMT Offset has been changed
for Daylight Saving Time (DST) or a physical move:.
Adjustment of Reservation Time should only be performed after adjustment of Collaboration Server
Time is completed as a separate procedure.

1 On the Collaboration Server menu, click Setup > RMX Time.
The RMX Time dialog box opens.
2 Click the Adjust Reservations Time button.

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The Adjust Reservations Time dialog box opens.

3 Click the arrows of the Offset - Hours box to indicate the number of hours to add or subtract from the
current start time; a positive value indicates adding time, while minus (-) indicates subtracting time.
4 Click the arrows of the Offset - minutes box to indicate the number of minutes to add or subtract
from the current start time of the reservations. Increments or decrements are by 15 minutes.
For example, to subtract 30 minutes from the start time of all the reservation, enter 0 in the hours box,
and -30 in the minutes box.
To add one hour and 30 minutes to the start time, enter 1 in the hours box and 30 in the minutes box.
5 Click the Adjust button to apply the change to all the reoccurring reservations currently scheduled
on the Collaboration Server.
When adjusting the start time of 1000 - 2000 reservations, an “Internal communication error” message
may appear. Ignore this message as the process completes successfully.

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Deleting Reservations
To delete a single reservation:
1 In the Reservations List, navigate to the reservation you want to delete, using the Show Day, Show
Week, Today, Back, Next or List buttons.
2 Click to select the reservation to be deleted.
3 Click the Delete Reservation ( ) button.
or
Place the mouse pointer within the Reservation block, right-click and select Delete Reservation.
4 Click OK in the confirmation dialog box.
The Reservation is deleted.

To delete all recurrences of a reservation:
1 In the Reservations List, navigate to the Reservation or any of its recurrences, using the Show Day,
Show Week, Today, Back, Next or List buttons.
2 Click the Delete Reservation ( ) button.
or
Place the mouse pointer within the Reservation or any of its recurrences, right-click and select
Delete Reservation.
A confirmation dialog box is displayed.

3 Select Delete the series.
4 Click OK.
All occurrences of the Reservation are deleted.

Searching for Reservations using Quick Search
Quick Search is available only in List View. It enables you to search for Reservations by Display Name.

To search for reservations:
1 In the Reservation Calendar toolbar, click in the Quick Search field.
The field clears and a cursor is displayed indicating that the field is active.

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2 Type all or part of the reservation’s Display Name into the field and click Search.
The closest matching Reservation entries are displayed.

3 Optional. Double- click the Reservation’s entry in the list to open the Reservations Properties dialog
box to view or modify the Reservation.
or
Right -click the Reservation’s entry in the list and select a menu option to view, modify or delete the
Reservation.

To clear the search and display all reservations:
1 Clear the Quick Search field.
2 Click Search.
All Reservations are displayed.

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Operator Assistance & Participant Move
User assistance to participants is available when:
● Participants have requested individual help (using *0 DTMF code) during the conference.
● Participants have requested help for the conference (using 00 DTMF code) during the conference.
● Participants have problems connecting to conferences, for example, when they enter the wrong
conference ID or password.
In addition, the user (operator) can join the ongoing conference and assist all conference participants.
Operator assistance is available only when an Operator conference is running on the MCU.
The Operator conference offers additional conference management capabilities to the Collaboration Server
users, enabling them to attend to participants with special requirements and acquire participant details for
billing and statistics. This service is designed usually for large conferences that require the personal touch.
In the RealPresence CloudAxis Solution, the Conference Profiles are defined in the RealPresence
DMA system component and should not be defined directly in the Polycom® RealPresence®
Collaboration Server 800s and Polycom® RealPresence® Collaboration Server Virtual Edition
component.

Operator conferences and participant move are supported in AVC CP Conferencing Mode only.

Operator Conferences
An Operator conference is a special conference that enables the Collaboration Server user acting as an
operator to assist participants without disturbing the ongoing conferences and without being heard by other
conference participants. The operator can move a participant from the Entry Queue or ongoing conference
to a private, one-on-one conversation in the Operator conference.
In attended mode, the Collaboration Server user (operator) can perform one of the following actions:
● Participants connected to the Entry Queue who fail to enter the correct destination ID or conference
password can be moved by the user to the Operator conference for assistance.
● After a short conversation, the operator can move the participant from the Operator conference to the
appropriate destination conference (Home conference).
● The operator can connect participants belonging to the same destination conference to their
conference simultaneously by selecting the appropriate participants and moving them to the Home
conference (interactively or using the right-click menu).
● The operator can move one or several participants from an ongoing conference to the Operator
conference for a private conversation.
● The operator can move participants between ongoing Continuous Presence conferences.

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Operator Conference Guidelines
● An Operator conference can only run in Continuous Presence mode.
● Operator conference is defined in the Conference Profile.
● An Operator conference can only be created by a User with Operator or Administrator Authorization
level.
● Operator conference name is derived from the User Login Name and it cannot be modified.
● Only one Operator conference per User Login Name can be created.
● When created, the Operator conference must include one and only one participant - the Operator
participant.
● Only a defined dial-out participant can be added to an Operator conference as an Operator participant
● Once running, the Collaboration Server user can add new participants or move participants from
other conferences to this conference. The maximum number of participants in an Operator
conference is the same as in standard conferences.
● Special icons are used to indicate an Operator conference in the Ongoing Conferences list and the
operator participant in the Participants list.
● An Operator conference cannot be defined as a Reservation.
● An Operator conference can be saved to a Conference Template. An ongoing Operator conference
can be started from a Conference Template.
● The Operator participant cannot be deleted from the Operator conference or from any other
conference to which she/he was moved to, but it can be disconnected from the conference.
● When deleting or terminating the Operator conference, the operator participant is automatically
disconnected from the MCU, even if participating in a conference other than the Operator conference.
● Participants in Telepresence conferences cannot be moved from their conference, but an operator
can join their conference and help them if assistance is required.
● Moving participants from/to an Operator conference follows the same guidelines as moving
participants between conferences. For move guidelines, see Move Guidelines.
● When a participant is moved from the Entry Queue to the Operator conference, the option to move
back to the source (Home) conference is disabled as the Entry Queue is not considered as a source
conference.
● The conference chairperson cannot be moved to the Operator conference following the individual
help request if the Auto Terminate When Chairperson Exits option is enabled, to prevent the
conference from automatically ending prematurely. In such a case, the assistance request is treated
by the system as a conference assistance request, and the operator can join the conference.

Defining the Components Enabling Operator Assistance
To enable operator assistance for conferences, the following conferencing entities must be adjusted or
created:
● IVR Service (Entry Queue and Conference) in which Operator Assistance options are enabled.
● A Conference Profile with the Operator Conference option enabled.
● An active Operator conference with a connected Operator participant.

Defining a Conference IVR Service with Operator Assistance Options
In the RMX Management pane, expand the Rarely Used list and click the IVR Services (
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1 On the IVR Services toolbar, click the New Conference IVR Service (

) button.

The New Conference IVR Service - Global dialog box opens.

2 Enter the Conference IVR Service Name.
3 Define the Conference IVR Service - Global parameters. For more information, see Conference IVR
Service Properties - Global Parameters.
4 Click the Welcome tab.
The New Conference IVR Service - Welcome dialog box opens.
5 Define the system behavior when the participant enters the Conference IVR queue. For more
information, see Defining a New Conference IVR Service.
6 Click the Conference Chairperson tab.
The New Conference IVR Service - Conference Chairperson dialog box opens.
7 If required, enable the chairperson functionality and select the various voice messages and options
for the chairperson connection. For more information, see New Conference IVR Service Properties Conference Chairperson Options and Messages.
8 Click the Conference Password tab.
The New Conference IVR Service - Conference Password dialog box opens.
9 If required, enable the request for conference password before moving the participant from the
conference IVR queue to the conference and set the MCU behavior for password request for Dial-in
and Dial-out participant connections. For more information, see New Conference IVR Service
Properties - Conference Password Parameters.
10 Select the various audio messages that will be played in each case. For more information, see For
more information, see New Conference IVR Service Properties - Conference Password Parameters.
11 Click the General tab.
The New Conference IVR Service - General dialog box opens.
12 Select the messages that will be played during the conference. For more information, see
Conference IVR Service Properties - General Voice Messages.
13 Click the Roll Call/Notifications tab.
The New Conference IVR Service - Roll Call dialog box opens.
14 Enable the Roll Call feature and assign the appropriate audio file to each message type. For more
information, see Conference IVR Service Properties - Roll Call Messages.

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15 Click the Video Services tab.
The New Conference IVR Service - Video Services dialog box opens.
16 Define the Video Services parameters. For more information, see New Conference IVR Service
Properties - Video Services Parameters.
17 Click the DTMF Codes tab.
The New Conference IVR Service - DTMF Codes dialog box opens.

The default DTMF codes for the various functions that can be performed during the conference by all
participants or by the chairperson are listed. For the full list of the available DTMF codes, see New
Conference IVR Service Properties - DTMF Codes.
18 If required, modify the default DTMF codes and the permissions for various operations including
Operator Assistance options:
 *0 for individual help - the participant requested help for himself or herself. In such a case, the
participant requesting help is moved to the Operator conference for one-on-one conversation. By
default, all participants can use this code.
 00 for conference help - the conference chairperson (default) can request help for the conference.
In such a case, the operator joins the conference.

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19 Click the Operator Assistance tab.
The Operator Assistance dialog box opens.

20 Select Enable Operator Assistance to enable operator assistance when the participant requires or
requests help during the connection process to the conference or during the conference.
21 In the Operator Assistance Indication Message field, select the audio message to be played
when the participant requests or is waiting for the operator’s assistance.
If the audio file was not uploaded prior to the definition of the IVR Service or if you want to add new
audio files, click Add Message File to upload the appropriate audio file to the Collaboration Server.

22 Click OK to complete the IVR Service definition.
The new Conference IVR Service is added to the IVR Services list.

Defining an Entry Queue IVR Service with Operator Assistance Options
1 In the RMX Management pane, click IVR Services (

).

2 In the IVR Services list, click the New Entry Queue IVR Service (

) button.

The New Entry Queue IVR Service - Global dialog box opens.
3 Define the Entry Queue Service Name.
4 Define the Entry Queue IVR Service Global parameters. For more information, see Entry Queue
IVR Service Properties - Global Parameters.
5 Click the Welcome tab.
The New Entry Queue IVR Service - Welcome dialog box opens.
6 Define the system behavior when the participant enters the Entry Queue. This dialog box contains
options that are identical to those in the Conference IVR Service - Welcome Message dialog box.
7 Click the Conference ID tab.
The New Entry Queue IVR Service - Conference ID dialog box opens.
8 Select the required voice messages. For more information, see Entry Queue IVR Service Properties
- Conference ID.

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9 Click the Video Services tab.
The New Entry Queue IVR Service - Video Services dialog box opens.
10 In the Video Welcome Slide list, select the video slide that will be displayed to participants
connecting to the Entry Queue. The slide list includes the video slides that were previously uploaded
to the MCU memory.
11 Click the Operator Assistance tab.
The Operator Assistance dialog box opens.

12 Select Enable Operator Assistance to enable operator assistance when the participant requires or
requests help during the connection process.
13 In the Operator Assistance Indication Message field, select the audio message to be played when
the participant requests or is waiting for operator’s assistance.
If the audio file was not uploaded prior to the definition of the IVR Service or if you want to add new
audio files, click Add Message File to upload the appropriate audio file to the Collaboration Server.

14 Click OK to complete the Entry Queue IVR Service definition.
The new Entry Queue IVR Service is added to the IVR Services list.

Defining a Conference Profile for an Operator Conference
In the RealPresence CloudAxis Solution, the Conference Profiles are defined in the RealPresence
DMA system component and should not be defined directly in the Polycom® RealPresence®
Collaboration Server 800s and Polycom® RealPresence® Collaboration Server Virtual Edition
component.

1 In the RMX Management pane, click Conference Profiles.
2 In the Conference Profiles pane, click the New Profile button.
The New Profile – General dialog box opens.
3 Define the Profile name and, if required, the Profile general parameters.
For more details, see New AVC CP Profile - General Parameters.

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4 Click the Operator Conference check box.

5 Click the Advanced tab.
The New Profile – Advanced dialog box opens.

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6 Define the Profile Advanced parameters. For more details, see New AVC CP Profile - Advanced
Parameters.
Note that when Operator Conference is selected, the Auto Terminate selection is automatically
cleared and disabled and the Operator conference cannot automatically end unless it is terminated
by the Collaboration Server User.
7 Click the Video Quality tab.
The New Profile – Video Quality dialog box opens.
8 Define the Video Quality parameters. For more details, see New AVC CP Profile - Video Quality
Parameters.
9 Click the Video Settings tab.
The New Profile - Video Settings dialog box opens.
10 Define the video display mode and layout. For more details, see New AVC CP Profile - Video
Settings Parameters.
11 Define the remaining Profile parameters. For more details, see Defining AVC CP Conferencing
Profiles.
12 Click OK to complete the Profile definition.
A new Profile is created and added to the Conference Profiles list.

Starting an Ongoing Operator Conference
To start a conference from the Conference pane:
1 In the Conferences pane, click the New Conference ( ) button.
The New Conference – General dialog box opens.

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2 In the Profile field, select a Profile in which the Operator Conference option is selected.

Upon selection of the Operator Conference Profile, the Display Name is automatically taken from the
Collaboration Server User Login Name. This name cannot be modified.
Only one Operator conference can be created for each User Login name.
3 Define the following parameters:
New Conference – General Options
Field

Description

Duration

Define the duration of the conference in hours using the format HH:MM (default
01:00).
Notes:
• The Operator conference is automatically extended up to a maximum of 168
hours. Therefore, the default duration can be used.
• This field is displayed in all tabs.

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Field

Description

Routing Name

Routing Name is the name with which ongoing conferences, Meeting Rooms, Entry
Queues and SIP Factories register with various devices on the network such as
gatekeepers and SIP servers. This name must be defined using ASCII characters.
Comma, colon and semicolon characters cannot be used in the Routing Name.
The Routing Name can be defined by the user or automatically generated by the
system if no Routing Name is entered as follows:
• If ASCII characters are entered as the Display Name, it is used also as the
Routing Name
• If a combination of Unicode and ASCII characters (or full Unicode text) is entered
as the Display Name, the ID (such as Conference ID) is used as the Routing
Name.
If the same name is already used by another conference, Meeting Room or Entry
Queue, the Collaboration Server displays an error message and requests that you
to enter a different name.

ID

Enter the unique-per-MCU conference ID. If left blank, the MCU automatically
assigns a number once the conference is launched.
This ID must be communicated to conference participants to enable them to dial in
to the conference.

Conference
Password

Leave this field empty when defining an Operator conference.

Chairperson
Password

Leave this field empty when defining an Operator conference.

Maximum Number of
Participants

Enter the maximum number of participants that can connect to an Operator
conference (you can have more than two), or leave the default selection
(Automatic).
Maximum number of participants that can connect to an Operator conference:

4 Click the Participants tab.
The New Conference - Participants dialog box opens.
You must define or add the Operator participant to the Operator conference.
This participant must be defined as a dial-out participant.
Define the parameters of the endpoint that will be used by the Collaboration Server User to connect
to the Operator conference and to other conference to assist participants.
For more details, see the Polycom® RealPresence Collaboration Server 800s / Virtual Edition Getting
Started Guide, Participants Tab.
5 Optional. Click the Information tab.
The Information dialog box opens.
6 Enter the required information. For more details, see the Polycom® RealPresence Collaboration
Server 800s / Virtual Edition Getting Started Guide, Information Tab.
7 Click OK.
The new Operator conference is added to the ongoing Conferences list with a special icon
The Operator participant is displayed in the Participants list with an Operator participant icon
and the system automatically dials out to the Operator participant.
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Saving an Operator Conference to a Template
The Operator conference that is ongoing can be saved as a template.

To save an ongoing Operator conference as a template:
1 In the Conferences List, select the Operator conference you want to save as a Template.
2 Click the Save Conference to Template ( ) button.
or
Right-click and select Save Conference to Template.

The conference is saved to a template whose name is taken from the ongoing conference Display
Name (the Login name of the Collaboration Server User). The Template is displayed with the
Operator Conference icon.

Starting an Operator Conference from a Template
An ongoing Operator conference can be started from an Operator Template saved in the Conference
Templates list.

To start an ongoing Operator conference from an Operator Template:
1 In the Conference Templates list, select the Operator Template to start as an ongoing Operator
conference.
•

•

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You can only start an Operator conference from a template whose name is identical to your Login
Name. For example, if your Login name is Polycom, you can only start an Operator conference from
a template whose name is Polycom.
If an ongoing Operator conference with the same name or any other conference with the same ID
is already running, you cannot start another Operator conference with the same login name.

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2 Click the Start Conference from Template ( ) button.
or
Right-click and select Start Conference from Template.

The conference is started.
The name of the ongoing conference in the Conferences list is taken from the Conference Template
Display Name.

Monitoring Operator Conferences and Participants Requiring
Assistance
Operator conferences are monitored in the same way as standard ongoing conferences.
Each Operator conference includes at least one participant - the Operator.

You can view the properties of the Operator conference by double-clicking the conference entry in the
Conferences list or by right-clicking the conference entry and selecting Conference Properties. For more
information, see the Polycom® RealPresence Collaboration Server 800s / Virtual Edition Getting Started
Guide, Participant Level Monitoring.

Requesting Help
A participant can request help using the appropriate DTMF code from his/her touch tone telephone or the
endpoint’s DTMF input device. The participant can request Individual Assistance (default DTMF code *0) or
Conference Assistance (default DTMF code 00).

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Participants in Entry Queues who failed to enter the correct destination conference ID or the conference
password will wait for operator assistance (provided that an Operator conference is active).
When requiring or requesting operator assistance, the Collaboration Server management application
displays the following:

● The participant’s connection Status changes, reflecting the help request. For more information, see
Table 5-2.
● The conference status changes and it is displayed with the exclamation point icon and the status
“Awaiting Operator”.
● The appropriate voice message is played to the relevant participants indicating that assistance will
be provided shortly.
The following icons and statuses are displayed in the Participant Status column:
Participants List Status Column Icons and Indications

Icon

Status Indication

Description

Awaiting Individual
Assistance

The participant has requested the operator’s
assistance for himself/herself.

Awaiting Conference
Assistance

The participant has requested the operator’s
assistance for the conference. Usually this means
that the operator is requested to join the conference.

When the Operator moves the participant to the Operator conference for individual assistance the
participant Status indications are cleared.

Participant Alerts List
The Participant Alerts list contains all the participants who are currently waiting for operator assistance.

Participants are automatically added to the Participants Alerts list in the following circumstances:
● The participant fails to connect to the conference by entering the wrong conference ID or conference
password and waits for the operator’s assistance
● The participant requests Operator’s Assistance during the ongoing conference

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This list is used as reference only. Participants can be assisted and moved to the Operator conference or
the destination conference only from the Participants list of the Entry Queues or ongoing conference where
they are awaiting assistance.
The participants are automatically removed from the Participant Alerts list when moved to any conference
(including the Operator conference).

Audible Alarms
In addition to the visual cues used to detect events occurring on the Collaboration Server, an audible alarm
can be activated and played when participants request Operator Assistance.

Using Audible Alarms
The Audible Alarm functionality for Operator Assistance requests is enabled for each MCU in either the
Collaboration Server Web Client or RMX Manager.
The Audible Alarm played when Operator Assistance is requested is enabled and selected in the Setup >
Audible Alarm > User Customization. When the Audible Alarm is activated, the *.wav file selected in the
User Customization is played, and it is repeated according to the number of repetitions defined in the User
Customization.
If more than one Collaboration Server is monitored in the RMX Manager, the Audible Alarm must be enabled
separately for each Collaboration Server installed in the site/configuration. A different *.wav file can be
selected for each MCU.
When multiple Audible Alarms are activated in different conferences or by multiple MCUs, the Audible
Alarms are synchronized and played one after the other. It is important to note that when Stop Repeating
Alarm is selected from the toolbar from the Collaboration Server Web Client or RMX Manager, all activated
Audible Alarms are immediately halted.
For more details on Audible alarms and their configuration, see Audible Alarms.

Moving Participants Between Conferences
The Collaboration Server User can move participants between ongoing conferences, including the Operator
conference, and from the Entry Queue to the destination conference if help is required.
When moving between conferences or when a participant is moved from an Entry Queue to a conference
by the Collaboration Server user (after failure to enter the correct destination ID or conference password),
the IVR messages and slide display are skipped.

Move Guidelines
● Move is available only between CP conferences.
● Move between conferences can be performed without an active Operator conference.
● When moving the conference chairperson from his/her conference to another conference, the source
conference will automatically end if the Auto Terminate When Chairperson Exits option is enabled
and that participant is the only conference chairperson.
● When moving the Operator to any conference (following assistance request), the IVR messages and
slide display are skipped.

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● Participants cannot be moved from a Telepresence conference.
● Participants cannot be moved from LPR-enabled conferences to non-LPR conferences. Move from
non-LPR conferences to LPR-enabled conferences is available.
● Move between encrypted and non-encrypted conferences depends on the
ALLOW_NON_ENCRYPT_PARTY_IN_ENCRYPT_CONF flag setting, as described in the following
table:
Participant Move Capabilities vs. ALLOW_NON_ENCRYPT_PARTY_IN_ENCRYPT_CONF flag

Flag Setting

Source Conference/EQ
Encrypted

Destination
Conference Encrypted

Move Enabled?

NO

Yes

Yes

Yes

NO

Yes

No

Yes

NO

No

Yes

No

NO

No

No

Yes

YES

Yes

Yes

Yes

YES

Yes

No

Yes

YES

No

Yes

Yes

YES

No

No

Yes

setting

● When moving dial-out participants who are disconnected to another conference, the system
automatically dials out to connect them to the destination conference.
● Cascaded links cannot be moved between conferences.
● Participants cannot be moved to a conference if the move will cause the number of participants to
exceed the maximum number of participants allowed for the destination conference.

Moving Participants Options
Collaboration Server users can assist participants by performing the following operations:
● Move a participant to an Operator conference (Attend a participant).
● Move a participant to the Home (destination) conference.
● Move participant from one ongoing conference to another
A move can be performed using the following methods:
● Using the participant right-click menu
● Using drag and drop
To move a participant from the ongoing conference using the right-click menu options:
1 In the Conferences list, click the conference where there are participants waiting for Operator’s
Assistance to display the list of participants.

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2 In the Participants list, right-click the icon of the participant to move and select one of the following
options:

 Move to Operator Conference - to move the participant to the Operator conference.
 Move to Conference - to move the participant to any ongoing conference.
When selected, the Move to Conference dialog box opens, letting you select the name of the
destination conference.

 Back to Home Conference - if the participant was moved to another conference or to the
Operator conference, this options moves the participant back to his/her source conference.
This option is not available if the participant was moved from the Entry Queue to the Operator
conference or the destination conference.

Moving a Participant Interactively
You can drag and drop a participant from the Entry Queue or ongoing conference to the Operator or
destination (Home) conference:
1 Display the participants list of the Entry Queue or the source conference by clicking its entry in the
Conferences list.
2 In the Participants list, drag the icon of the participant to the Conferences List pane and drop it on
the Operator Conference icon or another ongoing conference.

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Conference Templates
In the RealPresence CloudAxis Solution, the conference templates are defined in the RealPresence
DMA system component and should not be defined directly in the Polycom® RealPresence®
Collaboration Server 800s and Polycom® RealPresence® Collaboration Server Virtual Edition
component.

Conference Templates enable administrators and operators to create, save, schedule and activate identical
conferences.
A Conference Template:
● Saves the conference Profile.
● Saves all participant parameters including their Personal Layout and Video Forcing settings.
● Simplifies the setting up Telepresence conferences where precise participant layout and video forcing
settings are crucial.

Guidelines
● The maximum number of templates is 100.A maximum of 200 participants can be saved in a
Conference Template.
Trying to start a Conference Template that exceeds the allowed maximum number of participants will
result in participants being disconnected due to resource deficiency.
● If the Profile assigned to a conference is deleted while the conference is ongoing the conference
cannot be saved as a template.
● A Profile assigned to a Conference Template cannot be deleted. The system does not permit such a
deletion.
● Profile parameters are not embedded in the Conference Template, and are taken from the Profile
when the Conference Template becomes an ongoing conference. Therefore, any changes to the
Profile parameters between the time the Conference Template was created and the time that it is
activated (and becomes an ongoing conference) will be applied to the conference.
● Only defined participants can be saved to the Conference Template. Before saving a conference to
a template ensure that all undefined participants have disconnected.
● Undefined participants are not saved in Conference Templates.
● Participant properties are embedded in the Conference Template and therefore, if the participant
properties are modified in the Address Book after the Conference Template has been created they
are not applied to the participant whether the Template becomes an ongoing conference or not.
● The Conference Template display name, routing name or ID can be the same as an Ongoing
Conference, reservation, Meeting Room or Entry Queue as it is not active. However, an ongoing
conference cannot be launched from the Conference Template if an ongoing conference, Meeting
Room or Entry Queue already has the same name or ID. Therefore, it is recommended to modify the
template ID, display name, routing name to be unique.
● A Reservation that has become an ongoing conference can be saved as Conference Template.
● SIP Factories and Entry Queues cannot be saved as Conference Templates.

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● The conference specified in the Conference Template can be designated as a Permanent
Conference. For more information see Permanent Conference.

Using Conference Templates
In the RealPresence CloudAxis Solution, templates are used in the RealPresence DMA system
component and should not be defined directly in the RealPresence Collaboration Server component.

The Conference Templates list is initially displayed as a closed tab in the Collaboration Server Web Client
main window. The number of saved Conference Templates is indicated on the tab.

Clicking the tab opens the Conference Templates list.

The Conference Templates are listed by Conference Template Display Name and ID and can be sorted by
either field. The list can be customized by re-sizing the pane, adjusting the column widths or changing the
order of the column headings.
For more information see Polycom® RealPresence Collaboration Server 800s / Virtual Edition Getting
Started Guide, Customizing the Main Screen.
Clicking the anchor pin ( ) button hides the Conference Templates list as a closed tab.

Toolbar Buttons
The Conference Template toolbar includes the following buttons:

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Conference Templates – Toolbar Buttons
Button

Description

New Conference Template

Creates a new Conference Template.

Delete Conference Template

Deletes the Conference Template(s) that are selected in the list.

Start Conference from Template

Starts an ongoing conference from the Conference Template that has
an identical name, ID parameters and participants as the template.

Schedule Reservation from Template

Creates a conference Reservation from the Conference Template
with the same name, ID, parameters and participants as the
Template.
Opens the Scheduler dialog box enabling you to modify the fields
required to create a single or recurring Reservation based on the
template. For more information see Scheduling Reservations.

The Conferences List toolbar includes the following button:
Conferences List – Toolbar Button
Button

Description

Save Conference to Template

Saves the selected ongoing conference as a Conference Template.

Creating a New Conference Template
In the RealPresence CloudAxis Solution, the Conference Templates are defined in the RealPresence
DMA system component and should not be defined directly in the Polycom® RealPresence®
Collaboration Server 800s and Polycom® RealPresence® Collaboration Server Virtual Edition
component.

There are two methods to create a Conference Template:
● From scratch - defining the conference parameters and participants
● Saving an ongoing conference as Template

Creating a new Conference Template from Scratch
To create a new Conference Template:
1 In the Collaboration Server main screen, click the Conference Templates tab.

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2 Click the New Conference Template (

) button.

The New Conference Template - General dialog box opens.

The fields of the New Template – General dialog box are identical to those of the New Conference
– General dialog box. For a full description of the fields see the Polycom® RealPresence
Collaboration Server 800s / Virtual Edition Getting Started Guide, General Tab.
3 Modify the fields of the General dialog box.

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4 Click the Participants tab.
The New Template – Participants dialog box opens.

The fields of the New Template – Participants dialog box are the same as those of the New
Conference – Participant dialog box.
For a full description of these fields see the Polycom® RealPresence Collaboration Server 800s /
Virtual Edition Getting Started Guide, Participants Tab.
5 Optional. Add participants to the template from the Address Book.

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6 Click the New button.
The New Participant – General dialog box opens.
The New Template – Participant dialog box remains open in the background.

For a full description of the General tab fields see Adding a New participant to the Address Book
Directly.
7 Modify the fields of the General dialog box.

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8 Click the Advanced tab.
The New Participant – Advanced dialog box opens.

.

9 Modify the fields of the Advanced dialog box.

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10 Click the Media Sources tab.
The Media Sources dialog box opens.

The Media Sources dialog box enables you to set up and save Personal Layout and Video Forcing
settings for each participant. This is especially important when setting up Telepresence conferences.
For a full description of Personal Layout and Video Forcing settings see the Polycom® RealPresence
Collaboration Server 800s / Virtual Edition Getting Started Guide, Changing the Video Layout of a
Conference (AVC-Based CP and Mixed CP and SVC Conferences) and Video Forcing (AVC-Based
CP and Mixed CP and SVC Conferences).
11 Modify the Personal Layout and Video Forcing settings for the participant.
12 Optional. Click the Information tab.
The New Participant – Information dialog box opens.

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For a full description of the Information fields see the Polycom® RealPresence Collaboration Server
800s / Virtual Edition Getting Started Guide, Information Tab.
13 Click the OK button.
The participant you have defined is added to the Participants List.
The New Participant dialog box closes and you are returned to the New Template – Participant
dialog box (which has remained open since step 6).

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14 Optional. In the New Conference Template dialog box, click the Information tab.
The New Conference Template – Information dialog box opens.

For a full description of the Information fields see the Polycom® RealPresence Collaboration Server
800s / Virtual Edition Getting Started Guide, Information Tab.
15 Click the OK button.
The New Conference Template is created and its name is added to the Conference Templates list.

Saving an Ongoing or AVC-based CP Operator Conference as a
Template
Any ongoing or AVC-based CP Operator Conference can be saved as a template.

To save an ongoing or AVC-based CP Operator Conference as a template:
1 In the Conferences List, select the conference or Operator Conference to be saved as a Template.

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2 Click the Save Conference to Template (

) button.

or
Right-click and select Save Conference to Template.

The conference is saved to a template whose name is taken from the ongoing conference Display
Name (the Login name of the Collaboration Server User). The Template is displayed with the
Operator Conference icon.

Starting an Ongoing Conference From a Template
Conference Templates saved from an ongoing conference does not include Message Overlay text
messages.

In the RealPresence CloudAxis Solution, Conferences should be started in the RealPresence DMA
system component and should not be started directly in the Polycom® RealPresence® Collaboration
Server 800s and Polycom® RealPresence® Collaboration Server Virtual Edition component.

An ongoing conference can be started from any Template saved in the Conference Templates list. In
SVC-based templates, only defined dial-in participants may be part of the conference.

To start an ongoing conference from a Template:
1 In the Conference Templates list, select the Template you want to start as an ongoing conference.

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2 Click the Start Conference from Template (

) button.

or
Right-click and select Start Conference from Template.

The conference is started.
The name of the ongoing conference in the Conferences list is taken from the Conference Template
Display Name.
Participants that are connected to other ongoing conferences when the template becomes an
ongoing conference are not connected.
If an ongoing conference, Meeting Room or Entry Queue with the same Display Name, Routing Name
or ID already exists in the system, the conference will not be started.

Starting an Operator Conference from a Template (AVC Conferencing)
An ongoing Operator conference can be started from an Operator Template saved in the Conference
Templates list.

To start an ongoing Operator conference from an Operator Template:
1 In the Conference Templates list, select the Operator Template to start as an ongoing Operator
conference.
•

•

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You can only start an Operator conference from a template whose name is identical to your Login
Name. For example, if your Login name is Polycom, you can only start an Operator conference from
a template whose name is Polycom.
If an ongoing Operator conference with the same name or any other conference with the same ID
is already running, you cannot start another Operator conference with the same login name.

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2 Click the Start Conference from Template (

) button.

or
Right-click and select Start Conference from Template.

The conference is started.
The name of the ongoing conference in the Conferences list is taken from the Conference Template Display
Name.

Scheduling a Reservation From a Conference
Template
A Conference Template can be used to schedule a single or recurring Reservation.

To schedule a Reservation from a Conference Template:
1 In the Conference Templates list, select the Conference Template you want to schedule as a
Reservation.

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2 Click the Schedule Reservation from Template (

) button.

or
Right-click and select Schedule Reservation from Template.

The Reservation Properties dialog box is displayed.
The Display Name of the Reservation is taken from the Conference Template Display Name.

For a full description of the Reservation Properties fields see Creating a New Reservation.
3 Modify the fields of the Reservation Properties.

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4 Click the OK button.
A Reservation is created based on the Conference Template. The Reservation can be viewed and
modified along with all other Reservations using the Reservations - Calendar View and Reservations
List.
If you create a recurring reservation all occurrences have the same ID.The series number (_0000n)
of each reservation is appended to its Display Name.
Example:
Conference Template name: Sales
Display Name for single scheduled occurrence:Sales
If 3 recurrences of the reservation are created:
Display Name for occurrence 1: Sales_00001
Display Name for occurrence 2: Sales_00002
Display Name for occurrence 3: Sales_00003

Deleting a Conference Template
One or several Conference Templates can be deleted at a time.

To delete Conference Templates:
1 In the Conference Templates list, select the Template(s) you want to delete.
2 Click the Delete Conference Template (

) button.

or
Right-click and select Delete Conference Template.

A confirmation dialog box is displayed.
3 Click the OK button to delete the Conference Template(s).

Exporting and Importing Conference Templates
Conference Templates can be exported from one MCU and imported to multiple MCUs in your environment.
Additionally, you can export Conference Templates and their associated Conference Profiles
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simultaneously. Using this option can save configuration time and ensures that identical settings are used
for conferences running on different MCUs. This is especially important in environments using cascading
conferences that are running on different MCUs.
● Administrators can export and import Conference Templates. Operators are only allowed to export
Conference Templates.
● You can select a single, multiple or all Conference Templates to be exported.
● Both Conference Templates and their associated Conference Profiles can be exported and imported
simultaneously when enabling the Export includes conference profiles or Import includes
conference profiles options.
● Exporting and importing Conference Templates only can be used when you want to export and import
individual Conference Templates without their associated Conference Profiles. This option enables
you to import Conference Templates when Conference Profiles already exist on an MCU.

Exporting Conference Templates
Conference Templates are exported to a single XML file that can be used to import the Conference
Templates on multiple MCUs.
Using the Export Conference Templates option, you can:
● Export all Conference Templates from an MCU
● Export selected Conference Templates

Exporting All Conference Templates from an MCU
To export all Conference Templates from an MCU:
1 In the Collaboration Server Web Client main window, click the Conference Templates tab.
The Conference Templates list pane is displayed.

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2 Click the Export Conference Templates
button, or right-click the Conference Templates list,
and then click Export Conference Templates.

The Conference Templates - Export dialog box is displayed.

3 In the Export Path field, type the path name to the location where you want to save the exported file
or click Browse to select the desired path.
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4 Optional. Clear the Export includes conference profiles check box when you only want to export
Conference Templates.
When this check box is cleared, the Conference Templates - Export dialog box is displayed without
the Profiles file name field.

5 In the Templates file name field, type the file name prefix. The file name suffix
(_confTemplates.xml) is predefined by the system. For example, if you type Templates01, the
exported file name is defined as Templates01_confTemplates.xml.
The system automatically defines the Profiles file name field with the same file name prefix as the
Templates file name field. For example, if you type Templates01 in the Templates file name field, the
exported profiles file name is defined as Templates01_confProfiles.xml.
6 Click OK to export the Conference Templates and Conference Profiles to a file.

Exporting Selected Conference Templates
You can export a single Conference Template or multiple Conference Templates to other MCUs in your
environment.

To export selected Conference Templates:
1 In the Conference Templates list, select the templates you want to export.
2 Right-click the Conference Templates to be exported, and then click Export Selected Conference
Templates.

.

The Conference Templates - Export dialog box is displayed.

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3 In the Export Path field, type the path name to the location where you want to save the exported file
or click Browse to select the desired path.
4 Optional. Clear the Export includes conference profiles check box when you only want to export
Conference Templates.
When this check box is cleared, the Conference Templates - Export dialog box is displayed without
the Profiles file name field.

5 In the Templates file name field, type the file name prefix. The file name suffix
(_confTemplates.xml) is predefined by the system. For example, if you type, Templates01, the
exported file name is defined as Templates01_confTemplates.xml.
The system automatically defines the Profiles file name field with the same file name prefix as the
Templates file name field. For example, if you type Templates01 in the Templates file name field, the
exported profiles file name is defined as Templates01_confProfiles.xml.
6 Click OK to export the Conference Templates and Conference Profiles to a file.

Importing Conference Templates
You can import Conference Templates and Conference Profiles from one MCU to multiple MCUs in your
environment.

To import Conference Templates:
1 In the Collaboration Server Web Client main window, click the Conference Templates tab.
The Conference Templates are displayed.
2 Click the Import Conference Templates
button or right-click the Conference Templates pane,
and then click Import Conference Templates.

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The Conference Templates - Import dialog box is displayed.

3 Optional. Clear the Import includes conference profiles check box when you only want to import
Conference Templates.
When this check box is cleared, the Conference Templates - Import dialog box is displayed without
the Profiles file name field.

4 In the Import Path field, click Browse to navigate to the path and file name of the Conference
Templates you want to import.
When clicking the exported templates file you want to import, the system automatically displays the
appropriate files in the Templates file name field and the Profiles file name field (when the Import
includes conference profiles check box is selected).
5 Click OK to import the Conference Templates and their associated Conference Profiles, if selected.
Conference Templates are not imported when:
 A Conference Template already exists
 An associated Conference Profile is not defined in the Conference Profiles list
When one or more Conference Templates are not imported, a Message Alert window is displayed
with the templates that were not imported.
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6 Click Cancel to exit the Message Alerts window.
The imported Conference Templates are added to the Conference Templates list. When the Import
includes conference profiles check box is selected, the imported Conference Profiles are added to
the Conference Profiles list.

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Outlook®
Polycom Conferencing for Microsoft Outlook is supported in AVC CP Conferencing Mode only

Polycom Conferencing for Microsoft Outlook is an add-in that enables users to easily organize and invite
attendees to Video Enabled meetings via Microsoft Outlook®.
Polycom Conferencing for Microsoft Outlook is implemented by installing the Polycom Conferencing Add-in
for Microsoft Outlook on Microsoft Outlook® e-mail clients. It enables meetings to be scheduled with video
endpoints from within Outlook. The add-in also adds a Polycom Conference button in the Meeting tab of the
Microsoft Outlook e-mail client ribbon.
The meeting organizer clicks the Polycom Conference button to add Conference Information to the
meeting invitation.
Attendees call the meeting at the scheduled Start Time using the link or the dial-in number provided in the
meeting invitation.

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A Gathering Slide is displayed to connected participants until the conference starts.

The Gathering Slide displays live video along with information taken from the meeting invitation such as the
subject, meeting organizer, duration, dial-in numbers etc. At the end of the Gathering Phase, the conference
layout is displayed.
For more information see Video Preview (AVC Participants Only).

Setting up the Calendaring Solution
The following steps are performed to set up the Calendaring solution:
a The administrator installs the Polycom Conferencing Add-in for Microsoft for Microsoft Outlook
e-mail clients. For more information, see the Polycom Unified Communications Deployment
Guide for Microsoft Environments.
b The administrator creates an Microsoft Outlook e-mail-account for the Collaboration Server.
If included in the solution, Polycom RealPresence DMA system and calendaring-enabled
endpoints share this e-mail account. For more information, see the Polycom Unified
Communications Deployment Guide for Microsoft Environments.
c The administrator configures the Collaboration Server for Calendaring using the Exchange
Integration Configuration dialog box, providing it with the Microsoft Exchange Server Name,
User Name and Password and optional Primary SMTP Mail box information needed to access the
e-mail account.

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To configure the Collaboration Server’s Exchange Integration Configuration:
1 On the Collaboration Server menu, click Setup > Exchange Integration Configuration.
The Exchange Integration Configuration dialog box is displayed.

There are three options that can be used to configure the Exchange Integration Configuration. The
option you choose will depend on the configuration of the mailbox in the Exchange Server and the
configuration of the Exchange Server itself.
 Option 1 - Use this option if the Exchange Server settings have been left at their default values.
 Option 2 - Use this option if the Primary SMTP Mailbox is not the default mailbox.
 Option 3 - Use this option if the Exchange Server settings have been modified by the
administrator.

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Option 1 - Using default Exchange Server settings

a Define the following fields:
Exchange Integration Configuration - Option 1
Field

Description

Enable Calendaring
Service

Select or clear this check box to enable or disable the
Calendaring Service using the Polycom Add-in for Microsoft
Outlook. When this check box is cleared all fields in the dialog
box are disabled.

Exchange Server
Address

Enter the IP address of the Exchange Server.

User Name

Enter the User Name of the Collaboration Server, as registered
in the Microsoft Exchange Server, that the Collaboration Server
uses to login to its e-mail account.
Field length: Up to 80 characters.

Password

Enter the Password the Collaboration Server uses to login to its
e-mail account as registered in the Microsoft Exchange Server.
Field length: Up to 80 characters.

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Exchange Integration Configuration - Option 1
Field

Description

Domain

Enter the name of the network domain where the Collaboration
Server is installed as defined in the Microsoft Exchange Server.

Primary SMTP
Mailbox (Optional)

This field is left empty.

Accept Appointments

Select this check box to enable the Collaboration Server to send
replies to meeting invitations.
Clear this check box when the Collaboration Server is part of a
Unified Conferencing solution that includes a RealPresence
DMA system, as the RealPresence DMA system will send a
reply to the meeting invitation.

b Click OK.

Option 2 - Using an alternate Primary SMTP Mailbox

a Define the following fields:

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Exchange Integration Configuration - Option 2
Field

Description

Enable Calendaring
Service

These fields are defined as for Option 1 above.

Exchange Server
Address
User Name
Password
Domain
Accept Appointments
Primary SMTP
Mailbox (Optional)

Enter the name of the SMTP Mailbox in the Microsoft Exchange
Server to be monitored by the Collaboration Server.
Note: Although several mailboxes can be assigned to each user
in the Microsoft Exchange Server, only the Primary SMTP
Mailbox is monitored. The Primary SMTP Mailbox name does
not have to contain either the Collaboration Server’s User Name
or Domain name.

b Click OK.

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Option 3 - Using modified Exchange Server settings

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a Define the following fields:
Exchange Integration Configuration - Option 3
Field

Description

Exchange Server
Address

If Exchange Server settings have been modified, enter the full
path to the Microsoft Exchange Server where the Collaboration
Server’s Microsoft Outlook e-mail account is registered, for
example if the EWS folder has been renamed EWD:
https://labexch01/EWD/Exchange.asmx
Note: If a server name is entered, the Collaboration Server and
the Microsoft Exchange Server must be registered to the same
Domain. (The Domain name entered in this dialog box must
match the Local Domain Name entry in the Management
Network - DNS Properties dialog box.)
For more information see Modifying the Default IP Network
Service in the RealPresence Collaboration Server 800s on
page 8.
Field length: Up to 80 characters.

Enable Calendaring
Service

These fields are defined as for Option 1 above.

User Name
Password
Domain
Primary SMTP
Mailbox (Optional)
Accept Appointments

b Click the OK button.
If applicable, RSS, VMC, RealPresence DMA system, and calendaring-enabled endpoints are
configured with the Exchange Server Name, User Names and Passwords needed to access their
accounts.
For more information see the Polycom Unified Communications Deployment Guide for Microsoft
Environments.
1 The administrator configures the Collaboration Server to have a default Ad-hoc Entry Queue service
enabled.
For more information see Defining a New Entry Queue.

Calendaring Guidelines
● The Collaboration Server must have its MCU prefix registered in the gatekeeper.
For more information see Modifying the Default IP Network Service.
● The Collaboration Server must be configured as a Static Route.
For more information see Modifying the Default IP Network Service.
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● The Collaboration Server’s Default Entry Queue must be configured as an Ad Hoc Entry Queue and
must be designated as the Transit Entry Queue.
For more information see the Entry Queues.
● The meeting organizer can enable recording and/or streaming of the meeting.
● If meeting is to be recorded, the Ad Hoc Entry Queue must have recording enabled in its Profile.
For more information see Defining AVC CP Conferencing Profiles.
● Meetings can be single instance or have multiple occurrences.
● Attendees that do not have video devices may be invited to the meeting.
● Attendees using e-mail applications that use the iCalendar format may be invited to meetings via the
Calendaring Service.
● Meeting invitations sent by Polycom Conferencing for Microsoft Outlook can be in a different
language to the Collaboration Server Web Client. The following languages are supported:
 English
 French
 German
 International Spanish
 Korean
 Japanese
 Simplified Chinese
● Collaboration Server resource management is the responsibility of the system administrator:
 Conferences initiated by Polycom Conferencing for Microsoft Outlook are ad hoc and therefore
resources are not reserved in advance.
 Polycom Conferencing for Microsoft Outlook Add-in assumes that sufficient resources are
available and does not check resource availability. Sufficient resources are therefore not
guaranteed.
 A meeting invitation that is automatically accepted by the Collaboration Server is not guaranteed
availability of resources.
 If the Collaboration Server runs out of resources, attendees will not be able to connect to their
conferences.
● By using RealPresence DMA system to load-balance resources between several Collaboration
Servers, resource capacity can be increased, alleviating resource availability problems.

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Creating and Connecting to a Conference
Creating a Conference
Meetings are organized using the Microsoft Outlook client in the normal manner.
If the meeting organizer decides that video participants are to be included in a multipoint video conference,
he/she clicks the Polycom Conference button. Conference Information such as the Meeting ID and
connection information is automatically added to the existing appointment information.

The meeting organizer can add a meeting agenda or personal text to the invitation before it is sent. The
meeting organizer can update or cancel the video enabled meeting in the same manner as for any other
meeting.
When the meeting organizer sends the meeting invitation a meeting record is saved in the Microsoft
Exchange Server, the RealPresence Collaboration Server, RealPresence DMA system, RSS and
calendaring-enabled endpoints.
RealPresence Collaboration Servers, RealPresence DMA system, and calendaring-enabled endpoints poll
the Microsoft Exchange Server to retrieve new meeting records and updates to existing meeting records.

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The table below summarizes the Collaboration Server’s usage of Microsoft Outlook data fields included in
the meeting invitation.
Microsoft Outlook Field Usage
Microsoft
Outlook Field

Usage by the Collaboration Server / RealPresence DMAsystem
Conference / Meeting Room

Gathering Slide

Subject

Display Name of Conference / Meeting
Room.

Meeting Name.

Start/End Time

Used to calculate the Conference’s Duration.

Record

Enable Recording in the Conference or
Meeting Room Profile.

Display Recording option.

Video Access
Number

Comprised of:  .
Note: It is important that MCU Prefix in
Gatekeeper field in the Collaboration
Server’s IP Network Service Gatekeeper tab and the Dial-in prefix
field in the Polycom Conferencing
Add-in for Microsoft Outlook - Video
Network tab contain the same prefix
information.

Displayed as the IP dial in number
in the Access Number section of
the Gathering Slide.

Video Access
Number (Cont.)

If Recording and Streaming are enabled
in the Conference Profile, this number
is used as part of the recording file
name.

Streaming
recording link

Enables the recording of the
conference to the Polycom RSS using
the recording link.
Enables streaming of the recording of
the conference from the Polycom RSS.

If recording is enabled, a REC
indicator is displayed in the top left
corner of the slide.

Connecting to a Conference
Participants can connect to the conference in the following ways:
● Participants with Polycom CMA/RealPresence Desktop™ or a Microsoft Office Communicator client
running on their PCs can click on a link in the meeting invitation to connect to the meeting.
● Participants with a HDX or a room system will receive a prompt from the endpoint’s calendaring
system along with a button that can be clicked in order to connect.
Participants with endpoints that are not calendaring-enabled can connect to the meeting by dialing
the meeting number manually.

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Collaboration Server Standalone Deployment
When using a single Collaboration Server in a standalone deployment, connection is via an Ad Hoc Entry
Queue. The meeting is started when the first participant connects to the Collaboration Server.
When the first participant connects, a conference is created and named according to the information
contained in the dial string. Subsequent participants connecting with the same dial string are routed from
the Ad Hoc Entry Queue to the conference.
After the conference has been created the Conference Name, Organizer, Time, Duration and Password (if
enabled) are retrieved from the conference parameters for display during the Gathering Phase.

Collaboration Server and Polycom RealPresence DMA System Deployment
In a RealPresence DMA system deployment a Virtual Meeting Room is activated when the first participant
connects to the RealPresence DMA system. The RealPresence DMA system receives the dial string to
activate a Virtual Meeting Room on the Collaboration Server.
The RealPresence DMA system uses the Meeting ID contained in the dial-in string to access meeting
information stored in the Exchange Server database.
When the meeting information is found on the Exchange Server, the Conference Name, Organizer, Time,
Duration and Password (if enabled) are retrieved from the Exchange Server database for display during the
Gathering Phase.
If enabled, automatically generated passwords are ignored.
For more information see Automatic Password Generation Flags.

Polycom Solution Support
Polycom Implementation and Maintenance services provide support for Polycom solution components only.
Additional services for supported third-party Unified Communications (UC) environments integrated with
Polycom solutions are available from Polycom Global Services and its certified Partners. These additional
services will help customers successfully design, deploy, optimize and manage Polycom visual
communications within their UC environments.
Professional Services for Microsoft Integration is mandatory for Polycom Conferencing for Microsoft Outlook
and Microsoft Office Communications Server integrations. For additional information and details please see
http://www.polycom.com/services/professional_services/index.html or contact your local Polycom
representative.

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You can monitor ongoing conferences and perform various operations while conferences are running.
Three levels of monitoring are available with the Collaboration Server:
● General Monitoring - You can monitor the general status of all ongoing conferences and their
participants in the main window.
● Conference Level Monitoring - You can view additional information regarding a specific conference
and modify its parameters if required, using the Conference Properties option.
● Participant Level Monitoring - You can view detailed information on the participant's status, using the
Participant Properties option.

General Monitoring
Users can monitor a conference or keep track of its participants and progress. For more information, see
Polycom® RealPresence Collaboration Server 800s / Virtual Edition Getting Started Guide, Monitoring
Ongoing Conferences on page 166.

You can click the blinking Participant Alerts indication bar to view participants that require attention. For
more information, see System and Participant Alerts.

Conference Level Monitoring
In addition to the general conference information that is displayed in the Conference list pane, you can view
the details of the conference’s current status and setup parameters, using the Conference Properties dialog
box.

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Conference monitoring - Tab list per conferencing mode and user
Admin

Chairperson

Operator

Tab Name

CP

SVC

Mixed

VSW

CP

SVC

Mixed

VSW

CP

SVC

Mixed

VSW

General

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Advanced

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Gathering
Settings



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

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





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



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Video
Quality



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











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Video
Settings



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Audio
Settings



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



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

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

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



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Customized
Polling





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Skins

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IVR

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Information

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Recording

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

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Site Names

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Message
Overlay









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











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Network
Services

























Viewing the Properties of Ongoing CP and Mixed CP and SVC
Conferences
To view the parameters of an ongoing CP conference:
1 In the Conference list pane, double-click the CP conference or right-click the CP conference and
then click Conference Properties.

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The Conference Properties - General dialog box with the General tab opens.

The following information is displayed in the General tab:
Conference Properties - General
Field

Description

Display Name

The Display Name is the conference name in native language and Unicode
character sets to be displayed in the Collaboration Server Web Client.
Note: This field is displayed in all tabs.

Duration

The expected duration of the conference using the format HH:MM.
Note: This field is displayed in all tabs.

Permanent Conference

Indicates whether the conference is set as a Permanent Conference, with no
pre-determined End Time. This conference continues until it is terminated by
an administrator, operator or chairperson.
Note: This field is displayed in all tabs.

Routing Name

The ASCII name of the conference. It can be used by H.323 and SIP
participants for dialing in directly to the conference. It is used to register the
conference in the gatekeeper and the SIP server.

Conferencing Mode

The conferencing mode set for the conference: CP, VSW, SVC only or CP and
SVC.

Start Time

The time the conference started.

End Time

The expected conference end time.
Note: This field is not shown when the conference is set as a Permanent
Conference.

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Conference Properties - General
Field

Description

Conference Password

A numeric password for participants to access the conference.

Chairperson Password

A numeric password used by participants to identify themselves as the
conference chairperson.

ID

The conference ID.

Profile

The name of the conference Profile from which conference parameters were
taken.

Line Rate

The maximum transfer rate, in kilobytes per second (Kbps) of the call (video
and audio streams).

Max Number of
Participants

Indicates the total number of participants that can be connected to the
conference. The Automatic setting indicates the maximum number of
participants that can be connected to the MCU according to resource
availability.

2 Click the Advanced tab.
The Conference Properties - Advanced dialog box opens.

3 The following information is displayed in the Advanced tab:

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Conference Properties - Advanced Parameters
Field/Option

Description

Encryption

Indicates whether the conference is encrypted.

Packet Loss
Compensation (LPR and
DBA)

Indicates wether Packet Loss Compensation (LPR and DBA) is enabled for the
conference.

Auto Terminate

When selected, indicates that the MCU will automatically terminate the
conference when Before First Joins, At the End-After Last Quits and At the End
- When Last Participant Remains parameters apply.

Auto Redialing

Indicates whether dial-out participants are automatically (when selected) or
manually (when cleared) connected to the conference.
This option is disabled in mixed CP and SVC conferences.

Exclusive Content Mode

When selected, Content is limited to one participant.

TIP Compatibility

Indicates the TIP Compatibility mode implemented for the conference, when
the environment implements the Collaboration Server and Cisco Telepresence
Systems (CTS) Integration solution.
• None
• Video Only
• Video & Content
• Prefer TIP
For more information, see Collaboration With Cisco’s Telepresence
Interoperability Protocol (TIP).

Enable FECC

When selected, Far End Camera Control is enabled.

FW NAT Keep Alive

When selected, sends a FW NAT Keep Alive message at specific Intervals for
the RTP, UDP and BFCP channels. The interval specifies how often a FW NAT
Keep Alive message is sent. For more information, see RealPresence
Collaboration Server 800s/Virtual Edition Network Port Usage.

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4 Click the Gathering Settings tab.
The Conference Properties - Gathering Settings dialog box opens.

The following information is displayed:
Profile - Gathering Settings
Field/Options

Description

Enable Gathering

Indicates whether the Gathering Phase has been enabled.

Display Language

Indicates the language of the Gathering Slide field headings.
Note: When working with the Polycom Conferencing Add-in for Microsoft Outlook,
the language selected should match the language selected for the conference in
the Polycom Conferencing Add-in for Microsoft Outlook to ensure that the
Gathering Phase slide displays correctly.

Info 1

Additional information to be displayed during the Gathering Phase.

Info 2
Info 3

5 Click the Video Quality tab.

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The Conference Properties - Video Quality dialog box opens.

The following information is displayed:
Conference Properties - Video Quality Parameters
Field/Option

Description

People Video Definition
Video Quality

Indicates the resolution and frame rate that determine the video quality set for
the conference. This is always Sharpness. For more information, see Video
Resolutions in AVC-based CP Conferencing.

Maximum Resolution

Indicates the Maximum Resolution setting for the conference.
• Auto (default) - indicates that the Maximum Resolution is as selected in the
Resolution Configuration dialog box.
The Maximum Resolution settings for conferences and participants cannot be
changed during an ongoing conference.

Content Video Definition
AS-SIP

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Conference Properties - Video Quality Parameters
Field/Option

Description

Multiple Content
Resolutions

Indicates if Multiple Content Resolutions mode for content sharing is enabled. In
this mode, content is shared in multiple streams, one for each video protocol:
H.263 and H.264. This allows endpoints with different protocols to connect and
disconnect without having to restart Content sharing in the middle of a
conference. For more information, see Sharing Content Using Multiple Content
Resolutions Mode.

Content Settings

Indicates the Content channel resolution set for the conference. Possible
resolutions are:
• Graphics – default mode
• Hi-res Graphics – requiring a higher bit rate
• Live Video – content channel is live video
• Customized Content Rate - resolution is manually defined.

Content Protocol

Indicates the Content Protocol used for content sharing in Highest Common
Content Sharing Mode.
For more information, see Content Protocols.

Content Resolution

Indicates the Content Resolution and frame rate according to the selected
Content Sharing Mode (Highest common Content or Multiple Resolution
Contents) and the video protocol. For more information, see Defining Content
Sharing Parameters for a Conference.

Send Content to Legacy
Endpoints
(CP only)

Indicates if the Send Content to Legacy Endpoints is enabled. If enabled,
Content can be sent to H.323/SIP endpoints that do not support H.239 Content
(legacy endpoints) over the video (people) channel. For more information see
Sending Content to Legacy Endpoints (AVC Only).

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6 Click the Video Settings tab to list the video parameters.

Conference Properties - Video Settings Parameters
Field

Description

Presentation Mode

When checked, indicates that the Presentations Mode is active.
This option is disabled in a mixed CP and SVC conference.
For more information, see Supplemental Conferencing Features.

Lecturer View
Switching

When checked, the Lecturer View Switching enables automatic random switching
between the conference participants in the lecturer video window.
This option is disabled in a mixed CP and SVC conference.

Same Layout

When checked, forces the selected layout on all conference participants, and the
Personal Layout option is disabled.
This option is disabled in a mixed CP and SVC conference.

Auto Layout

When enabled, the system automatically selects the conference layout based on
the number of participants in the conference.

Lecturer

Indicates the name of the lecturer (if one is selected). Selecting a lecturer enables
the Lecture Mode.
This option is disabled in a mixed CP and SVC conference.

Auto Scan Interval(s)

The time interval, 10 - 300 seconds, that Auto Scan uses to cycle the display of
participants that are not in the conference layout in the selected cell.
This option is disabled in a mixed CP and SVC conference.

Video Layouts
(graphic)

Indicates the currently selected video layout.

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7 Click the Audio Settings tab to view the audio setting for the conference.

8 If needed, you can enable or disable the Mute participants except lecturer setting.

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9 CP Only Conferences: Click the Customized Polling tab to view and modify the customized
polling for the conference.

All conference participants are listed in the left pane (All Participants) while the participants that are
to be displayed in the Auto Scan enabled cell of the video layout are listed in the right pane (Scanning
Order).
The dialog box buttons are summarized in the table below.
Customized Polling - Buttons
Button

Description

Add

Select a participant and click this button to Add a the participant to the list of
participants to be Auto Scanned.
The participants name is removed from the All Participants pane.

Delete

Select a participant and click this button to Delete the participant from the list of
participants to be Auto Scanned.
The participants name is moved back to the All Participants pane.

Add All

Add all participants to the list of participants to be Auto Scanned.
All participants’ names are removed from the All Participants pane.

Delete All

Delete all participant from the list of participants to be Auto Scanned.
All participants’ names are moved back to the All Participants pane.

Up

Select a participant and click this button to move the participant Up in the Scanning
Order.

Down

Select a participant and click this button to move the participant Down in the
Scanning Order.

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10 Click Apply to confirm and keep the Conference Properties dialog box open.
or
Click OK to confirm and return to the Collaboration Server Web Client Main Screen.
11 Click the Skins tab to view the skin selected for the conference.
You cannot select another skin during an ongoing conference.
12 Click the IVR tab to view the IVR settings.
13 Click the Information tab to view general information defined for the conference. Changes made to
this information once the conference is running are not saved to the CDR.
14 Click the Recording tab to review the recording settings for the conference.
15 Click the Site Names tab to enable or disable the display of site names during the conference, and
adjust the display properties.
16 Click the Message Overlay tab to send text messages to the conference participants during the
conference, and adjust the display properties of the text messages.

For more information, see Sending Text Messages During a Conference Using Message Overlay on
page 71.
17 Click the Network Services tab to verify the SIP registration for the conference.
18 Click OK to close the Conference Properties dialog box.

Viewing the Properties of Ongoing SVC-based Conferences
To view the parameters of an ongoing SVC conference:
1 In the Conference list pane, double-click the SVC conference or right-click the SVC conference and
then click Conference Properties.

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The Conference Properties - General dialog box with the General tab opens.

2 The following information is displayed in the General tab:
Conference Properties - General Parameters
Field

Description

Display Name

The Display Name is the conference name in native language and Unicode
character sets to be displayed in the Collaboration Server Web Client.
Note: This field is displayed in all tabs.

Duration

The expected duration of the conference using the format HH:MM.
Note: This field is displayed in all tabs.

Conferencing Mode

The conferencing mode for the conference.

Routing Name

The ASCII name of the conference. It can be used by H.323 and SIP participants
for dialing in directly to the conference. It is used to register the conference in the
gatekeeper and the SIP server.

Start Time

The time the conference started.

End Time

The expected conference end time.

Conference
Password

A numeric password for participants to access the conference.

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Conference Properties - General Parameters
Field

Description

Chairperson
Password

A numeric password used by participants to identify themselves as the conference
chairperson.

ID

The conference ID.

Profile

The name of the conference Profile from which conference parameters were taken.

Line Rate

The maximum transfer rate, in kilobytes per second (Kbps) of the call (video and
audio streams).

Max Number of
Participants

Indicates the total number of participants that can be connected to the conference.
The Automatic setting indicates the maximum number of participants that can be
connected to the MCU according to resource availability.

3 Click the Advanced tab.
The Conference Properties - Advanced dialog box opens.

4 The following information is displayed in the Advanced tab:

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Conference Properties - Advanced Parameters
Field/Option

Description

Auto Terminate

When selected, indicates that the MCU will automatically terminate the conference
when Before First Joins, At the End-After Last Quits and At the End - When Last
Participant Remains parameters apply.

Auto Redialing

Dial-out is not supported in SVC conferences.

Exclusive Content
Mode

When selected, Content is limited to one participant.

Enable FECC

Far End Camera Control is not supported in SVC conferences.

FW NAT Keep Alive

When selected, sends a FW NAT Keep Alive message at specific Intervals for the
RTP, UDP and BFCP channels. The interval specifies how often a FW NAT Keep
Alive message is sent. For more information, see RealPresence Collaboration
Server 800s/Virtual Edition Network Port Usage on page 37.

5 Click the Video Quality tab.
The Conference Properties - Video Quality dialog box opens.

The following information is displayed:

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Conference Properties - Video Quality Parameters
Field/Option

Description

People Video Definition
Video Quality

Indicates the resolution and frame rate that determine the video quality set for the
conference. Only Sharpness is supported.

Maximum Resolution

In SVC conferencing, this is always Auto (default) - The Maximum Resolution
remains as selected in the Resolution Configuration dialog box.

Content Video Definition
AS-SIP

This option is not supported with Polycom® RealPresence® Collaboration Server
800s and Polycom® RealPresence® Collaboration Server Virtual Edition.

Multiple Content
Resolutions

Multiple Content Resolutions is not supported in SVC conferences.

Content Settings

In SVC conferencing, this is always set to Graphics

Content Protocol

In SVC conferencing this is always set to H.264 Cascade and SVC Optimized.

Content Resolution

Resolution is fixed in SVC conferences.

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6 Click the Video Settings tab to view the video parameters defined for the conference.

In SVC conferences, only Auto Layout is enabled and cannot be disabled. All other video settings are
disabled.

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7 Click the Audio Settings tab to view the audio parameters defined for the conference.

In SVC conferences, all Audio Settings options are disabled.
8 Click the Information tab to view general information defined for the conference. Changes made to
this information once the conference is running are not saved to the CDR.
9 Click OK to close the Conference Properties dialog box.

Monitoring of Operator Conferences and Participants Requiring
Assistance (CP and Mixed CP and SVC Conferences)
Operator conferences are monitored in the same way as standard ongoing conferences.
Each Operator conference includes at least one participant - the Operator.

You can view the properties of the Operator conference by double-clicking the conference entry in the
Conferences list or by right-clicking the conference entry and selecting Conference Properties. For more
information, see the Polycom® RealPresence Collaboration Server 800s / Virtual Edition Getting Started
Guide, Conference Level Monitoring.

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Requesting Help
A participant can request help using the appropriate DTMF code from his/her touch tone telephone or the
endpoint’s DTMF input device. The participant can request Individual Assistance (default DTMF code *0) or
Conference Assistance (default DTMF code 00).
Participants in Entry Queues who failed to enter the correct destination conference ID or the conference
password will wait for operator assistance (provided that an Operator conference is active).
When requiring or requesting operator assistance, the Collaboration Server management application
displays the following:

● The participant’s connection Status changes, reflecting the help request. For details, see Table 5-9.
● The conference status changes and it is displayed with the exclamation point icon and the status
“Awaiting Operator”.
● The appropriate voice message is played to the relevant participants indicating that assistance will
be provided shortly.
The following icons and statuses are displayed in the Participant Status column:
Participants List Status Column Icons and Indications
Icon

Status indication

Description

Awaiting Individual Assistance

The participant has requested the operator’s assistance for
himself/herself.

Awaiting Conference
Assistance

The participant has requested the operator’s assistance for the
conference. Usually this means that the operator is requested to
join the conference.

When the Operator moves the participant to the Operator conference for individual assistance the
participant Status indications are cleared.

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Request to Speak
Participants that were muted by the conference organizer/system operator can indicate that they want to be
unmuted by entering the appropriate DTMF code.
An icon is displayed in the Role column of the Participants list for 30 seconds.

Request to Speak is:
● Activated when the participant enters the appropriate DTMF code (default: 99).
The DTMF code can be modified in the conference IVR Service Properties - DTMF Codes dialog box.
● Available for dial-in and dial-out participants.
● A participant can request to speak more than once during the conference.
● Supported in all conference types.
● Supported in H.323 and SIP environments.
● The duration of the icon display cannot be modified.

Participant Alerts List
The Participant Alerts list contains all the participants who are currently waiting for operator assistance.

Participants are automatically added to the Participants Alerts list in the following circumstances:
● The participant fails to connect to the conference by entering the wrong conference ID or conference
password and waits for the operator’s assistance.
● The participant requests Operator’s Assistance during the ongoing conference.
This list is used as reference only. Participants can be assisted and moved to the Operator conference or
the destination conference only from the Participants list of the Entry Queues or ongoing conference where
they are awaiting assistance.
The participants are automatically removed from the Participant Alerts list when moved to any conference
(including the Operator conference).

Participant Level Monitoring
In addition to conference information, you can view detailed information regarding the status and
parameters of each listed participant, using the Participant Properties dialog box. Participant properties can
be displayed for all participants currently connected to a conference and for defined participants that have
been disconnected.

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SIP SVC-based participant properties are similar to SIP AVC-based participant properties.

The table below lists the tabs in the Participant Properties dialog box, as viewed by each user type, for each
participant connection types.
Participant monitoring - Tab list per participant connection type and user
Admin

Chairperson

Operator

Tab Name

AVC H.323

AVC/SVC SIP

AVC H.323

AVC/SVC SIP

AVC H.323

AVC/SVC SIP

General

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





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

Advanced













Information













Media Sources













H.245













SDP













Connection
Status













Channel Status













Channel Status Advanced













Gatekeeper
Status













Call Admission
Control













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Viewing the Properties of Participants
To view the participant Properties:
» In the Participant List pane double-click the participant entry. Alternatively, right-click a participant and
then click Participant Properties.
The Participant Properties dialog box opens, displaying the last opened tab.
Media Sources properties are not available for SVC participants.

The Media Sources dialog box enables you to mute participant’s audio, suspend participant’s video
transmission and select a personal Video Layout for the participant.

Monitoring IP Participants
The following parameters are displayed for an IP participant.

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Participant Properties - Media Sources Parameters
Field

Description

Name

Indicates the participant’s name.
Note: This field is displayed in all tabs.

Endpoint Website
(link)

Click the Endpoint Website hyperlink to connect to the internal website of the participant’s
endpoint. It enables you to perform administrative, configuration and troubleshooting
activities on the endpoint.
The connection is available only if the IP address of the endpoint’s internal site is filled in
the Website IP Address field in the Participant Properties - General dialog box.
Note: This field is displayed in all tabs

Endpoint Type

Indicates whether the participant is using an AVC-based or SVC-based endpoint.
Fields, tabs and options are enabled or disabled according to the endpoint type.
Note: This field is displayed in all tabs.

Layout Type

Indicates whether the video layout currently viewed by the participant is the Conference or
Personal Layout.
If Personal Layout is selected, you can select a Video Layout that will be viewed only by
this participant.

Video Layout

Indicates the video layout currently viewed by the participant. When Personal Layout is
selected in the Layout Type you can force participants to the video windows in a layout
that is specific to the participant. For more information, see Polycom® RealPresence
Collaboration Server 800s / Virtual Edition Getting Started Guide, Changing the Video
Layout of a Conference (AVC-Based CP and Mixed CP and SVC Conferences).

Mute/Suspend

Indicates if the endpoint’s audio and/or video channels have been muted/suspended. The
entity that initiated audio mute or video suspend is also indicated.
• MCU – Audio or Video channel has been muted/suspended by the MCU.
• User – Channels have been muted/suspended by the Collaboration Server user.
• Participant – Channels have been muted/suspended by the participant from the
endpoint.
You can also cancel or perform mute and suspend operation using these check boxes.
Note: If the participant muted his/her audio channel, the system displays the mute icon
only for H.323.

Block

When checked, the audio transmission from the conference to the participant’s endpoint is
blocked, but the participant will still be heard by other participants.

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1 Click the Connection Status tab to view the connection status, and if disconnected the cause of the
disconnection.

This dialog box is the same for AVC-based and SVC-based participants.
The following parameters are displayed:
Participant Properties - Connection Status Parameters
Field

Description

Participant Status
Status

Indicates the connection status of the participant.

Connection Time

The date and time the participant connected to the conference.
Note: The time format is derived from the MCU’s operating system
time format.

Disconnection Time

The date and time the defined participant disconnected from the
conference.

Connection Retries
Left

Indicates the number of retries left for the system to connect defined
participant to the conference.

Call Disconnection
Cause

Displays the cause for the defined participant’s disconnection from
the conference. See Appendix A: Appendix A - Disconnection Causes
on page 923.

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Participant Properties - Connection Status Parameters
Field

Description

Video Disconnection
Cause

Displays the cause the video channel could not be connected. For
more information, see Appendix A: Appendix A - Disconnection
Causes on page 923.

Possible Solution

In some cases, a possible solution is indicated to the cause of the
video disconnection.

2 Click the H.245 (H.323) or SDP (SIP) tab during or after the participant’s connection process to view
information that can help in resolving connection issues.

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Participant Properties - H.245/SDP Parameters
Field

Description

Remote Capabilities

Lists the participant’s capabilities as declared by the endpoint.

Remote
Communication Mode

Displays the actual capabilities used by the endpoint when
establishing the connection with the MCU (Endpoint to MCU).

Local Communication
Mode

Displays the actual capabilities used by the MCU when establishing
the connection with the participant’s endpoint (MCU to Endpoint).

3 Click on the Channel Status tab to view the status of the various channels.

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The following parameters are displayed:
Participant Properties - Channel Status Parameters
Field

Description

Channels Used

When checked, indicates the channel type used by the participant to connect to the
conference: Incoming channels are endpoint to MCU, Outgoing channels are from
MCU to endpoint.
Channels:
• H.225/Signaling - The call-signaling channel.
• H.245/SDP - The Control channel.
• Audio in - Incoming audio channel
• Audio out - Outgoing audio channel
• Video in - Incoming video channel
• Video out - Outgoing video channel
• Content in - H.239/People+Content conferences
• Content out - H.239/People+Content conferences
• FECC in - The incoming FECC channel is open.
• FECC out - The outgoing FECC channel is open.
Columns:
• Faulty – A red exclamation point indicates a faulty channel condition. This is a
real-time indication; when resolved the indication disappears. An exclamation
point indicates that further investigation may be required using additional
parameters displayed in the Advanced Channel Status tab.
• Bit Rate – The actual transfer rate for the channel. When channel is inactive, bit
rate value is 0. For example, if the participant is connected without video, the bit
rate for the video channel is 0.
Note: The CTS Audio Auxiliary channel is used only for Content. In all other
cases, the bit rate shown in this column for this channel is 0.
• Packet Loss – The accumulated count of all packets that are missing according
to the RTCP report since the channel was opened. This field is relevant only
during the connection stage and does not display faulty indications.
• Fraction Loss (Peak) – The ratio between the number of lost packets and the
total number of transmitted packets since the last RTCP report. Peak (in
parentheses) indicates the highest ratio recorded since the channel was opened.
• Number of Packets – The number of received or transmitted packets since the
channel has opened. This field does not cause the display of the faulty indicator.
• Jitter (Peak) – Displays the network jitter (the deviation in time between the
packets) as reported in the last RTCP report (in milliseconds). Peak (in
parentheses) reflects the maximum network jitter since the channel was opened.
• Latency – Indicates the time it takes a packet to travel from one end to another
in milliseconds (derived from the RTCP report).
High latency value may indicate that there is a problem in the network, or that
the endpoint is sending an incorrect RTCP values.

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Participant Properties - Channel Status Parameters
Field

Description

Sync Status

•
•
•
•
•
•

Channel - The channel type: Video or Content.
Source - The name of the participant currently viewed by this participant.
Position - The video layout position indicating the place of each participant as
they appear in a conference.
Protocol Sync Loss - Indicates whether the system was able to synchronize the
bits order according to the selected video protocol.
Video Intra Sync - Indicates whether the synchronization on a video Intra frame
was successful.
Video Resolution - The video resolution of the participant.

Rx - Rate

The received line rate.

Tx - Rate

The transmitted line rate.

Tx - Video Sync Loss

When checked, indicates a video synchronization problem in the outgoing channel
from the MCU.
The counter indicates the sync-loss count.

Rx - Video Sync Loss

When checked, indicates a video synchronization problem in the incoming channel
from the endpoint.
The counter indicates the sync-loss count.

Tx - LPR Activation

When checked, indicates LPR activation in the outgoing channel.

Rx - LPR Activation

When checked, indicates LPR activation in the incoming channel.

FECC Token

When checked, indicates that the participant is the holder of the FECC Token.

Content Token

When checked, indicates that the participant is the holder of the Content Token.

4 Click the Channel Status Advanced tab to view additional information for selected audio and video
channels.

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In the Channel Status - Advanced tab, channels can be selected for viewing additional information:

Participant Properties - Channel Status Advanced Parameters
Field

Description

Channel Info

Select a channel to view its information:
• H.225
• H.245
• Audio in
• Audio out
• Video in
• Video out
• Content in
• Content Out
• SIP BFCP TCP

Collaboration Server
IP Address

The IP address and the transport protocol (TCP/UDP) of the MCU to
which the participant is connected and the port number allocated to
the participant incoming media stream on the MCU side.

Participant IP
Address

The IP address and the transport protocol (TCP/UDP) of the
participant and the port number allocated to the media stream on the
participant side.

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Participant Properties - Channel Status Advanced Parameters
Field

Description

ICE Polycom®
RealPresence®
Collaboration Server
800s and Polycom®
RealPresence®
Collaboration Server
Virtual Edition IP
Address

The IP address, port number, and transport protocol of the MCU used
to pass through the media when ICE is functional. See Participant
Properties - ICE Connection Parameters.

ICE Participant IP
Address

The IP address, port number, and transport protocol of the endpoint
used to pass through the media when ICE is functional. See
Monitoring the Participant Connection in ICE Environment.

ICE Connection Type

Indicates the type of connection between the Collaboration Server
and the participant in the ICE environment:
• Local (or Host) - The endpoint (Remote) is on the same network
as the Collaboration Server and the media connection is direct,
using local addresses.
• Relay - Media between the Collaboration Server and the
participant passes through a media relay server.
• Firewall - Media connection between the Collaboration Server and
the participant is done using their external IP addresses (the IP
addresses as seen outside of the local network).

Media Info

This table provides information about the audio and video parameters,
such as video algorithm, resolution, etc. For more information, see
Appendix E - Participant Properties Advanced Channel Information.

RTP Statistics

This information may indicate problems with the network which can
affect the audio and video quality. For more information, see Appendix
E - Participant Properties Advanced Channel Information.

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5 Optional for H.323 AVC-based participants. Click the Gatekeeper Status tab to view its
parameters.

Participant Properties - Gatekeeper Status Parameters
Field

Description

Requested
Bandwidth

The bandwidth requested by the MCU from the gatekeeper.

Allocated Bandwidth

The actual bandwidth allocated by the gatekeeper to the MCU.

Required Info Interval

Indicates the interval, in seconds, between registration messages
that the MCU sends to the gatekeeper to indicate that it is still
connected.

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Participant Properties - Gatekeeper Status Parameters
Field

Description

Gatekeeper State

Indicates the status of the participant’s registration with the
gatekeeper and the bandwidth allocated to the participant. The
following statuses may be displayed:
• ARQ – Admission Request - indicates that the participant has
requested the gatekeeper to allocate the required bandwidth on
the LAN.
• Admitted – indicates that the gatekeeper has allocated the
required bandwidth to the participant.
• DRQ – Disengage Request – the endpoint informs the gatekeeper
that the connection to the conference is terminated and requests
to disconnect the call and free the resources.
• None – indicates that there is no connection to the gatekeeper.

6 Optional for SIP AVC-based and SVC-based participants. Click the Call Admission Control tab
to view its parameters.

Participant Properties - Gatekeeper Status Parameters
Field

Description

Requested
Bandwidth

The bandwidth requested by the MCU from the SIP server.

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Participant Properties - Gatekeeper Status Parameters
Field

Description

Allocated Bandwidth

The actual bandwidth allocated by the SIP server to the MCU.

Monitoring SIP BFCP Content
In the SIP Participant Properties dialog box, BFCP status information appears in:
● All three panes of the SDP tab.
● The Channel Status tab.
● The Channel Status -Advanced tab.

For more information see Participant Level Monitoring.

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Detecting Endpoint Disconnection
Endpoint detection can be detected for SIP and H323 protocols.
Note: H323 endpoint disconnection detection is not supported in the RealPresence Collaboration
Server 800s.

Detecting SIP Endpoint Disconnection
When an abnormal disconnection of SIP endpoints occurs because of network problems or client
application failures, SIP endpoints remain connected to the conference causing connection disruptions. For
example, the video freezes in the layout or blocks content for SIP endpoints when a quick re-connection is
performed. It can take several minutes to detect the SIP endpoint disconnection using the SIP standard
behavior.
In a normal SIP video call, audio and video (RTP and RTCP) messages are sent from the endpoints to the
MCU to detect the signaling of connected endpoints. Conversely, SVC endpoints might not send video RTP
messages to the MCU when a participant is not displayed in the video layout of any of the participants in the
conference. For SVC endpoints, the MCU will only verify audio RTP and RTCP messages and video RTCP
messages. Video RTP messages will not be checked.
To detect the disconnection of SIP endpoints in a reasonable amount of time, a new system flag can be
defined to specify the amount of time that the MCU should wait for an RTCP or RTP message from the SIP
endpoint before the endpoint starts the disconnection process. The system default value is automatically
set to 20 seconds.
The system flag, DETECT_SIP_EP_DISCONNECT _TIMER, contains the amount of time in seconds to
wait for an RTCP or RTP message to be received from the endpoint. When the time that was set in the
system flag has elapsed and no RTCP or RTP audio or video message has been received on either the
audio or the video channel, the MCU disconnects the SIP endpoint from the conference. A CDR event
record is created with a Call Disconnection Cause of “SIP remote stopped responding”.
The Microsoft Lync add-in endpoint opens audio and content channels. Lync endpoints can send
RTCP/RTP messages and empty RTP audio messages. When the time that was set in the system flag has
elapsed and no RTCP or RTP message has been received on the audio channel, the MCU disconnects the
endpoint from the conference.
SIP audio only endpoints use the audio channel only. When the time that was set in the system flag has
elapsed and no RTCP or RTP message has been received on the audio channel, the MCU disconnects the
SIP audio endpoint from the conference.

H323 Endpoint Disconnection Detection
In versions previous to version 8.4, when an H.323 endpoint disconnected, round trip messages were still
received, even when a gatekeeper was deployed. Because of this, the MCU failed to detect the endpoint
disconnection.
From version 8.4 onward, the MCU detects H.323 endpoint connection or disconnection by monitoring
RTCP/RTP messages reception using either the audio or video channels. When these messages are not
received within a predefined timeout interval, the endpoint is considered disconnected. Therefore as long
as either RTCP or RTP messages (interchangeably) are received on either of the video or audio channels,
the endpoint is considered connected.

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If no messages are received through either channel within the predefined time out interval, the endpoint is
disconnected, and a disconnection message, H.323 remote stopped responding, is sent to the
endpoint.
No channel disconnection detection occurs while an endpoint’s video or audio are muted, or while the
endpoint is put on hold.
In audio calls, only the audio channel is polled for RTCP/RTP messages.

Configuring the System Flag
The time out used for SIP endpoint disconnection detection is controlled by the
DETECT_SIP_EP_DISCONNECT_TIMER System Flag which must be added to the System Configuration
to view or modify its value.
Range: 0 - 300
When the value is set between 0 and 14, the feature is disabled and SIP endpoints are not detected for
disconnection. When the value is set between 15 and 300, the feature is enabled.
Default: 20
For more information see Manually Adding and Deleting System Flags, and Modifying System Flags in the
RealPresence Collaboration Server (RMX) Administrator’s Guide.

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Monitoring Telepresence Participant Properties
A Telepresence status indicator is displayed in the Participant Properties - Advanced tab when monitoring
conference participants.

The Telepresence mode of the participant is indicated:
● RPX - the participant’s endpoint is transmitting 4:3 video format.
● TPX - the participant’s endpoint is transmitting 16:9 video format.
None - the participant’s endpoint is neither RPX nor TPX.

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Recording Conferences
Conference recording is not available in SVC Conferencing Mode.

Conferences running on the Collaboration Server can be recorded using Polycom RealPresence® Capture
Server in the following ways:
● Capture Server dials in MCU for conference recording via an SIP interface from Capture Server. From
the Capture Server Admin UI, you can start a conference recording by dialing out to a Collaboration
Server. After the Collaboration Server gets the request from the Capture Server, the recording will
start if the recording is enabled on the Collaboration Server. During the conference recording, you
can pause or stop the recording from the Capture Server Admin UI. Refer to the Polycom
RealPresence Capture Server User’s Guide on how to start a recording by dialing out to an
interoperable endpoint.
● Collaboration Server dials out to Capture Server for a conference recording. Recording conferences
is enabled via a dial-out Recording Link, which is a dial-out connection from the conference to the
recording system.
The recording system can be installed at the same site as the conferencing MCU or at a remote site. Several
MCU’s can share the same recording system.
Recording conferences is enabled via a Dial Out Recording Link, which is a dial-out connection from the
conference to the recording system.
Recording can start automatically, when the first participant connects to a conference, or on request, when
the Collaboration Server user or conference chairperson initiates it.
Multiple Dial Out Recording Links may be defined.
Conference Dial Out Recording Links can be associated on the Collaboration Server with Virtual Recording
Rooms (VRR), created and saved on the Polycom® RSS™ 4000 Version 8.5 Recording and Streaming
Server (RSS).
Each Dial Out Recording Link defined on the Collaboration Server can be given a descriptive name and can
be associated with one VRR saved on the Polycom RSS 4000

Creating Multiple Virtual Recording Rooms on the RSS
If the environment includes a Polycom® RSS™ 4000 Version 8.5 Recording and Streaming Server (RSS)
and you want to associate Recording Links on the Collaboration Server with Virtual Recording Rooms
(VRR), created and saved on the Polycom® RSS™ 4000 Version 8.5 perform the following operations on
the RSS:
1 Modify the parameters of a recording Template to meet the recording requirements.
2 Assign the modified recording Template to a VRR. The recording and streaming server will assign a
number to the VRR.

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3 Repeat step 1 and step 2 for each VRR to create additional VRRs.
For more information see the RSS 4000 User Guide.

Configuring the Collaboration Server to Enable
Recording
To make recording possible the following components you must be configured on the Collaboration Server:
● Recording Link – defines the connection between the conference and the recording system.
● Recording-enabled Conference IVR Service – recording DTMF codes and messages must be set in
the Conference IVR Service to enable “recording-related” voice messages to be played and to allow
the conference chairperson to control the recording process using DTMF codes.
● Recording-enabled Profile – recording must be enabled in the Conference Profile assigned to the
recorded conference.
If Multiple Recording Links are being defined for Virtual Recording Rooms (VRRs), created and saved on
the Polycom® RSS™ 4000 Version 8.5, the MAXIMUM_RECORDING_LINKS System Flag in system.cfg
can be modified to determine the number of Recording Links available for selection.
● Range: 20 - 100
● Default: 20
The flag value can be modified by selecting the System Configuration option from the Setup menu. For more
information, see Modifying System Flags.

Defining the Recording Link
The Recording Link is defined once and can be updated when the H.323 alias or the IP address (of the
recording system) is changed. Only one Recording Link can be defined in the Collaboration Server. Its type
must be H.323.
In Multiple Networks Configuration, Recording Links use the default Network Service to connect to
conferences, therefore the recording system must be defined on the default IP Network Service to
enable the recording.

To define a Recording Link:
1 In the RMX Management pane, click Recording Links (

).

2 In the Recording Links list, click the New Recording Link (

) button.

The New Recording Link dialog box is displayed.

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3 Define the following parameters:
Recording Link Parameters
Parameter

Description

Name

Displays the default name that is assigned to the Recording Link.
If multiple Recording Links are defined, it is recommended to use a descriptive
name to be indicate the VRR to which it will be associated.
Default: Recording Link

Type

Select the network environment:
• H.323
• SIP

IP Address

•
•

If no gatekeeper is configured, enter the IP Address of the RSS. Example: If the
RSS IP address is 173.26.120.2 enter 173.26.120.2.
If a gatekeeper is configured, you can either enter the IP address or an alias (see
the alias description).

Alias Name

If using the endpoint’s alias instead of IP address, first select the alias type and
then enter the endpoint’s alias.
If you are associating this recording link to a VRR on the RSS, define the alias as
follows:
• If you are using the RSS IP address, enter the VRR number in the Alias field. For
example, if the VRR number is 5555, enter 5555.
• Alternatively, if the Alias Type is set to H.323 ID, enter the RSS IP address and
the VRR number in the format:
##
For example: If the RSS IP is 173.26.120.2 and the VRR number is 5555, enter
173.26.120.2##5555

Alias Type

Depending on the format used to enter the information in the IP address and Alias
fields, select H.323 ID or E.164 (for multiple Recording links). E-mail ID and
Participant Number are also available.

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4 Click OK.
The Recording Link is added to the Collaboration Server unit.

Enabling the Recording Features in a Conference IVR Service
To record a conference, a Conference IVR Service in which the recording messages and DTMF codes are
activated must be assigned to the conference. The default Conference IVR Service shipped with the
Collaboration Server includes the recording-related voice messages and default DTMF codes that enable
the conference chairperson to control the recording process from the endpoint. You can modify these default
settings.

To modify the default recording settings for an existing Conference IVR Service:
1 In the RMX Management pane, click the IVR Services (

) button.

The IVR Services are listed in the IVR Services list pane.
2 To modify the default recording settings, double-click the Conference IVR Service or right-click and
select Properties.
The Conference IVR Service Properties dialog box is displayed.
3 To assign voice messages other than the default, click the General tab and scroll down the list of
messages to the recording messages.

4 Select the Recording In Progress message, and then select the appropriate message file (by
default, Recording_in_Progress.wav) from the file list to the right of the field.
5 Select the Recording Failed message, and then select the appropriate message file (by default,
Recording_Failed.wav) from the file list to the right of the field.
6 To modify the default DTMF codes, click the DTMF Codes tab.
7 To modify the DTMF code or permission for a recording function:
a Select the desired DTMF name (Start, Stop or Pause Recording), click the DTMF code entry and
type a new code.

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Default DTMF Codes assigned to the recording process
Recording Operation

DTMF Code

Permission

Start or Resume Recording

*2

Chairperson

Stop Recording

*3

Chairperson

Pause Recording

*1

Chairperson

b In the Permission entry, select whether this function can be used by all conference participants
or only the chairperson.
8 Click OK.

Enabling the Recording in the Conference Profile
To be able to record a conference, the recording options must be enabled in the Conference Profile
assigned to it. You can add recording to existing Profiles by modifying them.

To enable recording for a conference:
1 In the Collaboration Server Management pane, click the Conference Profiles (

) button.

The Conference Profiles list is displayed.
2 Create a new profile by clicking the New Profile ( ) button or modify an existing profile by
double-clicking or right-clicking an existing profile and then selecting Profile Properties.
If creating a new profile, complete the conference definition. For more information on creating Profiles
see Defining AVC CP Conferencing Profiles.

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3 In the Profile Properties dialog box, click the Recording tab.

4 Select the Enable Recording check box.
5 Define the following parameters
Conference Profile Recording Parameters:
Parameter

Description

Enable Recording

Select to enable Recording Settings in the dialog box.

Recording Link

Select a recording link for the conference from the list.

Start recording

Select one of the following:
• Immediately – conference recording is automatically started upon
connection of the first participant.
• Upon Request – the operator or chairperson must initiate the
recording (manual).

Audio only

Select this option to record only the audio channel of the conference.
Note:
An Audio Only Recording Link cannot be used to record a conference
if there are no Voice resources allocated in the Video/Voice Port
Configuration.

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Parameter

Description

Display Recording
Icon

Select this option to display Recording Indication to all conference
participants informing them that the conference is being recorded.
The recording icon is replaced by a Paused icon when conference
recording is paused.

6 Click OK.
Recording is enabled in the Conference Profile.

Recording Link Encryption
The Recording Link can be encrypted when recording an encrypted conference. The encryption of the
Recording Link is enabled when Encryption is selected in the Conference Profile on the Collaboration
Server and on the RSS, and the system flag
ALLOW_NON_ENCRYPT_RECORDING_LINK_IN_ENCRYPT_CONF is set to NO.

Recording Link Encryption Guidelines:
● The Recording Link connection type must be H.323.
● The Recording Link uses the AES encryption format.
● The RSS 4000 recorder must be set to support encryption. For more information see the RSS 4000
User Manual.
● Encryption must be selected in the Conference Profile.

Recording Link Encryption Flag Setting
Recording Links are treated as regular participants, however if the
ALLOW_NON_ENCRYPT_RECORDING_LINK_IN_ENCRYPT_CONF System Flag is set to YES a
non-encrypted Recording Link is to be allowed to connect to an encrypted conference.
The following table summarizes the connection possibilities for a Recording Link that is to be connected to
a conference for each of the conference profile and Entry Queue encryption options.
Connections by Recording Link and Conference Encryption Settings

Conference Profile
Setting

Recording Link Connection Status according to flag:
ALLOW_NON_ENCRYPT_RECORDING_
LINK_IN_ENCRYPT_CONF
YES

NO

Encrypt All

Connected encrypted if possible,
otherwise connected
non-encrypted.

Connected only if encrypted,
otherwise disconnected.

No Encryption

Connected non-encrypted.

Connected non-encrypted.

Encrypt when possible

Connected encrypted if possible,
otherwise connected
non-encrypted.

Connected encrypted if possible,
otherwise connected
non-encrypted.

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Recording Link Settings
The recording of encrypted conferences via an encrypted Recording Link is enabled in the Conference
Profile by:
● Selecting the Encryption option (Encrypt All or Encrypt when Possible) in the Advanced tab.
For more details, see Packet Loss Compensation (LPR and DBA) AVC CP Conferences.
● Setting the Recording options in the Recording tab. For more details, see Enabling the Recording in
the Conference Profile.

Managing the Recording Process
When a conference is started and recording is enabled in its Profile, the system will automatically start the
recording if the Start Recording parameter is set to immediately. If it is set to Upon Request, the system waits
for the chairperson or Collaboration Server user’s request. Once the recording is initiated for a
conference, the MCU connects to the recording device (Capture Server). The connection that is created
between the conference and the recording device is represented as a special participant (Recording) whose
name is the Recording Link. Once the recording has started, the recording process can be stopped and
restarted from the Chairperson’s endpoint (using DTMF codes) or from the Collaboration Server Web Client.
After the recording process has finished, the recording can be identified in the Capture Server by its
Collaboration Server conference name.
.

A conference participant and the Recording Link cannot have identical names, otherwise the
recording process will fail.

Recording Layout
When the video layout of the conference is set to Auto Layout, the recording of the conference will now
include all the conference participants and not n-1 participants as in previous versions.
In the new Auto Layout algorithm, the Recording Link is counted as a “participant” and therefore it is
excluded from the layout display used for the recording. The layout used for the other participants will
behave as in the “standard” Auto Layout behavior.
The Recording Link Layout can be changed during an ongoing conference in the same manner as for any
other conference participant. For more information see the Participant Level Monitoring.
The default settings for Auto Layout for the conference and the Recording Link are summarized in the
following table:
Recording Link Default Layout Settings (Auto Layout Mode)

Participants

Conference Auto Layout Default
Settings

Recording Link Auto Layout
Settings

0

Not applicable

Not applicable

1

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Participants

Conference Auto Layout Default
Settings

Recording Link Auto Layout
Settings

2

3

4

5
6
7
8
9
10 or more

When Capture Server dials in MCU for conference recording, you can choose the video layout from the
Capture Server Admin UI. The following layout is supported by the Collaboration Server.
● Auto: automatic layout setting at the Collaboration Server side.
● Single View: record 1*1 layout
● Dual View: record 1*2 layout.The default settings for Auto Layout of the Recording Link cannot be
changed, and the Auto Layout flags do not apply to the Recording Link Auto Layout default settings.

Using the Collaboration Server Web Client to Manage the Recording
Process
To manage the recording process using the right-click menu:
● Right-click the Recording participant in the conference and select from one of the following options:

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Recording Participant Right-click Options
Name

Description

Suspend Video

The Suspend Video option prevents the incoming video of the recording link
participant to be part of the conference layout.
The Recording Link participant is set by default to Suspend Video. The Suspend
Video option toggles with the Resume Video option.

Resume Video

The Resume Video option enables the incoming video of the recording link
participant to be part of the conference layout.
This feature may be used to play back previously recorded video or audio feeds in
the conference layout. For more information, see the RSS 4000 User Guide.

Participant Properties

The Participant Properties option displays viewing only information for monitoring,
e.g. communication capabilities and channels used to connect to the conference.
Users will not be able to perform any functional requests from this window, i.e.
disconnect, change layout and mute.

To manage the recording process using the Conference toolbar:
In the Conferences pane, click one of the following buttons in the Conference tool bar.

The recording buttons will only be displayed in the conference tool bar for a conference that is
recording-enabled.

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Conferences List - Recording Tool bar buttons
Button

Description
Start/Resume recording. This button toggles with the Pause button.
Stop recording.
Pause recording. This button toggles with the Start/Resume button.

Using DTMF Codes to Manage the Recording Process
By entering the appropriate DTMF code on the endpoint, the chairperson can Stop the recording (*74),
Pause it (*75), or Start/Resume the recording (*73). For more information on managing the recording
process via DTMF codes, see the RSS 2000 User’s Guide.

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Collaboration Server Users
Collaboration Server Web Client users are defined in the User’s table and can connect to the MCU to
perform various operations.
A maximum of 100 users can be defined per MCU.

User Types
The MCU supports the following user Authorization Levels:
 Administrator
 Operator
 Machine Account (Application-user)
 Administrator Read-only
 Chairperson
 Auditor
Users with Auditor authorization level cannot connect to the Collaboration Server via the RMX Manager
application and must use the Collaboration Server Web Client.

The authorization level determines a user’s capabilities within the system.

Administrator
An administrator can define and delete other users, and perform all configuration and maintenance tasks.

Administrator Read-only
A user with Administrator permission with the same viewing and monitoring permissions of a regular
Administrator. However, this user is limited to creating system backups and cannot perform any other
configuration or conference related operation.

Operator
An Operator can manage Meeting Rooms, Profiles, Entry Queues, and SIP Factories, and can also view
the Collaboration Server configurations, but cannot change them.
Administrator and Operator users can verify which users are defined in the system. Neither of them can view
the user passwords, but an Administrator can change a password.

Chairperson
A Chairperson can only manage ongoing conferences and participants. The Chairperson does not have
access to the Collaboration Server configurations and utilities.
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Auditor
An Auditor can only view Auditor Files and audit the system.

Machine Account
User names can be associated with servers (machines) to ensure that all users are subject to the same
account and password policies. For more details, see Machine Account.

Listing Users
The Users pane lists the currently defined users in the system and their authorization levels. The pane also
enables the administrators to add and delete users.
The system is shipped with a default Administrator user called POLYCOM, whose password is POLYCOM.
However, once you have defined other authorized Administrator users, it is recommended to remove the
default user.
You can view the list of users that are currently defined in the system.

To view the users currently defined in the system:
1 In the RMX Management pane, click the Users (

) button.

The Users pane is displayed.

The list includes three columns: User Name, Authorization Level and Disabled.
The User Name is the login name used by the user to connect to the MCU.
The Authorization indicates the Authorization Level assigned to the User: Administrator,
Administrator Read-only, Operator, Chairperson or Auditor.
Disabled indicates whether the user is disabled and cannot access the system unless enabled by
the administrator. For more details, see Disabling a User.
Locked indicates whether the user has been locked out and cannot access the system unless
enabled by the administrator.

Adding a New User
Administrators can add new users to the system.
The User Name and Password must be in ASCII.

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To add a new user to the system:
1 In the RMX Management pane, click the Users (

) button.

2 The Users pane is displayed.
3 Click the New User (

) button or right-click anywhere in the pane and then click New User.

The User Properties dialog box opens.

4 In the User Name text box, enter the name of the new user. This is the login name used by the user
when logging into the system.
5 In the Password text box, enter the new user’s password. This will be the user’s password when
logging into the system.
6 In the Authorization Level list, select the user type: Administrator, Administrator Read-Only,
Operator, Chairperson or Auditor.
7 Optional. To associate a user with a machine:
a In the User Properties dialog box, select the Associate with a machine check box.
b Enter the FQDN of the server that hosts the application who’s application-user name is being
added. Example: cma1.polycom.com
8 Click OK.
The User Properties dialog box closes and the new user is added to the system.

Deleting a User
To delete a user, you must have Administrator authorization. The last remaining Administrator in the
Users list cannot be deleted.

1 In the RMX Management pane, click the Users (
2 Select the user and click the Delete (

) button.

) button or right-click the user and then click Delete User.

The system displays a confirmation message.
3 In the confirmation dialog box, select Yes to confirm or No to cancel the operation.
If you select Yes, the user name and icon are removed from the system.

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Changing a User’s Password
Users with Administrator authorization can change their own password and other users’ passwords. Users
with Operator authorization can change their own password.

To change a user’s password:
1 In the RMX Management pane, click the Users (

) option.

2 Right-click the user and click Change User Password.
The Change Password dialog box opens.

3 Enter the Old Password (current), New Password and Confirm the New Password.
The Password must be in ASCII.

4 Click OK.
The user’s password is changed.

Disabling a User
An administrator can disable an enabled user. An indication is displayed in the Users List when the User is
disabled. An administrator can enable a disabled User.

To disable a user:
1 In the RMX Management pane, click the Users (

) button.

The Users pane is displayed.
2 In the Users pane, right-click the user to be disabled and select Disable User in the menu.

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A confirmation box is displayed.

3 Click YES.
The User status in the Users list - Disabled column changes to Yes.

Enabling a User
An administrator can enable a User who was disabled manually by the administrator.

To enable a user:
1 In the RMX Management pane, click the Users (

) button.

The Users pane is displayed.
2 Right-click the user to be enabled and select Enable User.

A confirmation box is displayed.
3 Click YES.
The User status in the Users list - Disabled column changes to NO.

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Renaming a User
To rename a user:
1 In the RMX Management pane, click the Users (

) button.

The Users pane is displayed.
2 Right-click the user to be renamed and select Rename User.

The Rename User dialog box is displayed.

3 Enter the user’s new name in the New User Name field and click OK.
The user is renamed and is forced to change his/her password.

Machine Account
User names can be associated with servers (machines) to ensure that all users are subject to the same
account and password policies.
For enhanced security reasons it is necessary for the Collaboration Server to process user connection
requests in the same manner, whether they be from regular users accessing the Collaboration Server via
the Collaboration Server Web Browser / RMX Manager or from application-users representing applications
such as CMA and RealPresence DMA system.
Regular users can connect from any workstation having a valid certificate while application-users
representing applications can only connect from specific servers. This policy ensures that a regular user
cannot impersonate an application-user to gain access to the Collaboration Server in order to initiate an
attack that would result in a Denial of Service (DoS) to the impersonated application.
The connection process for an application-user connecting to the Collaboration Server is as follows:

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1 The application-user sends a connection request, including its TLS certificate, to the Collaboration
Server.
2 The Collaboration Server searches its records to find the FQDN that is associated with the
application-user’s name.
3 If the FQDN in the received certificate matches that associated with application-user, and the
password is correct, the connection proceeds.

Guidelines for defining a machine account
● Application-users are only supported when TLS security is enabled and Request peer certificate is
selected. TLS security cannot be disabled until all application-user accounts have been deleted from
the system.
● For Secure Communications, an administrator must set up on the Collaboration Server system a
machine account for the RealPresence CMA/DMA system with which it interacts. This machine
account must include a fully-qualified domain name (FQDN) for the RealPresence CMA/DMA
system.
● Application-user names are the same as regular user names.
Example: the CMA application could have an application-user name of CMA1.
● The FQDN can be used to associate all user types: Administrator, Operator with the FQDN of a
server.
● Multiple application-users can be configured the same FQDN name if multiple applications are
hosted on the same server
● If the system is downgraded the application-user’s FQDN information is not deleted from the
Collaboration Server’s user records.
● A System Flag, PASS_EXP_DAYS_MACHINE, enables the administrator to change the password
expiration period of application-user’s independently of regular users. The default flag value is 365
days.
● The server hosting an application-user whose password is about to expire will receive a login
response stating the number of days until the application-user’s password expires. This is determined
by the value of the PASSWORD_EXPIRATION_WARNING_DAYS System Flag. The earliest
warning can be displayed 14 days before the password is due to expire and the latest warning can
be displayed 7 days before passwords are due to expire. An Active Alarm is created stating the
number of days before the password is due to expire.
● The MIN_PWD_CHANGE_FREQUENCY_IN_DAYS System Flag does not effect application-user
accounts. Applications typically manage their own password change frequency.
● If an application-user identifies itself with an incorrect FQDN, its account will not be locked, however
the event is written to the Auditor Event File.
● If an application-user identifies itself with a correct FQDN and an incorrect password, its account will
be locked and the event written to the Auditor Event File.
● An application-user cannot be the last administrator in the system. The last administrator must be
regular user.
● User names are not case sensitive.

Monitoring
● An application-user and its connection is represented by a specific icon.

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Active Directory
● When working with Active Directory, CMA, RealPresence DMA system, and cannot be registered
within Active Directory as regular users. CMA and RealPresence DMA system application-users
must be manually.
● The only restriction is that TLS mode is enabled together with client certificate validation.
● If the above configuration are set off it will not be possible to add machine accounts.
● When setting the TLS mode off the system should check the existence of a machine account and
block this operation until all machine accounts are removed.

Connections
The Collaboration Server enables you to list all connections that are currently logged into the MCU, e.g.
users, servers or API users. The MCU issues an ID number for each login. The ID numbers are reset
whenever the MCU is reset.
A maximum of 50 users can be concurrently logged in to the MCU.

Viewing the Connections List
To list the users who are currently connected to the MCU:
1 In the Collaboration Server Management pane, click the Connections (
A list of connected users is displayed in the Connections pane.

) button.

The information includes:
 The user’s login name.
 The user’s authorization level (Chairperson, Operator, Administrator or Auditor).
 The time the user logged in.
 The name/identification of the computer used for the user’s connection.

Notes
Notes are the electronic equivalent of paper sticky notes. You can use notes to write down questions,
important phone numbers, names of contact persons, ideas, reminders, and anything you would write on
note paper. Notes can be left open on the screen while you work.
Notes can be read by all system Users concurrently connected to the MCU. Notes that are added to the
Notes list are updated on all workstations by closing and re-opening the Notes window. Notes can be written
in any Unicode language.
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Using Notes
To create a note:
1 On the Collaboration Server menu, click Administration > Notes.
The Notes window opens.

2 In the Notes toolbar, click the New Note (
window and select New Note.

) button, or right-click anywhere inside the Notes

3 In the Note dialog box, type the required text and click OK.
The new note is saved and closed. The Notes list is updated, listing the new note and its properties:
 Note – The beginning of the note’s text.
 Last Modified – The date of creation or last modification.
 Modified By – The Login Name of the user who last modified the note.
 Modified From – The Client Application and Workstation from which the note was created or
modified.

To open or edit a note:
● Double-click the entry to edit, or right-click the entry and select Note Properties.
The note opens for viewing or editing.
To delete a note:
1 In the Notes list, select the entry for the note to delete and click the Delete Note button (
right-click the entry and select Delete Note.

), or

A delete confirmation dialog box is displayed.
2 Click OK to delete the note, or click Cancel to keep the note.

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To enable the Collaboration Server to function within IP network environments, network parameters must
be defined for the IP Network Services.
The Polycom® RealPresence® Collaboration Server 800s and Polycom® RealPresence®
Collaboration Server Virtual Edition allows you only to view the parameters of the IP Network
Services in the RealPresence Collaboration Server Web Client or the RMX Manager, but you cannot
define a new IP Network Service or modify the parameters of an existing Network Service.
Attempting to do so may cause unexpected results, including complete inability to use or access the
RealPresence Collaboration Server. These settings can be modified only using the text user
interface. For more information, see the Manual IP Configuration.

The configuration dialog boxes for the IP network services are accessed via the Collaboration Server
Management pane of the Collaboration Server Web Client.

IP Network Services
Two IP Services are defined for the Collaboration Server:
● Management Network
● Default IP Service (Conferencing Service)
Dial in, dial out connections and Collaboration Server management are supported within the following IP
addressing environments:
● IPv4
● IPv6

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● IPv6 & IPv4
When IPv4 is selected, IPv6 fields are not displayed and conversely when IPv6 is selected, IPv4 fields are
not displayed. When IPv6 & IPv4 is selected both IPv6 and IPv4 fields are displayed.
For the purposes of comprehensive documentation, all screen captures in this chapter show the dialog
boxes as displayed with IPv6 & IPv4 selected.

Management Network (Primary)
The Management Network is used to control the Collaboration Server, mainly via the Collaboration Server
Web Client application. The Management Network contains the network parameters, such as the IP address
of the Control Unit, needed for connection between the Collaboration Server unit and the Collaboration
Server Web Client. This IP address can be used by the administrator or service personnel to connect to the
Control Unit should the MCU become corrupted or inaccessible.
In the RealPresence Collaboration Server 800s, the Management Network parameters can be set either via
a USB memory stick or by using a cable to create a private network, during First Time Power-up. For more
information, see thePolycom® RealPresence Collaboration Server 800s / Virtual Edition Getting Started
Guide, Installing the RealPresence Collaboration Server 800s and Appendix G - Configuring Direct
Connections to the Collaboration Server.
In the RealPresence Collaboration Server Virtual Edition with DHCP available, the Management Network
parameters are automatically set during First Time Power-up and whenever the Collaboration Server is
restarted. In the RealPresence Collaboration Server Virtual Edition without DHCP available, the
Management Network properties must be set manually. For more information, see the Polycom®
RealPresence Collaboration Server 800s / Virtual Edition Getting Started Guide, Manual IP Configuration.

Default IP Service (Conferencing Service - Media and signaling)
The Default IP Service (media and signaling) is used to configure and manage communications between
the Collaboration Server and conferencing devices such as endpoints, gatekeepers, SIP servers, etc.
The Default IP Service contains parameters for:
● Signaling Host IP Address
● External conferencing devices
Calls from all external IP entities are made to the Signaling Host, which initiates call set-up.
Conferencing related definitions such as environment (H.323 or SIP) are also defined in this service.
On the RealPresence Collaboration Server 800s, most of the Default IP Service is configured by the Fast
Configuration Wizard, which runs automatically should the following occur:
● First time power-up.
● Deletion of the Default IP Service, followed by a system reset.
On the RealPresence Collaboration Server Virtual Edition with DHCP available, the Default IP Service
parameters are automatically set during First Time Power-up and whenever the Collaboration Server is
restarted. On the RealPresence Collaboration Server Virtual Edition without DHCP available, the Default IP
Service properties must be set manually. For more information, see the Polycom® RealPresence
Collaboration Server 800s / Virtual Edition Getting Started Guide, Manual IP Configuration.

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Changes made to any of these parameters only take effect when the Collaboration Server is reset. An
Active Alarm is created when changes made to the system have not yet been implemented and the
MCU must be reset.

Modifying the Management Network in the RealPresence Collaboration
Server 800s
The Management Network parameters need to be modified if you want to:
● Connect directly to the Collaboration Server from a workstation
● Modify routes
● Modify DNS information

To view or modify the Management Network Service:
1 In the RMX Management pane, click the IP Network Services (

) button.

2 In the IP Network Services list pane, double-click the Management Network (

) entry.

The Management Network Properties - IP dialog box opens.
3 Modify the following fields:
RealPresence Collaboration Server 800s - Default Management Network Service – IP
Field

Description

Network Service Name

Displays the name of the Management Network. This name cannot be modified.
Note: This field is displayed in all Management Network Properties tabs.

IP Version

IPv4

Select this option for IPv4 addressing only.

IPv6

Select this option for IPv6 addressing only.

IPv4 & IPv6

Select this option for both IPv4 and IPv6 addressing.
Note: If the gatekeeper cannot operate in IPv6
addressing mode, the H323_RAS_IPV6 System Flag
should be set to NO.
For more information see Manually Added Flags CS_MODULE_PARAMETERS Tab.

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Field

Description

IPv6 Configuration
Method
Manual Configuration
Method is
recommended with
IPv6.

Auto (Stateless)

Select this option to allow automatic generation of the
following addresses:
• Link-Local (For internal use only)
• Site-Local
• Global

Manual

Select his option to enable manual entry of the
following addresses:
• Site-Local
• Global
Manual configuration of the following address types is
not permitted:
• Link-Local
• Multicast
• Anycast

IPv4

The IPv4 address of the Collaboration Server. This IP
address is used by the Collaboration Server Web
Client to connect to the Collaboration Server.

IPv6

The IPv6 address of the MCU. This IP address is used
by the Collaboration Server Web Client to connect to
the Collaboration Server.
Note: Internet Explorer 7™ is required for the
Collaboration Server Web Client to connect to the
MCU using IPv6.

Control Unit IP Address

All

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Click the All button to display the IPv6
addresses as follows:
• Auto - If selected, Site-Local and Global
site addresses are displayed.
• Manual - If selected, only the Manual site
address is displayed.

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Field

Description

Shelf Management
IP Address

IPv4

The IPv4 address of the RMX Shelf Management
Server. This IP address is used by the Collaboration
Server Web Client for Hardware Monitoring
purposes.

IPv6

The IPv6 address of the RMX Shelf Management
Server. This IP address is used by the Collaboration
Server Web Client for Hardware Monitoring
purposes.
Note: Internet Explorer 7™ is required for the
Collaboration Server Web Client to connect to the
MCU using IPv6.
All

Subnet Mask

Click the All button to display the IPv6
addresses as follows:
• Auto - If selected, Site-Local and Global
site addresses are displayed.
• Manual - If selected, only the Manual site
address is displayed.

Enter the subnet mask of the Control Unit.
Note: This field is specific to IPv4 and is not displayed in IPv6 only mode.

4 Click the Routers tab.

5 Modify the following fields:

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RealPresence Collaboration Server 800s - Default Management Network Service – Routers
Field

Description

Default Router
IP Address

IPv4
IPv6

The system uses Static Routes to search other networks
for endpoint addresses that are not found on the local LAN.
Up to five routers can be defined in addition to the Default
Router. The order in which the routers appear in the list
determines the order in which the system looks for the
endpoints on the various networks. If the address is in the
local subnet, no router is used.
To define a static route (starting with the first), click the
appropriate column and enter the required value.

Static Routes IPv4 Only
Table

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Enter the IP address of the default router. The default router
is used whenever the defined static routers are not able to
route packets to their destination. The default router is also
used when host access is restricted to one default router.

Router IP
Address

Enter the IP address of the router.

Remote IP
Address

Enter the IP address of the entity to be reached outside the
local network. The Remote Type determines whether this
entity is a specific component (Host) or a network.
• If Host is selected in the Remote Type field, enter the IP
address of the endpoint.
• If Network is selected in the Remote Type field, enter of
the segment of the other network.

Remote Subnet
Mask

Enter the subnet mask of the remote network.

Remote Type

Select the type of router connection:
• Network – defines a connection to a router segment in
another network.
• Host – defines a direct connection to an endpoint found
on another network.

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6 Click the DNS tab.

7 Modify the following fields:

RealPresence Collaboration Server 800s - Default Management Network
Service – DNS
Field

Description

MCU Host Name

Enter the name of the MCU on the network.
Default name is RMX

DNS

Select:
• Off – if DNS servers are not used in the network.
• Specify – to enter the IP addresses of the DNS servers.
Note: The IP address fields are enabled only if Specify is selected.

Register Host Names
Automatically to DNS
Servers

Select this option to automatically register the MCU Signaling Host and Shelf
Management with the DNS server.

Local Domain Name

Enter the name of the domain where the MCU is installed.

DNS Servers Addresses:
Primary Server
Secondary Server

The static IP addresses of the DNS servers.
A maximum of three servers can be defined.

Tertiary Server
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The Secuity and Whitelist tabs are not supported in the RealPresence Collaboration Server 800s.

8 Click OK.
9 If you have modified the Management Network Properties, reset the MCU.

Modifying the Default IP Network Service in the RealPresence
Collaboration Server 800s
The Default IP Service parameters need to be modified if you want to change the:
● Network type that the Collaboration Server connects to
● IP address of the Collaboration Server Signaling Host
● Gatekeeper parameters or add gatekeepers to the Alternate Gatekeepers list
● SIP server parameters

Fast Configuration Wizard
The Fast Configuration Wizard enables you to configure the Default IP Service. It starts automatically if no
Default IP Network Service is defined. This happens during First Time Power-up, before the service has
been defined or if the Default IP Service has been deleted, followed by an Collaboration Server restart.
The IP Management Service tab in the Fast Configuration Wizard is enabled only if the factory default
Management IP addresses were not modified.
If the Fast Configuration Wizard does not start automatically, the Default IP Service must be modified
through the IP Network Properties dialog boxes.

To view or modify the Default IP Service:
1 In the Collaboration Server Management pane, click IP Network Services (

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2 In the Network list pane, double-click the Default IP Service (

,

, or

) entry.

The Default IP Service - Networking IP dialog box opens.

3 Modify the following fields:
RealPresence Collaboration Server 800s - Default IP Network Service – IP
Field

Description

Network Service
Name

The name Default IP Service is assigned to the IP Network Service by the Fast
Configuration Wizard. This name can be changed.
Note: This field is displayed in all IP Signaling dialog boxes and can contain
character sets that use Unicode encoding.

IP Network Type

Displays the network type selected during the First Entry configuration. The Default
IP Network icon indicates the selected environment.
You can select:
• H.323: For an H.323-only Network Service.
• SIP: For a SIP-only Network Service.
• H.323 & SIP: For an integrated IP Service. Both H.323 and SIP participants can
connect to the MCU using this service.
Note: This field is displayed in all Default IP Service tabs.

Signaling Host IP
Address

Enter the address to be used by IP endpoints when dialing in to the MCU.
Dial out calls from the Collaboration Server are initiated from this address.
This address is used to register the RMX with a Gatekeeper or a SIP Proxy server.

Media Card 1
IP Address

Enter the address to be used by IP endpoints when dialing in to the MCU.

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Field

Description

Subnet Mask

Enter the subnet mask of the MCU.
Default value: 255.255.255.0.

4 Click the Routers tab.

With the exception of IP Network Type, the field definitions of the Routers tab are the same as for the
Default Management Network. For more information see step 3.
5 Optional. Click the DNS tab.
Settings in this dialog box are relevant to Multiple Network Services only.
For more information see NAT (Network Address Translation) Traversal.

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6 Click the Gatekeeper tab.

7

Modify the following fields:

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RealPresence Collaboration Server 800s - Default IP Service – Conferencing – Gatekeeper
Field

Description

Gatekeeper

Select Specify to enable configuration of the gatekeeper IP address.
When Off is selected, all gatekeeper options are disabled.

Primary Gatekeeper
IP Address or Name

Enter either the gatekeeper’s host name as registered in the
DNS or IP address.

Alternate Gatekeeper
IP Address or Name

Enter the DNS host name or IP address of the gatekeeper
used as a fallback gatekeeper used when the primary
gatekeeper is not functioning properly.

MCU Prefix in
Gatekeeper

Enter the number with which this Network Service registers in the gatekeeper. This
number is used by H.323 endpoints as the first part of their dial-in string when
dialing the MCU.

Register as Gateway

Select this check box if the Collaboration Server is to be seen as a gateway, for
example, when using a Cisco gatekeeper.

Refresh Registration
every __ seconds

The frequency with which the system informs the gatekeeper that it is active by
re-sending the IP address and aliases of the system to the gatekeeper. If the
system does not register within the defined time interval, the gatekeeper will not
refer calls to he system until it re-registers. If set to 0, re-registration is disabled.
Note:
• It is recommended to use default settings.
• This is a re-registration and not a ‘keep alive’ operation – an alternate gatekeeper
address may be returned.

Note: When in
IPv4&IPv6 or in
IPv6 mode, it is
easier to use
Names instead of
IP Addresses.

Aliases:
Alias

The alias that identifies the Collaboration Server’s Signaling Host within the
network. Up to five aliases can be defined for each Collaboration Server.
Note: When a gatekeeper is specified, at least one alias must be entered in the
table.
Additional aliases or prefixes may also be entered.

Type

The type defines the format in which the system alias is sent to the gatekeeper.
Each alias can be of a different type:
• H.323 ID (alphanumeric ID)
• E.164 (digits 0-9)
• Email ID (email address format,
e.g. abc@example.com)
• Participant Number (digits 0-9, * and #)
Note: Although all types are supported, the type of alias to be used depends on the
gatekeeper’s capabilities.

Parameters

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8 Click the Ports tab.
Settings in the Ports tab allow specific ports in the firewall to be allocated to multimedia conference
calls.

The port range recommended by IANA (Internet Assigned Numbers Authority) is 49152 to 65535.
The Collaboration Server uses this recommendation along with the number of licensed ports to
calculate the port range.
9 Modify the following fields:

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RealPresence Collaboration Server 800s - Default IP Service – Conferencing – Ports
Field

Description

Fixed Ports

Leave this check box cleared if you are defining a Network Service for local calls
that do not require configuring the firewall to accept calls from external entities.
When cleared, the system uses the default port range and allocates 4 RTP and
4 RTCP ports for media channels (Audio, Video, Content and FECC).
Note: When ICE Environment is enabled, 8 additional ports are allocated to
each call.
Click this check box to manually define the port ranges or to limit the number of
ports to be left open.

TCP Port from - to

Displays the default settings for port numbers used for signaling and control.
To modify the number of TCP ports, enter the first and last port numbers in the
range.
The number of ports is calculated as follows:
Number of simultaneous calls x 2 ports (1 signaling + 1 control).

UDP Port from - to

Displays the default settings for port numbers used for audio and video.
To modify the number of UDP ports:
Enter the first and last port numbers in the range, and the range must be
1024 ports.
When ICE environment is enabled, the range must be 2048 ports per media
card.

Parameters
If the network administrator does not specify an adequate port range, the system will accept the
settings and issue a warning. Calls will be rejected when the Collaboration Server’s ports are
exceeded.

10 If required, click the QoS tab.

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Quality of Service (QoS) is important when transmitting high bandwidth audio and video information.
QoS can be measured and guaranteed in terms of:
• Average delay between packets
• Variation in delay (jitter)
• Transmission error rate
DiffServ and Precedence are the two QoS methods supported by the Collaboration Server. These
methods differ in the way the packet’s priority is encoded in the packet header.
The Collaboration Server’s implementation of QoS is defined per Network Service, not per endpoint.
The routers must support QoS in order for IP packets to get higher priority.

11 View or modify the following fields:

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RealPresence Collaboration Server 800s - Default IP Service – Conferencing – QoS
Field

Description

Enable

Select to enable the configuration and use of the QoS settings.
When un-checked, the values of the DSCP (Differentiated Services Code Point)
bits in the IP packet headers are zero.

Type

DiffServ and Precedence are two methods for encoding packet priority. The
priority set here for audio video and IP Signaling packets should match the
priority set in the router.
• DiffServ: Select when the network router uses DiffServ for priority encoding.
The default priorities for both audio and video packets is 0x31. These values
are determined by the QOS_IP_VIDEO and QOS_IP_AUDIO flags in the
system.cfg file.
The default priority for Signaling IP traffic is 0x00 and is determined by the
QOS_IP_SIGNALING flag in the system.cfg file.
For more information Modifying System Flags
• Precedence: Select when the network router uses Precedence for priority
encoding, or when you are not sure which method is used by the router.
Precedence should be combined with None in the TOS field.
The default priority is 5 for audio and 4 for video packets.
Note: Precedence is the default mode as it is capable of providing priority
services to all types of routers, as well as being currently the most common
mechanism.

Audio / Video

You can prioritize audio and video IP packets to ensure that all participants in
the conference hear and see each other clearly. Select the desired priority. The
scale is from 0 to 5, where 0 is the lowest priority and 5 is the highest. The
recommended priority is 4 for audio and 4 for video to ensure that the delay for
both packet types is the same and that audio and video packets are
synchronized and to ensure lip sync.

TOS

Select the type of Service (TOS) that defines optimization tagging for routing the
conferences audio and video packets.
• Delay: The recommended default for video conferencing; prioritized audio
and video packets tagged with this definition are delivered with minimal delay
(the throughput of IP packets minimizes the queue sequence and the delay
between packets).
• None: No optimization definition is applied. This is a compatibility mode in
which routing is based on Precedence priority settings only. Select None if
you do not know which standard your router supports.

Parameters

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12 Click the SIP Servers tab.

13 Modify the following fields:
RealPresence Collaboration Server 800s - Default IP Network Service – SIP Servers
Field

Description

SIP Server

Select:
• Specify – to manually configure SIP servers.
• Off – if SIP servers are not present in the network.

SIP Server Type

Select:
• Generic - for non Microsoft environments.
• Microsoft - for Microsoft environments.

Refresh Registration

This defines the time in seconds, in which the Collaboration Server refreshes it’s
registration on the SIP server. For example, if “3600” is entered the
Collaboration Server will refresh it’s registration on the SIP server every 3600
seconds.

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Field

Description

Transport Type

Select the protocol that is used for signaling between the Collaboration Server
and the SIP Server or the endpoints according to the protocol supported by the
SIP Server:
UDP – Select this option to use UDP for signaling.
TCP – Select this option to use TCP for signaling.
TLS – The Signaling Host listens on secured port 5061 only and all outgoing
connections are established on secured connections. Calls from SIP clients or
servers to non secured ports are rejected.
The following protocols are supported: TLS 1.0, SSL 2.0 and SSL 3.0.

Skip Certificate Validation

When checked, no Certificate Validation is performed.

Revocation Method
Global Responder URL
Use Responder Specified
in Certificate
Allow Incomplete
Revocation Checks

For a detailed description of these fields see the Ultra Secure Mode chapter,
Certificate Management and Certificate Revocation.

Skip Certificate Validation
for OSCP Responder
SIP Servers: Primary / Alternate Server Parameter
Server IP Address

Enter the IP address of the preferred SIP server.
If a DNS is used, you can enter the SIP server name.
Note: When in IPv4&IPv6 or in IPv6 mode, it is easier to use Names instead of
IP Addresses.

Server Domain Name

Enter the name of the domain that you are using for conferences, for example:
user_name@domain name
The domain name is used for identifying the SIP server in the appropriate
domain according to the host part in the dialed string.
For example, when a call to EQ1@polycom.com reaches its outbound proxy,
this proxy looks for the SIP server in the polycom.com domain, to which it will
forward the call.
When this call arrives at the SIP server in polycom.com, the server looks for
the registered user (EQ1) and forwards the call to this Entry Queue or
conference.

Port

Enter the number of the TCP or UDP port used for listening. The port number
must match the port number configured in the SIP server.
Default port is 5060.

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Field

Description

Outbound Proxy Servers: Primary / Alternate Server Parameter
Server IP Address

By default, the Outbound Proxy Server is the same as the SIP Server. If they
differ, modify the IP address of the Outbound Proxy and the listening port
number (if required).
Note: When in IPv4&IPv6 or in IPv6 mode, it is easier to use Names instead of
IP Addresses.

Port

Enter the port number the outbound proxy is listening to.
The default port is 5060.

When updating the parameters of the SIP Server in the IP Network Service - SIP Servers dialog box,
the Collaboration Server must be reset to implement the change.

14 Click the Security tab.

15 Modify the following fields:

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RealPresence Collaboration Server 800s - Default IP Network Service – Security (SIP Digest)
Field

Description

SIP Authentication

Click this check box to enable SIP proxy
authentication.
Select this check box only if the authentication is
enabled on the SIP proxy, to enable the Collaboration
Server to register with the SIP proxy. If the
authentication is enabled on the SIP proxy and
disabled on the RMX, calls will fail to connect to the
conferences.
Leave this check box cleared if the authentication
option is disabled on the SIP proxy.

User Name

Enter the user name the Collaboration Server will use
to authenticate itself with the SIP proxy. This name
must be defined in the SIP Proxy.

Password

Enter the password the Collaboration Server will use
to authenticate itself with the SIP proxy. This
password must be defined in the SIP proxy.

H.323 Authentication

Click this check box to enable H.323 server
authentication.
Select this check box only if the authentication is
enabled on the gatekeeper, to enable the
Collaboration Server to register with the gatekeeper. If
the authentication is enabled on the gatekeeper and
disabled on the RMX, calls will fail to connect to the
conferences.
Leave this check box cleared if the authentication
option is disabled on the gatekeeper.

User Name

Enter the user name the Collaboration Server will use
to authenticate itself with the gatekeeper. This name
must be defined in the gatekeeper.

Password

Enter the password the Collaboration Server will use
to authenticate itself with the gatekeeper. This
password must be defined in the gatekeeper.

These fields can
contain up to 20 ASCII
characters.

These fields can
contain up to 64 ASCII
characters.

If the Authentication User Name and Authentication Password fields are left empty, the SIP Digest
authentication request is rejected. For registration without authentication, the Collaboration Server
must be registered as a trusted entity on the SIP server.

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16 Optional. To configure the ICE environment, click the SIP Advanced tab.

17 Modify the following fields:
RealPresence Collaboration Server 800s - Default IP Network Service – SIP Advanced
Field

Description

ICE Environment

Select MS (for Microsoft ICE implementation) to enable the ICE integration.

Server User Name

Enter the Collaboration Server User name as defined in the Active Directory.
For example, enter rmx1234.
This field is disabled if the ICE Environment field is set to None.

18 Click the OK button.
When updating the parameters of the SIP Server in the IP Network Service - SIP Servers dialog box,
the Collaboration Server must be reset to implement the change.

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Viewing the Management Network in the RealPresence Collaboration
Server Virtual Edition
In the Polycom® RealPresence® Collaboration Server 800s and Polycom® RealPresence®
Collaboration Server Virtual Edition, these settings can only be changed in the console Text User
Interface. For more information, see the Polycom® RealPresence Collaboration Server 800s /
Virtual Edition Getting Started Guide, Manual IP Configuration on page 21.

To view the Management Network Service:
1 In the Collaboration Server RMX Management pane, click the IP Network Services (
2 In the IP Network Services list pane, double-click the Management Network (

) button.

) entry.

The Management Network Properties - IP dialog box opens.

The following fields can be viewed, but can not be modified:
RealPresence Collaboration Server Virtual Edition - Default Management Network Service – IP
Field

Description

Network Service
Name

Displays the name of the Management Network. This name cannot be modified.
Note: This field is displayed in all Management Network Properties tabs.

Control Unit IP
Address

IPv4

Subnet Mask

The subnet mask of the Management Network Service.

Polycom®, Inc.

The IPv4 address of the Collaboration Server. This IP
address is used by the Collaboration Server Web Client
to connect to the Collaboration Server.

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If an attempt is made to modify these settings, the message below will be displayed:

3 Click the Routers tab.

The following fields can be viewed but not modified.

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RealPresence Collaboration Server Virtual Edition - Default Management Network Service –
Field

Description

Default Router
IP Address

IPv4
IPv6

Static Routes

The IP address of the default router. The default router is used
whenever the defined static routers are not able to route
packets to their destination. The default router is also used
when host access is restricted to one default router.
The system uses Static Routes to search other networks for
endpoint addresses that are not found on the local LAN.
Up to five routers can be defined in addition to the Default
Router. The order in which the routers appear in the list
determines the order in which the system looks for the
endpoints on the various networks. If the address is in the local
subnet, no router is used.
To define a static route (starting with the first), click the
appropriate column and enter the required value.

Router IP
Address

The IP address of the router.

Remote IP
Address

The IP address of the entity to be reached outside the local
network. The Remote Type determines whether this entity is a
specific component (Host) or a network.
• If Host is selected in the Remote Type field, enter the IP
address of the endpoint.
• If Network is selected in the Remote Type field, enter of the
segment of the other network.

Remote Subnet
Mask

The subnet mask of the remote network.

Remote Type

The type of router connection:
• Network – defines a connection to a router segment in
another network.
• Host – defines a direct connection to an endpoint found on
another network.

Routers

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4 Click the DNS tab.

The following fields can be modified, but their values will not be applied:
RealPresence Collaboration Server Virtual Edition - Default Management Network Service –
DNS
Field

Description

MCU Host Name

The name of the MCU on the network.
Default name is PolycomMCU

DNS

• Off – if DNS servers are not used in the network.
• Specify – to enter the IP addresses of the DNS servers.
Note: The IP address fields are enabled only if Specify is selected.

Register Host Names
Automatically to DNS Servers

Select this option to automatically register the MCU Signaling Host with
the DNS server.

Local Domain Name

Enter the name of the domain where the MCU is installed.

DNS Servers Addresses:
Primary Server
Secondary Server

The static IP addresses of the DNS servers.
A maximum of three servers can be defined.

Tertiary Server

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If an attempt is made to modify these settings, the message below will be displayed:

5 Click OK.

IP Network Monitoring
The Signaling Monitor is the Collaboration Server entity used for monitoring the status of external network
entities such as the gatekeeper, DNS, SIP proxy and Outbound proxy and their interaction with the MCU.

To monitor signaling status:
1 In the RMX Management pane, click Signaling Monitor (

).

2 In the Signaling Monitor pane, double-click Default IP Service.
The IP Network Services Properties – RMX CS IP tab opens:

The RMX CS IP tab displays the following fields:

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IP Network Services Properties – RMX CS IP
Field

Description

Service Name

In the RealPresence Collaboration Server 800s, the name assigned to the IP
Network Service by the Fast Configuration Wizard.
In the RealPresence Collaboration Server Virtual Edition, this is always, “IP
Network Service.”
Note: This field is displayed in all tabs.

IPv4

IP Address

IPv6

Default Router
IP Address

The IP address of the default router. The default router is
used whenever the defined static routers are not able to
route packets to their destination. The default router is also
used when host access is restricted to one default router.

Subnet Mask

The subnet mask of the MCU.
Default value: 255.255.255.0.

Scope

IP Address
Note:

Default Router
IP Address

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Global

The Global Unicast IP address of the
Collaboration Server.

Site-Local

The IP address of the Collaboration Server
within the local site or organization.

The IP address of the default router. The default router is
used whenever the defined static routers are not able to
route packets to their destination. The default router is also
used when host access is restricted to one default router.

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3 Click the H.323 tab.

The H.323 tab displays the following fields:
IP Network Services Properties – H.323
Field

Description

Connection State

The state of the connection between the Signaling Host and the gatekeeper:
Discovery - The Signaling Host is attempting to locate the gatekeeper.
Registration - The Signaling Host is in the process of registering with the
gatekeeper.
Registered - The Signaling Host is registered with the gatekeeper.
Not Registered - The registration of the Signaling Host with the gatekeeper
failed.

Registration Interval

The interval in seconds between the Signaling Host’s registration messages to
the gatekeeper. This value is taken from either the IP Network Service or from
the gatekeeper during registration. The lesser value of the two is chosen.

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Role

Active - The active gatekeeper.
Backup - The backup gatekeeper that can be used if
the connection to the preferred gatekeeper fails.

ID

The gatekeeper ID retrieved from the gatekeeper during
the registration process.

Name

The gatekeeper’s host’s name.

IP Address

The gatekeeper’s IP address.

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4 Click the SIP Servers tab.

The SIP Servers tab displays the following fields:
IP Network Services Properties – SIP Servers
Field

Description

Role

Active -The default SIP Server is used for SIP traffic.
Backup -The SIP Server is used for SIP traffic if the preferred proxy fails.

Name

The name of the SIP Server.

IP Address

The SIP Server's IP address.

Status

The connection state between the SIP Server and the Signaling Host.
Not Available - No SIP server is available.
Auto - Gets information from DHCP, if used.

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5 Click the ICE Servers tab.

The ICE Servers tab displays the following fields:
IP Network Services Properties – ICE Servers
Field

Description

Role

The ICE Server’s role is displayed:
• STUN password server
• STUN Server UDP
• STUN Server TCP
• Relay Server UDP
• Relay Server TCP

IP Address

The ICE Server’s IP Address.

Status 1/2/3/4

A status is displayed for each media card installed in the Collaboration Server:
• Connection O.K.
• MS – register fail
• MS – subscribe fail
• MS – service fail
• Connection failed
• User/password failed
• Channel didn't receive any packets for 5 seconds
• Channel exceeded allotted bandwidth
• Unknown failure
In systems with multiple media cards, Status 1 refers to the uppermost media
card.

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IP Network Services Properties – ICE Servers
Field

Description

FW Detection

The Firewall Detection status is displayed:
• Unknown
• UDP enabled
• TCP enabled
• Proxy -TCP is possible only through proxy
• Block – both UDP & TCP blocked
• None

Using IPv6 Networking Addresses for Collaboration Server Internal
and External Entities
IPv6 addresses can be assigned to both Collaboration Server (Internal) and External Entity addresses.

Collaboration Server Internal Addresses
Default Management Network Service
● Control Unit
● Signaling Host
● Shelf Management

External Entities
● Gatekeepers (Primary & Secondary)
● SIP Proxies
● DNS Servers
● Default Router
● Defined participants

IPv6 Guidelines
● Internet Explorer 7™ is required for the Collaboration Server Web Client and RMX Manager to
connect to the Collaboration Server using IPv6.
● The default IP address version is IPv4.
● The IP address field in the Address Book entry for a defined participant can be either IPv4 or IPv6. A
participant with an IPv4 address cannot be added to an ongoing conference while the Collaboration
Server is in IPv6 mode nor can a participant with an IPv6 address be added while the Collaboration
Server is in IPv4 mode.
An error message, Bad IP address version, is displayed and the New Participant dialog box remains
open so that the participant’s address can be entered in the correct format.

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● Participants that do not use the same IP address version as the Collaboration Server in ongoing
conferences launched from Meeting Rooms, Reservations and Conference Templates, and are
disconnected. An error message, Bad IP address version, is displayed.
IP Security (IPSec) Protocols are not supported.

Ethernet Settings
In the RealPresence Collaboration Server 800s, the automatically identified speed and transmit/receive
mode of each LAN port used by the system can be manually modified if the specific switch requires it.

To modify the automatic LAN port configuration:
1 On the Collaboration Server menu, click Setup > Ethernet Settings.
The Ethernet Settings dialog box opens.

The Collaboration Server has 2 LAN ports. You can select the speed and transmit/receive mode
manually for these ports.
2 In the Speed column, click the drop-down arrow of the table entry to modify and select the speed
and the transmit/receive mode for each port:

When Auto (default) is selected, the negotiation of speed and transmit/receive mode starts at 1000
Mbits/second Full Duplex, proceeding downward to 100 Mbits/second Half Duplex.
Note: To maximize conferencing performance, especially in high bit rate call environments, a 1Gb
connection is recommended.

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3

Click the OK button.

NAT (Network Address Translation) Traversal
NAT Traversal is a set of techniques enabling participants behind firewalls to connect to conferences,
hosted on the Collaboration Server, remotely using the internet.

Session Border Controller (SBC)
All signaling and media for both SIP and H.323 will be routed through an SBC.
The following SBC environments are supported:
● SAM - a Polycom SBC
● Acme Packet - a 3rd party SBC
● VBP - Polycom Video Border Proxy

Deployment Architectures
The following NAT Traversal topologies are given as examples. Actual deployments will depend on user
requirements and available infrastructure:

Remote Connection Using the Internet

The following Remote Connection call flow options are supported:

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Remote Connections
Enterprise Client

CMA Client

Environment

Registered

SBC

Registered

Environment

SIP / H.323

Yes

SAM / Acme Packet



Yes

SIP

SIP / H.323

No

SAM / Acme Packet



No

SIP

SIP / H.323

No

SAM Only



No

H.323

Business to Business Connections

The following Business to Business connection call flow options are supported:
Business to Business Connections
Enterprise A Client

Enterprise B Client

Environment

Registered

SBC

H.323

Yes

Access
Director

H.323

Yes

SIP
SIP

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SBC

Registered

Environment



Access
Director

Yes

H.323

Access
Director



VBP

Yes

H.323

Yes

Access
Director



Access
Director

Yes

H.323

Yes

Acme Packet



Acme Packet

Yes

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FW (Firewall) NAT Keep Alive
The Collaboration Server can be configured to send a FW NAT keep alive message at specific Intervals
for the RTP, UDP and BFCP channels.
This is necessary because port mappings in the firewall are kept open only if there is network traffic in both
directions. The firewall will only allow UDP packets into the network through ports that have been used to
send packets out.
By default the Collaboration Server sends a FW NAT Keep Alive message every 30 seconds. As there is
no traffic on the Content and FECC channels as a call begins, the firewall will not allow any incoming
packets from the Content and FECC channels in until the Collaboration Server sends out the first of the FW
NAT Keep Alive messages 30 seconds after the call starts.
If Content or FECC are required within the first 30 seconds of a call the FW NAT Keep Alive Interval should
be modified to a lower value.

To enable and modify FW NAT Keep Alive:
FW NAT Keep Alive is enabled in the New Profile - Advanced dialog box.

» Select the FW NAT Keep Alive check box and if required, modify the Interval field within the range
of 5 - 86400 seconds.

System Configuration in SBC environments
In an environment that includes SAM (a Polycom SBC), to ensure that a RealPresence Mobile endpoint
can send content to a conference the value of the system flag
NUM_OF_INITIATE_HELLO_MESSAGE_IN_CALL_ESTABLISHMENT must be set to at least 3.

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For more details on modifying the values of system flags, see Manually Adding and Deleting System Flags.

SIP Proxy Failover With Polycom® Distributed Media
Application™ (DMA™) 7000
Collaboration Server systems that are part of a RealPresence DMA system environment can benefit from
the RealPresence DMA system’s SIP Proxy Failover functionality.
SIP Proxy Failover is supported in the RealPresence DMA system’s Local Clustering mode with redundancy
achieved by configuring two DMA servers to share a single virtual IP address.
The virtual IP address is used by the Collaboration Server as the IP address of its SIP Proxy.
No additional configuration is needed on the Collaboration Server.
Should a SIP Proxy failure occur in one of the RealPresence DMA system servers:
● The other RealPresence DMA system server takes over as SIP Proxy.
● Ongoing calls may be disconnected.
● Previously ongoing calls will have to be re-connected using the original IP address, registration and
connection parameters.
● New calls will connect using the original IP address, registration and connection parameters.

Polycom® RealPresence® Collaboration Server 800s
and Polycom® RealPresence® Collaboration Server
Virtual Edition Network Port Usage
The following table summarizes the port numbers and their usage in the Polycom® RealPresence®
Collaboration Server 800s and Polycom® RealPresence® Collaboration Server Virtual Edition:
Collaboration Server Network Port Usage Summary
Connection
Type

Port Number

Protocol

Description

Configurable

HTTP

80

TCP

Management between the
Collaboration Server and
Collaboration Server Web Client.

No

HTTPS

443

TCP

Secured Management between the
Collaboration Server and
Collaboration Server Web Client.

No

DNS

53

TCP

Domain name server.

Can be disabled in the
IP Network Service.

DHCP

68

TCP

Dynamic Host Configuration
Protocol.

Can be disabled in the
IP Network Service.

SSH

22

TCP

Secured shell. It is the
Collaboration Server terminal.

No

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Connection
Type

Port Number

Protocol

Description

Configurable

NTP

123

UDP

Network Time Protocol. Enables
access to a time server on the
network.

No

H.323 GK
RAS

1719

UDP

Gatekeeper RAS messages traffic.

No

H.323 Q.931

1720 - incoming;
49152-59999 outgoing

TCP

H.323 Q.931 call signaling.
Each outgoing call has a separate
port.
The port for each outgoing call is
allocated dynamically.

Yes - for outgoing calls
only.
It is configured in the
Fixed Ports section of
the IP service.

H.323 H.245

49152 - 59999

TCP

H.245 control.
Each outgoingx call has a separate
port.
The port for each outgoing call is
allocated dynamically. It can be
avoided by tunneling.

Yes - for outgoing calls
only.
It is configured in the
Fixed Ports section of
the IP service.

SIP server

5060
60000

UDP,
TCP

Connection to the SIP Server.
Sometimes port 60000 is used
when the system cannot reuse the
TCP port. This port can be set in
the Central signaling (CS)
configuration file.

Yes - in the IP service.

Alternative
SIP server

5060
60000

UDP,
TCP

Connection to the alternate SIP
Server.
Sometimes port 60000 is used
when the system cannot reuse the
TCP port. This port can be set in
the Central signaling (CS)
configuration file.

Yes - in the IP service.

SIP Outbound
proxy

5060
60000

UDP,
TCP

Connection to the SIP outbound
proxy.
Sometimes port 60000 is used
when the system cannot reuse the
TCP port. This port can be set in
the Central signaling (CS)
configuration file.

Yes - in the IP service.

Alternative
SIP Outbound
proxy

5060
60000

UDP,
TCP

Connection to the alternate SIP
outbound proxy.
Sometimes port 60000 is used
when the system cannot reuse the
TCP port. This port can be set in
the Central signaling (CS)
configuration file.

Yes - in the IP service.

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Connection
Type

Port Number

Protocol

Description

Configurable

SIP-TLS

60002

TCP

Required for Binary Floor Control
Protocol (BFCP) functionality for
SIP People+Content content
sharing.

No - port is not opened
if SIP People+Content
is disabled.

RTP

49152 - 59999

UDP

RTP media packets.
The ports are dynamically
allocated.

Yes - It is configured in
the Fixed Ports section
of the IP service.

RTCP

49152 - 59999

UDP

RTP control.
The ports are dynamically
allocated.

Yes - It is configured in
the Fixed Ports section
of the IP service.

SIP -TLS

5061

TCP

SIP -TLS for SIP server, alternate
SIP server, outbound proxy and
alternate outbound proxy.

No

LAN Redundancy
LAN Redundancy is applicable to the RealPresence Collaboration Server 800s only.

LAN Redundancy enables the redundant LAN port connection to automatically replace the failed port by
using another physical connection and NIC (Network Interface Card). When a LAN port fails, IP network
traffic failure is averted and network or endpoints disconnections do not occur. When LAN cables are
connected to both LAN 2 and LAN 3 ports, the Collaboration Server automatically selects which port is
active and which is redundant.

Configuration Requirements
LAN Redundancy is available by default and is enabled by connecting the additional LAN cable to LAN 3,
or LAN 4.

Signaling and Media Redundancy
On the RealPresence Collaboration Server 800s, LAN2 is used for media and signaling and LAN 3, and
LAN 4 are the redundant media ports:

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Collaboration Server - Rear View

Media Redundancy on the Collaboration Server is dependent on the settings of the LAN_REDUNDANCY
and MULTIPLE_SERVICES System Flags as summarized in the table below.
Collaboration Server - Media Redundancy - System Flags
System Flag / Value

Collaboration Server

LAN_REDUNDANCY = NO
MULTIPLE_SERVICES = NO

No redundancy

LAN_REDUNDANCY = NO
MULTIPLE_SERVICES = YES
LAN_REDUNDANCY = YES
MULTIPLE_SERVICES = NO

Full signaling and media redundancy

Hardware Monitor Indications
With LAN redundancy, when LAN LEDs are lit they indicate that a physical connection of the cables is
present but does not indicate their activity status.
In the Hardware Monitor pane the Lan List displays the Collaboration Server LAN ports together with their
Status indication.

LAN Indications
Status

Description

Active

The LAN port cable is connected.

Inactive

The LAN port cable is not connected.

Standby

The LAN Redundancy option is enabled and this LAN port is the redundant and in standby
mode. In case of failure, this port becomes active.

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Multiple Network Services
Multiple Network Services are applicable to the RealPresence Collaboration Server 800s only.

Media, signaling and management networks can be physically separated on the Collaboration Server
system to provide enhanced security. This addresses the requirement in an organization that different
groups of participants be supported on different networks. For example, some participants may be internal
to the organization while others are external.
Up to three media and signaling networks can be defined for the RealPresence Collaboration Server 800s
for each media and signaling network connected to the Collaboration Server.
The Management Network is logically and physically separated from the media and signaling networks.
There can be one Management Network defined per Collaboration Server system.
Each conference on the Collaboration Server can host participants from the different IP Network networks
simultaneously.

Guidelines
● Multiple Services system mode is a purchasable option and it is enabled in the MCU license.
● Multiple Services system mode is enabled when the system configuration flag
MULTIPLE_SERVICES is added and set to YES.
● On the RealPresence Collaboration Server 800s, LAN redundancy cannot be enabled in parallel to
Multiple Networks and the LAN_REDUNDANCY flag must be set to NO when the Multiple Networks
option is enabled.
● Participants on different networks can connect to the same conference with full audio, video and
content capabilities.
● Traffic on one network does not influence or affect the traffic on other networks connected to the same
MCU. If one network fails, it will not affect the traffic in the other connected networks.
● The Signaling Host IP Address and the Media IP Address cannot be different.
● Maximum number of services that can be defined per Collaboration Server platform:
Maximum Number of Network Services

● A DNS server can be specified for each IP Network Service and for the Collaboration Server
Management Network Service.
 In the Network Services that do not include the DNS server, use the IP addresses of the various
devices to define them in the Network Services.
● Participants are associated with a Network Service and use it resources as follows:
 Dial-in participants - according to the network used to place the call and connect to the
Collaboration Server.
 Dial-out participant - according to the Network Service selected during the participant properties
definition or during conference definition, according to the Network Service selected as default.

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Resource Allocation and Capacity
The Resolution Configuration settings are configured per MCU and affect the resource capacity of the MCU.
They are reflected in the port gauges displayed on the Collaboration Server management application’s main
screen.
In Multiple Networks mode, the port gauges do not reflect the resource availability per Network Service.
In Multiple Networks mode, the resources of the Network Services are not split between the network
services but are used per their availability by all Network Services equally.

First Time Installation and Configuration
First Time Installation and Configuration of the Polycom® RealPresence® Collaboration Server 800s and
Polycom® RealPresence® Collaboration Server Virtual Edition consists of the following procedures:
1 Preparations
 Gather Network Equipment and Address Information - get the information needed for integrating
the Collaboration Server into the local network for each of the networks that will be connected to
the Collaboration Server unit. For a list of required address, see the RealPresence Collaboration
Server Getting Started Guide, Gather Network Equipment and Address Information.
 Unpack the Collaboration Server. For more details see the Polycom® RealPresence
Collaboration Server 800s / Virtual Edition Getting Started Guide, Unpacking the RealPresence
Collaboration Server 800s.
 Modify the Management Network parameters on the USB Key. For more details see the
Polycom® RealPresence Collaboration Server 800s / Virtual Edition Getting Started Guide,
Modifying the Factory Default Management Network Settings on the USB Memory Stick.
2 Hardware Installation and Setup
 Mount the Collaboration Server in a rack. For more details see the Polycom® RealPresence
Collaboration Server 800s / Virtual Edition Getting Started Guide, Hardware Installation and Rack
Mounting.
 Connect the necessary cables. For details, see Polycom® RealPresence Collaboration Server
800s / Virtual Edition Getting Started Guide, Connecting the Cables to the MCU.
3 First Entry Power-up and Configuration
 Power up the Collaboration Server. For more details see the Polycom® RealPresence
Collaboration Server 800s / Virtual Edition Getting Started Guide, Procedure 1: First-time
Power-up.
 Register the Collaboration Server. For more details see the Polycom® RealPresence
Collaboration Server 800s / Virtual Edition Getting Started Guide, Procedure 2: Product
Registration.
 Connect to the Collaboration Server. For more details see the Polycom® RealPresence
Collaboration Server 800s / Virtual Edition Getting Started Guide, Procedure 3: Connection to
MCU.
 Configure the Default IP Network Service using the information for one of the networks connected
to the system. For more details see the Polycom® RealPresence Collaboration Server 800s /
Virtual Edition Getting Started Guide, Procedure 4: Modifying the Default IP Service Settings.
4 Modify the required System Flag to enable Multiple Services and reset the MCU.
5 Add the required IP Network Services to accommodate the networks connected to the Collaboration
Server.

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6 Select a Network Service to act as default for dial out and gateway calls for which the Network
Service was not selected.
7 Place several calls and run conferences to ensure that the system is configured correctly.

Connecting the Cables to the RealPresence Collaboration Server 800s
On the Collaboration Server LAN2 is used for media and signaling and LAN 3 and LAN 4 can be used for
multiple Networks configuration.
Collaboration Server - Rear View

Multiple Network LAN Port Usage
LAN Port

Description

1

Management Network LAN connection (mandatory)

2

Signaling and media - first IP Network Service (mandatory)

3

Signaling and media - second IP Network Service (optional)

4

Signaling and media - third IP Network Service (optional)

If LAN Redundancy is configured, Multiple Networks on LAN 3 and 4 cannot be defined.

Collaboration Server Configuration
Once the network cables are connected to the Collaboration Server, you can modify the default IP Network
Service and add additional Network Services.

System Flags and License Settings
The MULTIPLE_SERVICES System Flag determines whether the Multiple Services option will be activated
once the appropriate license is installed. Possible Values: YES / NO Default: NO
This flag must be manually added to the system configuration and set to YES to enable this option. For more
information see the RealPresence Collaboration Server 800s Administrator’s Guide, Manually Adding and
Deleting System Flags.

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IP Network Service Definition
Use this procedure to define Network Services in addition to the Network Service already defined during
first entry installation and configuration. Each of the defined Network Service can be associated with one or
more media cards installed in the system (depending on the system type).

To add new/additional Network Services:
1 In the Device Management pane, click IP Network Services (
2 In the Network Services list toolbar, click the

).

Add Network Service button.

The New IP Service - Networking IP dialog box opens.
3 Define the following fields:
IP Network Service - IP Parameters
Field

Description

Network Service
Name

Enter the IP Network Service name.
Note: This field is displayed in all IP Signaling dialog boxes and can contain
character sets that use Unicode encoding.

IP Network Type

Select the IP Network environment. You can select:
• H.323: For an H.323-only Network Service.
• SIP: For a SIP-only Network Service.
• H.323 & SIP: For an integrated IP Service. Both H.323 and SIP participants can
connect to the Collaboration Server using this service.
Note: This field is displayed in all Default IP Service tabs.

Signaling Host IP
Address

This field is disabled as only one IP address is used for the signaling.

Media Card IP
address

If each of LAN ports designated for signaling and media (LAN2, LAN3 and LAN4)
on the system can be used with a different network, each port is assigned to its own
Network Service.
In such a case, enter the IP address of the port to be assigned to this Network
Service.
A LAN port that is already assigned to a different Network Service, displays the IP
Address of the assigned port and it cannot be assigned to this Network Service (it is
disabled).

Subnet Mask

Enter the subnet mask of the Collaboration Server in that network service.
Default value: 255.255.255.0.

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4 Optional. Some system flags can be defined per Network Service, depending on the network
environment.
To modify these flags, click the Service Configuration button.
The Service Configuration dialog box opens.

All the flags must be manually added to this dialog box. For a detailed description of the flags and
how to add them, see the Manually Adding and Deleting System Flags.
Flags defined per Network Service override their general definition in the System Configuration.

The following flags can be defined per service:
 ALLOW_NON_ENCRYPT_PARTY_IN_ENCRYPT_CONF
 ENABLE_H239
 SIP_ENABLE_FECC
 ENABLE_CLOSED_CAPTION
 ALLOW_NON_ENCRYPT_RECORDING_LINK_IN_ENCRYPT_CONF
 NUMERIC_CONF_ID_LEN
 NUMERIC_CONF_ID_MIN_LEN
 NUMERIC_CONF_ID_MAX_LEN
 ENABLE_CASCADED_LINK_TO_JOIN_WITHOUT_PASSWORD
 MAX_CP_RESOLUTION
 QOS_IP_AUDIO
 QOS_IP_VIDEO
 QOS_IP_SIGNALING
 ENABLE_CISCO_GK
 SIP_FREE_VIDEO_RESOURCES
 FORCE_CIF_PORT_ALLOCATION
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 MS_ENVIRONMENT
 SIP_FAST_UPDATE_INTERVAL_ENV
 SIP_FAST_UPDATE_INTERVAL_EP
 H263_ANNEX_T
 H239_FORCE_CAPABILITIES
 MIX_LINK_ENVIRONMENT
 IP_LINK_ENVIRONMENT
 FORCE_STATIC_MB_ENCODING
 FORCE_RESOLUTION
 SEND_WIDE_RES_TO_IP
 DISABLE_WIDE_RES_TO_SIP_DIAL_OUT
 SEND_SIP_BUSY_UPONRESOURCE_THRESHOLD
5 Click the Routers tab.
6 Define the routers used in this network and that are other than the routers defined in the
Management Network. The field definitions of the Routers tab are the same as for the Default
Management Network. For more information see the RealPresence Collaboration Server 800s Default Management Network Service – Routers.
7 Click the DNS tab.

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8 Modify the following fields:
Default Management Network Service – DNS
Field

Description

Service Host Name

Enter the host name of this network Service. Each Network Service must have a
unique Host Name otherwise an error message is displayed.

DNS

Select:
• Off – if no DNS server is used in this network.
• Specify – to enter the IP address of the DNS server used by this network
service.
Notes:
• The IP address field is enabled only if Specify is selected.
• Only one DNS can be define for the entire topology (that is, only one Network
Service can include the DNS definition).

Register Host Names
Automatically to DNS
Servers

Select this option to automatically register this Network Service Signaling Host with
the DNS server.

Local Domain Name

Enter the name of the domain for this network service.

DNS Server Address

Enter the static IP address of the DNS server that is part of this network.

9 Click the Gatekeeper tab.
10 Define the Primary and Alternate Gatekeepers and at least one Alias for this network Service. The
field definitions of the Gatekeeper tab are the same as for the Default IP Network Service. For more
information see the RealPresence Collaboration Server 800s - Default IP Service – Conferencing –
Gatekeeper Parameters.
In Multiple Services mode, an Alias must be defined for the specified gatekeeper.

11 Optional. Click the Ports tab.
Settings in the Ports tab allow specific ports in the firewall to be allocated to multimedia conference
calls. If required, defined the ports to be used multimedia conference calls handled by this Network
Service. The field definitions of the Ports tab are the same as for the Default IP Network Service.
For more information see the RealPresence Collaboration Server 800s - Default IP Service –
Conferencing – Ports Parameters.

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12 If required, click the QoS tab.
The Collaboration Server’s implementation of QoS is defined per Network Service, not per endpoint.
The routers must support QoS in order for IP packets to get higher priority.

The field definitions of the QoS tab are the same as for the Default IP Network Service. For more
information see the RealPresence Collaboration Server 800s - Default IP Service – Conferencing –
QoS Parameters.
13 Click the SIP Servers tab.
14 Define the Primary and Alternate SIP Server for this network Service.
•

•

If Microsoft Office Communications or Lync server are part of this network service, a certificate must
be created for this network service. If each network connected to the Collaboration Server includes
Microsoft Office Communications or Lync server, separate certificates must be created and sent to
the Collaboration Server for each of these networks.
If the Network Service does not include a DNS, you must use the IP address of the SIP Server
instead of its name.

The field definitions of the SIP Servers tab are the same as for the Default IP Network Service. For
more information see the RealPresence Collaboration Server 800s - Default IP Network Service –
SIP Servers.
15 Click the Security tab.
The field definitions of the Security tab are the same as for the Default IP Network Service. For more
information see the RealPresence Collaboration Server 800s - Default IP Network Service – Security
(SIP Digest).
16 Optional. To configure the ICE environment, click the SIP Advanced tab.
17 Modify the following fields:
Default IP Network Service – SIP Advanced
Field

Description

Server User Name

Enter the User name for this service as defined in the Active Directory. For
example, enter rmxNet2.
This field is disabled if the ICE Environment field is set to None.

ICE Environment

Select MS (for Microsoft ICE implementation) to enable the ICE integration.

18 Click the OK button.
The new Network Service is added to the IP Network Services list pane.

Setting a Network Service as Default
The default Network Service is used when no Network Service is selected for dial out participants.In
addition, the Signaling Host IP address and the MCU Prefix in GK displayed on the Collaboration Server
Web Client main screen are taken from the default H.323 Network Service.

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One IP Network Service can be defined as default for H.323 connections and another Network Service as
default for SIP connections. If the IP Network Service supports both H.323 and SIP connections, you can
set the same Network Service as default for both H.323 and SIP, or for H.323-only or for SIP-only.

To designate an IP Network Service as the default IP Network Service:
1 In the Device Management pane, click IP Network Services (

).

2 In the Network Services list pane right-click the IP Network Service to be set as the default, and then
click Set As H.323 Default, or Set As SIP Default.
The next time you access this menu, a check mark is added next to the network service type to
indicate its selection as default.
To set this IP Network Service for both H.323 and SIP connections, repeat step 2 and select the
option you need.
The following icons are used to indicate the default IP Network Service type:
Default IP Network Service Icons
Icon

Description
This Network Service supports both SIP and H.323 connections and
is designated as default for both SIP and H.323 connections.
This Network Service supports both SIP and H.323 connections and
is designated as default for H.323 connections.
This Network Service supports both SIP and H.323 connections and
is designated as default for SIP connections.
This Network Service supports only H.323 connections and is set as
default for H.323 connections.
This Network Service supports only SIP connections and is set as
default for SIP connections.

Signaling Host IP Address and MCU Prefix in GK Indications
The Collaboration Server Web Client displays the Signaling Host IP Address and MCU Prefix in GK
parameters as defined in the Default H.323 Network Service.

Resolution Configuration
These configurations are set for the system and are applied to all the Network Services.

Conference Profile
Registration of conferencing entities such as ongoing conferences, Meeting Rooms, Entry Queues, SIP
Factories and Gateway Sessions with SIP servers is done per conferencing entity. This allows better control
on the number of entities that register with each SIP server by selecting for each of the conferencing entities
whether it will register with the SIP server.
The registration is defined in the Conference Profile - Network Services tab.

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In the IP Network Services table, the system lists all the defined Network Services (one or several
depending on the system configuration).
● To register the conferencing entity to which this profile is assigned to a Network Service, in the
Registration column click the check box of that Network Service.
● You can also prevent dial in participants from connecting to that conferencing entities when
connecting via a Network Service.
In the Accept Calls column, clear the check box of the Network Service from which calls cannot
connect to the conference.

Signaling Monitor
The Signaling Monitor pane includes the list of the signaling and media IP Network Services defined in the
system (up to three in the Polycom® RealPresence® Collaboration Server 800s and Polycom®
RealPresence® Collaboration Server Virtual Edition). Double-clicking a Network Service, displays it
properties and status.

Conferencing
Each conference on the Collaboration Server can host participants from the different IP Network networks
simultaneously.

Defining AVC Dial Out Participants
When defining AVC dial out participants, you can select the Network Service to place the call according to
the network to which the endpoint pertains. If the endpoint is located on a network other than the selected
network, the participant will not be able to connect.
If no Network is selected, the system uses the default IP Network Service.
If no Network is selected, the system uses the IP Network Service selected for reserving the conference
resources, and if none is set for the conference it uses the Network Service set as default.
The IP Network Service is selected in the New Participant - Advanced dialog box.

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Monitoring Conferences
The Conference Properties - Network Services dialog box shows for each Network Service with which
Network Service’s SIP proxy the conference should be registered and if the dial in call will be connected to
the conference.
In the Participant pane, a new column - Service Name was added, indicating the name of Network Service
used for the participant’s connection.

Resource Report
The Resource Report displays the resource usage in total and per Network Service in a table format. The
Resources per Service table provides the actual information on resource usage and availability per network
Service and provides an accurate snapshot of resources usage in the system.

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You can select the graph to display: select either Totals (default) or the Network Service.

Port Gauge Indications
The port Gauges displays the total resource usage for the Collaboration Server and not per Network
Service. Therefore, it may not be an accurate representation of the availability of resources for conferencing,
as one Network Service may run out of available resources while another Network Service may have all of
it resources available. In such a case, the port gauges may show that half of the system resources are
available for conferencing, while calls via the Network Service with no available resources will fail to
connect.

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Interactive Voice Response (IVR) is an application that allows participants to communicate with the
conferencing system via their endpoint’s input device (such as a remote control). The IVR Service includes
a set of voice prompts and a video slide used to automate the participants connection to a conference or
Entry Queue. It allows customization of menu driven scripts and voice prompts to meet different needs and
languages.
The IVR module includes two types of services:
● Conference IVR Service that is used with conferences
● Entry Queue IVR Service that is used with Entry Queues
The system is shipped with two default Conference IVR Services (one for the conferences and the other for
gateway calls) and one default Entry Queue IVR Service. The default services include voice messages and
video slides in English.
To customize the IVR messages and video slide perform the following operations:
● Record the required voice messages and create a new video slide.
For more information, see Creating a Welcome Video Slide.
● Optional. Add the language to the list of languages supported by the system.
● Upload the voice messages to the MCU (This can be done as part of the language definition or during
the IVR Service definition).
● Create the Conference IVR Service and upload the video slide, and if required any additional voice
messages.
● Optional. Create the Entry Queue IVR Service and upload the required video slide and voice
messages.

IVR Services List
You can view the currently defined Conference IVR and Entry Queue IVR Services in the IVR Services list
pane.

To view the IVR Services list:
1 In the Collaboration Server Management pane, expand the Rarely Used list.
2 Click the IVR Services (

) entry.

The list pane displays the Conference IVR Services list and the total number of IVR services currently
defined in the system.

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IVR Services Toolbar
The IVR Services toolbar provides quick access to the IVR Service definitions as follows:
IVR Toolbar buttons
Button

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Button Name

Descriptions

New Conference IVR Service

To create a new Conference IVR Service.

New Entry Queue IVR Service

To create a new Entry Queue IVR Service.

Delete Service

Deletes the selected IVR service(s).

Set Default Conference IVR
Service

Sets the selected Conference IVR Service as default. When
creating a new conference Profile the default IVR Service is
automatically selected for the Profile (but can be modified).

Set Default Entry Queue Service

Sets the selected Entry Queue IVR Service as default. When
creating a new Entry Queue the default Entry Queue IVR
Service is automatically selected.

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IVR Toolbar buttons
Button

Button Name

Descriptions

Add Supported
Languages

Adds languages to the IVR module, enabling you to download
voice prompts and messages for various languages.

Replace/Change Music File

To replace the currently loaded music file that is used to play
background music, the MCU is shipped with a default music
file.

Adding Languages
You can define different sets of audio prompts in different languages, allowing the participants to hear the
messages in their preferred language.
The Collaboration Server is shipped with a default language (English) and all the prompts and messages
required for the default IVR Services, conference and Entry Queues shipped with the system.
You can add languages to the list of languages for which different messages are downloaded to the MCU
and IVR Services are created. This step is required before the creation of additional IVR messages using
languages that are different from English, or if you want to download additional voice files to existing files in
one operation and not during the IVR service definition.

To add a language:
1 In the Collaboration Server Management pane, expand the Rarely Used list.
2 Click the IVR Services (

) entry.

3 In the Conference IVR Services list, click the Add Supported Languages (
The Supported Languages dialog box opens.

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4 Click the Add Language button.
The New Language dialog box opens.

5 In the New Language box, enter the name of the new language. The language name can be typed
in Unicode and cannot start with a digit. Maximum field length is 31 characters.
6 Click OK.
The new language is added to the list of Supported Languages.

Uploading a Message File to the Collaboration Server
You can upload audio files for the new language or additional files for an existing language now, or you can
do it during the definition of the IVR Service. In the latter case, you can skip the next steps.
•
•

Voice messages should not exceed 3 minutes.
It is not recommended to upload more than 1000 audio files to the MCU memory.

To upload messages to the MCU:
1 To upload the files to the MCU, in the Supported Languages dialog box, click the Add Message File
button.
The Add Message File dialog box opens.

Audio files are uploaded to the MCU one-by-one.
2 In the IVR Message Language list, select the language for which the audio file will be uploaded to
the MCU.
3 In the IVR Message Category list, select the category for which the audio file is uploaded.

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4 In the Message Type list, select the message type for which the uploaded message is to be played.
You can upload several audio files for each Message Type. Each file is downloaded separately.
Table 5-2 lists the Message Types for each category:
IVR Message Types by Message Category
Message
Category

Message Type

Message

Conference
Password

Request Conference
Password

Requests the participant to enter the conference password.

Request Conference
Password Retry

A participant who enters an incorrect password is requested
to enter it again.

Request Digit

Requests the participant to enter any digit in order to connect
to the conference. Used for dial-out participants to avoid
answering machines in the conference.

Welcome
Message

Welcome Message

The first message played when the participant connects to
the conference or Entry Queue.

Conference
Chairperson

Request Chairperson
Identifier

Requests the participants to enter the chairperson identifier
key.

Request Chairperson
Password

Requests the participant to enter the chairperson password.

Request Chairperson
Password Retry

When the participant enters an incorrect chairperson
password, requests the participant to enter it again.

General

Messages played for system related event notifications, for example, notification that the
conference is locked. Upload the files for the voice messages that are played when an
event occurs during the conference. For more information, see Conference IVR Service
Properties - General Voice Messages.

Billing Code

Requests the chairperson to enter the conference Billing Code.

Roll Call

Roll call related messages, such as the message played when a participant joins the
conference. Messages are listed in the Conference IVR Service - Roll Call dialog box.

Conference ID

Requests the participant to enter the required Conference ID to be routed to the
destination conference.

5 Click Upload File to upload the appropriate audio file to the MCU.
The Install File dialog box opens.

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6 Enter the file name or click the Browse button to select the audio file to upload.
The Select Source File dialog box opens.
7 Select the appropriate *.wav audio file, and then click the Open button.
The name of the selected file is displayed in the Install field in the Install File dialog box.
8 Optional. You can play a .wav file by selecting the Play button (

).

9 Click Yes to upload the file to the MCU.
The system returns to the Add Message File dialog box.
10 Repeat step 6 to Click Yes to upload the file to the MCU. The system returns to the Add Message
File dialog box. for each additional audio file to be uploaded to the MCU.
11 Once all the audio files are uploaded to the MCU, close the Add Message File dialog box and
return to the Add Language dialog box.
12 Click OK.

Defining a New Conference IVR Service
The Collaboration Server is shipped with two default Conference IVR Services and all its audio messages
and video slide. You can define new Conference IVR Services or modify the default Conference IVR
Service.
Up to 80 IVR Services (Conference IVR Services and Entry Queue IVR Services) can be defined
per Collaboration Server.

Defining a New Conference IVR Service
To define a new Conference IVR Service:
1 On the IVR Services toolbar, click the New Conference IVR Service (

) button.

The New Conference IVR Service - Global dialog box opens.

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2 Define the following parameters:
Conference IVR Service Properties - Global Parameters
Field/Option

Description

Conference IVR Service
Name

Enter the name of the Conference IVR Service. The maximum field length is 20
characters and may be typed in Unicode.

Language For IVR

Select the language of the audio messages and prompts from the list of
languages defined in the Supported languages. The default language is
English. For more information, see Adding Languages.

External Server
Authentication

This option is not supported with Collaboration Server 800s/Virtual Edition.
You can configure the IVR Service to use an external database application to
verify a participant’s right to join the conference. For more information, see
Conference Access with External Database Authentication.
Select one of the following options:
• Never – The participant’s right to join the conference will not be verified with
an external database application (default).
• Always – Any participant request to join the conference is validated with the
external database application using a password.
• Upon Request – Only the participant request to join the conference as
chairperson is validated with the external database application using a
password. The validation process occurs only when the participant enters the
chairperson identifier key.

Number of User Input
Retries

Enter the number of times the participant will be able to respond to each menu
prompt before being disconnected from the conference. Range is between 1-4,
and the default is 3.

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Conference IVR Service Properties - Global Parameters
Field/Option

Description

Timeout for User Input
(Sec)

Enter the duration in seconds that the system will wait for the participant’s input
before prompting for another input. Range is between 1-10, and the default
value is 5 seconds.

DTMF Delimiter

Enter the key that indicates the last input key. Possible values are the pound (#)
and star (*) keys. The default is #.

3 Click the Welcome tab.
The New Conference IVR Service - Welcome dialog box opens.

4 Select the Enable Welcome Messages check box to define the system behavior when the
participant enters the Conference IVR queue. When participants access a conference through an
Entry Queue, they hear messages included in both the Entry Queue Service and Conference IVR
Service. To avoid playing the Welcome Message twice, disable the Welcome Message in the
Conference IVR Service.
5 Select the General Welcome Message, to be played when the participant enters the conference
IVR queue.
6 To upload an audio file for an IVR message, click Add Message File.
The Install File dialog box opens.

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The Collaboration Server unit is bundled with default audio IVR message files. To upload a
customized audio file, see Creating Audio Prompts and Video Slides.

a Click the Browse button to select the audio file (*.wav) to upload.
The Select Source File dialog box opens.
b Select the appropriate *.wav audio file and then click the Open button.
c Optional. You can play a .wav file by selecting the Play button (

).

d In the Install File dialog box, click Yes to upload the file to the MCU memory.
The Done dialog box opens.
e Once the upload is complete, click OK and return to the IVR dialog box. The new audio file can
now be selected from the list of audio messages.
7 Click the Conference Chairperson tab.
The New Conference IVR Service - Conference Chairperson dialog box opens.

8 Select the Enable Chairperson Messages check box to enable the chairperson functionality. If this
feature is disabled, participants are not able to connect as the chairperson.
When both Conference Password and Chairperson Password options are enabled and defined, the
system first plays the prompt "Enter conference password". However, if the participant enters the
chairperson password, the participant becomes the chairperson.
To play the prompt requesting the Chairperson password, "For conference chairperson services...",
do not select the Enable Password Messages option.

9 Select the various voice messages and options for the chairperson connection.
If the files were not uploaded prior to the definition of the IVR Service or if you want to add new audio
files, click Add Message File to upload the appropriate audio file to the Collaboration Server.

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New Conference IVR Service Properties - Conference Chairperson Options and Messages
Field/Option

Description

Chairperson Identifier
Request

Select the audio file that requests the participants to enter the key that identifies
them as the conference chairperson.

Request Chairperson
Password

Select the audio file that prompts the participant for the chairperson password.

Retry Chairperson
Password

Select the audio file that prompts participants to re-enter the chairperson
password if they enter it incorrectly.

Chairperson Identifier
Key

Enter the key to be used for identifying the participant as a chairperson.
Possible keys are: pound key (#) or star (*).

Billing Code

The prompt requesting the chairperson billing code selected in the General tab.

10 Click the Conference Password tab.
The New Conference IVR Service - Conference Password dialog box opens.

11 Select the Enable Password Messages check box to request the conference password before
moving the participant from the conference IVR queue to the conference.
When both Conference Password and Chairperson Password are enabled and defined, the system
first plays the prompt "Enter conference password". However, if the participant enters the
chairperson password, the participant becomes the chairperson.
To play the prompt requesting the Chairperson password, "For conference chairperson services...",
do not select the Enable Password Messages option.

12 Select the MCU behavior for password request for Dial-in and Dial-out participant connections.
Select the required system behavior as follows:
 Request password - The system requests the participant to enter the conference password.

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 None - The participant is moved to the conference without any password request.
 Request Digit - The system requests the participant to enter any key. This option is used mainly
for dial-out participants and to prevent an answering machine from entering the conference.
13 Select the various audio messages that will be played in each case.
New Conference IVR Service Properties - Conference Password Parameters
Option

Description

Request Password

Select the audio file that prompts the participant for the conference password.

Retry Password

Select the audio file that requests the participant to enter the conference
password again when failing to enter the correct password.

Request Digit

Select the audio file that prompts the participant to press any key when the
Request Digit option is selected.

14 Click the General tab.
The New Conference IVR Service - General dialog box opens.

The General dialog box lists messages that are played during the conference. These messages are
played when participants or the conference chairperson perform various operations or when a
change occurs.
15 To assign the appropriate audio file to the message type, click the appropriate table entry, in the
Message File column. A drop-down list is enabled.
16 From the list, select the audio file to be assigned to the event/indication.
17 Repeat steps 15 and 16 to select the audio files for the required messages.
The following types of messages and prompts can be enabled:

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Conference IVR Service Properties - General Voice Messages
Message Type

Description

Blip on Cascade Link

Indicates that the link to the cascaded conference connected successfully.

Chairperson Exit

Informs all the conference participants that the chairperson has left the
conference, causing the conference to automatically terminate after a short
interval.
Note: This message is played only when the Requires Chairperson option is
selected in the Conference Profile - IVR dialog box.

Chairperson Help Menu

A voice menu is played upon a request from the chairperson, listing the
operations and their respective DTMF codes that can be performed by the
chairperson. The playback can be stopped any time.
Note: If you modify the default DTMF codes used to perform various operations,
the default voice files for the help menus must be replaced.

Change Chairperson
Password

Requests the participant to enter a new chairperson password when the
participant is attempting to modify the chairperson password.

Change Conference
Password

Requests the participant to enter a new conference password when the
participant is attempting to modify the conference password.

Change Password
Failure

A message played when the participant enters an invalid password, for example
when a password is already in use.

Change Passwords
Menu

This voice menu is played when the participants requests to change the
conference password. This message details the steps required to complete the
procedure.

Conference is Locked

This message is played to participants attempting to join a Secured conference.

Conference is Secured

This message is played when the conference status changes to Secure as
initiated by the conference chairperson or participant (using DTMF code *71).

Conference is unsecured

This message is played when the conference status changes to Unsecured as
initiated by the conference chairperson or participant (using DTMF code #71).

Confirm Password
Change

Requests the participant to re-enter the new password.

Enter Destination ID

Prompts the calling participant for the destination number. Default message
prompts the participant for the conference ID (same message as in the Entry
Queue IVR Service).
Note: This option is not available in SVC conferences and for SVC participants
in mixed CP and SVC conferences.

First to Join

Informs the participant that he or she is the first person to join the conference.

Incorrect Destination ID

If the participant entered an incorrect conference ID (in gateway calls it is the
destination number), requests the participant to enter the number again.
Note: This option is not available in SVC conferences and for SVC participants
in mixed CP and SVC conferences.

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Conference IVR Service Properties - General Voice Messages
Message Type

Description

Maximum Number of
Participants Exceeded

Indicates the participant cannot join the destination conference as the maximum
allowed number of participants will be exceeded.

Mute All Off

This message is played to the conference to inform all participants that they are
unmuted (when Mute All is cancelled).

Mute All On

Informs all participants that they are muted, with the exception of the conference
chairperson.
Note: This message is played only when the Mute All Except Me option is
activated.

No Video Resources
Audio Only.

Informs the participant of the lack of Video Resources in the Collaboration
Server and that he/she is being connected as Audio Only.

Participant Help Menu

A voice menu that is played upon request from a participant, listing the
operations and their DTMF codes that can be performed by any participant.

Password Changed
Successfully

A message is played when the password was successfully changed.

Recording Failed

This message is played when the conference recording initiated by the
chairperson or the participant (depending on the configuration) fails to start.

Recording in Progress

This message is played to participant joining a conference that is being recorded
indicating the recording status of the conference.

Request Billing Code

Requests the participant to enter a code for billing purposes.

Requires Chairperson

The message is played when the conference is on hold and the chairperson
joins the conference. For this message to be played the Conference Requires
Chairperson option must be selected in the Conference Profile - IVR dialog
box.

Self Mute

A confirmation message that is played when participants request to mute their
line.

Self Unmute

A confirmation message that is played when participants request to unmute their
line.

18 Click the Roll Call/Notifications tab.
The New Conference IVR Service - Roll Call dialog box opens.
The Roll Call and Tone Notification options are disabled in SVC and mixed CP and SVC
conferences.

The Roll Call feature of the Conference IVR Service is used to record the participants’ names for
playback when the participants join and leave a conference.
Roll Call announcements played upon a participant’s connection or disconnection from a conference
(Entry and Exit announcements) can be replaced by tones. These tones can be used as notification
when participants join or leave the conference but the identification of the participant is not required.
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The system is shipped with two default tones: Entry Tone and Exit tone. When the Tone Notifications
option is enabled, no recording of the participant names will occur and the conference chairperson
will not be able to ask for a name review during the conference.
19 Select one of the following options to determine the announcement mode:
a To enable the Roll Call feature, select the Enable Roll Call option.

b Select Enable Tones to enable the Tone Notifications option.
The dialog box changes to display the tone notification options and all Roll Call options are
disabled. In such a case, skip to step Select the Entry Tone or Exit tone:.
c Select None to disable the Roll Call and Tone Notifications features.
If Enable Roll Call option is selected:
20 To assign the audio file to the message type, in the Message File column, click the appropriate table
entry. An arrow appears in the Message File column.
If the Roll Call option is enabled, you must assign the appropriate audio files to all message types.

21 Click the arrow to open the Message File list and select the appropriate audio file.
Conference IVR Service Properties - Roll Call Messages
Roll Call Message

Description

Roll Call Record

Requests participants to state their name for recording, when they connect to
the conference.
Note: The recording is automatically terminated after two seconds.

Roll Call Joined

A voice message stating that the participant has joined the conference.

Roll Call Left

A voice message stating that the participant has left the conference.

Roll Call Review

Played when Roll Call is requested by the chairperson, introducing the names of
the conference participants in the order they joined the conference.

If Enable Tone Notifications option is selected:
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22 Select the Entry Tone or Exit tone:

a Click the appropriate table entry in the Message File column.
A drop-down list is enabled.
b From the list, select the audio file to be assigned to the event/indication.
If the Tones option is enabled, you must assign the appropriate audio files to all notification types.
The Collaboration Server system is shipped with two default tones: Entry_tone.wav and
Exit_tone.wav.
If required, you can upload customized audio files that will be played when participants join or leave
the conference.
If the option to play a tone when a cascading link connection is established, make sure that the tone
selected for Entry or Exit notification differ from the cascading link tone as the latter one cannot be
customized.

23 Click the Video Services tab.
The New Conference IVR Service - Video Services dialog box opens.

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.

The Click&View and Invite Participants features are disabled in SVC and mixed CP and SVC
conferences.

In addition to the low and high resolution slides included in the default slide set, customized low and
high resolution slides are supported.
The following guidelines apply:
 Two customized slides can be loaded per IVR Service:
 A low resolution slide, to be used with low resolution endpoints.
 A high resolution slide, to be used with high resolution endpoints.
The following table summarizes the recommended input slide formats and the resulting slides that
are generated:
IVR Slide - Input / Output Formats
Format
Slide Resolution

Input Slides

Generated Slides
HD1080p
HD720p

High

HD1080p (16:9)
or
HD720p (16:9)

Low

4CIF (4:3)
or
CIF (4:3)

4SIF
SIF
CIF

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 The source images for the high resolution slides must be in *.bmp or *.jpg format.
 If the uploaded slides are not of the exact SD or HD resolution, an error message is displayed and
the slides are automatically cropped or enlarged to the right size.
 If a slide that is selected in an IVR Service is deleted, a warning is displayed listing the IVR
Services in which it is selected. If deleted, it will be replaced with a default Collaboration Server
slide.
 The generated slides are not deleted if the system is downgraded to a lower software version.
 The first custom source file uploaded, whatever its format, is used to generate both high and low
resolution custom slides. High resolution source files uploaded after the first upload will be used
to generate and replace high resolution custom slides. Likewise, low resolution source files
uploaded after the first upload will be used to generate and replace low resolution custom slides.
 If there are two custom source files in the folder, one high resolution, one low resolution, and a
new high resolution custom source file is uploaded, new high resolution custom slides are
created. The existing low resolution custom slides are not deleted.
 If there are two custom source files in the folder, one high resolution, one low resolution, and a
new low resolution custom source file is uploaded, new low resolution custom slides are created.
The existing high resolution custom slides are not deleted.
24 Define the following parameters
:

New Conference IVR Service Properties - Video Services Parameters
Video
Services

Description

Video
Welcome
Slide

Select the Low Resolution and High Resolution video slides to be displayed when
participants connect to the conference.
To view any slide, click the Preview Slide (
) button.
Notes:
• When using one of the default Polycom slides, the slide will be displayed in the resolution
defined in the profile, i.e. CIF, SD, HD 720p
• Customized H.261 slides are not supported.
When Collaboration Server is configured to IPv6, the IVR slide is displayed without taking
into account the MTU Size.

Invite
Participant

See Inviting Participants using DTMF .
Note: The Invite Participant feature is not available in SVC conferences and for SVC
participants in mixed CP and SVC conferences.

25 If the video slide file was not uploaded to the MCU prior to the IVR Service definition, click the:
 Add Slide - Low Resolution button to upload a Low Resolution Slide.
 Add Slide - High Resolution button to upload a High Resolution Slide.
The Install File dialog box opens. The uploading process is similar to the uploading of audio files.
For more information, see To upload an audio file for an IVR message, click Add Message File..
•
•

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The video slide must be in a .jpg or .bmp file format. For more information, see Creating a
Welcome Video Slide.
Customized H.261 slides are not supported.

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26 Click the DTMF Codes tab.
The New Conference IVR Service - DTMF Codes dialog box opens.

● This dialog box lists the default DTMF codes for the various functions that can be performed during
the conference by all participants or by the chairperson
.

New Conference IVR Service Properties - DTMF Codes
Operation

DTMF String

Permission

Mute My Line

*6

Everyone

Unmute My Line

#6

Everyone

Mute All Except Me

*5

Chairperson

Cancel Mute All Except Me

#5

Chairperson

Change Password

*77

Chairperson

Mute Incoming Participants

*86

Chairperson

Unmute Incoming Participants

#86

Chairperson

Play Help Menu

*83

Everyone

Terminate Conference

*87

Chairperson

Change To Chairperson

*78

Everyone

Override Mute All

Configurable

Everyone

Start Recording

*2

Chairperson

Stop Recording

*3

Chairperson

Pause Recording

*1

Chairperson

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New Conference IVR Service Properties - DTMF Codes
Operation

DTMF String

Permission

Secure Conference

*71

Chairperson

Unsecured Conference

#71

Chairperson

Request individual assistance
Note: This option is not available for SVC participants.

*0

Everyone

Request assistance for conference
Note: This option is not available for SVC participants.

00

Chairperson

Request to Speak

99

Everyone

27 To modify the DTMF code or permission:
a In the DTMF Code column, in the appropriate entry enter the new code.
b In the Permission column, select from the list who can use this feature (Everyone or just the
Chairperson).
By default, the Secure, Unsecure Conference and Show Number of Participants options are enabled
in the Conference IVR Service. These options can be disabled by removing their codes from the
Conference IVR Service.
• To disable the Text Indication option in the DTMF Code column, clear the DTMF code (*88) of
Show Number of Participants from the table.
• To disable the Secure Conference options, in the DTMF Code column, clear the DTMF codes of
both Secured Conference (*71) and Unsecured Conference (#71) from the table.

28 Click the Operator Assistance tab.
The Operator Assistance dialog box opens.

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29 Select Enable Operator Assistance to enable operator assistance when the participant requires or
requests help during the connection process to the conference or during the conference.
30 In the Operator Assistance Indication Message field, select the audio message to be played
when the participant requests or is waiting for the operator’s assistance.
If the audio file was not uploaded prior to the definition of the IVR Service or if you want to add new
audio files, click Add Message File to upload the appropriate audio file to the Collaboration Server.

31 Click OK to complete the IVR Service definition.
The new Conference IVR Service is added to the IVR Services list.

Change to Chairperson
Regular participants can request to become the conference chairperson using the appropriate DTMF code
(default: *78), which enabled them to perform operations designated for chairpersons only.
The Change to Chairperson via the DTMF code (default: *78) is executed only if the following settings were
configured for the MCU and the conference:
● In the Conference IVR Service - Conference Chairperson dialog box, select the Enable
Chairperson Messages check box, and select the appropriate voice messages.
For more information, see the Polycom® RealPresence Collaboration Server 800s/Virtual Edition
Administrator’s Guide, New Conference IVR Service Properties - Conference Chairperson Options
and Messages.
● When starting a new conference or defining a new Meeting Room, define the Chairperson
Password in the conference General dialog box.
For more information, see Creating a New Meeting Room.

Controlling the receipt of in-band and out-of-band DTMF Codes
The RFC2833_DTMF System Flag controls the receipt of in-band or out-of-band DTMF Codes.
When set to YES (default), the RMX will receive DTMF Codes sent in-band. When set to NO the RMX
receives DTMF Codes sent out-of-band. The RMX always sends DTMF Codes in-band (as part of the Audio
Media stream). If you wish to modify the flag value, the flag must be added to the System Configuration
file. For more information see Modifying System Flags.

Entry Queue IVR Service
An Entry Queue (EQ) is a routing lobby for conferences. Participants are routed to the appropriate
conference according to the conference ID they enter.
An Entry Queue IVR Service must be assigned to the Entry Queue to enable the voice prompts and video
slide guiding the participants through the connection process.
An Entry Queue IVR Service is a subset of an IVR Service. You can create different Entry Queue Services
for different languages and personalized voice messages.
The Collaboration Server is shipped with a default Entry Queue IVR Service and all its audio messages and
video slide. You can define new Entry Queue IVR Services or modify the default Entry Queue IVR Service.

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Defining a New Entry Queue IVR Service
To set up a new Entry Queue IVR Service:
1 In the RMX Management pane, click IVR Services (

).

2 In the IVR Services list, click the New Entry Queue IVR Service (

) button.

The New Entry Queue IVR Service - Global dialog box opens.

3 Fill in the following parameters:
Entry Queue IVR Service Properties - Global Parameters
Option

Description

Entry Queue Service
Name

(Mandatory) Enter the name of the Entry Queue Service. The name can be
typed in Unicode. Maximum field length is 80 ASCII characters.

Language

Select the language in which the Audio Messages and prompts will be heard.
The languages are defined in the Supported Languages function.

External Server
Authentication

This option is used for Ad Hoc conferencing, to verify the participant’s
permission to initiate a new conference. For a detailed description see Appendix
D: Appendix D - Ad Hoc Conferencing and External Database Authentication.
Select one of the following options:
• None to start a new conference without verifying with an external database
the user right to start it.
• Conference ID to verify the user’s right to start a new conference with an
external database application using the conference ID.

Number of User Input
Retries

Enter the number of times the participant is able to respond to each menu
prompt before the participant is disconnected from the MCU.

Timeout for User Input
(Sec.)

Enter the duration in seconds that the system waits for input from the participant
before it is considered as an input error.

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Entry Queue IVR Service Properties - Global Parameters
Option

Description

DTMF Delimiter

The interaction between the caller and the system is done via touch-tone
signals (DTMF codes). Enter the key that will be used to indicate a DTMF
command sent by the participant or the conference chairperson. Possible keys
are the pound key (#) or star (*).

4 Click the Welcome tab.
The New Entry Queue IVR Service - Welcome dialog box opens.

If the files were not uploaded prior to the definition of the IVR Service or if you want to add new audio
files, click Add Message File to upload the appropriate audio file to the Collaboration Server.

5 Define the appropriate parameters. This dialog box contains options that are identical to those in the
Conference IVR Service - Welcome Message dialog box. For more information about these
parameters, see New Conference IVR Service Properties - Conference Chairperson Options and
Messages.

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6 Click the Conference ID tab.
The New Entry Queue IVR Service - Conference ID dialog box opens.

7 Select the voice messages:
Entry Queue IVR Service Properties - Conference ID
Field/Option

Description

Request Conference ID

Prompts the participant for the conference ID.

Retry Conference ID

When the participant entered an incorrect conference ID, requests the
participant to enter the ID again.

8 Assign an audio file to each message type, as follows:
 In the Message File column, click the table entry, and then select the appropriate audio message.

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9 Click the General tab.
The New Entry Queue IVR Service - General dialog box opens.

The administrator can enable an audio message that informs the participant of the lack of Video
Resources in the Collaboration Server and that he/she is being connected as Audio Only. The
message states: All video resources are currently in use. Connecting using audio only.
The following guidelines apply:
 The IVR message applies to video participants only. Audio Only participants will not receive the
message.
 Only H.323 and SIP participants receive the audio message.
 The audio message is the first message after the call is connected, preceding all other IVR
messages.
 The message is called No Video Resources-Audio Only and the message file (.wav) is called
No video resources audio only.wav.
 The audio message must be added to the Conference and Entry Queue IVR Services
separately.
 The IVR message can be enabled/disabled by the administrator using the ENABLE_
NO_VIDEO_RESOURCES_ AUDIO_ONLY_MESSAGE System Flag in system.cfg.
Possible values: YES / NO, default: YES
If you wish to modify the flag value, the flag must be added to the System Configuration file. For
more information see the Modifying System Flags.
10 Enter the message Name and Message File name for the Audio Only message:
 Message Name: No Video Resources-Audio Only
 Message File name: No_Video_Resources_Audio_Only.wav

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11 Click the Video Services tab.
The New Entry Queue IVR Service - Video Services dialog box opens.

12 In the Video Welcome Slide list, select the video slide that will be displayed to participants
connecting to the Entry Queue. The slide list includes the video slides that were previously uploaded
to the MCU memory.
13 To view any slide, click the Preview Slide (

) button.

14 If the video slide file was not uploaded to the MCU prior to the IVR Service definition, click the:
 Add Slide - Low Resolution button to upload a Low Resolution Slide.
 Add Slide - High Resolution button to upload a High Resolution Slide.
The Install File dialog box opens. The uploading process is similar to the uploading of audio files.
For more information, see step 6 .
The video slide must be in a .jpg or .bmp file format. For more information, see Creating a Welcome
Video Slide.

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15 Click the Operator Assistance tab.
The Operator Assistance dialog box opens.

16 Select Enable Operator Assistance to enable operator assistance when the participant requires or
requests help during the connection process.
17 In the Operator Assistance Indication Message field, select the audio message to be played
when the participant requests or is waiting for operator’s assistance.
If the audio file was not uploaded prior to the definition of the IVR Service or if you want to add new
audio files, click Add Message File to upload the appropriate audio file to the Collaboration Server.

18 Click OK to complete the Entry Queue Service definition.
The new Entry Queue IVR Service is added to the IVR Services list. For more information, see IVR
Services List.

Setting a Conference IVR Service or Entry Queue IVR Service as the
Default Service
The first Conference IVR Service and Entry Queue IVR Service are automatically selected by default. The
IVR Services (Conference and Entry Queue) shipped with the system are also set as default. If additional
Conference IVR Services and Entry Queue IVR Services are defined, you can set another service as the
default for each service type.

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To select the default Conference IVR Service:
● In the IVR Services list, select the Conference IVR Service to be defined as the default, and then
click the Set Default Conference IVR Service (
) button.
Alternatively, in the IVR Services list, right-click the Conference IVR Service and then select Set
Default Conference IVR Service.

The IVR Service is displayed in bold, indicating that it is the current default service.
To select the Default Entry Queue IVR Service:

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● In the IVR Services list, select the Entry Queue IVR Service to be defined as the default, and then
click Set Default Entry Queue IVR Service (
) button.
Alternatively, in the Conference IVR Services list, right-click the Entry Queue IVR Service and then
select Set Default Entry Queue IVR Service.

The default Entry Queue IVR Service is displayed in bold, indicating that it is the current default
service.

Modifying the Conference or Entry Queue IVR Service
Properties
You can modify the properties of an existing IVR Service, except the service name and language.

To modify the properties of an IVR Service:
1 In the RMX Management pane, click IVR Services.
2 In the IVR Services list, Click the IVR Service to modify.
For more information about the tabs and options of this dialog box, see Defining a New Conference
IVR Service.
3 Modify the required parameters or upload the required audio files.
4 Click OK.

Replacing the Music File
The Collaboration Server is shipped with a default music file that is played when participants are placed on
hold, for example, while waiting for the chairperson to connect to the conference (if the conference requires
a chairperson), or when a single participant is connected to the conference. You can replace the default
music file with your own recorded music.
Music file guidelines:
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● The file must be in *.wav format.
● Music length cannot exceed one hour.
● The music recording must be in the range of (-12dB) to (-9dB).

Adding a Music File
To replace the Music file:
1 In the RMX Management pane, click IVR Services.
2 In the IVR Services list toolbar, click the Replace/Change Music File (
The Install Music File window opens.

) button.

3 Click the Browse button to select the audio file (*.wav) to upload.
The Open dialog box opens.

4 Select the appropriate audio *.wav file and then click the Open button.
The selected file name is displayed in the Install Music File dialog box.
5 Optional. You can play the selected file by clicking the Play (

) button.

a Click Play Selected File to play a file on your computer.
b Click Play RMX File to play a file already uploaded on the RMX.
6 In the Install Music File dialog box, click OK to upload the file to the MCU.
The new file replaces the previously uploaded file and this file is used for all background music
played by the MCU.

Creating Audio Prompts and Video Slides
The Collaboration Server is shipped with default voice messages (in WAV format) and video slides that are
used for the default IVR services. You can create your own video slides and record the voice messages for
different languages or customize them to your needs.

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Recording an Audio Message
To record audio messages, use any sound recording utility available in your computer or record them
professionally in a recording studio. Make sure that recorded message can be saved as a Wave file (*.wav
format) and that the recorded format settings are as defined in steps 4 and 5 on the following procedure.
The files are converted into the Collaboration Server internal format during the upload process.
This section describes the use of the Sound Recorder utility delivered with Windows 95/98/2000/XP.

To define the format settings for audio messages:
•
•

The format settings for audio messages need to be set only once. The settings will then be
applied to any new audio messages recorded.
The utility or facility used to record audio messages must be capable of producing audio files with
the formats and attributes as shown in the following procedure, namely, PCM, 16.000kHz, 16Bit,
Mono.
Windows® XP® Sound Recorder is one of the utilities that can be used.

1 On your PC, click Start > Programs > Accessories > Entertainment > Sound Recorder.
The Sound–Sound Recorder dialog box opens.

2 To define the recording format, click File > Properties.
The Properties for Sound dialog box opens.

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3 Click the Convert Now button.

The Sound Selection dialog box opens.
4 In the Format field, select PCM.
5 In the Attributes list, select 16.000 kHz, 16Bit, Mono.

6 To save this format, click the Save As button.
The Save As dialog box opens.
7 Select the location where the format will reside, enter a name and then click OK.

The system returns to the Sound Selection dialog box.
8 Click OK.
The system returns to the Properties for Sound dialog box.

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9 Click OK.
The system returns to the Sound–Sound Recorder dialog box. You are now ready to record your
voice message.

To record a new audio message:
Regardless of the recording utility you are using, verify that any new audio message
recorded adheres to the following format settings: 16.000kHz, 16Bit, Mono.

Make sure that a microphone or a sound input device is connected to
your PC.
1 On your PC, click Start > Programs > Accessories > Entertainment > Sound Recorder.
The Sound–Sound Recorder dialog box opens.
2 Click File > New.
3 Click the Record button.
The system starts recording.
4 Start narrating the desired message.
For all audio IVR messages, stop the recording anytime up to 3 minutes (which is the maximum
duration allowed for an IVR voice message). If the message exceeds 3 minutes it will be rejected by
the Collaboration Server unit.

5 Click the Stop Recording button.
6 Save the recorded message as a wave file, click File > Save As.
The Save As dialog box opens.

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7 Verify that the Format reads: PCM 16.000 kHz, 16Bit, Mono. If the format is correct, continue with
step 10. If the format is incorrect, click the Change button.
The Sound Selection dialog box is displayed.

8 In the Name field, select the name of the format created in step 7 on Select the location where the
format will reside, enter a name and then click OK..
9 Click OK.
The system returns to the Save As dialog box.
10 In the Save in field, select the directory where the file will be stored.
11 In the Save as Type field, select the *.wav file format.
12 In the File name box, type a name for the message file, and then click the Save button.
13 To record additional messages, repeat steps 1 to 10.
To upload your recorded *.wav file to the Collaboration Server, see step 6.

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Creating a Welcome Video Slide
The video slide is a still picture that can be created in any graphic application.

To create a welcome video slide:
1 Using any graphic application, save your image in either *.jpg or *.bmp file format.
2 For optimum quality, ensure that the image dimensions adhere to the Collaboration Server
recommended values (width x height in pixels):
 640 x 480
 704 x 480
 848 x 480
 720 x 576
 704 x 576
 1024 x 576
 960 x 720
 1280 x 720
 1440 x 1088
 1920 x 1088
The Collaboration Server can accommodate small deviations from the recommended slide
resolutions.
3 Save your file.
Customized H.261 slides are not supported..

If using a default Polycom slide, the slide’s resolution will be as defined in the profile, i.e. SD, HD or
CIF.
If the display of the Welcome slide is cut in the upper area of the screen, change the settings of the
endpoint’s monitor to People "Stretch" instead of "Zoom".

To upload your video slide to the Collaboration Server, see step 12.

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Inviting Participants using DTMF
This feature is disabled in SVC conferences and for SVC participants in mixed CP and SVC
conferences.

A participant in a video or audio conference can invite another participant to the conference using the
touch-tone DTMF numeric keypad on the participant’s endpoint. You can invite a participant using various
communication devices, such as a mobile phone, an IP phone, PSTN phones, laptops, or connect to
another conference running on another PBX or MCU.

Invite Call Flow
The following flow describes how a participant is invited to the conference using the DTMF codes:
1 During the conference, the participant enters the DTMF code (default is *72) on the numeric keypad
to invite another participant.
2 The participant is prompted to enter the invited participant’s destination number (a number or IP
address) including the prefix (if required) and the DTMF delimiter digit (‘*’ or ‘#’) at the end. The
asterisk (‘*’) is used to denote the dot in the IP address.
For example: To enter an IP address such as 10.245.22.19, on the DTMF keypad press
10*245*22*19 and then the DTMF delimiter.
Digits that are entered after the DTMF delimiter and before the participant is connected are ignored.

3 The system automatically dials to the destination according to the protocol order as defined in the
IVR Services Properties - Video Services tab.
When the call cannot be completed by the current protocol, the system attempts to connect to the
destination using the next protocol according to the protocol order.
The Collaboration Server connects the participant when the call is answered.
4 The last invited participant can be disconnected when the inviting participant enters the DTMF code
(default is #72) on the numeric keypad.

Entering Additional DTMF Codes
In some environments, the call is answered by an IVR system (for example when connecting to another
conference or PBX), requesting a password or a destination number to complete the connection process.
In such a case, additional DTMF digits must be entered before the DTMF forward duration time has
expired and are forwarded to the invited destination. When the additional DTMF codes are entered, they
are heard by all the conference participants.
If the DTMF code is not entered on time or if the wrong DTMF code is entered, the participant is prompted
for a new input. After the defined number of retries have elapsed, the call is ended.

Error Handling
● If the destination endpoint is busy or the participant did not answer, the system ends the call.

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● When an incorrect number is entered, the call fails and an error message is displayed.
● If the destination number is not entered in a specific amount of time (defined in Timeout for user
input in the IVR Services - Global tab), the participant is prompted to enter a destination number
again. Depending on the Number of user input retries as defined in the IVR Services - Global tab,
the system will attempt to receive the required input. When all the retries have failed, the call to the
invited participant is cancelled.

Guidelines
● Inviting other participants is available to AVC-enabled participants only.
● Participants can be invited to Event Mode, and CP and VSW conferences.
● All network protocols are supported (H.323, SIP, ISDN, and PSTN). It is recommended to select
PSTN and not ISDN if PSTN is the only destination protocol. If both PSTN and ISDN are enabled, it
is recommended to select the PSTN before ISDN as the connection process for PSTN endpoints will
be quicker.
● In an Multiple IP Networks environment, the system will try to connect the participant using each of
the IP Network Services listed in the Conference Profile - Network Services dialog box. Network
services that are excluded from this list are skipped during the dialing sequence.
● In Event Mode conferences, the invited participant connection parameters must match one of the
conference levels.
● In CP conferences, the participant initiating the invitation to another participant is able to view the
dialing information and connection status. During the dialing process, the dialing string is displayed
as the participant name which is replaced by the site name when connected to the conference.
● By default, all participants (Everyone) are granted permission to invite a participant to join a
conference. To change the permission to the Chairperson, modify the Permission column in the IVR
Service - DTMF Codes tab.

Enabling the Invite Participants using DTMF Option
The option to invite participants to a conference using the DTMF keypad is enabled in the following
Conference IVR Services dialog boxes:
● General
● Video Services
● DTMF Codes

To enable the Invite Participant using DTMF on the Collaboration Server:
1 Open an existing or define a new Conference IVR Service.
Conference IVR Service - Global dialog box opens.

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2 Click the General tab.
The Conference IVR Services - General tab is displayed.

3 In the Message File column of the Invite Participant entry, click the drop-down arrow and select the
required voice message. The file Enter_Destination_Number.wav that is shipped with the system
can be used for this message.
To upload a new file, click the Add Message File. For more details, see Creating Audio Prompts and
Video Slides.

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4 Click the Video Services tab.
The IVR Services - Video Services tab is displayed.

5 Define the following parameters
:

IVR Services Properties - Video Services Parameters - Invite Participants
Video Services

Description

Dial out protocols order

Select the order of the network protocols that will be used by the system to dial
the destination number. The system will start dialing using the first protocol, and
if the call is not answered it will continue with the second, third and fourth
protocols (if they are enabled) until the call is answered. By default, H.323 is set
as the first protocol and SIP as the second while the remaining protocols are
disabled (set to Off).

DTMF forward duration

Use this field when connecting to another conferencing entity with an IVR,
requiring the input of a password, destination number or ID. Enter the number of
seconds that the system will wait for the input of additional DTMF digits such as
a password or conference number. The range can be from 10 seconds to 600
seconds. Default is 60 seconds.

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6 Click the DTMF Codes tab.
The IVR Services - DTMF Codes tab is displayed.

7 Make sure that Invite Participant and Disconnect Invited Participant have DTMF Codes
assigned to them. Default system values are *72 (Invite Participant) and #72 (Disconnect Invited
Participant), however you can enter your own values.
When upgrading from a previous version, default system values may not be assigned if these IVR
entries were not defined in your existing IVR Service and have to be manually added to the DTMF
Codes table.
8 If required, determine who can invite other participants to the conference using DTMF codes by
changing the permissions to either Chairperson or Everyone.
9 Click OK.

Disabling the Invite Participant Option
To disable the Invite Participant option:
1 From the IVR Services - DTMF Codes tab, delete the DTMF digits from the DTMF Code column.
2 Click OK.

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External IVR Service Control
IVR Services can be controlled externally from an application server supporting the MCCF-IVR (Media
Control Channel Framework-Interactive Voice Response) package. The external IVR service is currently
being implemented with the integration of the Polycom RealPresence Virtualization Manager (DMA) as the
application server. When the application server is deployed in the enterprise environment and the Polycom
RealPresence Collaboration Server (MCU) is deployed as a media server, the external IVR service can be
used to play audio messages, display slides, and collect DTMF input from the participant.
For more information, see Using External IVR Services via the MCCF-IVR Package.

IVR Services Support with TIP Protocol
From Version 8.1, Conference IVR and Entry Queue IVR Services are supported with AVC TIP protocol in
conferences that include both TIP-enabled and non-TIP-enabled endpoints. TIP-enabled endpoints can be
moved from the Entry Queue to the destination conference if the TIP Compatibility Modes settings in the
Profile are identical for both conferencing entities (it is recommended to use the same Profile for both
entities).
The IVR services can be enabled for all TIP Compatibility Modes:
● Video only
● Video and Content
● Prefer TIP
IVR media files, WAV for voice messages and JPG for video slides, are all stored on the RealPresence
Collaboration Server (RMX).

Guidelines for TIP Support with IVR Services
● IVR default audio files are enabled for all TIP Compatibility Modes.
● Only Polycom default Welcome slides are available. Custom Welcome slides are not supported.
● TIP-enabled endpoints can send DTMF digits to MCU.
● In an mixed TIP environment, there is no support for content in cascaded conferences. Additionally,
Legacy and Lync endpoints cannot view content.

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Default IVR Prompts and Messages
The system is shipped with the following audio prompts and messages:
Default IVR Messages
Message Type

Message Text

When Played

File Name

General
Welcome
Message

“Welcome to unified conferencing.”

The participant
enters the
conference IVR
queue

General_Welcome.wav

Chairperson
Identifier
Request

“For conference Chairperson
Services, Press the Pound Key. All
other participants please wait...”

The participant is
asked to
self-identify as the
chairperson

Chairperson_ Identifier.wav

Request
Chairperson
Password

“Please enter the Conference
Chairperson Password. Press the
pound key when complete.”

The participant is
asked for the
chairperson
password

Chairperson_Password.wav

Retry
Chairperson
Password

“Invalid chairperson password.
Please try again.”

A participant
enters an
incorrect
Chairperson
password

Chairperson_
Password_Failure.wav

Request
Password

“Please enter the conference
password. Press the pound key when
complete.”

A participant is
requested to
enter the
conference
password

Conference_ Password.wav

Retry Password

“Invalid conference password. Please
try again.”

An incorrect
conference
password is
entered

Retry_ Conference_
Password.wav

Request Digit

“Press any key to enter the
conference.”

A participant is
requested to
press any key

Request_Digit.wav

Request Billing
Code

“Please enter the Billing code. Press
the pound key when complete.”

A participant is
asked to enter a
billing code

Billing_Code.wav

Requires
Chairperson

“Please wait for the chairperson to
join the conference.”

A participant
attempts to join a
conference prior
to the
Chairperson
joining

Requires Chairperson.wav

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Default IVR Messages
Message Type

Message Text

When Played

File Name

Chairperson Exit

“The chairperson has left the
conference.”
Note: The
TERMINATE_CONF_AFTER_CHAIR
_DROPPED flag must be enabled to
play this message.

The chairperson
has left the
conference.

Chairperson_Exit.wav

First to Join

“You are the first person to join the
conference.”

The first
participant joins a
conference

First to Join.wav

Mute All On

“All conference participants are now
muted.”

When all
participants are
muted by the
operator or
chairperson.

Mute_All_On.wav

Mute All Off

“All conference participants are now
unmuted.”

When all
participants are
unmuted by the
operator or
chairperson.

Mute_All_Off.wav

End Time Alert

“The conference is about to end.”

The conference is
about it end

End_Time_Alert.wav

Change
Password Menu

“Press one to change conference
password.
Press two to change chairperson
password.
Press nine to exit the menu.”

A participant
requests a
conference
password change

Change_Password_ Menu.wav

“Please enter the new conference
password. Press the pound key when
complete.”

A participant
presses two in the
Change
Password IVR
menu.

Change_ Conference_
Password.wav

“Please enter the new chairperson
password. Press the pound key when
complete.”

A participant
presses one in
the Change
Password IVR
menu.

Change_ Chairperson_
Password.wav

“Please re-enter the new password.
Press the pound key when complete.”

A participant
enters a new
conference or
chairperson
password

Confirm_ Password_
Change.wav

Change
Conference
Password

Change
Chairperson
Password
Confirm
Password
Change

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Default IVR Messages
Message Type

Message Text

When Played

File Name

Change
Password
Failure

“The new password is invalid.”

A participant
enters an invalid
password

Change_ Password_
Failure.wav

Password
Changed
Successfully

“The password has been successfully
changed.”

A participant has
confirmed a
password change

Password_ Changed_
Successfully.wav

Self Mute

“You are now muted.”

A participant
mutes his or her
audio

Self_Mute.wav

Self Unmute

“You are no longer muted.”

A participant
unmutes his or
her audio

Self_Unmute.wav

A chairperson
requests the
chairperson help
menu

Chairperson_ Help_Menu.wav

Chairperson
Help Menu

“The available touch-tone keypad
actions are as follows:
• To exit this menu press any key.
• To request private assistance,
press star, zero.
• To request operator’s assistance
for the conference, press zero,
zero.
• To mute your line, press star, six.
• To unmute your line, press pound,
six.”

A participant
requests the
participant help
menu

Participant_Help_Menu.wav

Participant Help
Menu

“The available touch-tone keypad
actions are as follows:
• To exit this menu press any key.
• To request private assistance,
press star, zero.
• To mute your line, press star, six.
• To unmute your line, press pound,
six.
• To increase your volume, press
star, nine.
• To decrease your volume, press
pound, nine.

Maximum
Participants
Exceeded

“The conference is full. You cannot
join at this time.”

A participant
attempts to join a
full conference

Maximum_ Participants_
Exceeded.wav

Request
Conference
NID

“Please enter your conference NID.
Press the pound key when complete.”

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Default IVR Messages
Message Type

Message Text

When Played

File Name

Retry
Conference
NID

“Invalid conference NID. Please try
again.”

A participant
enters an invalid
conference NID

Retry_Conference_NID.wav

Secured
Conference

“The conference is now secured.”

A chairperson or
participant
secures a
conference

Conference_Secured.wav

Unsecured
Conference

“The conference is now in an
unsecured mode”

A chairperson or
participant
unsecures a
conference

Conference_Unsecured.wav

Locked
Conference

“Conference you are trying to join is
locked”

Conference_Locked.wav

Conference
Recording

“The conference is being recorded”

Recording_ in_Progress.wav

Conference
Recording Failed

“The conference recording has failed”

Recording_Failed.wav

No Video
Resources Audio
Only.

"All video resources are currently in
use. Connecting using audio only"

No_Video_Resources_Audio_
Only.wav

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Volume Control of IVR Messages, Roll Call and Music
The volume of IVR music, and IVR messages and Roll Call is controlled by the following system flags:
● IVR_MUSIC_VOLUME
● IVR_MESSAGE_VOLUME
● IVR_ROLL_CALL_VOLUME

To control the volume of IVR music, Roll Call and messages:
● Modify the values of the System Flags listed in the following table by clicking the menu Setup >
System Configuration.
If these flags do not appear in the System Flags list, they must be manually added. For more information
see System Configuration Flags.
.

Default IVR Messages
Flag

Description

IVR_MUSIC_VOLUME

The volume of the IVR music played when a single participant is connected
to the conference varies according to the value of this flag.
Possible value range: 0-10 (Default: 2).
0 – disables playing the music
1 – lowest volume
10 – highest volume

IVR_MESSAGE_VOLUME

The volume of IVR messages varies according to the value of this flag.
Possible value range: 0-10 (Default: 2).
0 – disables playing the IVR messages
1 – lowest volume
10 – highest volume
Note: It is not recommended to disable IVR messages by setting the flag value
to 0.

The following System Flags do not require an MCU reset:
• IVR_MESSAGE_VOLUME
• IVR_MUSIC_VOLUME
• IVR_ROLL_CALL_VOLUME
For all other flag changes, the MCU must be reset for the modified flag settings (including deletion) to
take effect.

IVR Services in TIP-Enabled Conferences
Conference IVR and Entry Queue/Virtual Entry Queues are supported with AVC TIP protocol in conferences
that include both TIP-enabled and non-TIP-enabled endpoints.
A Virtual Entry Queue can be configured to either IVR Only Service Provider or External IVR Control mode.

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TIP-enabled endpoints can be moved from the Entry Queue to the destination conference if the TIP
Compatibility Modes settings in the Profile are identical for both conferencing entities (it is recommended
to use the same Profile for both entities).
TIP IVR users can access the conference directly or enter the Entry Queue/Virtual Entry Queue and provide
a password to access the conference.
The IVR services can be enabled for all TIP Compatibility Modes:
● Video only
● Video and Content
● Prefer TIP
IVR media files, WAV for voice messages and JPG for video slides, are all stored on the RealPresence
Collaboration Server (RMX).

IVR Services in TIP-Enabled Conferences Guidelines
● IVR default audio files are enabled for all TIP Compatibility Modes.
● Only Polycom default Welcome slides are available. Custom Welcome slides are not supported.
● TIP-enabled endpoints can send DTMF digits to MCU.
● In a mixed TIP environment there is no support for content in cascaded conferences. Additionally,
Legacy and Lync endpoints cannot view content.

Entry Queue and Virtual Entry Queue Access
TIP endpoints can dial-in to conferences directly using the IVR, Entry Queue/Virtual Entry Queue and IVR
Only Service Provider. For more information see Defining a New Entry Queue.
For more information on Multipoint see the Collaboration With Cisco’s Telepresence Interoperability
Protocol (TIP).

Configuring the Conference and Entry Queue IVR Services
The IVR module includes two types of services:
● Conference IVR Service that is used with conferences
● Entry Queue IVR Service that is used with Entry Queues
The configuration process is the same for TIP and non-TIP enabled Conferences and Entry Queues.
For more information about IVR Services see, Defining a New Conference IVR Service.
For more information about Entry Queues see, Entry Queues.
For more information see Appendix I - Polycom Open Collaboration Network (POCN).

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Call Detail Record (CDR) Utility
The Call Detail Record (CDR) utility enables you to view summary information about conferences, and
retrieve full conference information and archive it to a file. The file can be used to produce reports or can be
exported to external billing programs.
The value of the fields that support Unicode values, such as the info fields, will be stored in the CDR
file in UTF8. The application that reads the CDR must support Unicode.

The Collaboration Server can store details of up to 2000 conferences. When this number is exceeded, the
system overwrites conferences, starting with the earliest conference. To save the conferences’ information,
their data must be retrieved and archived. The frequency with which the archiving should be performed
depends on the volume of conferences run by the MCU.
The Collaboration Server displays Active Alarms before overwriting the older files, enabling the users to
backup the older files before they are deleted. The display of Active Alarms is controlled by the
ENABLE_CYCLIC_FILE_SYSTEM_ALARMS system flag.
If the ENABLE_CYCLIC_FILE_SYSTEM_ALARMS is set to YES and a Cyclic File reaches a file storage
capacity limit, an Active Alarm is created: “Backup of CDR files is required”.
Each conference is a separate record in the MCU memory. Each conference is archived as a separate file.
Each conference CDR file contains general information about the conference, such as the conference
name, ID, start time and duration, as well as information about events occurring during the conference, such
as adding a new participant, disconnecting a participant or extending the length of the conference.

The CDR File Properties
The output of a CDR file depends on the format in which the file was archived and the size of the file.

CDR File Formats
The conference CDR records can be retrieved and archived in the following two formats:
● Unformatted data – Unformatted CDR files contain multiple records in “raw data” format. The first
record in each file contains general conference data. The remaining records contain event data, one
record for each event. Each record contains field values separated by commas. This data can be
transferred to an external program such as Microsoft Excel for billing purposes. The following is a
sample of an unformatted CDR file.
Unformatted CDR File

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● Formatted text – Formatted CDR files contain multiple sections. The first section in each file contains
general conference data. The remaining sections contain event data, one section for each event.
Each field value is displayed in a separate line, together with its name. This data can be used to
generate a summary report for a conference. The following is an example of a formatted CDR file.
Formatted CDR File

The field names and values in the formatted file will appear in the language being used for the
Collaboration Server Web Client user interface at the time when the CDR information is retrieved.

Multi-Part CDR Files
By default, the maximum CDR (Call Data Record) file size is limited to 1MB. When a CDR file reaches a
size of 1MB the file is saved and further call data recording is stopped and the additional data is lost.
The Collaboration Server can be configured to keep recording the data in multiple CDR file set of 1MB each.
Multi-Part CDR ensures that conference call data from long duration or permanent conferences is recorded
and not lost.

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Enabling the Multi-Part CDR Option
● Multi-Part CDR is enabled by setting the value of the ENABLE_MULTI_PART_CDR system flag to
YES.
The flag’s default value is NO.
When the flag value is NO, CDR file size is limited to one file of 1MB and further call data recording
is stopped.
To modify the default setting, the flag must be manually added to the System Configuration. For more
information see, Modifying System Flags.
● If the flag value is set to YES, when a CDR file reaches 1MB, an additional CDR file is created and
added to the CDR file set for that conference.
● If the flag value is changed from YES to NO (or visa versa) all existing CDR files are retained.

CDR File Contents
The general conference section or record contains information such as the Routing Name and ID, and the
conference starting date and time.
The event sections or records contain an event type heading or event type code, followed by event data.
For example, an event type may be that a participant connects to the conference, and the event data will
list the date and time the participant connects to the conference, the participant name and ID, and the
participant capabilities used to connect to the conference.
To enable compatibility for applications that written for the MGC family, the Collaboration Server CDR file
structure is based on the MGC CDR file structure.
The unformatted and formatted text files contain basically the same information. The following differences
should be noted between the contents of the unformatted and formatted text files:
● In many cases a formatted text file field contains a textual value, whereas the equivalent unformatted
file field contains a numeric value that represents the textual value.
● For reading clarity, in a few instances, a single field in the unformatted file is converted to multiple
fields in the formatted text file, and in other cases, multiple fields in the unformatted file are combined
into one field in the formatted file.
● To enable compatibility between MGC CDR files and Collaboration Server CDR files, the unformatted
file contains fields that were applicable to the MGC MCUs, but are not supported by the Collaboration
Server MCUs. These fields are omitted from the formatted text file.
Appendix C - CDR Fields, Unformatted File Appendix contains a full list of the events, fields and
values that appear in the unformatted file. This appendix can be referred to for information regarding
the contents of fields in the unformatted text file, but does not reflect the exact contents of the
formatted text file.

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Viewing, Retrieving and Archiving Conference
Information
You can view the list of CDR files and retrieve them to your local workstation. These files can then be used
to generqate billing information, resource usage reports and more by any third party application.

Viewing the Conference Records
You can list all the CDR files that are currently saved on the MCU.

To open the CDR utility:
● On the Collaboration Server Menu, click Administration > CDR.
The CDR List pane opens, displaying a list of the conference CDR records stored in the MCU
memory.

The following fields are displayed:
Conference Record Fields
Field

Description

Display Name

The Display Name of the conference and an icon indicating whether or not the
CDR record has been retrieved and saved to a formatted text file.
The following icons are used:
The CDR record has not been saved.
The CDR record has been saved.

Start Time

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Conference Record Fields (Continued)
Field

Description

GMT Start Time

The actual time the conference started according to Greenwich Mean Time
(GMT).

Duration

The actual conference duration.

Reserved Start Time

The reserved start time of the conference. If the conference started immediately
this is the same as the Start Time.

Reserved Duration

The time the conference was scheduled to last. Discrepancy between the
scheduled and the actual duration may indicate that the conference duration was
prolonged or shortened.

Status

The conference status. The following values may be displayed:
• Ongoing Conference
• Terminated by User
• Terminated when end time passed
• Automatically terminated when conference was empty – The conference
ended automatically because no participants joined the conference for a
predefined time period, or all the participants disconnected from the
conference and the conference was empty for a predefined time period.
• Conference never became ongoing due to a problem
• Unknown error
Note: If the conference was terminated by an MCU reset, the status Ongoing
Conference will be displayed.

File Retrieved

Indicates if the conference record was downloaded using any of the file retrieval
buttons in the CDR List pane or the API.
• Yes - when the conference record was retrieved to any file or using the API.
• No - when the conference record was not retrieved at all.
The File Retrieved field is updated whenever the record is downloaded.

Multi-part CDR File display
When the Multi-Part CDR is configured on the Collaboration Server, an additional column, Part Index is
added to the CDR list.

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The Part Index column displays the CDR file’s sequence in the CDR file set:
● CDRs that are up to 1MB consist of a single file. Each file has a unique Display Name and a Part
Index of 1.
● Files included in a Multi-Part CDR file sets have the same Display Name. The first file of the set is
numbered 1 with each additional CDR file numbered in an ascending numeric sequence.

Refreshing the CDR List
If the CDR file list is displayed for sometime and you want the latest CDR files to be displayed, you can
refresh the list.

To refresh the CDR list:
● Click the Refresh
button, or right-click on any record and then select Refresh.
Updated conference CDR records are retrieved from the MCU memory.

Retrieving and Archiving Conference CDR Records
You can retrieve the CDR files and store them on your workstation for later use.

To retrieve and archive CDR records:
1 To retrieve a single CDR record, right-click the record to retrieve and then select the required format
or select the record to retrieve, and then click the appropriate button on the toolbar as detailed in the
following table.
To retrieve multiple CDR records simultaneously, use standard Windows multi-selection methods.
Menu Option

Button

Action

Retrieve

Retrieves the conference information as unformatted data
into a file whose extension is .cdr.

Retrieve Formatted
XML

Retrieves the conference information as formatted text
into a file whose extension is .xml.
Note: Viewed when logged in as a special support user.

Retrieve Formatted

Retrieves the conference information as formatted text
into a file whose extension is .txt.

The Retrieve dialog box opens.
The dialog box displays the names of the destination CDR files.
2 Select the destination folder for the CDR files and then click OK.
If the destination file already exists, you will be asked if you want to overwrite the file or specify a new
name for the destination file.
The files are saved to the selected folder.
CDR files are not included in the backup process and should be backed up manually by saving the
CDR files to a destination device.

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The RMX Manager is the Windows version of the Collaboration Server Web Client. It can be used instead
of the Collaboration Server Web Client for routine Collaboration Server management and for Collaboration
Server management
Using the RMX Manager application, a single user can control a single or multiple MCU units as well as
conferences from multiple MCUs. The Polycom® RealPresence® Collaboration Server 800s and Polycom®
RealPresence® Collaboration Server Virtual Edition system can be managed and controlled by the RMX
Manager application.
The RMX Manager can list and monitor:
● Up to 20 Collaboration Server systems in the MCUs pane
● Up to 800 conferences in the Conferences pane
● Up to 1600 participants in the Participants pane
The RMX Manager is faster than the RMX Web Client and can give added efficiency to Collaboration Server
management tasks, especially when deployed on workstations affected by:
● Lack of performance due to bandwidth constraints within the LAN/WAN environment.
● Slow operation and disconnections that can be caused by the anti-phishing component of various
antivirus applications.
Users with Auditor authorization level cannot connect to the RealPresence Collaboration Server via
the RMX Manager application and must use the RMX Web Client.

The RMX Manager application can be installed in your local workstation or accessed directly on the
RealPresence Collaboration Server system without installing it in your workstation.

Installing the RMX Manager Application
The installation of the RMX Manager Application includes two main stages:
● Accessing or downloading the RMX Manager Installer
● Installing the RMX Manager application

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Upgrade Notes
• When upgrading the RMX Manager application, it is recommended to backup the MCU list using
the Export RMX Manager Configuration option. For more details, see Import/Export RMX
Manager Configuration.
• When upgrading the RMX Manager from a major version (for example, version 8.7.0) to a
maintenance version of that version (for example, 8.7.0.x), the installation must be performed from
the same MCU (IP address) from which the major version (for example, version 8.7.0) was
installed.
If you are upgrading from another MCU (different IP address), you must first uninstall the RMX
Manager application using Control Panel > Add or Remove Programs.

New RealPresence Collaboration Server Installation Note
When managing the RealPresence Collaboration Server, upgrade/install the latest MCU version and
then install the latest RMX Manager application.
The Collaboration Server Installation and First Entry Configuration must be completed before
installing the RMX Manager application. For more details, see the Polycom® RealPresence
Collaboration Server 800s / Virtual Edition Getting Started Guide, Software Installation.
Once the connection to the Collaboration Server unit is established and the Login window is
displayed, the RMX Manager application can be installed.

Upgrade Notes
When upgrading the RMX Manager application, it is recommended to backup the MCU list using the
Export RMX Manager Configuration option.
When upgrading the RMX Manager from a major version (for example, version 8.7.0) to a
maintenance version of that version (for example, 8.7.0.x), the installation must be performed from
the same MCU (IP address) from which the major version (for example, version 8.7.0) was installed.
If you are upgrading from another MCU (different IP address), you must first uninstall the RMX
Manager application using Control Panel > Add or Remove Programs.

Accessing or downloading the RMX Manager Installer
The RMX Manager installer can be downloaded or accessed and installed on your workstations using one
of the following methods:
● Accessing the RMX Manager Application Installer Directly from the MCU
● Downloading the RMX Manager application from the Polycom web site at
http://www.polycom.com/support and installing it. The Installation procedure is the same as if you
have downloaded the application from the Login screen.
● Accessing the RMX Manager Installer from the Login screen

Accessing the RMX Manager Application Installer Directly from the MCU
1 Start Internet Explorer and in your browser enter:
http:///RMXManager.html.
For example, if the Collaboration Server IP address is 10.226.10.46, enter in the browser the
following address: http://10.226.10.46/RMXManager.html.
The RMX Manager Version nnnn page is displayed.

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2 Click the Install button.

The installer verifies the application’s requirements on the workstation.

3 Continue the Installation as described in Installing the RMX Manager on Your Workstation.

Downloading the Installation files from Polycom Support Site
1 Access the Polycom web site at http://www.polycom.com/support.
2 Click on Documents and Downloads and then select UC Infrastructure from the drop-down list
3 Select the appropriate RMX/Collaboration Server product.
4 Click the Web Client (RMX Manager) link.
The file download dialog box opens.
5 Follow the standard download procedure to either run the installer directly or save the files to your
local computer.
6 Continue the Installation as described in Installing the RMX Manager on Your Workstation.

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Accessing the RMX Manager Installer from the Login screen
1 Start Internet Explorer and connect to one of the Collaboration Server units in your site. It is
recommended to connect to the Collaboration Server installed with the latest software version.
The Login screen is displayed.
There is a link to the RMX Manager Installer at the top of the right edge of the screen.

2 Click the Install RMX Manager link.
The installer verifies the application’s requirements on the workstation.

The Install dialog box is displayed.

Installing the RMX Manager on Your Workstation
Once the installer has verified that the application’s requirements on the workstation are met, the The Install
dialog box is displayed.
1 Click the Install button.

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The installation proceeds.

The installation completes, the application loads and the RMX Manager - MCUs screen is displayed.

The first time you start the RMX Manager application, the MCUs pane is empty.

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Installing the RMX Manager for Multi-User Capability
The RMX Manager can be installed to be available to all users of a shared computer during the initial
installation.
The following procedure is performed after downloading the RMX Manager from the Polycom Support
website.
The RMX Manager can still be installed from the Collaboration Server Web Client, but the installation
will only be available to the current user.

To install the RMX Manager for Multiple Users:
1 Download the RMX Manager installation package from the Polycom Support website.
2 Unzip the installation package.

3 Double-click setup.exe to open the RMX Manager Setup Wizard.

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4 Click Next.

5 Select Everyone to install the RMX Manager for all users sharing the computer.
(Select Just me to install the RMX Manager just for the current user.)
6 Click Next.

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The installation begins.

When the installation is complete, a confirmation window is displayed.

7 Click Close to close the RMX Manager installer.
The installation is complete.

Starting the RMX Manager Application
Once installed, the RMX Manager can be run using the http:// (non-secured) command in the browser’s
address line or the Windows Start menu.

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To use the browser:
1 In the browser’s command line, enter:
http:///RMXManager.html
or
https:///RMXManager.html
2 Press Enter.

To use the Windows Start menu:
1 Click Start > Programs.
a If the RMX Manager is displayed in the recently used programs list, click RMX Manager in the list
to start the application.
or
b Click All Programs > Polycom > RMX Manager.

The MCUs screen is displayed, listing the MCUs currently defined in the RMX Manager.

This screen enables you to add additional MCUs or connect to any of the MCUs listed. For details on
adding MCUs, see Adding MCUs to the MCUs List.
For each listed MCU, the system displays the following information:
 MCU Display Name (as defined in the Add MCU dialog box).
 IP Address of the MCU’s control unit

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 Product Type - The MCU type: RealPresence Collaboration Server 800s, RMX VE, RealPresence
Collaboration Server (RMX) 1500, RealPresence Collaboration Server (RMX) 2000, or
RealPresence Collaboration Server (RMX) 4000.
Before connecting to the MCU for the first time, the Collaboration Server type is unknown so
“RMX” is displayed instead as a general indication.
To display the RMX Manager main screen you must connect to one of the listed Collaboration
Servers. For more details, see Connecting to the MCU.

Connecting to the MCU
Once an MCU is defined, the RMX Manager can be connected to it. This allows you to set up conferences,
make reservations, monitor On Going Conferences and perform other activities on several MCUs.
The first Collaboration Server unit that is connected to the RMX Manager dictates the Authorization
Level of Users that can connect to the other MCUs on the list. For example, if the Authorization level of
the User POLYCOM is Administrator, all Users connecting to the other MCUs on the list must be
Administrators. Each user can have a different login name and password for each of the listed MCUs
and they must be defined in the Users list of each of the listed MCUs.

To connect the RMX Manager to an MCU:
1 In the MCUs pane or screen, use one of the following methods:
a Double-click the MCU icon.
b Select the Collaboration Server to connect and click the Connect MCU

button.

c Right-click the MCU icon and then click Connect MCU.

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If you are connecting to the MCU from the MCUs opening screen and have defined the Username
and Password for the connecting MCU, the system connects to the Collaboration Server, and the
RMX Manager Main Screen is displayed.

If you are connecting to any MCU from the MCUs pane in the RMX Manager Main Screen and have
defined the Username and Password for the connecting MCU, the MCU icon changes to connected
and its status, type and number of audio and video resources are displayed in the MCUs pane.
If the Username and Password are missing from the MCU parameters, or if the Remember Me check
box has been cleared, the Connect dialog box opens.

2 In the Username field, enter the user name with which you will login to the MCU.
3 In the Password field, enter the password as defined for the user name with which you will login to
the MCU.
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4 To add the user name and password to the MCU properties so you will not have to enter them each
time you login to the MCU, make sure that the Remember Login check box is selected. Otherwise,
clear the Remember Login check box.
5 Click OK.
The system connects to the Collaboration Server, and the RMX Manager Main Screen is displayed.
If a User with the entered Username and Password is not defined in the Collaboration Server, an error
message is displayed and the system lets you re-enter the Username and Password.

RMX Manager Main Screen
The RMX Manager Main Screen is displayed only when at least one MCU is connected.
This screen is similar to the RMX Web Client Main Screen with the addition of the MCUs pane. As in the
RMX Web Client, the panes are displayed according to the Authorization Level of the logged in User. The
MCUs pane is displayed to all users.

Only one MCU can be selected in the MCUs pane. If only one MCU is connected, it is automatically
selected. The selected MCU is highlighted.
The menu items, the Collaboration Server Management features, the Address Book and the Conference
Templates are all properties of the selected MCU and apply to it.

MCUs Pane
The MCUs pane includes a list of MCUs and a toolbar.

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For each listed MCU, the system displays the following information:
● MCU Display Name - the name of the MCU and its icon according to its type and connection status.
The following icons are available:
MCU Icons and Statuses

Icon

Description
RealPresence Collaboration Server (RMX) 1500, disconnected.
RealPresence Collaboration Server (RMX) 1500, connected.
RealPresence Collaboration Server (RMX) 2000, disconnected.
RealPresence Collaboration Server (RMX) 2000, connected.
RealPresence Collaboration Server (RMX) 4000, disconnected.
RealPresence Collaboration Server (RMX) 4000, connected.
RealPresence Collaboration Server 1800, disconnected
RealPresence Collaboration Server 1800, connected
RealPresence Collaboration Server 800s, disconnected
RealPresence Collaboration Server 800s, connected
RealPresence Collaboration Server Virtual Edition, disconnected
RealPresence Collaboration Server Virtual Edition, connected

● IP Address - of the MCU’s control unit.
● Status - The status of the MCU:
 Connected - the MCU is connected to the RMX Manager and can be managed by the RMX
Manager user.

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 Disconnected - The MCU is disconnected from the RMX Manager
 Major - The MCU has a major problem. MCU behavior could be affected and attention is required.
● Product Type - The MCU type: RealPresence Collaboration Server 1500/2000/4000, RealPresence
Collaboration Server 1800, RealPresence Collaboration Server 800s, RealPresence Collaboration
Server Virtual Edition. Before connecting to the MCU for the first time, the Collaboration Server type
is unknown so RMX is displayed instead as a general indication.
● Monitored - When checked indicates that the conferences running on this MCU are automatically
added to the Conferences list and monitored. To stop monitoring the conferences running on this
MCU and their participants, clear the Monitored check box.
● Video Resources - The number of video resources that are available for conferencing.
● Audio Resources - The number of audio resources that are available for conferencing.

MCUs Toolbar
The MCUs toolbar contains the following buttons:

Conferences Pane
The Conferences pane lists all the ongoing conferences from all the MCUs that are connected and
monitored along with their MCU, Status, Conference ID, Start Time and End Time data. The number of
ongoing conferences is displayed in the pane’s title.
The Conferences list toolbar contains the following buttons:

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If Conference Recording is enabled the following buttons are enabled:
● Start/Resume Recording – start/resume recording.
● Stop Recording – stop recording.
Pause –

toggles with the Start/Resume button.Monitoring conferences

New conferences run on MCUs selected for Monitoring are automatically added to the Conferences list. You
can sort the conferences by MCU by clicking the MCU column heading in the Conferences table.
Conferences run on MCUs that are connected but not monitored are not listed.
Using Windows multiple selection methods to select conferences, participants from several conferences
running on different MCUs can be listed in the Participants list pane.

Starting a new conference
When starting a new conference, you must first select the MCU to run the conference in the MCUs pane.

Collaboration Server Management
The Collaboration Server Management pane lists the entities of the selected MCU that need to be
configured to enable the Collaboration Server to run conferences. Only users with Administrators
permission can modify these parameters.
The Collaboration Server Management pane is divided into two sections:
● Frequently Used – parameters often configured monitored or modified.
● Rarely Used – parameters configured during initial system set-up and rarely modified afterward.

List Pane
The List pane displays details of the participants connected to the conferences selected in the Conferences
pane or the item selected in Collaboration Server Management pane. The title of the pane changes
according to the selected item.
When selecting an item in the Collaboration Server Management pane it applies only to the MCU selected
in the MCUs list. In such a case, the system displays the name of the selected MCU in the List pane title.

Status Bar
The Status Bar at the bottom of the RMX Web Client contains System and Participant Alerts tabs as well as
Port Usage Gauges and an MCU State indicator.

System Alerts
Lists system problems of all connected MCUs (even if the MCU is not monitored). The alert indicator flashes
red when at least one system alert is active. The flashing continues until a user with Operator or
Administrator permission reviews the list.
The System Alerts can be sorted by MCU by clicking the MCU header in the System Alerts table.

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The System Alerts pane is opened and closed by clicking the System Alerts button in the left corner of the
Status Bar.

For more information about Active Alarms and Faults List, see System and Participant Alerts.

Participant Alerts
Lists the participants of all monitored MCUs that are experiencing connection problems. The list is sorted
by MCU and conference.
The Participant Alerts can be sorted by MCU by clicking the MCU header in the Participant Alerts table.
The Participant Alerts pane is opened and closed by clicking the Participant Alerts button in the left corner
of the Status Bar.

Port Usage Gauge
IThe Port Usage Gauge displays for the selected MCU:
● The total number of Video ports in the system.
● The number of Video ports in use.
● The High Port Usage threshold.
For more details, see the Polycom® RealPresence Collaboration Server 800s / Virtual Edition Getting
Started Guide, Port Usage Gauges.

MCU State
The MCU State indicator displays the status of the selected MCU.
For more details, see the Polycom® RealPresence Collaboration Server 800s / Virtual Edition Getting
Started Guide, The basic unit used for reporting resource usage in the Port Gauges is HD720p30. Results
are rounded to the nearest integer..

Address Book
Displays the Address Book of the selected MCU (regardless of its Monitored status). The Address Book is
a list of Participants and Groups that have been defined on the selected Collaboration Server.

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The information in the Address Book can be modified only by an administrator. All Collaboration Server
users can, however, view and use the Address Book to assign participants to conferences.
The name of the selected Collaboration Server is displayed in the title of the Address Book pane. For more
details, see the Polycom® RealPresence Collaboration Server 800s / Virtual Edition Getting Started Guide,
Address Book.

Conference Templates
Conference Templates enable administrators and operators to create, save, schedule and activate identical
conferences.
The Conference Templates pane lists the Conference Templates that have been defined on the selected
Collaboration Server (regardless of its Monitored status).
The Conference Templates pane is initially displayed as a closed tab. The name of the selected
Collaboration Server and the number of saved Conference Templates is indicated on the tab.
For more details, see the Polycom® RealPresence Collaboration Server 800s / Virtual Edition Getting
Started Guide, Conference Templates.

Adding MCUs to the MCUs List
The RMX Manager can connect to one or several Collaboration Servers simultaneously. If the site’s
configuration includes more than one MCU, or when a new MCU is added to your configuration, and you
want to monitor and control all MCUs from within the same window, you must add the MCU to the MCUs list.
The Collaboration Server must be installed and its IP addresses properly configured in the
Management Network Service before defining its connection parameters in the RMX Manager
application.

To add the MCU to the list of MCUs being managed, define the MCU’s connection parameters.

To add a Collaboration Server unit:
1 On the MCUs toolbar, click the Add MCU
The Add MCU dialog box opens.

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2 Define the following parameters:

MCU Properties

Field

Description

MCU Name

Enter the name of the MCU on the network.

MCU IP

Enter the IP address of the MCU’s Control Unit. The IP address must be identical to the
one configured in the MCU during first entry Configuration. For more details, see the
Polycom® RealPresence Collaboration Server 800s / Virtual Edition Getting Started
Guide, "To obtain the IP address of the Virtual Machine:” on page 2-21.

Port

Enter the number of the port used for communication and data transactions between
the Collaboration Server unit and the RMX Manager.
For standard connection, enter 80.
For a Secured connection (using TLS or SSL), enter 443.

Username

Enter the user name with which you will login to the MCU. A User with this name must
be defined in the Collaboration Server Users list. The system is shipped with a default
User whose name is POLYCOM.

Password

Enter the password as defined for the user name with which you will login to the MCU.
The system is shipped with a default User whose password is POLYCOM.

Secure Mode

Optional. Select this check box to connect to the Collaboration Server with SSL and
work in Secure Mode.

Remember Login

This check box is automatically selected, and it enables the usage of the user name
and password entered in this dialog box when connecting to the Collaboration Server.
If this check box is cleared, the user is prompted for the user name and password when
connecting to this Collaboration Server unit.

Auto
Reconnection

Select this check box to automatically reconnect to the Collaboration Server if the
connection between the RMX Manager and the MCU is broken.

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Field

Description

Interval

Enter time in seconds between reconnect ion attempts to the Collaboration Server. For
example, if you enter 10, the system will wait 10 seconds between the connection
attempts.

Max Time

Enter the maximum amount of time in seconds that the Collaboration Server is allowed
to try to reconnect. If the Collaboration Server reconnects before the allotted time frame
the count down timer is halted. For example, if you enter 100, the system will stop trying
to reconnect if it has failed to do so within 100 seconds.

3 Click OK.
The MCU is added to the MCUs pane.
4 If required, repeat steps 1-3 to define additional Collaboration Server units.
The MCUs pane contains the list of all defined MCUs.

Starting a Conference
There are several ways to start a conference:
● Clicking the New Conference button in the Conferences pane. For more information, see Starting a
Conference from the Conferences Pane.
● Dialing in to a Meeting Room defined on any of the MCUs.
 A Meeting Room is a conference that is saved on the MCU. It remains in passive mode until it is
activated by the first participant, or the meeting organizer dialing in. For more information about
Meeting Rooms, see Meeting Rooms.
● Dialing in to an Ad Hoc Entry Queue defined on one of the MCUs which is used as the access point
to the MCU.
For a detailed description of Ad Hoc Entry Queues, see Entry Queues.
● Start a Reservation:
 If the Start Time of the Reservation is past due the conference becomes ongoing immediately.
 If the Start Time of the Reservation is in the future the conference becomes ongoing, at the
specified time on the specified date.
For more information, see Starting a Reservation.
● Start any Conference Template saved in the Conference Templates list.
For more information, see Starting an Ongoing Conference or Reservation From a Template.

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Starting a Conference from the Conferences Pane
To start a conference from the Conference pane:
1 In the MCUs pane, select the MCU to run the conference.
2 In the Conferences pane, click the New Conference ( ) button.
The New Conference – General dialog box opens.

The system displays the conference’s default Name, Duration and the default Profile, which contains
the conference parameters and media settings.
The Collaboration Server automatically allocates the conference ID, when the conference starts.
In most cases, the default conference ID can be used and you can just click OK to launch the
conference. If required, you can enter a conference ID before clicking OK to launch the conference.
If you are the meeting chairperson or organizer using the RMX Web Client to start your own
meeting, you need to communicate the default conference ID (or the one you created) to the
other conference participants so they can dial in.
You can use the New Conference - General dialog box to modify the conference parameters.
If no defined participants are to be added to the conference, or you do not want to add
additional information, click OK.
For more details, see the Polycom® RealPresence Collaboration Server 800s / Virtual Edition Getting
Started Guide, Starting an AVC CP Conference from the Conferences Pane.

Starting a Reservation
To start a conference from the Reservation Calendar:
1 In the MCUs pane, select the MCU to run the conference.
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2 In the RMX Management pane, click the Reservation Calendar button (

).

The Reservation Calendar is displayed.

3 Click the New Reservation (

) button.

For more information, see the Polycom® RealPresence Collaboration Server 800s / Virtual Edition Getting
Started Guide, Scheduling an AVC-based Reservation.

Starting an Ongoing Conference or Reservation From a Template
An ongoing conference or a Reservation can be started from any Conference Template saved in the
Conference Templates list of the selected MCU.

To start an ongoing conference or a reservation from a Template:
1 In the MCUs pane, select the MCU to run the conference.
2 In the Conference Templates list, select the Template you want to start as an ongoing conference.

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3 Click the Start Conference from Template ( ) button to start a conference or Schedule
Reservation from Template ( ) button to schedule a reservation.
or
Right-click and select Start Conference from Template to start an ongoing conference or Schedule
Reservation from Template to schedule a reservation.

The conference is started.
For detailed description of Conference Templates, see Conference Templates.

Monitoring Conferences
When MCUs are connected to the RMX Manager they are automatically monitored, that is, any ongoing
conference that is started on that MCU is automatically added to the Conferences pane and its participants
are monitored.

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To list participants from several conferences (running on the same or different MCUs):
● In the Conferences pane, using Windows multiple selection methods, select the conferences whose
participants you want to list.
The participants are displayed in the Participants list pane.
By default, the participants are grouped by conferences, and the name of the MCU is displayed in the
first column of the properties table, enabling sorting according to MCU name.

Grouping the Participants by MCU
The Participants can be grouped by MCU and then by conferences.
To change the display mode for the Participants pane:
● On the Collaboration Server menu, click View > Group by MCU.

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The Participants pane display changes accordingly.

To toggle between the two display modes, click View > Group by MCU.

Start Monitoring/Stop Monitoring
By default, all conferences running on connected Collaboration Servers are monitored.
You can stop the automatic monitoring of conferences on a specific MCU in one of the following methods:
● By clearing the check box in the Monitored column in the MCUs pane.

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● Right-clicking the MCU icon and selecting Stop Monitoring.

The check box is cleared in the Monitored column.
To start monitoring again, click the check box in the Monitored column in the MCUs pane, or right-clicking
the MCU icon and selecting Start Monitoring.

Modifying the MCU Properties
You can view the currently defined MCU settings, and modify them when required, for example, change the
MCU name, IP address or Secured mode.
Use this procedure to add the Username and Password to the properties of the MCU that was automatically
added to the MCU list when installing the RMX Manager. This enables automatic login when connecting the
MCU to the RMX Manager.
You can modify the MCU properties when the MCU is connected or disconnected.

To view and/or modify the MCU Properties:
1 Use one of the following methods:
a Select the MCU to disconnect and click the MCU Properties

button.

b Right-click the MCU icon and then click MCU Properties.

The MCU Properties dialog box opens.
2 Define/modify the required parameters. For details, see MCU Properties.
3 Click OK.

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Disconnecting an MCU
An MCU can be disconnected from the RMX Manager, without removing it from the MCUs list.

To disconnect an MCU:
1 Use one of the following methods:
a Select the MCU to disconnect and click the Disconnect MCU

button.

b Right-click the MCU icon and then click Disconnect MCU.

The MCU icon changes to disconnected and any ongoing conference running on that MCU will not
be monitored in this RMX Manager; they are removed from the Conferences pane. This MCU can still
be monitored and controlled by other users.

Removing an MCU from the MCUs Pane
An MCU can be removed from the RMX Manager. This function should be used if the MCU hardware was
disconnected and removed from the network.

To Remove an MCU from the list:
1 Use one of the following methods:
a Select the MCU to disconnect and click the Delete

button.

b Right-click the MCU icon and then click Remove MCU.

A confirmation message is displayed.
2 Click OK to confirm or Cancel to abort the operation.
The MCU icon is removed from the MCUs pane.

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Changing the RMX Manager Language
You can change the language of the RMX Manager menus and dialog boxes. Only one language can be
selected at a time and the RMX Manager application must be restarted after changing the display language.

To select a language:
1 On the RMX Manager menu, click Setup > Customize Display Settings > Multilingual Settings.
The Multilingual Settings dialog box opens, displaying the current language selection.

2 Click the check box of the required language. Only one language can be selected.
3 Click OK.
4 Restart the RMX Manager application to implement the language change.

Import/Export RMX Manager Configuration
The RMX Manager configuration that includes the MCU list and the multilingual selection can be save to
any workstation/PC on the network and imported to any Multi-RMX Manager installed in the network. This
enables the creation of the MCUs list once and distributing it to all RMX Manager installations on the
network.
In addition, when upgrading to a previous version, the MCU list is deleted, and can be imported after
upgrade.
The exported file is save in XML format and can be edited in any text editor that can open XML files.

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To Export the RMX Manager Configuration:
1 In the RMX Manager, click the Export RMX Manager Configuration
button in the toolbar, or
right-click anywhere in the MCUs pane and then click Export RMX Manager Configuration.

The Export RMX Manager Configuration dialog box opens.
2 Click the Browse button to select the location of the save file, or enter the required path in the
Export Path box.

The selected file path is displayed in the Export Path box.
3 Click OK to export the RMX Manager configuration.

To Import the RMX Manager Configuration:
1 In the RMX Manager, click the Import RMX Manager Configuration
button in the toolbar, or
right-click anywhere in the MCUs pane and then click Import RMX Manager Configuration.

The Import RMX Manager Configuration dialog box opens.

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2 Click the Browse button to select the saved file, or enter the required path in the Export Path box.
The Open dialog box is displayed.

3 Select the XML file previously saved, and click the Open button.
The selected file path is displayed in the Import Path box.
4 Click OK to import the file.

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System and Participant Alerts
The MCU alerts users to any faults or errors the MCU encountered during operation. Two indication bars
labeled System Alerts and Participant Alerts signal users of system errors by blinking red in the event of an
alert.
Collaboration Server 800s Status Bar

The System Alerts indication bar blinks red prompting the user to view the active alarms. Once viewed, the
System Alerts indication bar becomes statically red until the errors have been resolved in the MCU.
The Participants Alerts indication bar blinks red indicating participant connection difficulties in conferences.
Once viewed, the Participant Alerts indication bar becomes statically red until the errors have been resolved
in the MCU.

System Alerts
System Alerts are activated when the system encounters errors such as a general or card error. The system
errors are recorded by the Collaboration Server and can be generated into a report that can be saved in *.txt
format.

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To view the System Alerts list:
1 Click the red blinking System Alerts indication bar.
The Active Alarms pane opens. This screen indicates what events have not been resolved.

The following columns appear in the Active Alarms pane:
Active Alarms Pane Columns
Field

Description

ID

An identifying number assigned to the system alert.

Time

Lists the local date and time that the error occurred. This column also
includes the icon indicating the error level (as listed in the level column).

GMT Time

Lists the date and time according to Greenwich Mean Time (GMT) that
the error occurred.

Category

Lists the type of error. The following categories may be listed:
• File indicates a problem in one of the files stored on the MCU’s hard disk.
• Card indicates problems with a card.
• Exception indicates software errors.
• General indicates a general error.
• Assert indicates internal software errors that are reported by the software
program.

Category (cont.)

•
•

Level

Indicates the severity of the problem, or the type of event. There are three fault
level indicators:

Startup indicates errors that occurred during system startup.
Unit indicates problems with a unit.

- Major Error
- System Message
- Startup Event
Code

Indicates the problem, as indicated by the error category.

Process Name

Lists the type of functional process involved.

Description

When applicable, displays a more detailed explanation of the cause of the
problem.

For more information about the Active Alarms, see Appendix B - Active Alarms.

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2 Click one of the following two buttons to view its report in the System Alerts pane:
System Alerts Buttons
Active Alarms (default) – this is the default reports list that is displayed when clicking the
System Alerts indication bar. It displays the current system errors and is a quick indicator of
the MCU status.
Faults Full List - A list of all system faults.
Note: Viewed when logged in as a special support user.
Faults List – a list of faults that occurred previously (whether they were solved or not) for
support or debugging purposes.

3 To save the Active Alarms, Faults Full List or Faults report:
 to a text file, click the Save to Text
 to an XML file, click the Save to XML

button
button

The Save to XML button is only available when logged in as a special support user.

The Save dialog window opens.
4 Select a destination folder and enter the file name.
5 Click Save.

Participant Alerts
Participant Alerts enables users, participants and conferences to be prompted and currently connected.
This includes all participants that are disconnected, idle, on standby or waiting for dial-in. Alerts are intended
for users or administrators to quickly see all participants that need their attention.

To view the Participants Alerts list:
1 Click the red blinking Participants Alerts indication bar.
The Participant Alerts pane opens.

The Participant Alerts pane displays similar properties to that of the Participant List pane. For more
information, see the Polycom® RealPresence Collaboration Server 800s / Virtual Edition Getting
Started Guide, Participant Level Monitoring.

2 To resolve participant issues that created the Participant Alerts, the administrator can either
Connect
, Disconnect
or Delete
a participant.

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RMX Time
To ensure accurate conference scheduling, the MCU has an internal clock that can function in standalone
mode, or in synchronization with up to three Network Time Protocol (NTP) servers.
NTP Servers can be used if:
● NTP servers use Version 4 as it is the only supported protocol.
● If applicable, daylight saving adjustments must be implemented by the administrator whether the
MCU is in standalone mode or synchronized with NTP Servers.

Altering the clock
The MCU’s date and time can be set manually or enabled to synchronize with external NTP servers.

To Alter the MCU Time:
1 On the Collaboration Server menu, click Setup > RMX Time.
The RMX Time dialog box opens.

2 View or modify the following fields:
RMX Time – Fields Properties
Field

Description

GMT Date

The date at Greenwich, UK.

Local Time

The MCU’s local time settings, are calculated from the GMT Time and the GMT
Offset.

GMT Time

The MCU’s current GMT Time settings.
Select the Up or Down arrows to alter the GMT Time on the MCU.

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RMX Time – Fields Properties
Field

Description

GMT Offset

The time zone difference between Greenwich and the MCU’s physical location
in hours and minutes.
Select the Up or Down arrows to alter the GMT Offset time on the MCU. To
enter a negative offset either type a minus in the hour box or use the down
arrow and decrease the offset below zero.

Retrieve Client Time

Click this button to automatically update the MCU’s GMT Date, Time and Offset
to match that of the workstation.

Use NTP Server

Select this check box to synchronize the time with up to three NTP servers.
When selected, the manual GMT Date and GMT Time setting options are
disabled. The GMT Offset fields are still active.
To implement this mode an external connection to an NTP server must be
enabled.
Enter the IP addresses of the required NTP servers in order of precedence.
The Status field indicates whether registration with the NTP Server failed or
succeeded.
Note: TheCollaboration Server will not use a time source such as a
Windows-based, W32Time service (SNTP) time service. Only full-featured
(below Stratum 16) NTP Servers are considered sufficiently reliable for
high-accuracy timing environments.

Adjust Reservations
Time
(Button)

Use this button to adjust the start time of all the reservations in one operation.
For more information see Adjusting the Start Times of all Reservations.
Not supported in the RealPresence Collaboration Server.

After resetting the MCU a delay may occur when synchronizing with the external NTP server.

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Resource Management
This section describes how the MCU resources are managed by the MCU and how they are used by the
MCU to connect participant to conferences.
This section describes:
Resource Capacity
AVC Conferencing - Voice
Displaying the Resource Report
MCU Resource Management by RealPresence Resource Manager, Polycom CMA and Polycom
RealPresence DMA System

Resource Capacity
The MCU resources are determined by the MCU type, allocated hardware (Virtual Edition), and the system
license you have purchased. The total number of licensed resources is shown in the System Information.

MCU Capacities in CP Only Conferencing and SVC Only Conferencing
The following table describes the resource capacity allocations for the Collaboration Server per Resource
type (voice or video protocol) and per resolution.
Resource Capacity Allocation per Resource Type
Resource Type

Number of Resources

VoIP Ports

120

CIF Ports

40

SD Ports (4CIF)

40

HD 720p30

20

HD 1080p30

10

VGA RTV

20

SVC Only

60

One HD720p30 port equals 3 SVC ports or 2 CIF ports.

Resource Usage in SVC Conferencing
During a SVC conference, each SVC-endpoint uses a video port that is equivalent to a third of HD720p30
port. When sharing content an additional video resource is used.

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MCU Capacities in Mixed CP and SVC Conferencing
In a mixed CP and SVC conference, video resources are used according to the amount of both AVC and
SVC participants in the conference.
The ratio of resources in a mixed conference is one AVC HD (720p30) video resource to three SVC video
resources, meaning for each AVC HD video resource, three SVC video resources can be allocated.
For example, in a mixed AVC/SVC conference, 10 HD AVC ports and 30 SVC ports can be used,
maintaining ratio of one HD port to three SVC ports.
The following diagram illustrates the amount of AVC to SVC port resources that are used in a mixed
AVC/SVC conference:

R

AVC Conferencing - Voice
One CIF video resource equals 3 voice resources. All resources are taken from the same pool of video
resources.

Forcing Video Resource Allocation to CIF Resolution
You can set the MCU to allocate one CIF video resource to an endpoint, regardless of the resolution
determined by the Conference Profile parameters. This forcing saves resources and enables more
endpoints to connect to conferences.
The forcing is done by modifying the system configuration and it applies to all conferences running on the
MCU.
You can specify the endpoint types for which resource allocation can be forced to CIF resource, enabling
other types of endpoints to use higher resolutions in the same conference. For example, you can force the
system to allocate one CIF video resource to CMAD and VSX endpoints while HDX endpoints can connect
using SD or HD video resources.
Once the endpoint connects to the conference, its type is identified by the Collaboration Server and, if
applicable, the Collaboration Server will connect it using one CIF resource, even if a higher resolution can
be used.

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To force CIF resource:
1 On the Collaboration Server menu, click Setup > System Configuration.
The System Flags dialog box opens.
2 In the MCMS_PARAMETERS tab, click the New Flag button.
The New Flag dialog box is displayed.

3 In the New Flag field enter the flag name: FORCE_CIF_PORT_ALLOCATION
4 In the Value field enter the product type to which the CIF resource should be allocated. Possible
values are:
 CMA Desktop for CMA desktop client
 VSX nnnn where nnnn represents the model number for example, VSX 8000.
You can define several endpoint types, listing them one after the other separated by semicolon (;).
For example, CMA Desktop;VSX 8000.
5 Click OK.
The new flag is added to the flags list.
Reset the MCU for changes to take effect. For more details, see the Resetting the RealPresence
Collaboration Server Virtual Edition.

To cancel the forcing of CIF resource:
1 On the Collaboration Server menu, click Setup > System Configuration.
The System Flags dialog box opens.
2 In the MCMS_PARAMETERS tab, double-click or select the flag FORCE_CIF_PORT_ALLOCATION
and click the Edit Flag button.
3 In the New Value field, clear the value entries.
4 Click OK.
Reset the MCU for changes to take effect. For more details, see the Resetting the RealPresence
Collaboration Server Virtual Edition.

Resource Reports
When viewing the Collaboration Server resource report, the resource allocations are described in AVC
HD720p30 units.
The Resource Report includes a graphic representation of the resource usage. One resource report is
available for all resource usage including SVC-based endpoints.

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Displaying the Resource Report
» In the main toolbar, click Administration > Resource Report.

For each resource type, the Resource Report includes the following columns:
Resource Report Fields Parameters
Column

Description

Type

This is always Video. This applies to both AVC and SVC-based endpoints (and
resources).

Occupied

The number of MCU resources that are used by connected AVC and
SVC-based participants or reserved for defined participants.

Free

The number of MCU resources available for connecting AVC and SVC-based
endpoints.

Total

The Total column displays the total number of resources of that type (Occupied
and Free).

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The Resource Report dialog box is displayed, showing the resource usage according to the Resource
Capacity Mode.

The actual number of occupied or free resources can also be displayed by moving the cursor over
the columns of the bar graph. Moving the cursor over the Video bar displays the following view:

Port Gauge
The Port Gauge in the Status Bar show the numbers as they appear in the resource report. In the following
example, 20 of the 400 system resources are shown as occupied.

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Resource Capacities in AVC CP, SVC and Mixed Mode Conferences
When viewing the Collaboration Server resource report for mixed CP and SVC conferences, the resource
allocations are described in AVC HD720p30 units. A port ratio of 1 AVC HD port will equal 2 AVC SD ports,
which equals 3 SVC ports (in a non-mixed conference). When the Collaboration Server is reporting the
available capacity, it will appropriately round up the remaining capacity to the nearest whole value of
available ports. For example, one SVC endpoint in a conference is equal to 1/3 of the resource value. The
resource report displays this as one full resource used. Two SVC endpoints is equal to 2/3 of the resource
value. Therefore, the resource report displays this as one full resource used, and so forth. The following
tables show the actual resource capacity utilization for both CP only and mixed CP and SVC conferences
in AVC HD720p30 units a single

MCU Resource Management by RealPresence Resource Manager,
Polycom CMA and Polycom RealPresence DMA System
When the RealPresence Resource Manager, Polycom CMA and Polycom RealPresence DMA system are
part of the solution, following a request by the RealPresence Resource Manager Polycom CMA or Polycom
RealPresence DMA system, the MCU will send updates on resource usage to both CMA and DMA, with
each application updating its own resource usage for the MCU. This provides better management of the
Collaboration Server resources by the RealPresence Resource Manager, Polycom CMA and Polycom
RealPresence DMA system.

Guidelines
● Following requests sent by CMA and RealPresence DMA system, the Collaboration Server will send
the number of occupied resources for a conference or total for the MCU.In Flexible Resource
Capacity Mode, CMA/DMA receive information about how many Video (CIF) and Audio resources are
occupied per conference or MCU according the request type sent by the CMA and DMA.
● Occupied resources are resources that are connected to ongoing conferences. Disconnected
endpoints in an ongoing conference are not counted as occupied resources.
● An ongoing conference that does not include participants and the Send Content to Legacy Endpoints
option is disabled does not occupy resources. If the Send Content to Legacy Endpoints option is
enabled, the conference occupies one SD resource.
● The Collaboration Server is unaware of the resource usage split between the CMA and
RealPresence DMA system.

Port Usage Threshold
The Collaboration Server can be set to alert the administrator to potential port capacity shortages. A
capacity usage threshold can be set as a percentage of the total number of licensed ports in the system.
When the threshold is exceeded, a System Alert is generated.
The default port capacity usage threshold is 80%.
The administrator can monitor the MCU’s port capacity usage via the Port Gauge in the Status Bar of the
Collaboration Server Web Client.

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Setting the Port Usage Threshold
To Set the Port Usage Threshold:
1 In the Setup menu, click Port Gauge to open the Port Gauge dialog box.

2 Enter the value for the percentage capacity usage threshold.
The high Port Usage threshold represents a percentage of the total number of video available. It is
set to indicate when resource usage is approaching its maximum, resulting in no free resources to
run additional conferences. When port usage reaches or exceeds the threshold, the red area of the
gauge flashes. The default port usage threshold is 80%.
3 Click OK.

SIP Dial-in Busy Notification
When the system flag SEND_SIP_BUSY_UPON_RESOURCE_THRESHOLD is set to YES (NO is the
default), it enables the MCU to send a busy notification to a SIP audio endpoint or a SIP device when dialing
in to the MCU whose audio resource usage exceeded the Port Usage threshold.
The Collaboration Server will send a SIP busy response to SIP audio endpoints when:
● The system flag SEND_SIP_BUSY_UPON_RESOURCE_THRESHOLD is set to YES (NO is the
default)
● The port usage threshold for Audio resources is exceeded. The threshold is defined in the Setup >
Port Gauge dialog box.

When the flag is set to YES, the system will allow SIP audio endpoints to connect to the MCU until the Port
Usage threshold is reached. Once this threshold is exceeded, the SIP audio endpoints will not be able to
connect, ensuring that the remaining system resources can be used by all other connections, including SIP

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video, and H.323 cascaded links. When the call is rejected by the MCU because of lack of resources, the
appropriate indication will be sent by the MCU to the SIP audio endpoint.
For example, if the Port Gauge threshold is set to 80%, when 80% of the Audio resources are used, the
system will not allow additional SIP audio endpoints to connect and will send a busy notification to the
endpoint.
This does not affect the video resources usage.

Port Usage Gauge
The Port Usage Gauge is displayed in the Status Bar at the bottom of the Collaboration Server Web Client
screen.

Port Usage Gauge

T

The Port Usage Gauge displays for the selected MCU:
● The total number of Video ports in the system.
● The number of Video ports in use.
● The High Port Usage threshold.

The basic unit used for reporting resource usage in the Port Gauges is HD720p30. Results are
rounded to the next integer.

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System Information
System Information includes License Information and general system information.

To view the System Information properties box:
● On the Collaboration Server menu, click Administration > System Information.
The System Information dialog box is displayed.

The following information is displayed:

System Information
Field

Description

Total Number of CP
(HD720p30) Resources

Displays the number of HD720p30 video resources licensed for the system.
Each HD720p30 resource represents 3 CIF video resources. Each SVC
resource is equivalent to one CIF video resource.

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System Information (Continued)
Field

Description

Total Number of Event Mode
Resources

Displays the number of video/voice resources licensed for a system in
Event Mode Licensing. It also determines the conference type that is
available on the system.
0 - indicates that this Licensing mode is disabled for this system.

RMX Version

Displays the System Software Version of the MCU.

Encryption

Indicates whether Encryption is included in the MCU license. Encryption is
not available in all countries.
Range: True / False

Telepresence Mode

The field value indicates whether the system is licensed to work with RPX
and TPX Telepresence room systems.
Range: True / False

Serial Number

Displays the Serial Number of the Collaboration Server unit.

HD

Indicates if the MCU is licensed to connect endpoints at HD resolutions in
Continuous Presence conferences.

SVC

Indicates if the MCU is licensed to run SVC-based conferences.

Polycom Partners

Indicates that the System Software contains features for the support of
specific Polycom Partner environments.

Memory Size [MB]

Indicates the MCU system memory size in Megabytes.
Note: If Memory size is 512MB (Collaboration Server 1500/2000/4000
only), Version 7.1 and later are not supported. DO NOT upgrade the system
to Version 7.1 and later.

Card Configuration Mode

Not applicable to Collaboration Server 800s/Virtual Edition.
•
•
•

SNMP (Simple Network Management Protocol)
SNMP enables managing and monitoring of the MCU status by external managing systems, such as HP
OpenView or through web applications.
The Collaboration Server’s implementation of SNMPv3 is FIPS 140 compliant.

MIBs (Management Information Base)
MIBs are a collection of definitions, which define the properties of the managed object within the device to
be managed. Every managed device keeps a database of values for each of the definitions written in the
MIB.
The SNMP systems poll the MCU according to the MIB definitions.

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Traps
The MCU is able to send Traps to different managers. Traps are messages that are sent by the MCU to the
SNMP Manager when an event such as MCU Reset occurs.

Guidelines
● Version 1, Version 2 and Version 3 traps are supported.
● When SNMPv3 is selected only SNMPv3 Queries and Traps receive responses.
● A mixture of Version 1, Version 2 and Version 3 traps is not permitted.

MIB Files
The H.341 standard defines the MIBs that H.320 and H.323 MCUs must comply with. In addition, other MIBs
should also be supported, such as MIB-II and the ENTITY MIB, which are common to all network entities.
The MIBs are contained in files in the SNMP MIBS sub-directory of the Collaboration Server root directory.
The files should be loaded to the SNMP external system and compiled within that application. Only then can
the SNMP external application perform the required monitoring tasks.
The MULTI-MEDIA_MIB_TC must be compiled before compiling the other MIBs.

Private MIBs
● RMX-MIB (RMX-MIB.MIB)
 Contains the statuses of the Collaboration Server: Startup, Normal and Major.
 Contains all the Alarms of the Collaboration Server that are sent to the SNMP Manager.

Support for MIB-II Sections
The following table details the MIB-II sections that are supported:
Supported MIB-II Sections
Section

Object Identifier

system

mib-2 1

interfaces

mib-2 2

ip

mib-2 4

The Alarm-MIB
MIB used to send alarms. When a trap is sent, the Alarm-MIB is used to send it.

H.341-MIB (H.341 – H.323)
● Gives the address of the gatekeeper.
● Supports H.341-MIB of SNMP events of H.323.
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Standard MIBs
This section describes the MIBs that are included with the Collaboration Server. These MIBs define the
various parameters that can be monitored, and their acceptable values.
Standard MIBs
MIB Name

Description

MULTI-MEDIA-MIB-TC
(MULTIMTC.MIB)

Defines a set of textual conventions used within the set of Multi Media MIB
modules.

H.320ENTITY-MIB
(H320-ENT.MIB)

This is a collection of common objects, which can be used in an H.320 terminal, an
H.320 MCU and an H.320/H.323 gateway. These objects are arranged in three
groups: Capability, Call Status, and H.221 Statistics.

H.320MCU-MIB
(H320-MCU.MIB)

Used to identify managed objects for an H.320 MCU. It consists of four groups:
System, Conference, Terminal, and Controls. The Conference group consists of the
active conferences. The Terminal group is used to describe terminals in active
MCU conferences. The Controls group enables remote management of the MCU.

H323MC-MIB
(H323-MC.MIB)

Used to identify objects defined for an H.323 Multipoint Controller. It consists of six
groups: System, Configuration, Conference, Statistics, Controls and Notifications.
The Conference group is used to identify the active conferences in the MCU. The
Notifications group allows an MCU, if enabled, to inform a remote management
client of its operational status.
Note: The Collaboration Server supports only one field in H.341-H323MC MIB. The
Collaboration Server reports the Gatekeeper address using H.341-H323MC MIB –
323McConfigGatekeeperAddress (0.0.8.341.1.1.4.2.1.1.4) in response to a query
from a manager.

MP-MIB (H323-MP.MIB)

Used to identify objects defined for an H.323 Multipoint Processor, and consists of
two groups: Configuration and Conference. The Configuration group is used to
identify audio/video mix configuration counts. The Conference group describes the
audio and video multi-processing operation.

MIB-II/RFC1213-MIB
(RFC1213.MIB)

Holds basic network information and statistics about the following protocols: TCP,
UDP, IP, ICMP and SNMP. In addition, it holds a table of interfaces that the Agent
has. MIB-II also contains basic identification information for the system, such as,
Product Name, Description, Location and Contact Person.

ENTITY-MIB (ENTITY.MIB)

Describes the unit physically: Number of slots, type of board in each slot, and
number of ports in each slot.

Unified MIB
The Collaboration Server uses the Polycom Unified MIB, in addition to the RMX specific MIB. The Polycom
Unified MIB is an MIB that is used by many Polycom products. The following table describes the information
provided by the Collaboration Server in the Unified MIB.

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Unified MIB SNMP Fields
Name

Type

Description

Debug

Boolean

Indicates whether the unit is in a debugging state.

IncomingCallsReqrGK

Boolean

Indicates whether a gatekeeper is required to receive incoming
H.323 calls.

OutgoingCallsReqrGK

Boolean

Indicates whether a gatekeeper is required to make outgoing
H.323 calls.

HDBitrateThrshld

Integer

The minimum bit rate required by endpoints in order to connect
to an HD conference.

MaxCPRstln

Integer

Maximum resolution of a CP conference.

MaxCPRstlnCfg

Integer

Configured resolution for a CP conference.

EndpointDispayName

String

The name of the MCU that is displayed on the screen of
endpoints that are connecting to the conference.

PALNTSC

NTSC/PAL/AUT
O

The video encoding of the RMX.

SeparateMgmtNet

Boolean

Indicates whether management network separation is enabled.

NumPorts

Integer

Total number of ports.

NumVideoPorts

Integer

Number of ports configured for video.

ServiceH323

Integer

Indicates the status of H.323 capabilities:
1 - The service is enabled and operational.
2 - The service is enabled but is not operational.
3 - The service is disabled.

ServiceSIP

Integer

Indicates the status of SIP capabilities:
1 - The service is enabled and operational.
2 - The service is enabled but is not operational.
3 - The service is disabled.

ServiceISDN

Integer

Indicates the status of SIP capabilities:
1 - The service is enabled and operational.
2 - The service is enabled but is not operational.
3 - The service is disabled.

RsrcAllocMode

Fixed/Flexible

The resource allocation method which determines how the
system resources are allocated to the connecting endpoints.

McuSystemStatus

Integer

System State.

FanStatus

Boolean

Status of the hardware fan.

PowerSupplyStatus

Boolean

Status of the power supply.

IntegratedBoardsStatus

Boolean

Status of the integrated boards.

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Unified MIB SNMP Fields (Continued)
Name

Type

Description

UltraSecureMode

Boolean

Indicates whether the RMX is operating in Ultra Secure Mode.

ChassisTemp

Integer

The temperature of the chasis.

NumPortsUsed

Integer

Number of ports currently in use.

NewCallsPerMinute

Integer

New calls in the last minute.

ScsfNewCallsPerMinute

Integer

Successful new calls in the last minute.

FldNewCallsPerMinute

Integer

Failed new calls in the last minute.

PctScsflNewCalls

Integer

Percentage of new calls in the last minute which were
successful.

CallsEndedScsflPerMin

Integer

Number of calls in the last minute which ended with a success
code.

CallsEndedFailedPerMin

Integer

Number of calls in the last minute which ended with a failure
code.

CallsEndedScsfl

Integer

Number of calls in the last minute which ended with a success
code.

CallsEndedFailed

Integer

Number of calls in the last minute which ended with a failure
code.

NumActvCnfrncs

Integer

Number of active conferences.

Traps
Three types of traps are sent as follows:
● ColdStart trap. This is a standard trap which is sent when the MCU is reset.
An Example of a ColdStart Trap

coldStart notification received from: 172.22.189.154 at 5/20/2007
7:03:12 PM
Time stamp: 0 days 00h:00m:00s.00th
Agent address: 172.22.189.154 Port: 32774 Transport: IP/UDP
Protocol: SNMPv2c Notification
Manager address: 172.22.172.34 Port: 162 Transport: IP/UDP
Community: public
Enterprise: enterprises.8072.3.2.10
Bindings (3)

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● Authentication failure trap. This is a standard trap which is sent when an unauthorized community
tries to enter.
An Example of an Authentication Failure Trap

authentication Failure notification received from: 172.22.189.154 at
5/20/2007 7:33:38 PM
Time stamp: 0 days 00h:30m:27s.64th
Agent address: 172.22.189.154 Port: 32777 Transport: IP/UDP
Protocol: SNMPv2c Notification
Manager address: 172.22.172.34 Port: 162 Transport: IP/UDP
Community: public
Enterprise: enterprises.8072.3.2.10
Bindings (3)

● Alarm Fault trap. The third trap type is a family of traps defined in the POLYCOM-RMX-MIB file,
these traps are associated with the Collaboration Server active alarm and clearance (proprietary
SNMP trap).
An Example of an Alarm Fault Trap

rmxFailedConfigUserListInLinuxAlarmFault notification received
from: 172.22.189.154 at 5/20/2007 7:04:22 PM
Time stamp: 0 days 00h:01m:11s.71th
Agent address: 172.22.189.154 Port: 32777 Transport: IP/UDP
Protocol: SNMPv2c Notification
Manager address: 172.22.172.34 Port: 162 Transport: IP/UDP
Community: public
Bindings (6)
Binding #1: sysUpTime.0 *** (timeticks) 0 days 00h:01m:11s.71th
Binding #2: snmpTrapOID.0 *** (oid)
rmxFailedConfigUserListInLinuxAlarmFault
Binding #3: rmxAlarmDescription *** (octets) Insufficient resources
Binding #4: rmxActiveAlarmDateAndTime *** (octets)
2007-6-19,16:7:15.0,0:0
Each trap is sent with a time stamp, the agent address, and the manager address.

Status Trap
The MCU sends status traps for the status MAJOR - a trap is sent when the card/MCU status is MAJOR.
All traps are considered “MAJOR”.

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RMX MIB entities that do not generate traps.
The following table lists the entities that appear in the RMX MIB of the SNMP that do not generate traps.
These traps will be displayed as Faults in the System Alerts pane (at the bottom of the Collaboration Server
(RMX) Web Client screen.
SNMP MIB entities that do not generate traps
Key

Description

5002

Resource process did not receive the Meeting Room list during startup.

5004

Task terminated

5008

Low Processing Memory

5009

Low system Memory

5010

High system CPU usage

5014

High CPU utilization

5016

Process idle

5107

Failed to open Apache server configuration file

5108

Failed to save Apache server configuration file

5110

A private version is loaded

5111

NTP synchronization failure

5112

Invalid date and time

5116

Incorrect Ethernet Settings

5117

Smart Report found errors on hard disk

5118

Invalid MCU Version

5150

Music file error

5205

Unspecified problem

5207

Unit not responding

5209

Failed to mount Card folder

5401

The Log file system is disabled

5450

Action redirection failure

5601

Process terminated

5602

Terminal initiated MCU reset

5603

User initiated MCU reset

5604

Internal MCU reset

5605

MCU reset

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SNMP MIB entities that do not generate traps (Continued)
Key

Description

Comment

5606

MCU Reset to enable Diagnostics mode

5607

Startup process failure

5801

Polycom default User exists. For security reasons, it is recommended to delete
this User and create your own User.

5904

Single clock source

5950

MCU is not configured for AVF gatekeeper mode

5652

Hard disk error /AA_HARD_DISK_FAILURE

Not in use

5551

Port configuration modified

Not in use

5011

Used for testing the Active Alarms mechanism

Not in use

5001

License not found

Not in use
(Product activation
failure is trapped)

Only in non-Ultra
Secure Mode

Defining the SNMP Parameters in the Collaboration Server
The SNMP option is enabled via the Collaboration Server Web Client application.
The addresses of the Managers monitoring the MCU and other security information are defined in the
Collaboration Server Web Client application and are saved on the MCU’s hard disk. Only users defined as
Administrator can define or modify the SNMP security parameters in the Collaboration Server Web Client
application.

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To enable SNMP option:
1 In the Collaboration Server Web Client menu bar, click Setup > SNMP.
The RMX SNMP Properties - Agent dialog box is displayed.

This dialog box is used to define the basic information for this MCU that will be used by the SNMP
system to identify it.
2 In the Agent dialog box, click the SNMP Enabled check box.
3 Click the Retrieve MIB Files button to obtain a file that lists the MIBs that define the properties of
the object being managed.
The Retrieve MIB Files dialog box is displayed.
4 Click the Browse button and navigate to the desired directory to save the MIB files.
5 Click OK.
The path of the selected directory is displayed in the Retrieve MIB Files dialog box.
6 Click the Save button.
The MIB files are saved to the selected directory.
7 Click Close to exit the Retrieve MIB Files dialog box.
8 In the Agent dialog box, define the parameters that allow the SNMP Management System and its
user to easily identify the MCU.
Collaboration Server-SNMP Properties - Agent Options
Field

Description

Contact person for
this MCU

Type the name of the person to be contacted in the event of problems with the
MCU.

MCU Location

Type the location of the MCU (address or any description).

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Collaboration Server-SNMP Properties - Agent Options
Field

Description

MCU System Name

Type the MCU’s system name.

9 Click the Traps tab.
The SNMP Properties – Traps dialog box opens.

Traps are messages sent by the MCU to the SNMP Managers when events such as MCU Startup or
Shutdown occur. Traps may be sent to several SNMP Managers whose IP addresses are specified
in the Trap Destinations box.

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10 Define the following parameters:
SNMPv3 - Traps
Field

Description

SNMP Trap Version

Specifies the version, either Version 1 2 or 3 of the traps being sent to the IP
Host. Polycom software supports the standard SNMP version 1 and 2 traps,
which are taken from RFC 1215, convention for defining traps for use with
SNMP.
Note: The SNMP Trap Version parameters must be defined identically in the
external SNMP application.

Trap Destination

This box lists the currently defined IP addresses of the Manager terminals to
which the message (trap) is sent.

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IP

Enter the IP address of the SNMP trap
recipient.

All Versions

Community Name

Enter the Community Name of the
manager terminal used to monitor the
MCU activity

Version 1
and
Version 2

User Name

Enter the name of the user who is to have
access to the trap.

Version 3

Authentication
Protocol

Enter the authentication protocol: MD5 or
SHA.

Privacy Protocol

Enter the privacy protocol: DES or AES.

Engine ID

Enter an Engine ID to be used for both
the Agent and the Trap.
Default: Empty

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11 Click the Add button to add a new Manager terminal.
Depending on the SNMP Trap Version selected, one of the two following New Trap Destination
dialog boxes opens.

12 Define the following parameters:
SNMPv3 - Traps
Field

Description

Version

IP Address

Enter the IP address of the SNMP trap recipient.

1,2,3

Enable Trap
Inform

An Inform is a Trap that requires receipt confirmation from the entity
receiving the Trap. If the Engine ID field (Version 3) is empty when Enable
Trap Inform has been selected, the Engine ID is set by the Client.

Community Name

Enter the Community Name of the manager terminal used to monitor the
MCU activity

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SNMPv3 - Traps (Continued)
Field

Description

Version

User Name

Enter the name of the user who is to have access to the trap.

3

Engine ID

Enter an Engine ID to be used for the Trap.
This field is enabled when the Enable Trap Inform check box is selected. If
the Enable Trap Inform check box is cleared the Engine ID of the Agent is
used. The Engine ID is comprised of up to 64 Hexadecimal characters.
Default: Empty

Security Level

Select a Security Level from the drop-down menu.
Range: No Auth, No Priv; Auth, No Priv; Auth, Priv
Default: Auth, Priv

Authentication
Protocol

Enter the authentication protocol: MD5 or SHA.
The availability of the MD5 Authentication Protocol as a selectable option
is controlled by adding the SNMP_FIPS_MODE System Flag to
system.cfg and setting its value. A value of YES means that MD5 will
neither be displayed as selectable option nor supported.
Range: YES/NO.
Default: NO.

Authentication Password
Privacy Protocol

Enter the privacy protocol: DES or AES.
The availability of the DES Privacy Protocol as a selectable option is
controlled by adding the SNMP_FIPS_MODE System Flag to system.cfg
and setting its value. A value of YES means that DES will neither be
displayed as a selectable option nor supported.
Range: YES/NO.
Default: NO.

Privacy Password

13 Type the IP Address and the Community name of the manager terminal used to monitor the MCU
activity, and then click OK.
The Community name is a string of characters that will be added to the message that is sent to the
external Manager terminals. This string is used to identify the message source by the external
Manager terminal.
The new IP Address and Community name is added to the Trap Destinations box.
a To delete the IP Address of a Manager terminal, select the address that you wish to delete, and
then click the Remove button.
The IP address in the Trap Destinations box is removed.

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14 Click the Security tab.
The SNMP Properties – Security dialog box opens.

This dialog box is used to define whether the query sent to the MCU is sent from an authorized
source. When the “Accept SNMP packets from all Hosts” is disabled, a valid query must contain the
appropriate community string and must be sent from one of the Manager terminals whose IP address
is listed in this dialog box.

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15 Define the following parameters:
SNMP - Security
Field

Description

Send Authentication
Trap

Select this check box to send a message to the SNMP Manager
when an unauthorized query is sent to the MCU. When cleared, no
indication will be sent to the SNMP Manager.

Accept Host Community
Name

Enter the string added to queries that are sent from the SNMP
Manager to indicate that they were sent from an authorized source.
Note: Queries sent with different strings will be regarded as a
violation of security, and, if the Send Authentication Trap check box
is selected, an appropriate message will be sent to the SNMP
Manager.

Accept SNMP Packets
from all Host

Select this option if a query sent from any Manager terminal is valid.
When selected, the Accept SNMP Packets from These Hosts option
is disabled.

Accept SNMP Packets
from the following Hosts

Lists specific Manager terminals whose queries will be considered
as valid. This option is enabled when the Accept SNMP Packets
from any Host option is cleared.

User Name

Enter a User Name of up to 48 characters
Default: Empty

Security Level

Select a Security Level from the drop-down menu.
Range: No Auth, No Priv; Auth, No Priv; Auth, Priv
Default: Auth, Priv

Authentication Protocol

Select the authentication protocol
Range: MD5, SHA
Default: MD5

Authentication
Password

Enter an Authentication Password.
Range: 8 - 48 characters
Default: Empty

Privacy Protocol

Select a Privacy Protocol.
Range: DES, AES
Default: DES

Privacy Password

Enter a Privacy Password.
Range: 8 - 48 characters
Default: Empty

Engine ID

Enter an Engine ID to be used for
both the Agent and the Trap.
Default: Empty

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Versions
1&2

Version3

These fields are enabled if
Authentication is selected in
the Security Level field.

These fields are enabled if
Privacy is selected in the
Security Level field.

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16 To specifically define one or more valid terminals, ensure that the Accept SNMP Packets from any
Host option is cleared and then click the Add button.
The Accepted Host IP Address dialog box opens.

17 Enter the IP Address of the Manager terminal from which valid queries may be sent to the MCU, and
then click OK.
Click the Add button to define additional IP Addresses.
The IP Address or Addresses are displayed in the Accept SNMP Packets from These Hosts box.
Queries sent from terminals not listed in the Accept SNMP Packets from These Hosts box are
regarded as a violation of the MCU security, and if the Send Authentication Trap check box is
selected, an appropriate message will be sent to all the terminals listed in the SNMP Properties –
Traps dialog box.

18 In the SNMP Properties - Security dialog box, click OK.

19
Audible Alarms
In addition to the visual cues used to detect events occurring on the Collaboration Server, an audible alarm
can be activated and played when participants request Operator Assistance.

Using Audible Alarms
The Audible Alarm functionality for Operator Assistance requests is enabled for each MCU in either the
Collaboration Server Web Client or RMX Manager.
The Audible Alarm played when Operator Assistance is requested is enabled and selected in the Setup >
Audible Alarm > User Customization.
When the Audible Alarm is activated, the *.wav file selected in the User Customization is played, and it is
repeated according to the number of repetitions defined in the User Customization.
If more than one Collaboration Server is monitored in the RMX Manager, the Audible Alarm must be enabled
separately for each Collaboration Server installed in the site/configuration. A different *.wav file can be
selected for each MCU.
When multiple Audible Alarms are activated in different conferences or by multiple MCUs, the Audible
Alarms are synchronized and played one after the other. It is important to note that when Stop Repeating
Alarm is selected from the toolbar from the Collaboration Server Web Client or RMX Manager, all activated
Audible Alarms are immediately halted.

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Audible Alarm Permissions
An operator/administrator can configure the Request Operator Assistance audible alarm, however Users
with different authorization level have different configuration capabilities as shown in the following table.
Audible Alarm Permissions
Option
User Customization

Operator

Administrator





Download Audible Alarm File
Stop Repeating Alarms







Stop Repeating Message
The Collaboration Server User can stop playing the audible alarm at any time. If more than one audible
alarm has been activated, all activated alarms are immediately stopped.
If after stopping the Audible Alarms a new Operator Assistance request event occurs, the audible alarm is
re-activated.

To stop the Audible Alarm on the Collaboration Server Client or RMX Manager:
● On the Collaboration Server menu, click Setup > Audible Alarms >Stop Repeating Alarm.
When selected all audible alarms are immediately stopped.

Configuring the Audible Alarms
User Customization
The operators and administrators can:
● Enable/Disable the Audible Alarm.
● Select whether to repeat the Audible Alarm.
● Define the number of repetitions and the interval between the repetitions.

To Customize the Audio Alert on the Collaboration Server Client or RMX Manager:
1 On the Collaboration Server menu, click Setup > Audible Alarms > User Customization.
The User Customization window opens.

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2 Define the following parameters:
Audible Alarm - User Customization Options
Option

Description

Enable Audible Alarm

Select this check box to enable the Audible Alarm feature and to define its
properties.
When this check box is cleared, the Audible Alarm functionality is disabled.

Repeat Audible Alarm

Select this check box to play the Audible Alarm repeatedly. When selected, it
enables the definition of the number of repetitions and the interval between
repetitions.
When cleared, the Audible Alarm will not be repeated and will be played only
once.

Number of Repetitions

Define the number of times the audible alarm will be played.
Default number of repetitions is 4.

Repetition interval in
seconds

Define the number of seconds that the system will wait before playing the
Audible Alarm again.
Default interval is 20 seconds.

3 Click OK.

Replacing the Audible Alarm File
Each Collaboration Server is shipped with a default tone file in *.wav format that plays a specific tone when
participants request Operator Assistance. This file can be replaced by a *.wav file with your own recording.
The file must be in *.wav format and its length cannot exceed one hour.
Only the User with Administrator permission can download the Audible Alarm file.

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To replace the Audio file on the Collaboration Server Client or RMX Manager:
1 On the Collaboration Server menu, click Setup > Audible Alarms > Download Audible Alarm
File.
The Download Audible Alarm File window opens.
2 Click the Browse button to select the audio file (*.wav) to download.
The Open dialog box opens.

3 Select the appropriate *.wav file and then click the Open button.
The selected file name is displayed in the Install Audible Alarm File dialog box.
4 Optional. You can play the selected file or the currently used file by clicking the Play (
follows:

) button as

a Click Play Selected File to play a file saved on your computer.
b Click Play Collaboration Server File to play the file currently saved on the Collaboration Server.
5 In the Download Audible Alarm File dialog box, click OK to download the file to the MCU.
The new file replaces the file stored on the MCU. If multiple Collaboration Servers are configured in the RMX
Manager, the file must be downloaded to each of the required MCUs separately.

Multilingual Setting
Each supported language is represented by a country flag in the Welcome Screen and can be selected as
the language for the Collaboration Server Web Client.

Customizing the Multilingual Setting
The languages available for selection in the Login screen of the Collaboration Server Web Client can be
modified using the Multilingual Setting option.

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To customize the Multilingual Setting:
1 On the Collaboration Server menu, click Setup > Customize Display Settings > Multilingual
Setting.
The Multilingual Setting dialog box is displayed.

2 Click the check boxes of the languages to be available for selection.
3 Click OK.
4 Log out from the Collaboration Server Web Client and Log in for the customization to take effect.

Banner Display and Customization
The Login Screen and Main Screen of the Collaboration Server Web Client and the RMX Manager can
display informative or warning text banners. These banners can include general information or they can be
cautioning users to the terms and conditions under which they may log into and access the system, as
required in many secured environments.
Banner display is enabled in the Setup > Customize Display Settings > Banners Configuration.
The administrator can choose one of four alternative login banners to be displayed. The four alternative
banners cannot be modified. A Custom banner (default) can also be defined.
The Main Page Banner is blank and can be defined.

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The Banner Configuration dialog box allows the administrator to select a Login Banner from a drop-down
menu.

One of the following Login Banners can be selected:
● Non-Modifiable Banners
 Sample 1
 Sample 2
 Sample 3
 Sample 4
● Modifiable Banner
 Custom (Default)

Guidelines
● The Login Banner must be acknowledged before the user is permitted to log in to the system.
● If a Custom banner has been created, and the user selects one of the alternative, non-modifiable
banners the Custom banner not deleted.
● The Custom Login Banner may contain up to 1300 characters.
● An empty Login Banner is not allowed.
● Any attempt to modify a non-modifiable banner results in it automatically being copied to the Custom
banner.

Non-Modifiable Banner Text
Sample 1 Banner
You are accessing a U.S. Government (USG) Information System (IS) that is provided
for USG-authorized use only.
By using this IS (which includes any device attached to this IS), you consent to the
following conditions:

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− The USG routinely intercepts and monitors communications on this IS for purposes
including, but not limited to, penetration testing, COMSEC monitoring, network
operations and defense, personnel misconduct (PM), law enforcement (LE), and
counterintelligence (CI) investigations.
− At any time, the USG may inspect and seize data stored on this IS.
− Communications using, or data stored on, this IS are not private, are subject to
routine monitoring, interception, and search, and may be disclosed or used for any
USG authorized purpose.
− This IS includes security measures (e.g., authentication and access controls) to
protect USG interests--not for your personal benefit or privacy.
− Notwithstanding the above, using this IS does not constitute consent to PM, LE or
CI investigative searching or monitoring of the content of privileged communications,
or work product, related to personal representation or services by attorneys,
psychotherapists, or clergy, and their assistants. Such communications and work
product are private and confidential. See User Agreement for details.

Sample 2 Banner
This system is for the use of authorized users only. Individuals using this computer
system without authority, or in excess of their authority, are subject to having all
of their activities on this system monitored and recorded by systems personnel. In
the course of monitoring individuals improperly using this system, or in the course
of system maintenance, the activities of authorized users also may be monitored.
Anyone using this system expressly consents to such monitoring and is advised that if
such monitoring reveals possible criminal activity, system personnel may provide the
evidence of such monitoring to law enforcement officials.

Sample 3 Banner
You are about to access a system that is intended for authorized users only. You
should have no expectation of privacy in your use of this system. Use of this system
constitutes consent to monitoring, retrieval, and disclosure of any information
stored within the system for any purpose including criminal prosecution.

Sample 4 Banner
This computer system including all related equipment, network devices (specifically
including Internet access), is provided only for authorized use. All computer systems
may be monitored for all lawful purposes, including ensuring that their use is
authorized, for management of the system, to facilitate protection against
unauthorized access, and to verify security procedures, survivability and operational
security. Monitoring includes active attacks by authorized personnel and their
entities to test or verify the security of the system. During monitoring, information
may be examined, recorded, copied and used for authorized purposes. All information
including personal information, placed on or sent over this system may be monitored.
Use of this system, authorized or unauthorized, constitutes consent to monitoring of
this system. Unauthorized use may subject you to criminal prosecution. Evidence of
any such unauthorized use collected during monitoring may be used for administrative,
criminal or other adverse action. Use of this system constitutes consent to
monitoring for these purposes.

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Customizing Banners
The Login and Main Screen banners can be customized to display conference information, assistance
information or warning text.

To customize the banners:
1 In the Collaboration Server menu, click Setup > Customize Display Settings > Banners
Configuration.
The Banners Configuration dialog box opens.

2 Customize the banners by modifying the following fields:
Banner Configuration
Description
Field

Check Box

Login Page
Banner
Select or clear the
check box to enable or
disable the display of
the banner.

Text Field
Edit the text in this field to meet
local requirements:
• Banner content is multilingual
and uses Unicode, UTF-8
encoding. All text and special
characters can be used.
• Maximum banner size is
100KB.

Restore Default
Button

Click the button to
restore the default text
to the banner

Main Page
Banner

3 Click the OK button.

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Banner Display
Login Screen Banner
The Login screen banner can display any text, for example the terms and conditions for system usage. The
user must acknowledge that the information was read and click the Accept button to proceed to the Login
screen as shown in the following screen:

Main Screen Banner
The Main Screen banner is displayed at the bottom of the screen:

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Software Management
The Software Management menu is used to backup and restore the Collaboration Server's configuration
files and to download MCU software.

Backup and Restore Guidelines
● System Backup can only be performed by an administrator.
● The System Backup procedure creates a single backup file that can be viewed or modified only by
developers.
● A System Backup file from one system can be restored on another system.
This applies only to one Polycom® RealPresence® Collaboration Server 800s and Polycom®
RealPresence® Collaboration Server Virtual Edition system to another. Do not use a backup file from
the Polycom® RealPresence® Collaboration Server 800s and Polycom® RealPresence®
Collaboration Server Virtual Edition on any other model.

● To ensure file system consistency, do not perform any configuration changes as the system does not
suspended them during the backup procedure.
● The following parameters, settings and files are backed up:
 MCMS configuration files (/mcms/Cfg):
 Network and service configurations,
 Rooms,
 Profiles
 Reservations
 System Flags
 Resource Allocation
 IVR messages, music
 Collaboration Server Web Client user setting - fonts, windows
 Collaboration Server Web Client global settings – notes, address book, language
 Private keys and certificates (TLS)
 Conference participant settings
 Operation DB (administrator list)
 SNMP settings
 Time configuration
● CDR files are not included in the backup process and should be backed up manually by saving the
CDR files to a destination device.

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Using Software Management
To backup configuration files:
1 On the Collaboration Server menu, click Administration > Software Management > Backup
Configuration.
The Backup Configuration dialog box opens.

2 Click the Browse button.
The Browse To File dialog box opens.
3 Select the Backup Directory Path and then click Backup.
When the Collaboration Server system backs up the current configuration, if any changes occur
immediately or during the request, then additional changes are not registered.

To restore configuration files:
1 On the Collaboration Server menu, click Administration > Software Management > Restore
Configuration.
2 Browse to the Restore Directory Path where the backed up configuration files are stored and then
click Restore.

To download MCU software files:
1 On the Collaboration Server menu, click Administration > Software Management > Software
Download.
2 Browse to the Install Path and then click Install.

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Ping the Collaboration Server
The Ping administration tool enables the Collaboration Server Signaling Host to test network connectivity
by Pinging IP addresses.

Guidelines
● The IP addressing mode can be either IPv4 or IPv6.
● Both explicit IP addresses and Host Names are supported.
● The Collaboration Server Web Client blocks any attempt to issue another Ping command before the
current Ping command has completed. Multiple Ping commands issued simultaneously from multiple
Collaboration Server Web Clients are also blocked.

Using Ping
To Ping a network entity from the Collaboration Server:
1 On the Collaboration Server menu, click Administration > Tools > Ping.
The Ping dialog box is displayed:

2 Modify or complete the following fields:
Ping Parameters
Field

Description

Host Name or Address

Enter the Host Name or IP Address of the network entity to be Pinged.

3 Click the Ping button.
The Ping request is sent to the Host Name or IP Address of the Collaboration Server entity.
The Answer is either:
 OK, or
 FAILED

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Notification Settings
The Collaboration Server can display notifications when:
● A new Collaboration Server user connects to the MCU.
● A new conference is started.
● Not all defined participants are connected to the conference or when a single participant is connected.
● A change in the MCU status occurs and an alarm is added to the alarms list.
A welcome message is displayed to the Collaboration Server user upon connection.

To configure the notifications:
1 On the Collaboration Server menu, select Setup > Notification Settings.
The Notification Settings dialog box is displayed.

The following notification options are displayed.
Notification Settings Parameters
Field

Description

New Connection

Notification of a new user/administrator connecting to the Collaboration Server.

New Conference Created

New conference has been created.

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Notification Settings Parameters (Continued)
Field

Description

Conference Not Full

The conference is not full and additional participants are defined for the
conference.

Welcome Message

A welcome message after user/administrator logon.

Active Alarms Update

Updates you of any new alarm that occurred.

Fault List Updated

Updates you when the faults list is updated (new faults are added or existing
faults are removed).

2 Enable/Disable All Notifications or Custom to select specific notifications to display.
3 Click OK.

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Logger Diagnostic Files
The Logger utility is a troubleshooting tool that continually records MCU system messages and saves them
to files in the MCU hard drive. For each time interval defined in the system, a different data file is created.
The files may be retrieved from the hard drive for off-line analysis and debugging purposes.
The Logger utility is activated at the MCU startup. The Logger is disabled when the MCU is reset manually
or when there is a problem with the Logger utility, e.g. errors on the hard drive where files are saved. In such
cases, data cannot be retrieved.
When the MCU is reset via the Collaboration Server, the files are saved on the MCU hard drive.

To access the Logger Diagnostic Files:
● On the Collaboration Server menu, click Administration > Tools > Logger Diagnostic Files.

The following tasks can be performed:
Diagnostic File Button Options
Button

Description

Refresh List

Refreshes the list and adds newly generated logger files.

Select All

Selects all the logger files listed.

Browse

Selects the destination folder for download.

Retrieve Files

Saves files to the destination folder.

When retrieved, the log file name structure is as follows:
● Sequence number (starting with 1)

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● Date and Time of first message
● Date and Time of last message
● File size
● Special information about the data, such as Startup
File name structure:
Log_SNxxxxxxxxxx_FMDddmmyyy_FMThhmm_LMDddmmyyyy_LMThhmm_SZxxxxxxxxxx_SUY.log
File name format:
● SN = Sequence Number
● FM = First Message, date and time
● LM = Last Message, date and time
● SZ = Size
● SU = Startup (Y/N) during the log file duration
Example:
Log_SN0000000002_FMD06032007_FMT083933_LMD06032007_LMT084356_SZ184951_SUY.log.

To Retrieve the Logger Files:
1 Select the log files to retrieve. Multiple selections of files are enabled using standard Windows
conventions.
2 In the Logger Diagnostic Files dialog box, click the Browse button.
3 In the Browse for Folder window, select the directory location to save the Logger files and click OK.
You will return to the Logger Diagnostic Files dialog box.

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4 Click the Retrieve Files button.

The log files (in *.txt format) are saved to the defined directory and a confirmation caption box is
displayed indicating a successful retrieval of the log files.

Viewing the Logger Files:
To analyze the log files generated by the system, open the retrieved *.txt files in any text editor application,
i.e. Notepad, Textpad or MS Word.
1 Using Windows Explorer, browse to the directory containing the retrieved log files.
2 Use any text editor application to open the log file(s).

Information Collector
Standard Security Mode
The Information Collector comprehensively attains all information from all the MCU internal entities for data
analysis. That data, stored in a central repository, is logged from the following system components:
•

System Log Files

•

Full faults

•

CDR

•

Apache logs

•

OS (Core dumps, CFG - DNS, DHCP, NTP, kernal state,
event logs

•

CFG directory (without IVR)

•

Signaling Trace files (H.323 & SIP)

•

Cards info: HW version, state and status

•

Central Signaling logs

•

SW version number

•

Processes internal state and statistics

•

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The data collected is saved into a single compressed file containing all the information from each system
component in its relative format (.txt, .xml, etc...). In case the disk is malfunctioning, the file will be written
to the RAM (involves only a small amount of information where the RAM size is 1/2 a gigabyte). The zipped
file (info.tgz) can be opened with the following applications: WinRAR and WinZip. The entire zipped file is
then sent to Polycom’s Network Systems Division for analysis and troubleshooting.

Using the Information Collector
When the Information Collector is used the following steps are performed:
● Step 1: Creating the Information Collector file.
● Step 2: Saving the Information Collector file.
● Step 3: Viewing the information in the Information Collector file.

Step 1: Creating the Information Collector Compressed File
To create the compressed file:
1 In the Collaboration Server menu, click Administration > Tools > Information Collector.
The Information Collector dialog box is displayed.

2 In the From Date and Until Date fields, use the arrow keys to define the date range of the data files
to be included in the compressed file.

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3 In the From Time and Until Time fields, use the arrow keys to define the time range of the data files
to be included in the compressed file.
If logs are being collected in order to troubleshoot a specific issue, it is important that the date and
time range include the time and date in which the issue occurred. The default date and time ranges
may not be sufficient.
For example, if a specific issue occurred on October 1, 2013 at 12:15, the From Date and Until Date
should be October 1, 2013, the From Time should be around 12:10, and the Until Time should be
around 12:20.

4 Select check boxes of the information to be collected.
5 In the Export Path field, click the Browse button and navigate to the directory path where the
compressed file is to be saved.
6 Click the Collect Information button.
A progress indicator is displayed in the Information Collector dialog box while the file is being
created.

Step 2: Saving the Compressed File
1 The compressed file is automatically saved in the directory selected in the Information Collector
dialog box. The file is named info.tgz.
A success information box is displayed.
2 Click the OK button.

Step 3: Viewing the Compressed File
The compressed file is saved in .tgz format and can be viewed with any utility that can open files of that
format, for example WinRAR® 3.80.

To view the compressed file:
1 Navigate to the directory on the workstation in which the file was saved.
2 Double click the info.tgz file to view the downloaded information.
Some browsers save the file as info.gz due to a browser bug. If this occurs, the file must be
manually renamed to info.tgz before it can be viewed.

Auditor
An Auditor is a user who can view Auditor and CDR files for system auditing purposes.
The Auditor user must connect to the Collaboration Server using the Collaboration Server Web
Client only.

The Event Auditor enables administrators and auditors to analyze configuration changes and unusual or
malicious activities in the Collaboration Server system.

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Auditor operates in real time, recording all administration activities and login attempts from the following
Collaboration Server modules:
● Control Unit
● Shelf Manager
For a full list of monitored activities, see Audit Events.
The Auditor must always be active in the system. A System Alert is displayed if it becomes inactive for any
reason.
The Auditor tool is composed of the Auditor Files and the Auditor File Viewer that enables you to view the
Auditor Files.
Time stamps of Audit Events are GMT.

Auditor Files
All audit events are saved to a buffer file on hard disk in real time and then written to a file on hard disk in
XML in an uncompressed format.
A new current auditor event file is created when:
● the system is started
● the size of the current auditor event file exceeds 2 MB
● the current auditor event file’s age exceeds 24 hours
Up to 1000 auditor event files are stored per Collaboration Server. These files are retained for at least one
year and require 1.05 GB of disk space. The files are automatically deleted by the system (oldest first) when
the system reaches the auditor event file limit of 1000.
A System Alert is displayed with Can't store data displayed in its Description field if:
● the system cannot store 1000 files
● the Collaboration Server does not have available disk space to retain files for one year
Audit Event Files are retained by the Collaboration Server for at least 1 year. Any attempt to delete an audit
event file that is less than one year old raises a System Alert with File was removed listed in the Description
field.
Using the Restore Factory Defaults of the System Restore procedure erases Audit Files.

Retrieving Auditor Files
You can open the Auditor file directly from the Auditor Files list or you can retrieve the files and save them
to a local workstation.

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To access Auditor Files:
● On the Collaboration Server menu, click Administration > Tools > Auditor Files.
The Auditor Files dialog box is displayed.

The Auditor Files dialogue box displays a file list containing the following file information:
 Name
 Size (Bytes)
 First Message – date and time of the first audit event in the file
 Last Message – date and time of the last audit event in the file
 StartUp:
 True – file was created when the system was started
 False – file was created when previous audit event file reached a size of 2 MB or was more
than 24 hours old
 File Retrieved:
 True - file was previously retrieved.
 False - file was never previously retrieved.
The order of the Auditor Files dialog box field header columns can be changed and the fields can be
filtered to enable searching.
For more information, see Auditor File Viewer.

To retrieve files for storage on a workstation:
1 Click Browse and select the folder on the workstation to receive the files and then click OK.
The folder name is displayed in the directory path field.

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2 Select the file(s) to be retrieved by clicking their names in the file list or click Select All to retrieve all
the files. (Windows multiple selection techniques can be used.)
3 Click Retrieve Files.
The selected files are copied to the selected directory on the workstation.

To open the file in the Auditor File Viewer:
● Double-click the file.

Auditor File Viewer
The Auditor File Viewer enables Auditors and Administrators to view the content of and perform detailed
analysis on auditor event data in a selected Auditor Event File.
You can view an Auditor Event File directly from the Auditor Files list or by opening the file from the Auditor
File Viewer.

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To open the Auditor File Viewer from the Administration Menu:
1 On the Collaboration Server menu, click Administration > Tools > Auditor File Viewer.
The Auditor File Viewer is displayed.
If you previously double clicked an Auditor Event File in the Auditor Files list, that file is automatically
opened.

The following fields are displayed for each event:
Auditor Event Columns
Field

Description

Event ID

The sequence number of the event generated by the Collaboration Server.

Date & Time

The date and time of the event taken from the Collaboration Server’s Local Time
setting.

User Name

The Username (Login Name) of the user who triggered the event.

Reporting Module

The Collaboration Server system internal module that reported the event:
• MCMS
• MPL
• Central Signaling
• MPL Simulation
• Collaboration Server Web Client
• CM Switch
• Video
• MUX

Workstation

The name (alias) of the workstation used to send the request that triggered the
event.

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Auditor Event Columns (Continued)
Field

Description

IP Address (Workstation)

The IP address of the workstation used to send the request that triggered the
event.

Event Type

Auditor events can be triggered by:
• API
• HTTP
• Collaboration Server Internal Event

Event

The process, action, request or transaction that was performed or rejected.
• POST:SET transactions (API)
• Configuration changes via XML (API)
• Login/Logout (API)
• GET (HTTP)
• PUT (HTTP)
• MKDIR (HTTP)
• RMDIR (HTTP)
• Startup (Collaboration Server Internal Event)
• Shutdown (Collaboration Server Internal Event)
• Reset (Collaboration Server Internal Event)
• Enter Diagnostic Mode (Collaboration Server Internal Event)
• IP address changes via USB (Collaboration Server Internal Event)

Process Completed

Status of the process, action, request or transaction returned by the system:
• Yes – performed by the system.
• No – rejected by the system.

Description

A text string describing the process, action, request or transaction.

Additional Information

An optional text string describing the process, action, request or transaction in
additional detail.

The order of the Auditor File Viewer field header columns can be changed and the fields can be
sorted and filtered to facilitate different analysis methods.
2 In the event list, click the events or use the keyboard’s Up-arrow and Down-arrow keys to display
the Request Transaction and Response Transaction XML trees for each audit event.
The transaction XML trees can be expanded and collapsed by clicking the expand
( ) and collapse ( ) buttons.

To open an auditor event file stored on the workstation:
1 Click the Local File button (

) to open the Open dialogue box.

2 Navigate to the folder on the workstation that contains the audit event file.
3 Select the audit event file to be opened.
4 Click Open.
The selected file is opened in the Auditor Viewer.
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Audit Events
Alerts and Faults
Alerts and Faults that are recorded by the Auditor.
Alerts and Faults recorded by the Auditor
Event
Attempt to exceed the maximum number of management session per user
Attempt to exceed the maximum number of management sessions per system
Central Signaling indicating Recovery status.
Failed login attempt
Failed to open Apache server configuration file.
Failed to save Apache server configuration file.
Fallback version is being used.
File system scan failure.
File system space shortage.
Internal MCU reset.
Internal System configuration during startup.
Invalid date and time.
Invalid MCU Version.
IP addresses of Signaling Host and Control Unit are the same.
IP Network Service configuration modified.
IP Network Service deleted.
Login
Logout
Management Session Time Out
MCU Reset to enable Diagnostics mode.
MCU reset.
Music file error.
New activation key was loaded.
New version was installed.
NTP synchronization failure.

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Alerts and Faults recorded by the Auditor (Continued)
Event
Polycom default User exists.
Private version is loaded.
Restoring Factory Defaults.
Secured SIP communication failed.
Session disconnected without logout
SSH is enabled.
System Configuration modified.
System is starting.
System Resets.
TCP disconnection
Terminal initiated MCU reset.
The Log file system is disabled.
The software contains patch(es).
USB key used to change system configuration.
User closed the browser
User initiated MCU reset.

Transactions
Transactions that are recorded by the Auditor.
Transactions recorded by the Auditor
Transaction
TRANS_CFG:SET_CFG
TRANS_IP_SERVICE:DEL_IP_SERVICE
TRANS_IP_SERVICE:NEW_IP_SERVICE
TRANS_IP_SERVICE:SET_DEFAULT_H323_SERVICE
TRANS_IP_SERVICE:SET_DEFAULT_SIP_SERVICE
TRANS_IP_SERVICE:UPDATE_IP_SERVICE
TRANS_IP_SERVICE:UPDATE_MANAGEMENT_NETWORK
TRANS_MCU:BEGIN_RECEIVING_VERSION
TRANS_MCU:COLLECT_INFO
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Transactions recorded by the Auditor (Continued)
Transaction
TRANS_MCU:CREATE_DIRECTORY
TRANS_MCU:FINISHED_TRANSFER_VERSION
TRANS_MCU:LOGIN
TRANS_MCU:LOGOUT
TRANS_MCU:REMOVE_DIRECTORY
TRANS_MCU:REMOVE_DIRECTORY_CONTENT
TRANS_MCU:RENAME
TRANS_MCU:RESET
TRANS_MCU:SET_PORT_CONFIGURATION
TRANS_MCU:SET_RESTORE_TYPE
TRANS_MCU:SET_TIME
TRANS_MCU:TURN_SSH
TRANS_MCU:UPDATE_KEY_CODE
TRANS_OPERATOR:CHANGE_PASSWORD
TRANS_OPERATOR:DELETE_OPERATOR
TRANS_OPERATOR:NEW_OPERATOR
TRANS_SNMP:UPDATE

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ActiveX Bypass
At sites that, for security reasons, do not permit Microsoft® ActiveX® to be installed, the MSI (Windows
Installer File) utility can be used to install .NET Framework and .NET Security Settings components on
workstations throughout the network.
All workstation that connect to Collaboration Server systems must have both.NET Framework and .NET
Security Settings running locally. These components are used for communication with the Collaboration
Server and can only be installed on workstations by users with administrator privileges.
The MSI utility requires the IP addresses of all the Collaboration Server systems (both control unit and Shelf
Management IP addresses) that each workstation is to connect to.
If the IP address of the any of the target Collaboration Servers is changed, the ActiveX components must
be reinstalled.

Installing ActiveX
To install ActiveX components on all workstations in the network:
1 Download the MSI file EMA.ClassLoaderInstaller.msi from the Polycom Resource Center.
The MSI file contains installation scripts for both .NET Framework and .NET Security Settings.
2 Create a text file to be used during the installation containing the IP addresses of all the
Collaboration Server systems (both control unit and Shelf Management IP addresses) that each
workstation in the network is to connect to.
The file must be named url_list.txt and must be saved in the same folder as the downloaded MSI file.

3 Install the ActiveX components on all workstations on the network that connect to Collaboration
Server systems.
The installation is done by the network administrator using a 3rd party network software installation
utility and is transparent to all other users.

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Resetting the Collaboration Server 800s
These instructions are applicable to the RealPresence Collaboration Server 800s only.

System Reset saves system configuration changes and restarts the system with the latest settings.

To reset the RMX:
1 In the RMX Management pane, click the Hardware Monitor button.
The Hardware Monitor pane is displayed.
2 Click the Reset (

) button.

When the Collaboration Server system is reset, during Collaboration Server startup the Progress Bar
appears at the bottom of the Collaboration Server Status pane, displaying the amount of time
remaining for the reset process to complete:

The Startup progress is also indicated by a green bar moving from left to right.
The duration of the Startup depends on the type of activity that preceded the MCU reset. For
example: Fast Configuration Wizard, New Version installation, Version Upgrade, Restore Last
Configuration etc.

Resetting the RealPresence Collaboration Server
Virtual Edition
To restart the MCU instance:
1 Click Start > Programs.
a If the VMware vSphere Client is displayed in the recently used programs list, click VMware
vSphere Client in the list to start the application.
or
b Click All Programs > VMware > VMware vSphere Client.

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The VMware vSphere Client login window opens.

2 In the IP address / Name field, enter the IP Address or the name of the vSphere host.
3 Enter the User Name and password by either:
a In the User name field, enter the user name of the with which you will log in to the vSphere host.
In the Password field, enter the password as defined for the user name with which you will log in
to the vSphere host.
or
b Click the Use Windows session credentials check box.

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4 Click Login.
The VMware vSphere Client opens.

5 In the Inventory Panel, click the Datastore that houses the MCU.

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6 Right-click on the MCU virtual machine, then click Power > Restart Guest

The virtual machine and the MCU instance restart.

DO NOT click Reset. Doing so may corrupt the Virtual Machine.

Upgrading and Downgrading
This procedure allows an administrator to update the MCU instance without requiring the administrator to
reregister the product.
Updating the MCU instance requires the previously used activation key. If you no longer have the
activation key, contact support before starting this procedure.

See
http://kb.vmware.com/selfservice/microsites/search.do?language=en_US&cmd=displayKC&externa
lId=1714 before proceeding.

To update the MCU instance:

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1 On the RealPresence Collaboration Server menu, click Administration > Software Management >
Backup Configuration.
The Backup Configuration dialog box opens.

2 Click the Browse button.
The Browse To File dialog box opens.
3 Select the Backup Directory Path and then click Backup.
When the RealPresence Collaboration Server backs up the current configuration, if any changes
occur immediately or during the request, then additional changes are not registered.

4 On the Windows taskbar, click the Start > Programs.
a If the VMware vSphere Client is displayed in the recently used programs list, click VMware
vSphere Client in the list to start the application.
or
b Select All Programs > VMware > VMware vSphere Client.

The VMware vSphere Client login window is displayed.

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5 In the IP address / Name field, enter the IP Address or the name of the vSphere host.
6 Either type your vSphere User Name and Password or select Use Windows sessions
credentials.
7 Click Login.
The VMware vSphere Client is displayed.

8 In the Inventory Panel, select the Datastore that houses the MCU.
The inventory of the Datastore appears.

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9 Right-click the MCU virtual machine, then click Power > Shut Down Guest.

10 When VM turns blue, the virtual machine has shut down.

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11 When the MCU has shut down, click the Summary tab.
12 Under Resources, right-click the datastore, and click Browse Datastore.

The Browse Datastore window appears.

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13 In the Folders tab, select the folder whose name matches that of the MCU.
If the same name has been used multiple times, there will be multiple folders with an underscore
and a number appended to the name. In such a case, select the folder with the name of the MCU
which ends with the highest number.

The contents of the folder are displayed.
14 Right-click the file ending with “.vmx” and click Download.

The Browse For Folder window appears.
15 Browse to a location and click OK.

The Upload/Download Operation Warning window may appear.

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16 If the Upload/Download Operation Warning window appears, click Yes.

If it does not appear, proceed to step 17.
The file downloads.
17 Open the file in any plain text editor.

18 Locate the line that starts with uuid.bios.
19 Copy the entire line and paste it into another text file.
20 Save the text file.
21 In the Inventory Panel, click the Datastore that houses the MCU.

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22 Right-click the MCU, and select Delete from Disk.

The Confirm Delete window appears.

23 Click Yes.
24 On the vSphere Client menu, select File > Deploy OVF Template.

The Deploy OVF Template wizard opens to the Source page.
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25 Click Browse.

The Open dialog box appears.
26 Browse to the new OVA file.

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27 Either double-click on the OVA file or click on the file, then click Open.

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28 Click Next.
The OVF Template Details page is displayed.

29 Click Next.
The Name and Location page is displayed.

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30 In the Name field, type the same name previously used for the MCU.

31 Click Next.
The Disk Format page is displayed.

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32 Select Thin Provision, then click Next.

The Network Mapping page is displayed.

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33 Select the appropriate network mappings, then click Next.

The Ready to Complete page is displayed.

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34 Verify that Power on after deployment is not selected.
35 Confirm that all the settings are correct, then click Finish.
The vSphere Client deploys the OVF file.

When the deployment is complete the following window appears:

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36 Click Close.
37 In the Inventory Panel, select the Datastore that used to house the MCU.
38 Click the Summary tab.
39 Under Resources, right-click the datastore, and click Browse Datastore.

The Browse Datastore window appears.

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40 In the Folders tab, select the folder whose name matches that of the MCU. The folder name will
have an underscore and a number at the end.
The contents of the folder are displayed.
41 Right-click the file ending with “.vmx” and click Download.

The Browse For Folder window appears.

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42 Browse to a location and click OK.
The Upload/Download Operation Warning window may appear.

43 If the Upload/Download Operation Warning window appears, click Yes. If it does not appear,
proceed to step 44.
The file downloads.
44 Open the file created in step 19.
45 Open the file in any plain text editor.

46 Locate the line that starts with, uuid.bios.
47 Replace that line with the line saved in the other text file.
48 Add the following as a separate line to the file, including the quotation marks:
uuid.action = “keep”

49 Save and close the file.
50 Optional. To back up the previous configuration:
a In the Datastore Browser window, right-click the .vmx file, then select Rename.

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b Change the file extension to “.bak”.
51 In the tool bar of the Datastore Browser window, click the Upload files to this datastore button.

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52 Click Upload File.
The Upload Items window appears.

53 Navigate to where you saved the “.vmx” file in step 49, select it, then click Open.
The Upload/Download Operation Warning window may appear.

54 If the Upload/Download Operation Warning window appears, click Yes.
The file is uploaded.
55 Close the Datastore Browser.
56 In the Inventory Panel, click the Datastore that houses the MCU.

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57 Right-click the MCU virtual machine, then click Power > Power On.

After a few minutes, the MCU turns on.
58 Start the Collaboration Server Web Client application on the workstation.
a In the browser’s address line, enter the IP address of the Control Unit in the format:
http://:8080.
b Click Enter.
The Collaboration Server Web Client Login screen is displayed.

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59 In the Collaboration Server Web Client Login screen, enter the default Username (POLYCOM) and
Password (POLYCOM) and click Login.
The Collaboration Server Web Client opens and the Product Activation dialog box appears with the
serial number filled in:

60 In the Activation Key field, enter or paste the Product Activation Key that was used on the previous
MCU.
61 Click OK.
A message indicating that the Product Activation Key was loaded successfully appears.
If the Product Activation Key fails to load, please contact your vendor.
62 Click OK.
If the Product Activation dialog box does not appear, go to Setup --> Product Activation to display
the dialog box.

63 On the RealPresence Collaboration Server menu, click Administration > Software Management >
Restore Configuration.
64 Browse to the Restore Directory Path where the backed up configuration files are stored and then
click Restore.

Upgrading or Downgrading the Polycom® RealPresence®
Collaboration Server 800s and Polycom® RealPresence®
Collaboration Server Virtual Edition
These instructions are applicable to the RealPresence Collaboration Server 800s only.

To upgrade or downgrade the RealPresence Collaboration Server 800s:
1 Inset the Polycom USB key that came with the RealPresence Collaboration Server 800s into your
computer.
The Polycom Documentation window is displayed.
In Windows XP:
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a The Polycom Documentation option is automatically selected. Click OK.
In Windows 7:
b Select Open Folder to view files using Windows Explorer.
c Double-click the index.hta file.
The Language Menu is displayed, offering a choice of several languages.

2 Click the documentation language of your choice.
An End-User Licence Agreement for Polycom Software is displayed.

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3 Read the agreement and click the Accept Agreement button.
4 In the Product Type dialog box, select RealPresence Collaboration Server 800s.

5 In the Initial Setup Utility dialog box, click the Upgrade / Downgrade link.

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The Collaboration Server Software Version Alignment Application dialog box is displayed.

6 Click Open.

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7 Navigate to the folder where the upgrade or downgrade software you have downloaded is saved and
click Open.

The software file is displayed in the Select the path where the downloaded version is located box.

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8 Click Add.
This step may take a few minutes.
The version is added to the Selected Version dropdown box.

9 Make sure the correct version is selected in the Selected Version dropdown box and click OK.
10 Remove the USB key from the PC workstation.
11 Insert the USB key in any USB port of the RealPresence Collaboration Server 800s.
12 Turn off the Collaboration Server, then turn it On.
This step may take up to ten minutes.
13 Start the Collaboration Server Web Client application on the workstation.
a In the browser’s address line, enter the IP address of the Control Unit in the following format:
http://.

b Click Enter.
When the Collaboration Server Web Client Login window is displayed, the version change was
successful.

2

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System Configuration Flags
The system’s overall behavior can be configured by modifying the default values of the System Flags.
For flag changes (including deletion) to take effect, the MCU must be reset.
For more information see Resetting the Collaboration Server (RMX).
The following System Flags do not require an MCU reset:
• IVR_MESSAGE_VOLUME
• IVR_MUSIC_VOLUME
• IVR_ROLL_CALL_VOLUME
• ENABLE_SELECTIVE_MIXING

Modifying System Flags
To modify system flags:
1 On the Collaboration Server menu, click Setup > System Configuration.
The System Flags dialog box opens.

2 In the MCMS_PARAMETERS_USER tab, the flags listed in the MCMS_PARAMETERS_USER Flags
table can be modified.
3 To modify a flag value, double-click or select the flag and click the Edit Flag button.

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4 In the New Value field, enter the flag’s new value.

5 Click OK to close the Update Flag dialog box.
6 Repeat steps 2–4 to modify additional flags.
7 Click OK to close the System Flags dialog box.
For flag changes (including deletion) to take effect, reset the MCU. For more information see
Resetting the Collaboration Server (RMX).

MCMS_PARAMETERS_USER Flags
Flag

Description

ALLOW_NON_ENCRYPT_PART
Y_IN_ENCRYPT_CONF

If YES, allows non-encrypted participants to connect to encrypted
conferences.
Default: No

CHANGE_AD_HOC_CONF_DUR
ATION

The duration of an ad-hoc conference* can be configured on a system
level by setting the flag to one of the following values (in minutes): 60
(default), 90, 180 and 270.
* An ad-hoc conference is automatically created when the participant
dials into an Ad-hoc Entry Queue and enters a conference ID that is
not being used by any other conferencing entity. It is based on the
Conference Profile assigned to the EQ.

CONTENT_SLAVE_LINKS_INTR
A_SUPPRESSION_IN_SECOND
S

Defines the interval, in seconds, during which the Collaboration Server
is allowed to forward an Intra Request received from any of the Slave
Cascading Links. The Slave Cascading Link can be connected to the
local Collaboration Server, to an MCU on a higher cascade level or to
the Content sharer.
The first Intra request that is received from any of the Slave MCUs
connected to the Collaboration Server starts the interval counter and is
forwarded to the next level MCU or to the Content sharer.
All other Intra requests that are received within this interval are
registered but ignored. After an interval of  seconds, the
system checks if during the last interval any additional Intra requests
were registered. If there is at least one Intra request it will be
forwarded. If there is no additional Intra request not no action is taken
other than to wait for the next cycle.
This filtering process is repeated every  seconds.
Default: 30

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System Configuration Flags

Flag

Description

CONTENT_SPEAKER_INTRA_S
UPPRESSION_IN_SECONDS

This flag controls the requests to refresh (intra) the content sent from
the Collaboration Server system to the content sender as a result of
refresh requests initiated by other conference participants.
Enter the interval in seconds between the Intra requests sent from the
Collaboration Server to the endpoint sending the content to refresh the
content display. Refresh requests that will be received from endpoints
within the defined interval will be postponed to the next interval.
Default setting: 5

CPU_TCP_KEEP_ALIVE_TIME_
SECONDS

This flag indicates when to send the first KeepAlive indication to check
the TCP connection.
Default value: 7200 second (120 minutes)
Range: 600-18000 seconds
When there are NAT problems, this default may be too long and the
TCP connection is lost. In such a case, the default value should be
changed to 3600 seconds (60 minutes) or less.

CPU_TCP_KEEP_INTERVAL_SE
CONDS

This flag indicates the interval in seconds between the KeepAlive
requests.
Default value: 75 second
Range: 10-720 seconds.

DISABLE_INACTIVE_USER

Users can be automatically disabled by the system when they do not
log into the Collaboration Server application for a predefined period.
Possible Values: 0 - 90 days.
Default: 0 (disables this option).

ENABLE_ACCEPTING_ICMP_RE
DIRECT

When set to YES, allows the RMX to accept ICMP Redirect Messages
(ICMP message type #5).
Possible values: YES / NO
• Default: YES

ENABLE_AGC

Set this flag to YES to enable the AGC option. (Default setting is NO.)
When disabled, selecting the AGC option in the Participant Properties
has not effect on the participant audio. For more information see
Managing the Address Book.
The Auto Gain Control mechanism regulates noise and audio volume
by keeping the received audio signals of all participants balanced.
Note:
Enabling AGC may result in amplification of background noise.

ENABLE_CASCADED_LINK_TO_
JOIN_WITHOUT_PASSWORD

Enables a cascaded link to enter a conference without a password.
Default: NO, for security reasons.

ENABLE_CYCLIC_FILE_SYSTE
M_ALARMS

Enables or disables the display of Active Alarms before overwriting the
older CDR/Auditor/Log files, enabling the users to backup the older
files before they are deleted.
Default: NO

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System Configuration Flags

Flag

Description

ENFORCE_SAFE_UPGRADE

Applicable to the RealPresence Collaboration Server 800s only.
When set to YES this flag enables the Collaboration Server system to
notify users when an incorrect version upgrade/downgrade or
upgrade/downgrade path is selected.
When set to NO, after initiating an upgrade or downgrade software
installation, the Collaboration Server activates a fault alert in the Faults
List: “Warning: Upgrade started and SAFE Upgrade protection is
turned OFF” and the upgrade/downgrade process continues.
Range: YES / NO
Default: YES

ENABLE_SENDING_ICMP_DEST
INATION_UNREACHABLE

Not supported with Polycom® RealPresence® Collaboration Server
800s and Polycom® RealPresence® Collaboration Server Virtual
Edition.

EXT_DB_IVR_PROV_TIME_SEC
ONDS

When an Entry Queue is set as IVR Service Provider for the
RealPresence DMA system, the value here indicates the time interval
in seconds in which the database is accesses for the ID.
Default: 300

FORCE_CIF_PORT_ALLOCATIO
N

Sets the MCU to allocate one CIF video resource to an endpoint,
regardless of the resolution determined by the Conference Profile
parameters. You can specify the endpoint types for which resource
allocation can be forced to CIF resource, enabling other types of
endpoints to use higher resolutions in the same conference.
Enter the product type to which the CIF resource should be allocated.
Possible values are:
• CMA Desktop - for CMA desktop client
• VSX nnnn - where nnnn represents the model number for example,
VSX 8000.

FORCE_STRONG_PASSWORD_
POLICY

When set to YES , implements the Strong Password rules. For more
details, Changing a User’s Password.
Default: NO

FORCE_SYSTEM_BROADCAST
_VOLUME

If set to YES, the level of broadcasting volume of the connected
participant is value taken from the system flag
SYSTEM_BROADCAST_VOLUME.
If set to NO (default), the broadcasting volume level is 5.

FORCE_SYSTEM_LISTENING_V
OLUME

If set to YES, the level of listening volume of the connected participant
is value taken from the system flag SYSTEM_LISTENING_VOLUME.
If set to NO (default), the listening volume level is 5.

GK_MANDATORY_FOR_CALLS_
IN

If set to YES, a gatekeeper is required to receive incoming H.323 calls.
If a gatekeeper is not configure in the Collaboration Server, the calls
will fail.
If set to NO (default), gatekeeper is not required to process H.323
incoming calls and H.323 participants can dial in with or without a
gatekeeper.

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System Configuration Flags

Flag

Description

GK_MANDATORY_FOR_CALLS_
OUT

If set to YES, a gatekeeper is required to perform H.323 outgoing calls.
If a gatekeeper is not configure on the Collaboration Server, the calls
will fail.
If set to NO (default), gatekeeper is not required to dial out to H.323
participants and calls can be dialed out with or without a gatekeeper.

H263_ANNEX_T

Set to NO to send the content stream without Annex T and enable
Aethra and Tandberg endpoints, that do not support Annex T, to
process the content.
Default: YES

HD_THRESHOLD_BITRATE

Sets the minimum bit rate required by endpoints to connect to an HD
Conference. Endpoints that cannot support this bit rate are connected
as audio only.
Range: 384kbps - 4Mbs (Default: 768)

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System Configuration Flags

Flag

Description

ITP_CROPPING

If the conference is set to TelePresence mode, cropping of the image
is done according to this flag value:
• ITP (default) - Cropping is done as follows:
 Left/right sides: no cropping
 Top/Bottom: the calculated area to be stripped will be split and
cropped equally from the top and the bottom of the display area.

•

CP - Cropping is done as follows:
 Left/right sides: the calculated area to be stripped will be split
and cropped equally from the top and bottom of the image
 Top/Bottom: the calculated area to be stripped will be split and
cropped equally from both sides.

•

MIXED - Cropping is done as follows:
 Left/right sides: the calculated area to be stripped will be split
and cropped equally from the top and bottom of the image
 Top/Bottom: the calculated area to be stripped will be cropped
84% of the calculated area to be stripped will be cropped from
the bottom, and 16%will be cropped from the top.

Note: If the flag was added with no value, and the conference is set to
TelePresence mode, cropping is done as follows:
• Left/right sides: no cropping
• Top/Bottom: the calculated area to be stripped will be cropped 84%
of the calculated area to be stripped will be cropped from the
bottom, and 16%will be cropped from the top.

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System Configuration Flags

Flag

Description

IVR_MESSAGE_VOLUME

The volume of IVR messages varies according to the value of this flag.
Possible value range: 0-10 (Default: 6).
0 – disables playing the IVR messages
1 – lowest volume
10 – highest volume
Notes:
• It is not recommended to disable IVR messages by setting the flag
value to 0.
• System reset is not required for flag changes to take effect.

IVR_MUSIC_VOLUME

The volume of the IVR music played when a single participant is
connected to the conference varies according to the value of this flag.
Possible value range: 0-10 (Default: 5).
0 – disables playing the music
1 – lowest volume
10 – highest volume
Note: System reset is not required for flag changes to take effect.

IVR_ROLL_CALL_VOLUME

The volume of the Roll Call varies according to the value of this flag.
Possible value range: 0-10 (Default: 6).
0 – disables playing the Roll Call
1 – lowest volume
10 – highest volume
Note:
• It is not recommended to disable the Roll Call by setting the flag
value to 0.
• System reset is not required for flag changes to take effect.

LAST_LOGIN_ATTEMPTS

If YES, the system displays a record of the last Login of the user.
Default: NO.
.

LEGACY_EP_CONTENT_DEFAU
LT_LAYOUT

Defines the video layout to be displayed on the screen of the legacy
endpoints when switching to Content mode.
Default value: CP_LAYOUT_1P7 (1+7).

MAX_CONF_PASSWORD_REPE
ATED_CHAR

Allows the administrator to configure the maximum number of
consecutive repeating characters that are to be allowed in a
conference password.
Range: 1 - 4
Default: 2

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System Configuration Flags

Flag

Description

MAX_CP_RESOLUTION

The MAX_CP_RESOLUTION flag value is applied to the system
during First Time Power-on and after a system upgrade. The default
value is HD720.
All subsequent changes to the Maximum CP Resolution of the system
are made using the Resolution Configuration dialog box.
Possible flag values:
• HD1080 - High Definition at 30 fps
• HD720 – High Definition at 60 fps
• HD – High Definition at 30 fps
• SD30 – Standard Definition at 30 fps
• SD15 – Standard Definition at 15 fps
• CIF – CIF resolution
Default: HD1080
For more information see Video Resolutions in AVC-based CP
Conferencing.

MAX_INTRA_REQUESTS_PER_I
NTERVAL_

Enter the maximum number of refresh (intra) requests for the Content
channel sent by the participant’s endpoint in a 10 seconds interval that
will be dealt by the Collaboration Server system. When this number is
exceeded, the Content sent by this participant will be identified as
noisy and his/her requests to refresh the Content display will be
suspended.
Default setting: 3

MAX_INTRA_SUPPRESSION_D
URATION_IN_SECONDS_

Enter the duration in seconds to ignore the participant’s requests to
refresh the Content display.
Default setting: 10

MAX_NUMBER_OF_MANAGEM
ENT_SESSIONS_PER_SYSTEM

Defines the maximum number of concurrent management sessions
(http and https connections) per system.
Value: 4 - 80
Default: 80

MAX_NUMBER_OF_MANAGEM
ENT_SESSIONS_PER_USER

Defines the maximum number of concurrent management sessions
(http and https connections) per user.
Value: 4 - 80
Default: 10

MAX_PASSWORD_REPEAPED_
CHAR

Allows the administrator to configure the maximum number of
consecutive repeating characters to be allowed in a user password.
Range: 1 - 4
Default: 2

MCU_DISPLAY_NAME

The name of the MCU that is displayed on the endpoint’s screen when
connecting to the conference.
Default: POLYCOM RealPresence Collaboration Server
800sPOLYCOM

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System Configuration Flags

Flag

Description

MIN_PASSWORD_LENGTH

The length of passwords.
Possible value: between 0 and 20.
0 means this rule is not enforced.

MIN_PWD_CHANGE_FREQUEN
CY_IN_DAYS

Defines the frequency with which a user can change a password.
Values: 0 -7.
0 (standard default) - users do not have to change their passwords.

MIN_SYSTEM_DISK_SPACE_TO
_ALERT

Defines a minimum remaining Collaboration Server disk capacity in
megabytes. If the remaining disk capacity falls below this level an
active alarm is raised.
Default: 2048

MIN_TIP_COMPATIBILITY_LINE_
RATE

This flag determines the minimum line rate at which conferencing
entities such as an Entry Queue or Meeting Room can be TIP-enabled
and TIP-enabled endpoints can connect to them.
CTS version 7 requires a minimum line rate of 1024 kbps and will
reject calls at lower line rates, therefore the System Flag value should
be 1024 kbps or higher.
0 means that no minimum line rate is enforced on the conference for
TIP connectivity.
Default: 1024

MS_ENVIRONMENT

If YES, sets the Collaboration Server SIP environment to integrate with
Microsoft OCS solution.
Default: NO

MULTIPLE_SERVICES

Applicable to the RealPresence Collaboration Server (RMX) 800s
only.
Determines whether the Multiple Services option is be activated once
the appropriate license is installed.
Possible Values: YES / NO
Default: NO

NUMERIC_CHAIR_PASS_DEFA
ULT_LEN

This flag enables or disables the automatic generation of chairperson
passwords and determines the number of digits in the chairperson
passwords assigned by the MCU.
Possible values are:
• 0 disables the automatic password generation.
Any value other than 0 enables the automatic generation of
chairperson passwords if the flag
HIDE_CONFERENCE_PASSWORD is set to NO.
• 1 – 16, default: 6 (Standard Security Mode)
If the default is used, in non-secured mode the system will
automatically generate chairperson passwords that contain 6
characters.

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System Configuration Flags

Flag

Description

NUMERIC_CHAIR_PASS_MAX_L
EN

The maximum number of digits that the user can enter when manually
assigning a password to the chairperson.
Range: 0 – 16
Default: 16

NUMERIC_CHAIR_PASS_MIN_L
EN

Defines the minimum length required for the Chairperson password.
Value: 0-16
Default: 0 - this rule is not enforced.

NUMERIC_CONF_ID_LEN

Defines the number of digits in the Conference ID that will be assigned
by the MCU. Enter 0 to disable the automatic assignment of IDs by the
MCU and let the Collaboration Server user manually assign them.
Range: 2-16 (Default: 4).

NUMERIC_CONF_ID_MAX_LEN

The maximum number of digits that the user can enter when manually
assigning an ID to a conference.
Range: 2-16 (Default: 8)
Note: Selecting 2 limits the number of simultaneous ongoing
conferences to 99.

NUMERIC_CONF_ID_MIN_LEN

The minimum number of digits that the user must enter when manually
assigning an ID to a conference.
Range: 2-16 (Default: 4)
Note: Selecting 2 limits the number of simultaneous ongoing
conferences to 99.

NUMERIC_CONF_PASS_DEFAU
LT_LEN

This flag enables or disables the automatic generation of conference
passwords and determines the number of digits in the conference
passwords assigned by the MCU.
Possible values are:
• 0 disables the automatic password generation.
Any value other than 0 enables the automatic generation of
conference passwords if the flag
HIDE_CONFERENCE_PASSWORD is set to NO.
• 1 – 16, default: 6
If the default is used, in non-secured mode the system will
automatically generate conference passwords that contain 6
characters.

NUMERIC_CONF_PASS_MAX_L
EN

The maximum number of digits that the user can enter when manually
assigning a password to the conference.
Range: 0 – 16
Default (both Modes): 16

NUMERIC_CONF_PASS_MIN_L
EN

Defines the minimum length required for the Conference password.
Value: 0-16
• Default: 0 - this rule is not enforced.

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System Configuration Flags

Flag

Description

PAL_NTSC_VIDEO_OUTPUT

When set to AUTO (default), the video output sent by the Collaboration
Server is either PAL or NTSC format, depending on the current
speaker in the layout. This ensures full synchronization between the
frame rate of the speaker and the video encoder, ensuring smoother
video.
In environments where the majority of endpoints are configured to
either NTSC or PAL, the flag can be set accordingly to change the
video encoding of the Collaboration Server to be compatible with the
majority of endpoints in the call.
Possible Values: AUTO, PAL, NTSC

PASSWORD_EXPIRATION_DAY
S

Determines the duration of password validity.
Value: between 0 and 90 days.
0 - user passwords do not expire.

PASSWORD_EXPIRATION_DAY
S_MACHINE

Enables the administrator to change the password expiration period of
Application-user’s independently of regular users. Default: 365 (days).

PASSWORD_EXPIRATION
_WARNING_DAYS

Determines the display of a warning to the user of the number of days
until password expiration.
Value: between 0 and 14 days.
0 - password expiry warnings are not displayed.

PASSWORD_HISTORY_SIZE

The number of passwords that are recorded to prevent users from
re-using their previous passwords.
Values are between 0 and 16.
0 (standard default) - the rule is not enforced.

RESTRICT_CONTENT_BROADC
AST_TO_LECTURER

If set to YES, only the conference lecturer may send content to the
conference.
If set to NO, any conference participant can send content.
Default: YES

RRQ_WITHOUT_GRQ

To enable registration, some gatekeepers require sending first RRQ
and not GRQ.
Set flag to YES, if this behavior is required by the gatekeeper in your
environment.
Default: NO.
GRQ (Gatekeeper Request) - Gatekeeper discovery is the process an
endpoint uses to determine which gatekeeper to register with.
RRQ - registration request sent to the gatekeeper.

SEPARATE_MANAGEMENT_NE
TWORK

Applicable to the RealPresence Collaboration Server (RMX) 800s
only.
Enables/disables the Network Separation
Default: NO.

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System Configuration Flags

Flag

Description

SESSION_TIMEOUT_IN_MINUT
ES

If there is no input from the user or if the connection is idle for longer
than the number of minutes specified by this flag, the connection to the
Collaboration Server is terminated.
Value: 0-999
0 - Session Timeout is disabled.
Default: 0

SIP_AUTO_SUFFIX_EXTENSIO
N

Used to automatically add a suffix to a SIP address (To Address)
instead of adding it manually in the Collaboration Server Web Client
(SIP address) when the SIP call is direct-dial and not through a Proxy.
Example:
Participant Name = john.smith
Company Domain = maincorp.com
SIP_AUTO_SUFFIX_EXTENSION flag value = @maincorp.com
Entering john.smith will generate a SIP URI =
john.smith@maincorp.com

STAR_DELIMITER_ALLOWED

When set to YES, an asterisk “*” can be used as a delimiter in
Conference and Meeting Room dial strings.
The dial string is first searched for “'#' first followed by “*”.
Default: NO

SYSTEM_BROADCAST_VOLUM
E

This value is used when the system flag
FORCE_SYSTEM_BROADCAST_VOLUME is set to YES.
Determines the default audio level with which the participants connects
and sends audio to the conference.
The volume scale is from 1 to 10, where 1 is the weakest and 10 is the
strongest. The default connection value is 5.
Each unit change represents an increase or decrease of 3 dB
(decibel).
Range: 1-10
Default: 5

SYSTEM_LISTENING_VOLUME

This value is used when the system flag
FORCE_SYSTEM_LISTENING_VOLUME is set to YES.
Determines the default audio level with which the participants connects
and receives audio from the conference.
The volume scale is from 1 to 10, where 1 is the weakest and 10 is the
strongest. The default value is 5. Each unit change represents an
increase or decrease of 3 dB (decibel).
Range: 1-10
Default: 5

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System Configuration Flags

Flag

Description

TERMINATE_CONF_AFTER_CH
AIR_DROPPED

From Version 8.1, this flag’s functionality is replaced by the Terminate
Conference after Chairperson Drops check box in the Profile - IVR
dialog box.
In versions prior to 8.1, if YES, sets conferences to automatically
terminate if the Chairperson disconnects from the conference. This
takes effect only if the Conference Requires Chairperson check box in
the Conference Profile Properties, IVR Tab, is selected.
Default: YES
Note: In order for the "Chairperson Exit" message to be played this
flag must be set to YES.

USER_LOCKOUT

If YES, a user is locked out of the system after three consecutive Login
failures with same User Name. The user is disabled and only the
administrator can enable the user within the system.
Default: NO

USER_LOCKOUT_DURATION_I
N_MINUTES

Defines the duration of the Lockout of the user.
Value: 0 - 480
0 means permanent User Lockout until the administrator re-enables
the user within the system.
Default: 0

USER_LOCKOUT_WINDOW_IN_
MINUTES

Defines the time period during which the three consecutive Login
failures occur.
Value: 0 - 45000
0 means that three consecutive Login failures in any time period will
result in User Lockout.
Default: 60

Manually Adding and Deleting System Flags
To add a flag:
1 In the System Flags dialog box, click the New Flag button.
The New Flag dialog box is displayed.

2 In the New Flag field enter the flag name.

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System Configuration Flags

3 In the Value field enter the flag value.
The flags in the Manually Added, Modified, Deleted System Flags table can be manually added
to the MCMS_PARAMETERS_USERS tab.
4 Click OK to close the New Flag dialog box.
The new flag is added to the flags list.
5 Click OK to close the System Flags dialog box.
For flag changes (including deletion) to take effect, reset the MCU. For more information see
Resetting the Collaboration Server.

:

Manually Added, Modified, Deleted System Flags
Flag

Description

802_1X_CERTIFICATE_MODE

Not supported with Polycom® RealPresence® Collaboration Server
800s and Polycom® RealPresence® Collaboration Server Virtual
Edition.

802_1X_SKIP_CERTIFICATE_VA
LIDATION

Not supported with Polycom® RealPresence® Collaboration Server
800s and Polycom® RealPresence® Collaboration Server Virtual
Edition.

802_FIPS_MODE

Not supported with Polycom® RealPresence® Collaboration Server
800s and Polycom® RealPresence® Collaboration Server Virtual
Edition.

ACCEPT_VOIP_DTMF_TYPE

Defines the type of DTMF tones (inband) or digits (outband) that the
Collaboration Server will accept in VOIP calls.
Range:
• 0 - Auto (default):
Inband or outband DTMF tones/digits are accepted depending on
the endpoint's current setting. If the endpoint switches from inband
to outband or visa versa the value of the
SET_DTMF_SOURCE_DIFF_IN_SEC flag determines the time
interval after which both inband and outband tones/digits will be
accepted.
• 1 - Outband (H.245) only
• 2 - Inband only

ANAT_IP_PROTOCOL

If YES, enables Alternative Network Address Types.
Range: DISABLED, AUTO, PREFER_IPv4, PREFER_IPv6
• Default: YES

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System Configuration Flags

Manually Added, Modified, Deleted System Flags
Flag

Description

APACHE_KEEP_ALIVE_TIMEOU
T

If the connection is idle for longer than the number of seconds specified
by this flag, the connection to the Collaboration Server is terminated.
Value: 0 - 999
Default: 15
Note: A value of 0 results in an unlimited keep-alive duration.

AVOID_VIDEO_LOOP_BACK_IN
_CASCADE

When set to YES the current speaker’s image is not sent back through
the participant link in cascaded conferences with conference layouts
other than 1x1.
Default: YES
Range: YES / NO

BLOCK_CONTENT_LEGACY_F
OR_LYNC

This flag is used to control the system behavior in an environment
where some Lync clients use the Polycom CCS plug-in and some do
not.
When set to NO (default), Content is sent to all Lync clients over the
video channel, including those with the plug-in installed, even when the
Send Content to Legacy Endpoints is disabled. Other, non-Lync legacy
endpoints will not be affected by this flag and will receive content
according to the Send Content to Legacy Endpoints settings in the
conference Profile.
When set to YES, Content is not sent to Lync clients over the video
channel including those with the Polycom CCS plug-in installed, even
when the Send Content to Legacy Endpoints is enabled. Other,
non-Lync legacy endpoints will not be affected by this flag and will
receive content according to the Send Content to Legacy Endpoints
settings in the conference Profile.

CAC_ENABLE

When set to YES, enables the Call Admission Control implementation
in the Collaboration Server.
Default: NO (CAC is disabled)

CASCADE_LINK_PLAY_TONE_
ON_CONNECTION

When set to YES, the RealPresence Collaboration Server plays a tone
when a cascading link between conferences is established. The tone is
played in both conferences.
This tone is not played when the cascading link disconnects from the
conferences.
The tone used to notify that the cascading link connection has been
established cannot be customized.
Default value: NO.
The tone volume is controlled by the same flag as the IVR messages
and tones: IVR_MESSAGE_VOLUME.

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System Configuration Flags

Manually Added, Modified, Deleted System Flags
Flag

Description

CELL_IND_LOCATION

Change the location of the display of Network Quality Indicators
displayed in the cells of the conference Video Layout.
Default: TOP_RIGHT
Range:
• BOTTOM_LEFT
• BOTTOM_RIGHT
• TOP_LEFT
• TOP_RIGHT

CFG_KEY_ENABLE_FLOW_CO
NTROL_REINVITE

Used to enable or disable sending a re-INVITE to endpoints to adjust
their data rate. When set to YES, re-INVITE is used for endpoints that
do not support flow control in SIP using either the Information or RTCP
Feedback mechanisms.
Default: NO.

CONF_GATHERING_DURATION
_SECONDS

The value of this System Flag sets the duration of the Gathering Phase
in seconds. The Gathering Phase duration of the conference is
measured from the scheduled start time of the conference.
Range: 0 - 3600
Default: 180
For more information see Video Preview (AVC Only Participants).

CP_REGARD_TO_INCOMING_S
ETUP_RATE

For use in the Avaya Environment.
If set to YES, the RealPresence Collaboration Server calculates the
line rate for incoming calls in CP conferences, according to the line rate
which is declared by the endpoint in the H.225 setup message.
If set to NO, the rate is calculated according to the conference line rate
regardless of the rate in the H.225 setup message.
Default: YES.

CPU_BONDING_LINK_MONITO
RING_FREQUENCY

Used when using the MII Monitor for troubleshooting networks.This
flag sets the MII Polling Interval in milliseconds. A value of zero
disables MII monitoring.
Default: 100

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System Configuration Flags

Manually Added, Modified, Deleted System Flags
Flag

Description

CPU_BONDING_MODE

Sets the Bonding Mode of the Signalling and Management network
interface controllers.
Mode=6, balance-alb,
(Adaptive Load Balancing) includes balance-tlb, (Transmit Load
Balancing) and balance-rlb (Receive Load Balancing) for IPV4 traffic.
No special switch support is required.
Receive Load Balancing is achieved by ARP negotiation.
Outbound ARP Replies are intercepted and their source hardware
address is overwritten with the unique hardware address of one of the
slaves in the bond. In this way different peers will use different
hardware addresses for the server.
Note: balance-alb is the only supported value. All other possible values
are for troubleshooting purposes only.
Default: balance-alb
Possible values:
• balance-alb
• balance-rr
• active-backup
• balance-xor
• broadcast
• 802.3ad
• balance-tlb

DETECT_H323_EP_DISCONNE
CT_TIMER

Allows you to specify the amount of time the MCU waits before
disconnecting H.323 endpoint.

DETECT_SIP_EP_DISCONNEC
T _TIMER

The flag value indicates the amount of time in seconds to wait for an
RTCP or RTP message to be received from the endpoint. When the
time that was set in the system flag has elapsed and no RTCP or RTP
audio or video message has been received on either the audio or the
video channel, the MCU disconnects the SIP endpoint from the
conference.
Default: 20 (seconds)
Range: 0 - 300
For more information see Detecting SIP Endpoint Disconnection.

DISABLE_CELLS_NETWORK_IN
D

Disable the display of Network Quality Indicators displayed in the cells
of the conference Video Layout.
Default: YES
Range: YES / NO

DISABLE_DUMMY_REGISTRATI
ON

Enables or disables SIP dummy registration on the domain.
Possible Values:
NO (Default) - Disables SIP dummy registration.
YES - Enables SIP dummy registration.
Note: For homologation and certification testing, the flag must be set to
YES.

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System Configuration Flags

Manually Added, Modified, Deleted System Flags
Flag

Description

DISABLE_GW_OVERLAY_INDIC
ATION

When set to NO (default), displays progress indication during the
connection phase of a gateway call.
Set the value to YES to hide the connection indications displayed on
the participant’s screen during the connection phase of a gateway call.

DISABLE_SELF_NETWORK_IN
D

Disable the display of the Network Quality Indicator of the participant’s
own endpoint.
Default: NO
Range: YES / NO

DISABLE_WIDE_RES_TO_SIP_
DIAL_OUT

When set to NO (default), the RealPresence Collaboration Server
sends wide screen resolution to dial-out SIP endpoints. Endpoint types
that do not support wide screen resolutions are automatically identified
by the Collaboration Server according to their product type and version
and will not receive the wide resolution even if the flag is set to YES.
When manually added and set to YES, the RealPresence
Collaboration Server does not send wide screen.
Default: NO.

DTMF_FORWARD_ANY_DIGIT_
TIMER_SECONDS

Used for DTMF code suppression in cascading conferences.
Determines the time period (in seconds) that MCU A will forward DTMF
inputs from conference A participants to MCU B.
Flag range (in seconds): 0 - 360000
This flag is defined on MCU A (the calling MCU).
For more information, see Video Layout in Cascading conferences (CP
and mixed CP and SVC).

ENABLE_CISCO_GK

When set to YES, it enables the use of an identical prefix for different
Collaboration Servers when registering with a Cisco MCM Gatekeeper.
Default: NO.

ENABLE_CLOSED_CAPTION

Enables or disables the Closed Captions option that allow endpoints to
endpoints to provide real-time text transcriptions or language
translations of the video conference.
When set to NO (default), Closed Captions are disabled.
When set to YES, Closed Captions are enabled.

ENABLE_EPC

When set to YES (default), enables Polycom proprietary People+.
When set to NO, disables this feature for all conferences and
participants.

ENABLE_EXTERNAL_DB_ACCE
SS

If YES, the Collaboration Server connects to an external database
application, to validate the participant’s right to start a new conference
or access a conference.
Default: NO

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595

System Configuration Flags

Manually Added, Modified, Deleted System Flags
Flag

Description

ENABLE_H239

When set to YES, Content is sent via a separate Content channel.
Endpoints that do not support H.239 Content sharing will not be able to
receive
When set to NO, the Content channel is closed. In such a case, H.239
Content is sent via the video channel (“people” video) enabling
endpoints that do not support H.239 Content sharing to receive the
Content in their video channel.
Default: YES.

ENABLE_H239_ANNEX_T

In H.239-enabled MIH Cascading, when MGC is on level 1, enables
sending Content using Annex T.

ENABLE_LYNC_RTCP_INTRA

When set to YES, RTCP FIR is used for sending Intra Requests. When
set to NO Intra Requests are sent using SIP INFO Messages.
Range: YES / NO
Default: NO

ENABLE_MCCF

Enables or disables the support of External IVR Services via the
MCCF-IVR package is enabled. In Ultra Secure Mode and in secured
environments where the External IVR Services via the MCCF-IVR
package is not required and unused ports should be closed, this flag
should be set to NO.
Range: YES / NO
Default: YES (in Standard security Mode) or NO (in Ultra Secure
Mode)
Note: Ultra Secure Mode is not supported by 800s and Virtual Edition
MCUs.

ENABLE_MS_FEC

Enables the Microsoft FEC (Forward Error Correction) support for RTV.
Range: Auto/No
Default: Auto
When set to Auto, FEC support is enabled. FEC uses the DV00 option
(DV=00 - one FEC per frame using XOR). When set to No, FEC
support is disabled.

ENABLE_NO_VIDEO_RESOUR
CES_AUDIO_ONLY_MESSAGE

Enables playing a voice message that Informs the participant of the
lack of Video Resources in the RealPresence Collaboration Server and
that he/she is being connected as Audio Only.
Default: YES

ENABLE_RECORDING_
OPERATION_VIA_SIPINFO

Enables or disables recording control operations to be performed using
either DTMF tones or a SIP INFO request.

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596

System Configuration Flags

Manually Added, Modified, Deleted System Flags
Flag

Description

ENABLE_SELECTIVE_MIXING

Enables (default) or disables the Automatic muting of noisy AVC
endpoints. For more details, see Automatic Suppression of Noisy
Endpoints (AVC Endpoints).
When set to YES, the automatic muting of noisy endpoints can be
enabled or disabled at the conference level in the Conference Profile Audio Settings dialog box.
When set to NO, the automatic muting of noisy endpoints is disabled at
the conference level and cannot be enabled in the Conference Profile Audio Settings dialog box.
Default: YES
Note: MCU reset is not required when changing the flag value.

ENABLE_SIP_PEOPLE_PLUS_C
ONTENT

If security is of higher priority than SIP Content sharing, SIP
People+Content can be disabled by setting this System Flag to NO.
(The content management control (BFCP) utilizes an unsecured
channel (60002/TCP) even when SIP TLS is enabled.)
Default: YES

ENABLE_SIP_PPC_FOR_ALL_U
SER_AGENT

When set to YES, SIP People+Content and BFCP capabilities are
declared with all vendors’ endpoints.
Default: YES
Range: YES / NO

ENABLE_SIRENLPR

Enable / disable SirenLPR Audio Algorithm for use in IP (H.323, SIP)
calls in both CP and VSW conferences.
Range: YES / NO
Default: YES

ENABLE_SIRENLPR_SIP_ENCR
YPTION

Enables the SirenLPR audio algorithm when using encryption with the
SIP protocol.
Range: YES / NO
Default: NO

ENABLE_TC_PACKAGE

Enables or disables Network Traffic Control.
Range: YES / NO
Default: NO

ENABLE_TEXTUAL_CONFEREN
CE_STATUS

Set the value of this flag to NO to disable Text Indication. This setting is
recommended for MCUs running Telepresence conferences.
Default: YES.

ENABLE_VIDEO_PREVIEW

Enables the Video Preview feature.
Default: YES.
For more details, see Video Preview (AVC Only Participants).

EXTERNAL_CONTENT_
DIRECTORY

The Web Server folder name. Change this name if you have changed
the default names used by the CMA application.
Default: /PlcmWebServices

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597

System Configuration Flags

Manually Added, Modified, Deleted System Flags
Flag

Description

EXTERNAL_CONTENT_IP

Enter the IP address of the CMA server in the format:
http://[IP address of the CMA server].
For example, http://172.22.185.89.
This flag is also the trigger for replacing the internal Collaboration
Server address book with the CMA global Address Book.
When empty, the integration of the CMA address book with the
Collaboration Server is disabled.

EXTERNAL_CONTENT_PASSW
ORD

The password associated with the user name defined for the
Collaboration Server in the CMA server.

EXTERNAL_CONTENT_PORT

The CMA port used by the Collaboration Server to send and receive
XML requests/responses.
Default: 80.

EXTERNAL_CONTENT_USER

The login name defined for the Collaboration Server in the CMA server
defined in the format:
domain name/user name.

EXTERNAL_DB_DIRECTORY

The URL of the external database application. For the sample script
application, the URL is:
/SubmitQuery.asp

EXTERNAL_DB_IP

The IP address of the external database server, if one is used.
Default: 0.0.0.0

EXTERNAL_DB_LOGIN

The login name defined for the Collaboration Server in the external
database server.
Default: POLYCOM

EXTERNAL_DB_PASSWORD

The password associated with the user name defined for the
Collaboration Server on the external database server.
Default: POLYCOM

EXTERNAL_DB_PORT

The external database server port used by the Collaboration Server to
send and receive XML requests/responses.
For secure communications set the value to 443.
Default: 5005.

FADE_IN_FADE_OUT

Not supported from version 8.1

FORCE_1X1_LAYOUT_ON_CAS
CADED_LINK_CONNECTION

When set to YES, the cascaded link is automatically set to Full Screen
(1x1) in CP conferences forcing the speaker in one cascaded
conference to display in full window in the video layout of the other
conference.
Set this flag to NO when connecting to an MGC using a cascaded link,
if the MGC is functioning as a Gateway and participant layouts on the
other network are not to be forced to 1X1.
Default: YES

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598

System Configuration Flags

Manually Added, Modified, Deleted System Flags
Flag

Description

FORCE_AUDIO_CODEC_FOR_
MS_SINGLE_CORE

This flag is used to force the use of a specific Audio algorithm when a
Microsoft Office Communicator R2 or Lync Client is hosted on a
workstation with a single core processor. The flag value overrides the
default audio algorithm selection (G.722.1) that may cause audio
quality problems when G.722.1 is used by Microsoft Clients running on
single processor workstations.
This flag can be set to:
• AUTO – No forcing occurs and the Collaboration Server negotiates
a full set of Audio algorithm during capabilities exchange.
• G711A/U or G722 – Set this flag value according to the hosting
workstation capabilities. If the Collaboration Server detects single
core host during capabilities exchange it will assign a G.711 or G.722
Audio algorithm according to the flag value.
Possible values: AUTO, G711A, G711U, G722
Default: G711A

FORCE_ENCRYPTION_FOR_U
NDEFINED_PARTICIPANT_IN_W
HEN_AVAILABLE_MODE

When set to YES, Undefined participants must connect encrypted,
otherwise they are disconnected.
When set to NO (default) and the conference Encryption in the Profile
is set to “Encrypt When Possible”, both Encrypted and Non-encrypted
Undefined participants can connect to the same conferences, where
encryption is the preferred setting.
Default: NO

FORCE_G711A

Setting this flag forces the use of the G711A Audio Codec.
Possible values: YES / NO
Default: NO

FORCE_RESOLUTION

Use this flag to specify IP (H.323 and SIP) endpoint types that cannot
receive wide screen resolution and that were not automatically
identified as such by the Collaboration Server.
Possible values are endpoint types, each type followed by a semicolon.
For example, when disabling Wide screen resolution in an HDX
endpoint enter the following string: HDX;
Note: Use this flag when the flag SEND_WIDE_RES_TO_IP is set to
YES.

FORCE_STATIC_MB_ENCODIN
G

This flag supports Tandberg MXP mode of sending and receiving video
by IP endpoint in HD 720p resolution and Video Quality set to Motion.
Default value: Tandberg MXP.
To disable this flag, enter NONE.

G728_IP

Enables or disables declaration of G.728 Audio Algorithm capabilities in
IP calls.
Range: YES / NO
Default: NO

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599

System Configuration Flags

Manually Added, Modified, Deleted System Flags
Flag

Description

H239_FORCE_CAPABILITIES

When the flag is set to NO, the Collaboration Server only verifies that
the endpoint supports the Content protocols: Up to H.264 or H.263.
When set to YES, the Collaboration Server checks frame rate,
resolution and all other parameters of the Content mode as declared by
an endpoint before receiving or transmitting Content.
Default: NO.

H264_HD_GRAPHICS_MIN_CO
NTENT_RATE

Determines the minimum content rate (in kbps) required for endpoints
to share H.264 high quality content via the Content channel When
Content Setting is Graphics.
Range: 0-1536
Default: 128

H264_HD_HIGHRES_MIN_CON
TENT_RATE

Determines the minimum content rate (in kbps) required for endpoints
to share H.264 high quality content via the Content channel When
Content Setting is Hi Resolution Graphics.
Range: 0-1536
Default: 256

H264_HD_LIVEVIDEO_MIN_CO
NTENT_RATE

Determines the minimum content rate (in kbps) required for endpoints
to share H.264 high quality content via the Content channel When
Content Setting is Live Video.
Range: 0-1536
Default: 384

H323_FREE_VIDEO_RESOURC
ES

For use in the Avaya Environment.
In the Avaya Environment there are features that involve converting
undefined dial-in participants’ connections from video to audio (or vice
versa). To ensure that the participants’ video resources remain
available for them, and are not released for use by Audio Only calls,
set this flag to NO.
If set to YES, the Collaboration Server will release video resources for
Audio Only calls.
Default: YES.

HIDE_CONFERENCE_PASSWO
RD

If set to YES:
• Conference and Chairperson Passwords that are displayed in the
Collaboration Server Web Client or RMX Manager are hidden when
viewing the properties of the conference.
• Automatic generation of passwords (both conference and
chairperson passwords) is disabled, regardless of the settings of the
flags:
 NUMERIC_CONF_PASS_DEFAULT _LEN
 NUMERIC_CHAIR_PASS_ DEFAULT_LEN.
For more information see Automatic Password Generation Flags.
Default: NO.

Polycom®, Inc.

600

System Configuration Flags

Manually Added, Modified, Deleted System Flags
Flag

Description

IP_LINK_ENVIRONMENT

In H.239-enabled MIH Cascading, when MGC is on level 1, setting this
flag to YES will adjust the line rate of conferences run on the
RealPresence® Collaboration Server 800s and RealPresence®
Collaboration Server Virtual Edition from 1920Kbps to 18432,
100bits/sec to match the actual rate of the IP Only HD Video Switching
conference running on the MGC.
Note: If the flag MIX_LINK_ENVIRONMENT is set to NO, the
IP_ENVIRONMENT_LINK flag must be set to YES.

IP_RESPONSE_ECHO

When the System Flag value is YES, the Collaboration Server will
respond to ping (IPv4) and ping6 (IPv6) commands. When set to NO,
the Collaboration Server will not respond to ping and ping6 commands.

ITP_CERTIFICATION

When set to NO (default), this flag disables the telepresence features
in the Conference Profile.
Set the flag to YES to enable the telepresence features in the
Conference Profile (provided that the appropriate License is installed).

LAN_REDUNDANCY

Applicable to the RealPresence Collaboration Server (RMX) 800s only.
Enables Local Area Network port redundancy.
Default: NO
Range: YES / NO
Note: If the flag value is set to YES and either of the LAN connections
(LAN1 or LAN2) experiences a problem, an active alarm is raised
stating that there is no LAN connection, specifying both the card and
port number.

LIMIT_SD_AND_CIF_BW_MPMR
X

When to YES (default), limits the maximum negotiated and opened bit
rate for resolutions equal or lower than SD to 1Mbps.
When set to NO no limitation is applicable to SD and CIF bit rates.
Range: YES/NO.
Default: YES.

MANAGE_TELEPRESENCE_RO
OM_SWITCH_LAYOUTS

Determines whether the MLA or the RMX controls the Room Switch
Telepresence Layouts.
• When set to NO, the RMX does not manage Telepresence Room
Switch Layouts and they continue to be managed by the MLA.
• When set to YES, the RMX manages Telepresence Room Switch
Layouts.
Default: NO
Range: YES / NO
Note: System re-start is not required for this flag’s settings to take
effect.
For more information see Room Switch Telepresence Layouts.

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601

System Configuration Flags

Manually Added, Modified, Deleted System Flags
Flag

Description

MAX_ALLOWED_RTV_HD_FRA
ME_RATE

Defines the threshold Frame Rate (fps) in which RTV Video Protocol
initiates HD resolutions.
Flag values are as follows:
Range: 0-30 (fps)
Default: 0 (fps) - Implements any Frame Rate based on Lync RTV
Client capabilities

MAX_RTV_RESOLUTION

Enables you to override the Collaboration Server resolution selection
and limit it to a lower resolution, hence minimizing the resource usage
to 1 or 1.5 video resources per call instead of 3 resources. Possible
flag values are:
AUTO (default), QCIF, CIF, VGA or HD720.

MAX_TRACE_LEVEL

This flag indicates the debugging level for system support.
Possible values:
TRACE = t, DEBUG = d, INFO_NORMAL = n, INFO_HIGH = i, WARN
= w, ERROR = e, FATAL = f, OFF = o.
Default: n

MAXIMUM_RECORDING_LINKS

The maximum number of Recording Links available for selection in the
Recording Links list and the Conference Profile - Recording dialog box.
Range: 1 - 100
Default: 20

MINIMUM_FRAME_RATE_THRE
SHOLD_FOR_SD

Low quality, low frame rate video is prevented from being sent to
endpoints by ensuring that an SD channel is not opened at frame rates
below the specified value.
Range: 0 -30
Default: 15

MIX_LINK_ENVIRONMENT

In H.239-enabled MIH Cascading, when MGC is on level 1, setting this
flag to YES will adjust the line rate of conferences run on the
RealPresence® Collaboration Server 800s and RealPresence®
Collaboration Server Virtual Edition from 1920Kbps to 17897,
100bits/sec to match the actual rate of the HD Video Switching
conference running on the MGC.
Note: If the flag MIX_LINK_ENVIRONMENT is set to YES, the
IP_ENVIRONMENT_LINK flag must be set to NO.

MS_CAC_AUDIO_MIN_BR

The minimum bit rate for audio using the Microsoft CAC (Call
Admission Control) protocol. When the bit rate is lower than the
MS_CAC_AUDIO_MIN_BR, the call is not connected.
Range: 0 - 384
Default: 30

MS_CAC_VTDEO_MIN_BR

The minimum bit rate for video using the Microsoft CAC (Call
Admission Control) protocol. When the bit rate is lower than the
MS_CAC_VIDEO_MIN_BR, the call is not connected as a video call..
Range: 0 - 384
Default: 40

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602

System Configuration Flags

Manually Added, Modified, Deleted System Flags
Flag

Description

MS_PROXY_REPLACE

Enables the proxy=replace parameter in the SIP Header. When set to
YES the outbound proxy to replaces the contact information in the
contact header with its own enabling other clients and servers to reach
the client using the proxy's IP address, even if the client is behind a
firewall.
Possible Values: YES / NO
Default: YES

NETWORK_IND_CRITICAL_PER
CENTAGE

The percentage degradation due to packet loss required to change the
indicator from Major to Critical.
Default: 5

NETWORK_IND_MAJOR_PERC
ENTAGE

The percentage degradation due to packet loss required to change the
indicator from Normal to Major.
Default: 1

NUM_OF_INITIATE_HELLO_ME
SSAGE_IN_CALL_ESTABLISHM
ENT

Indicates how many times the Hello (keep alive) message is sent from
the Collaboration Server to the endpoint in an environment that
includes a Session Border Controller (SBC) with a 3-second interval
between messages.
Range: 1 to 10.
Default:3

NUM_OF_PCM_IN_MPMX

In Collaboration Server 1500/2000/4000 systems with MPMx cards,
sets the maximum number of PCM sessions.
The default value of this flag is set according to the SVC license:
1 - If SVC is enabled in the license (the only possible value).
4 - If SVC is disabled in the license
Range: 1-4 (If SVC is disabled in the license).

NUMBER_OF_REDIAL

Enter the number re dialing attempts required. Dialing may continue
until the conference is terminated.
Default: 3

OCSP_RESPONDER_TIMEOUT

Determines the number of seconds the RMX is to wait for an OCSP
response from the OCSP Responder before failing the connection.
Network latency or slow WAN links can cause login problems when
logging in to the RMX’s Management Network.This System Flag’s
value determines the number of seconds the MCU is to wait for an
OCSP response from the OCSP Responder before failing the
connection.
Default: 3 (seconds)
Range: 1-20 (seconds)

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603

System Configuration Flags

Manually Added, Modified, Deleted System Flags
Flag

Description

PARTY_GATHERING_DURATIO
N_SECONDS

The value of this System Flag sets the duration, in seconds, of the
display of the Gathering slide for participants that connect to the
conference after the conference start time.
Range: 0 - 3600
Default: 15
For more information see Video Preview (AVC Only Participants).

PASSWORD_FAILURE_LIMIT

The number of unsuccessful Logins permitted in Ultra Secure Mode.
Default: 3
Note: Ultra Secure Mode is not supported by 800s and Virtual Edition
MCUs.

PCM_FECC

Determines whether the DTMF Code, ##, the Far/Arrow Keys (FECC)
or both will activate the PCM interface. This flag can be also be used in
combination with DTMF code definitions to disable PCM.
Possible Values: YES / NO
Default: YES.

PCM_LANGUAGE

Determines the language of the PCM interface.
Possible Values are: ENGLISH, CHINESE_SIMPLIFIED,
CHINESE_TRADITIONAL, JAPANESE, GERMAN, FRENCH,
SPANISH, KOREAN, PORTUGUESE, ITALIAN, RUSSIAN,
NORWEGIAN
Default: Current Collaboration Server Web Client language.

PORT_GAUGE_ALARM

When set to YES, if system resource usage reaches the High Port
Usage Threshold as defined for the Port Gauges, System Alerts in the
form of an Active Alarm and an SNMP trap are generated.

PRESERVE_ICE_CHANNEL_IN_
CASE_OF_LOCAL_MODE

When set to NO (default), local the ICE channel is closed after applying
CAC bandwidth management when Call Admission Control is enabled
in the local network.
When set to YES, the ICE channel is preserved open throughout the
call.
Default: NO

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604

System Configuration Flags

Manually Added, Modified, Deleted System Flags
Flag

Description

PRESERVE_PARTY_CELL_ON_
FORCE_LAYOUT

Used to prevent reassignment of cells in a forced layout that were
assigned to endpoints that have disconnected, paused their video, or
have been removed from the conference. The cell will remain black
until the endpoint reconnects or a new layout is used, or the
conference ends.
Range: YES / NO
Default: NO
• NO - Cells of dropped endpoints are reassigned. Endpoints that
reconnect will be treated as new endpoints.
• YES - Cells of dropped endpoints are not reassigned, but will be
reserved until the endpoint reconnects.
For information see the Polycom® RealPresence Collaboration Server
800s / Virtual Edition Getting Started Guide, Force Layout and
Preserve Participant Call.

QOS_IP_AUDIO

Used to select the priority of audio packets when DiffServ is the is the
selected method for packet priority encoding.
Default: 0x31

QOS_IP_VIDEO

Used to select the priority of video packets when DiffServ is the is the
selected method for packet priority encoding.
Default: 0x31

QOS_MANAGEMENT_NETWOR
K

Enter the DSCP value for the RMX Management Network.
Default: 0x10
Range: 0x00 - 0x3F

REDUCE_CAPS_FOR_REDCOM
_SIP

To accommodate deployments where some devices have limits on the
size of the SDP payload in SIP messages (such as LSCs from Redcom
running older software versions), when the flag value = YES, the SDP
size is less than 2kb and includes only one audio and one video media
line.
Default: NO

REDIAL_INTERVAL_IN_SECON
DS

Enter the number of seconds that the Collaboration Server should wait
before successive re dialing attempts.
Range: 0-30 (Default: 10)

REDUCE_CAPS_FOR_REDCOM
_SIP

To accommodate Redcom’s SDP size limit, when the flag value = YES,
the SDP size is less than 2kb and includes only one audio and one
video media line.
Default: NO

REJECT_INCORRECT_PRECED
ENCE_DOMAIN_NAME

When set to YES, when the Precedence Domain of a SIP dial-in call
does not match the Precedence Domain of the RMX, the call is
rejected. Possible values: YES/NO
Default: No

Polycom®, Inc.

605

System Configuration Flags

Manually Added, Modified, Deleted System Flags
Flag

Description

REMOVE_EP_FROM_LAYOUTƒ
n_ON_NO_VIDEO_TIMER

Enables the removal of empty video cells from the Video Layout.

Range:

0 – 19 (seconds): The feature is disabled.
20 – 300 (seconds): The feature is enabled.

Default:

20

For more information, see Remove Empty Cells From the Video
Layout.
REMOVE_H323_EPC_CAP_TO_
NON_POLYCOM_VENDOR

Used to disable EPC protocol. Use of Polycom’s proprietary protocol,
High Profile, EPC, may result in interoperability issues when used with
other vendors’ endpoints.
Possible values: YES / NO
Default: NO

REMOVE_H323_HIGH_PROFILE
_CAP_TO_NON_POLYCOM_VE
NDOR

Used to disable High Profile protocol. Use of Polycom’s proprietary
protocol, High Profile, may result in interoperability issues when used
with other vendors’ endpoints.
Possible values: YES / NO
Default: NO

REMOVE_H323_HIGH_QUALITY
_AUDIO_CAP_TO_NON_POLYC
OM_VENDOR

Used to disable the following Audio Codecs:
• G7221C
• G7221
• Siren22
• Siren14
Possible values: YES / NO
Default: NO

REMOVE_H323_LPR_CAP_TO_
NON_POLYCOM_VENDOR

Used to disable H.323 LPR protocol. Use of Polycom’s proprietary
protocol, H.323 LPR, may result in interoperability issues when used
with other vendors’ endpoints.
Possible values: YES / NO
Default: NO

REMOVE_IP_IF_NUMBER_EXIS
TS

Between the time a conference is scheduled and when it becomes
active, the IP of an endpoint may change, especially in an environment
that uses DHCP. This flag determines if the E.164 number is to be
substituted for the IP address in the dial string.
Range: YES / NO
Default: YES - The IP address will be substituted with the E.164
number.

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606

System Configuration Flags

Manually Added, Modified, Deleted System Flags
Flag

Description

RFC2833_DTMF

Controls the receipt of in-band and out-of-band DTMF Codes. When
set to:
• YES The RMX will receive DTMF Codes sent in-band.
• NO The RMX receive DTMF Codes sent out-of-band.
The RMX always sends DTMF Codes in-band (as part of the Audio
Media stream).
Range: YES/NO
Default YES

RMX_MANAGEMENT_SECURIT
Y_PROTOCOL

Enter the protocol to be used for secure communications.
Default: TLSV1_SSLV3 (both).

RTCP_FIR_ENABLE

When set to YES, the Full Intra Request (FIR) is sent as INFO (and not
RTCP).
Default = YES

RTCP_FLOW_CONTROL_TMMB
R_ENABLE

Enables/disables the SIP RTCP flow control parameter.
Default: YES

RTCP_FLOW_CONTROL_TMMB
R_INTERVAL

Modifies the interval (in seconds) of the TMMBR (Temporary Maximum
Media Stream Bit Rate) parameter for SIP RTCP flow control.
Range: 5 - 999 (seconds)
Default: 180

RTCP_PLI_ENABLE

When set to YES, the (Picture Loss Indication (PLI) is sent as INFO
(and not RTCP).
Default = YES

RTCP_QOS_IS_EQUAL_TO_RT
P

Range: YES/NO
Default: YES

RTV_MAX_BIT_RATE_FOR_FO
RCE_CIF_PARTICIPANT

Enables the removal of empty video cells from the Video Layout.

SELF_IND_LOCATION

Change the location of the display of the Network Quality Indicator of
the participant’s own endpoint.
Default: BOTTOM_RIGHT
Range:
• TOP_ LEFT
• TOP
• TOP_RIGHT
• BOTTOM_ LEFT
• BOTTOM
• BOTTOM_RIGHT

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607

System Configuration Flags

Manually Added, Modified, Deleted System Flags
Flag

Description

SEND_SIP_BUSY_UPON_RESO
URCE_THRESHOLD

When set to YES, it enables the Collaboration Server to send a busy
notification to a SIP audio endpoint or a SIP device when dialing in to
the Collaboration Server whose audio resource usage exceeded the
Port Usage threshold.
When set to NO, the system does limit the SIP audio endpoint
connections to a certain capacity and will not send a busy notification
when the resource capacity threshold is exceeded.
Default: NO

SEND_SRTP_MKI

Enables or disables the inclusion of the MKI field in SRTP packets sent
by the Collaboration Server. Setting the value to NO to disables the
inclusion of the MKI field in SRTP packets sent by the Collaboration
Server.
Set this flag to:
• NO
 When all conferences on the RMX will not have MS-Lync clients
participating and will have 3rd party endpoints participating.
 When using endpoints (eg. CounterPath Bria 3.2 Softphone) that
cannot decrypt SRTP-based audio and video streams if the MKI
(Master Key Identifier) field is included in SRTP packets sent by
the Collaboration Server.
This setting is recommended for Maximum Security Environments.
• YES
 When any conferences on the RMX will have both MS-Lync
clients and Polycom endpoints participating.
 Some 3rd party endpoints may be unsuccessful in participating
in conferences with this setting.
Notes:
• This System Flag must be added and set to YES (default) when
Microsoft Office Communicator and Lync Clients are used as they all
support SRTP with MKI.
• The system flag must be added and set to NO when Siemens
phones (Openstage and ODC WE) are used in the environment as
they do not support SRTP with MKI.
• Polycom endpoints function normally regardless of the setting of this
flag.
Default: YES

SEND_WIDE_RES_TO_IP

When set to YES (default), the Collaboration Server sends wide screen
resolution to IP endpoints. Endpoint types that do not support wide
screen resolutions are automatically identified by the Collaboration
Server according to their product type and version and will not receive
the wide resolution even when the flag is set to YES.
When manually added and set to NO, the Collaboration Server does
not send wide screen resolution to all IP endpoints.
Default: YES.

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Manually Added, Modified, Deleted System Flags
Flag

Description

SET_DTMF_SOURCE_DIFF_IN_
SEC

If the ACCEPT_VOIP_DTMF_TYPE flag is set to 0 (Auto) this flag
determines the interval, in seconds after which the Collaboration
Server will accept both DTMF tones (inband) and digits (outband).
Default: 120

SIP_BFCP_DIAL_OUT_MODE

Controls BFCP’s use of UDP and TCP protocols for dial-out SIP Client
connections according to its value:
• AUTO (Default)
If SIP Client supports UDP, TCP or UDP and TCP:
- BFCP/UDP is selected as Content sharing protocol.
• UDP
If SIP Client supports UDP or UDP and TCP:
- BFCP/UDP selected as Content sharing protocol.
If SIP Client supports TCP
- Cannot share Content.
• TCP
If SIP Client supports TCP or UDP and TCP
- BFCP/TCP selected as Content sharing protocol.
If SIP Client supports UDP
- Cannot share Content.

SIP_DUAL_DIRECTION_TCP_C
ON

In environments set to integration with Microsoft, if set to YES the
system sends a new request on the same TCP connection (instead of
opening a new one).

SIP_ENABLE_FECC

By default, FECC support for SIP endpoints is enabled at the MCU
level. You can disable it by manually adding this flag and setting it to
NO.

SIP_FAST_UPDATE_INTERVAL_
ENV

Default setting is 0 to prevent the Collaboration Server from
automatically sending an Intra request to all SIP endpoints.
Enter n (where n is any number of seconds other than 0) to let the
Collaboration Server automatically send an Intra request to all SIP
endpoints every n seconds.
It is recommended to set the flag to 0 and modify the frequency in
which the request is sent at the endpoint level (as defined in the next
flag).

SIP_FAST_UPDATE_INTERVAL_
EP

Default setting is 6 to let the Collaboration Server automatically send
an Intra request to Microsoft OC endpoints only, every 6 seconds.
Enter any other number of seconds to change the frequency in which
the Collaboration Server send the Intra request to Microsoft OC
endpoints only.
Enter 0 to disable this behavior at the endpoint level (not
recommended).

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Manually Added, Modified, Deleted System Flags
Flag

Description

SIP_FORMAT_GW_HEADERS_F
OR_REDCOM

Controls whether the RMX adds special gateway prefix and postfix
characters to the user portion of the SIP URI expressed in the “From”
and “Contact” headers of SIP messages sent during calls involving
Gateway Services. The addition of these characters can result in call
failures with some SIP call servers. It is recommended to set this flag to
YES whenever the RMX is deployed such that it registers its
conferences to a SIP call server.
Range: YES, NO
Default: NO

SIP_FREE_VIDEO_RESOURCE
S

For use in Avaya and Microsoft Environments.
When set to NO (required for Avaya and Microsoft environments),
video resources that were allocated to participants remain allocated to
the participants as long as they are connected to the conference even if
the call was changed to audio only. The system allocates the resources
according to the participant’s endpoint capabilities, with a minimum of 1
CIF video resource.
Enter YES to enable the system to free the video resources for
allocation to other conference participants. The call becomes an audio
only call and video resources are not guaranteed to participants if they
want to add video again.
Default value in Microsoft environment: NO.

SIP_OMIT_DOMAIN_FROM_PA
RTY_NAME

Provides option to remove Domain Names from SIP dial-in participants’
Site Names. This prevents long domain names being appended to SIP
participant names.

SIP_TCP_PORT_ADDR_STRAT
EGY

Setting the flag to 1 prevents the use of two sockets for one SIP call one for inbound traffic, one for outbound traffic. This is done by
inserting port “5060/5061” into the Route[0] header.
Possible values:
• 0 - Inbound traffic on port 5060/5061 outbound traffic on port 60000
• 1 - Both inbound and outbound traffic on port 5060/5061
Default: 1

SOCKET_ACTIVITY_TIMEOUT

For use in Microsoft environments.
When the MS_KEEP_ALIVE System Flag is set to YES, the value of
this flag is used as the MS Keep-Alive Timer value.

SUPPORT_HIGH _PROFILE

Enables or disables the support of High Profile video protocol in CP
conferences. This flag is specific to CP conferences and has no effect
on VSW conferences.
Range: YES / NO
Default: YES

TC_BURST_SIZE

This flag regulates the Traffic Control buffer or maxburst size as a
percentage of the participant line rate.
Range: 1-30.

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Manually Added, Modified, Deleted System Flags
Flag

Description

TC_LATENCY_SIZE

This flag limits the latency (in milliseconds) or the number of bytes that
can be present in a queue.
Range: 1-1000 (in milliseconds).

TCP_RETRANSMISSION_TIME
OUT

The number of seconds the server will wait for a TCP client to answer a
call before closing the connection.
Default = 5 (seconds)

V35_ULTRA_SECURED_SUPPO
RT

This flag must be set to YES when deploying a Serial Gateway S4GW
in Ultra Secure Mode.
Note: Ultra Secure Mode is not supported by 800s and Virtual Edition
MCUs.

VSW_CIF_HP_THRESHOLD_BI
TRATE

Controls the Minimum Threshold Line Rate (kbps) for CIF resolution for
High Profile-enabled VSW conferences.
Default: 64

VSW_HD1080p_HP_THRESHOL
D_BITRATE

Controls the Minimum Threshold Line Rate (kbps) for HD1080p
resolution for High Profile-enabled VSW conferences.
Default: 1024

VSW_HD720p30_HP_THRESHO
LD_BITRATE

Controls the Minimum Threshold Line Rate (kbps) for HD720p30
resolution for High Profile-enabled VSW conferences.
Default: 512

VSW_HD720p50-60_HP_THRES
HOLD_BITRATE

Controls the Minimum Threshold Line Rate (kbps) for HD720p50 and
HD720p50 resolutions for High Profile-enabled VSW conferences.
Default: 832

VSW_RATE_TOLERANCE_PER
ECENT

Determines the percentage of bandwidth that can be deducted from
the required bandwidth to allow participants to connect to the
conference.
For example, a value of 20 will allow a participant to connect to the
conference if the allocated line rate is up to 20% lower than the
conference line rate (or between 80% to 100% of the required
bandwidth).
Range: 0 - 75
Default: 0

VSW_SD_HP_THRESHOLD_BIT
RATE

Controls the Minimum Threshold Line Rate (kbps) for SD resolution for
High Profile-enabled VSW conferences.
Default: 128

WRONG_NUMBER_DIAL_
RETRIES

The number of re-dial attempts for a wrong destination number or a
wrong destination number time-out.
Range: 0 - 5
Default: 3
A flag value of 0 means that no redials are attempted.

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Manually Adding Flags to the CS_MODULE_PARAMETERS Tab
Using the procedure to manually add flags to the System Configuration, the following flags can be manually
added to the CS_MODULE_PARAMETERS tab:
Manually Added CS_MODULE_PARAMETERS System Flags
Flag

Description

CS_ENABLE_EPC

Add this flag with the value YES (default value is NO) to enable endpoints that support
People+Content and require a different signaling (for example, FX endpoints) to receive
Content.

H245_TUNNELING

For use in the Avaya Environment.
In the Avaya Environment, set the flag to YES to ensure that H.245 is tunneled through
H.225. Both H.245 and H.225 will use the same signaling port.
Default: NO.

H323_TIMERS_SET_I
NDEX

Enables or disables H.323 index timer according to standard or proprietary H.323
protocol.
Possible values:
0 (Default) - Sets the H.323 index timer to Polycom proprietary.
1 - Sets the H.323 index timer based on the H.323 Standard recommendation.
Note: For homologation and certification testing, this flag must be set to 1.

MS_UPDATE_CONTAC
T_REMOVE

When the flag value is set to:
• YES - The Contact Header is removed from the UPDATE message that is sent
periodically to the endpoints.
This is required when the SIP Server Type field of the IP Network Service is set as
Microsoft. Removal of the Contact Header from the UPDATE message is required
specifically by OCS R2.
• NO - The Contact Header is included in the UPDATE message. This is the system
behavior when the SIP Server Type is set as Generic.
This is required when the Collaboration Server is configured to accept calls from both
Microsoft LYNC and Cisco CUCM as CUCM requires the Contact Header.

QOS_IP_SIGNALING

Used to select the priority of IP packets when DiffServ is the is the selected method for
packet priority encoding.
Range: 0x##
Default: 0x28

SIP_DUAL_DIRECTIO
N_TCP_CON

For use in Microsoft environments.
When set to YES, sends a new request on the same TCP connection instead of opening
a new connection.
Range: YES/NO
Default: NO

SIP_ST_ENFORCE_VA
L

For use in Microsoft environments.
Session timer interval in seconds.
Default = YES

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Manually Added CS_MODULE_PARAMETERS System Flags
Flag

Description

SIP_TCP_TLS_TIMER
S

Determines the timeout characteristics of SIP TCP TLS connections.
Format: SIP_TCP_TLS_TIMERS = 
The string contains the following parameters:
Ct - Timeout of TCP CONNECT operation (seconds)
Cs - Timeout of TLS CONNECT operation (seconds)
A - Timeout of accept operation (seconds)
D - Timeout of disconnect operation (nanoseconds)
H - Timeout of handshake operation (seconds)
Default: <1,5, 4,500000,5>

SIP_TIMERS_SET_IND
EX

SIP Timer type timeout settings according to standard or proprietary protocol.
Possible values are:
0 - Default
1 - SIP Standard recommendation.
Note: For homologation and certification testing, this flag must be set to 1.

SIP_TO_TAG_CONFLI
CT

For use in Microsoft environments.
In case of forking, a tag conflict will be resolved when Status 200 OK is received from an
answering UA.
Default: YES

Deleting a Flag
To delete a flag:
1 In the System Flags dialog box, select the flag to delete and click the Delete Flag button.
2 In the confirmation message box, click Yes to confirm.
3 Click OK to close the System Flags dialog box.

Auto Layout Configuration
The Auto Layout option lets the Collaboration Server automatically select the conference video layout
based on the number of participants currently connected to the conference. You can modify the default
selection of the conference video layout to customize it to your conferencing preferences.

Customizing the Default Auto Layout
The default Auto Layout is controlled by 13 flags:
PREDEFINED_AUTO_LAYOUT_0, ... , PREDEFINED_AUTO_LAYOUT_12
Each of the 11 Auto Layout flags can be left at its default value, or set to any of the Possible Values listed
in the following Default Auto Layouts table.
The flag that controls the Auto Layout you wish to modify must be added to the System Configuration
file. For more information see Manually Adding and Deleting System Flags.

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The table below lists the available layouts.
Available Layouts
No. of Cells

Layout

Layout Flag Value

1

CP_LAYOUT_1X1

2

CP_LAYOUT_1X2
CP_LAYOUT_1X2HOR
CP_LAYOUT_1x2VER
CP_LAYOUT_2X1

3

CP_LAYOUT_1P2HOR
CP_LAYOUT_1P2HOR_UP
CP_LAYOUT_1P2VER

4

CP_LAYOUT_2X2
CP_LAYOUT_1P3HOR_UP
CP_LAYOUT_1P3VER

5

CP_LAYOUT_1P4HOR_UP
CP_LAYOUT_1P4HOR
CP_LAYOUT_1P4VER

6

CP_LAYOUT_1P5

8

CP_LAYOUT_1P7

9

CP_LAYOUT_1P8UP
CP_LAYOUT_1P8CENT
CP_LAYOUT_1P8HOR_UP
CP_LAYOUT_3X3
CP_LAYOUT_1TOP_LEFT_P8

10

CP_LAYOUT_2P8
CP_LAYOUT_2TOP_P8

13

CP_LAYOUT_1P12

16

CP_LAYOUT_4X4

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The table below lists the default layouts for according to the number of participants.
Default Auto Layouts
Flag Name: PREDEFINED_AUTO_LAYOUT_n (n = Number of Participants)
n

Layout

Layout Flag Value

0

CP_LAYOUT_1X1

1

CP_LAYOUT_1X1

2

CP_LAYOUT_1X1

3

CP_LAYOUT_1x2VER

4

CP_LAYOUT_2X2

5

CP_LAYOUT_2X2

6

CP_LAYOUT_1P5

7

CP_LAYOUT_1P5

8

CP_LAYOUT_1P7

9

CP_LAYOUT_1P7

10

CP_LAYOUT_2P8

11

CP_LAYOUT_2P8

12

CP_LAYOUT_1P12

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Example:
The following table illustrates the effect of modifying the PREDEFINED_AUTO_LAYOUT_5 flag in
conferences with fewer or more participants than the number of windows selected in the default layout.
Example of Predefined Auto Layouts
Flag

Set to
Possible Value

Number of
Participants

3
CP_LAYOUT_
1x2VER

Participant’s View

Voice activated switching displays the current
speaker in the left window of the video layout and
only the two last speakers are displayed.

7
PREDEFINED
_AUTO_LAYOUT_5
Default =

CP_LAYOUT
_IP5

3
Voice activated switching displays the current
speaker in the large (top left) window of the video
layout.

7
Voice activated switching displays the current
speaker in the top left window of the video layout.

LEGACY_EP_CONTENT_DEFAULT_LAYOUT Flag Values
The following table lists the value for each video layout that can be defined for the
LEGACY_EP_CONTENT_DEFAULT_LAYOUT Flag. It allows the selection of video layout that will be
displayed on the screen of the legacy endpoint when switching to Content mode.
For a list of available layouts see Available Layouts.

CS_ENABLE_EPC Flag
Endpoints that support People+Content may require a different signaling (for example, FX endpoints). For
these endpoints, manually add the flag CS_ENABLE_EPC with the value YES (default value is NO) to the
CS_MODULE_PARAMETERS tab.

Automatic Password Generation Flags
The Collaboration Server can be configured to automatically generate conference and chairperson
passwords when the Conference Password and Chairperson Password fields are left blank.

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Guidelines
● If the flag HIDE_CONFERENCE_PASSWORD is set to YES, the automatic generation of passwords
(both conference and chairperson passwords) is disabled, regardless of the settings of the flags
NUMERIC_CONF_PASS_DEFAULT_LEN and NUMERIC_CHAIR_PASS_ DEFAULT_LEN.
● The automatic generation of conference passwords is enabled/disabled by the flag
NUMERIC_CONF_PASS_DEFAULT_LEN.
● The automatic generation of chairperson passwords is enabled/disabled by the flag
NUMERIC_CHAIR_PASS_ DEFAULT _LEN.
● The automatically generated passwords will be numeric and random.
● The passwords are automatically assigned to ongoing conferences, Reservations, and Meeting
Rooms at the end of the creation process (once they are added to the Collaboration Server).
● Automatically assigned passwords can be manually changed through the Conference/Meeting
Room/Reservation Properties dialog boxes.
● Deleting an automatically created password will not cause the system to generate a new password
and the new password must be added manually or the field can be left blank.
● If a password was assigned to the conference via Microsoft Outlook using the PCO add-in, the
system does not change these passwords and additional passwords will not be generated (for
example, if only the conference password was assigned a chairperson password will not be
assigned).
● If the flag values (i.e. the password lengths) are changed, passwords that were already assigned to
conferences, Reservations, and Meeting Rooms will not change and they can be activated using the
existing passwords. Only new conferencing entities will be affected by the change.
Do not enable this option in an environment that includes a Polycom DMA system.

Enabling the Automatic Generation of Passwords
To enable the automatic generation of passwords, the following flags have to be defined:
Automatic Password Generation Flags
Flag

Description

HIDE_CONFERENCE_PASSWORD

NO (default) - Conference and chairperson passwords are
displayed when viewing the Conference/Meeting Room/
Reservation properties. It also enables the automatic generation
of passwords in general.
Yes - Conference and Chairperson Passwords are hidden (they
are replaced by asterisks). It also disables the automatic
generation of passwords.

NUMERIC_CONF_PASS_MIN_LEN

Enter the minimum number of characters required for conference
passwords.
Possible values: 0 – 16.
0 (default) means no minimum length.

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Flag

Description

NUMERIC_CHAIR_PASS_MIN_LEN

Enter the minimum number of characters required for chairperson
passwords.
Possible values: 0 – 16.
0 (default) means no minimum length. However this setting
cannot be applied when the Collaboration Server is in Ultra
Secure Mode.
Note: Ultra Secure Mode is not supported by 800s and Virtual
Edition MCUs.

NUMERIC_CONF_PASS_MAX_LEN

Enter the maximum number of characters permitted for
conference passwords.
Possible values: 0 – 16
16 (default) - Conference password maximum length is 16
characters.

NUMERIC_CHAIR_PASS_MAX_LEN

Enter the maximum number of characters permitted for
chairperson passwords.
Possible values: 0 – 16
16 (default) - chairperson password maximum length is 16
characters.

NUMERIC_CONF_PASS_DEFAULT_LEN

This flag enables or disables the automatic generation of
conference passwords. The length of the automatically generated
passwords is determined by the flag value.
Possible values: 0 – 16, 6 default
Enter 0 to disable the automatic generation of passwords.
Any value other than 0 enables the automatic generation of
conference passwords provided the flag
HIDE_CONFERENCE_PASSWORD is set to NO.
If the default is used, in non-secured mode the system will
automatically generate conference passwords that contain 6
characters.

NUMERIC_CHAIR_PASS_ DEFAULT _LEN

This flag enables or disables the automatic generation of
chairperson passwords. The length of the automatically
generated passwords is determined by the flag value.
Possible values: 0 – 16, 6 default
Enter 0 to disable the automatic generation of passwords.
Any value other than 0 enables the automatic generation of
chairperson passwords provided the flag
HIDE_CONFERENCE_PASSWORD is set to NO.
If the default is used, in non-secured mode the system will
automatically generate chairperson passwords that contain 6
characters.

If the default password length defined by the NUMERIC_CONF_PASS_DEFAULT_LEN or
NUMERIC_CHAIR_PASS_ DEFAULT LEN does not fall within the range defined by the minimum and
maximum length an appropriate fault is added to the Faults list.

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Hardware Monitoring
The status and properties of the MCU hardware components can be viewed and monitored in the Hardware
Monitor list pane.

Viewing the Status of the Hardware Components
The Hardware Monitor pane displays the hardware component, its present status, temperature and voltage.

To view the status of the Hardware Components on the Polycom® RealPresence®
Collaboration Server 800s and Polycom® RealPresence® Collaboration Server Virtual
Edition:
● In the RealPresence Collaboration ServerManagement pane, click the Hardware Monitor button.
The Hardware Monitor pane is displayed.

The Hardware Monitor pane displays the following information:
Hardware Monitor Pane Columns
Field

Description

Slot

Displays an icon according to the type of hardware component. The icon indicates
the hardware status as an exclamation point (!) indicates errors in the hardware
component.

Type

The type of the hardware component.

Status

The current status of the hardware component; Normal, Major, or Critical.

Temperature

Monitors the temperature of the hardware components; Normal, Major and Critical.
Note: Critical condition invokes a system shut down.

Voltage

The voltage threshold of the hardware component; either Normal or Major.

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Hardware Monitor Toolbar
The following buttons appear in the toolbar of the Hardware Monitor:
Button

Name

Description

System Reset

Resets and restarts the system. Resetting saves settings and information that
you changed in the system, i.e. IP Services, etc...

System Shut Down

Shuts down the system into a standby mode. When the user in the RMX
Manager Manager/Web Client presses the System Shut Down (red) button in
the Hardware Monitor toolbar, the system should enter a standby mode and
the LED turns ON. Only the media and control unit cards are in a standby
mode. Shelf Manager remains active.
Turn the system OFF/ON to exit the standby mode.

Viewing the Properties of Polycom® RealPresence®
Collaboration Server 800s and Polycom®
RealPresence® Collaboration Server Virtual Edition
Hardware Components
The properties displayed for the hardware components will vary according to the type of component viewed.
These component properties can be grouped as follows:
● MCU Properties (Polycom® RealPresence® Collaboration Server 800s and Polycom®
RealPresence® Collaboration Server Virtual Edition)
● Supporting Hardware Components Properties (FANS, LAN)
No properties are provided for Power Supply (PWR).

The Hardware Properties dialog box has the following structure:

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To view the MCU Properties:
1 In the Hardware Monitor pane, either double-click or right-click and select Properties for
RealPresence Collaboration Server, slot 4.

The following information is displayed:
MCU Properties - General Info
Field

Description

Hardware Version

The version of the server hardware.

Bios Version

The version of the BIOS installed on the server.

Hardware Serial
Number

The hardware serial number.

Hardware Model

Always blank.

Hardware Part
Number

The hardware part number.

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Hardware Monitoring

Field

Description

Network Interface Cards (NIC)
NIC number (1-4)

The number of the NIC.

NIC MAC Address

The physical address of the NIC.

2 Click the Active Alarms tab to view alarms related to the RealPresence Collaboration Server, i.e.
temperatures, CPU usage, memory usage, voltages and main power sensors.

The Active Alarms dialog box displays fields that relate to faults and errors detected on the
RealPresence Collaboration Server by sensors. The Active Alarms dialog box is divided into two
sections: Hardware Alarm List and Software Alarm List.
The Hardware Alarms list and software alarm list can each be saved to an Excel file (*.xls) by clicking
the Save hardware Alarm List and Save SW Alarm List buttons respectively. The severity of the
alarms is color coded: Critical (RED), Major (ORANGE) and Normal (GREEN).
3 Click Close to return to the Hardware Monitor pane.

FAN Properties:
The RealPresence Collaboration Server chassis contains 14 fans that regulate the unit’s temperature. If the
temperature increases, the fans speed will increase and vice-versa. A “Critical” condition in the fans
operation will result in a system shut down.

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To access the Fan properties:
» In the Hardware Monitor pane, either double-click or right-click and select Properties for FANS.

FANS Properties - General Info
Field

Description

General Settings
Min. Speed Level (rpm)

The minimum speed level of the fans.

Max. Speed Level (rpm)

The maximum speed level of the fans.

Normal Operating Level
(rpm)

The normal operating level defined for the fans.

Current Fan Level (rpm)

The current operating level of the fans.

Fans
Fan number (1-14)

Fan numbering

Speed (rpm)

Present speed of a fan (1-14).

LAN 1, LAN 2, LAN 3, LAN4 Properties:
The Collaboration Server unit’s chassis contains 4 external LAN connectors which register the following
information listed below. The information will be refreshed every second and also contains a peek detector
to log the maximal values, since the last peek values reset.

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To access the properties of LAN connector:
1 In the Hardware Monitor pane, double-click anywhere on the line for LANS.
The Lan List is displayed:

2 Either double-click or right-click and select Properties for the Lan connector.
The properties for the LAN connector are displayed:

Diagnostics
Your system comes with a Hardware Diagnostics USB memory stick. If required for troubleshooting,
Polycom Global Services personnel may ask you to run hardware diagnostics.
The appliance will not be operational during the running of diagnostics.

The following are required to run the diagnostics:
● Hardware diagnostics USB memory stick
● USB Keyboard
● VGA Monitor

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Hardware Monitoring

To perform diagnostics:
1 Turn off the Collaboration Server.
2 Connect a keyboard and a VGA monitor.
3 Insert the USB memory stick containing the hardware diagnostics utility into any of the system’s
USB ports.
4 Turn on the Collaboration Server.
The server boots up.

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5 Press F11.
The Boot Manager Main Menu loads.

6 Select Bios Boot Menu.
The Bios Boot Manager is displayed.

7 Using the Up/Down arrows go to Hard drive C:.
A window will display showing two options.

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8 Select Front USB.
If the USB memory stick is inserted into the back of the server this option will be called Back USB.

The Collaboration Server loads the hardware diagnostics utility.

9 Select the desired diagnostic mode.
If none is selected the Loop mpmemory and diagnostic in batch mode tests will be performed
automatically.
The diagnostics will continue running until stopped. One cycle of the diagnostics may take up to 30
minutes to complete.
The results are saved in a file called auto.txt to the USB memory stick.
10 Press Escape to stop the diagnostics.
11 Remove the USB memory stick.
12 Optional: Disconnect the keyboard and monitor.
If the RealPresence Collaboration Server is in secure mode the keyboard and monitor must

be disconnected, otherwise the RealPresence Collaboration Server will revert to non-secure
mode when the RealPresence Collaboration Server is started.

13 Turn off the Collaboration Server, then turn it On.
14 Contact Polycom Global Services for further instructions.

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Appendix A - Disconnection Causes
If a participant was unable to connect to a conference or was disconnected from a conference, the
Connection Status tab in the Participant Properties dialog box indicates the call disconnection cause. In
some cases, a possible solution may be displayed.
A video participant who is unable to connect the video channels, but is able to connect as an audio only
participant, is referred to as a Secondary participant. For Secondary participants, the Connection Status
tab in the Participant Properties dialog box indicates the video disconnection cause. In some cases, a
possible solution may be indicated.
The table below lists the call disconnection causes that can be displayed in the Call Disconnection Cause
field and provides an explanation of each message

IP Disconnection Causes
Call Disconnection Causes
Disconnection Cause

Description

Disconnected by User

The user disconnected the endpoint from the conference.

Remote device did not open the
encryption signaling channel

The endpoint did not open the encryption signaling channel.

Remote devices selected encryption
algorithm does not match the local
selected encryption algorithm

The encryption algorithm selected by the endpoint does not match the
MCU's encryption algorithm.

Resources deficiency

Insufficient resources available.

Call close. Call closed by MCU

The MCU disconnected the call.

H323 call close. No port left for audio

Insufficient audio ports.

H323 call close. No port left for video

The required video ports exceed the number of ports allocated to video
in fixed ports.

H323 call close. No port left for FECC

The required data ports exceed the number of ports allocated to data in
fixed ports.

H323 call close. No control port left

The required control ports exceed the number of ports allocated to
control data in fixed ports.

H323 call close. No port left for videocont

The required video content ports exceed the number of ports allocated
to video content in fixed ports.

H323 call closed. Small bandwidth

The gatekeeper allocated insufficient bandwidth to the connection with
the endpoint.

H323 call closed. No port left

There are no free ports left in the IP card.

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Appendix A - Disconnection Causes

Disconnection Cause

Description

Caller not registered

The calling endpoint is not registered in the gatekeeper.

H323 call closed. ARQ timeout

The endpoint sent an ARQ message to the gatekeeper, but the
gatekeeper did not respond before timeout.

H323 call closed. DRQ timeout

The endpoint sent a DRQ message to the gatekeeper, but the
gatekeeper did not respond before timeout.

H323 call closed. Alt Gatekeeper failure

An alternate gatekeeper failure occurred.

H323 call closed. Gatekeeper failure

A gatekeeper failure occurred.

H323 call closed. Remote busy

The endpoint was busy. (Applicable only to dial-out)

H323 call closed. Normal

The call ended normally, for example, the endpoint disconnected.

H323 call closed. Remote reject

The endpoint rejected the call.

H323 call closed. Remote unreachable

The call remained idle for more than 30 seconds and was
disconnected because the destination device did not answer. Possible
causes can be due to network problems, the gatekeeper could not find
the endpoint's address, or the endpoint was busy or unavailable (for
example, the “do not disturb” status is selected).

H323 call closed. Unknown reason

The reason for the disconnection is unknown, for example, the
endpoint disconnected without giving a reason.

H323 call closed. Faulty destination
address

Incorrect address format.

H323 call closed. Small bandwidth

The gatekeeper allocated insufficient bandwidth to the connection with
the endpoint.

H323 call closed. Gatekeeper reject
ARQ

The gatekeeper rejected the endpoint's ARQ.

H323 call closed. No port left

There are no ports left in the IP card.

H323 call closed. Gatekeeper DRQ

The gatekeeper sent a DRQ.

H323 call closed. No destination IP
address

For internal use.

H323 call. Call failed prior or during the
capabilities negotiation stage

The endpoint did not send its capabilities to the gatekeeper.

H323 call closed. Audio channels didn't
open before timeout

The endpoint did not open the audio channel.

H323 call closed. Remote sent bad
capability

There was a problem in the capabilities sent by the endpoint.

H323 call closed. Local capability wasn't
accepted by remote

The endpoint did not accept the capabilities sent by the gatekeeper.

H323 failure

Internal error occurred.

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Appendix A - Disconnection Causes

Disconnection Cause

Description

H323 call closed. Remote stop
responding

The endpoint stopped responding.

H323 call closed. Master slave problem

A People + Content cascading failure occurred.

SIP bad name

The conference name is incompatible with SIP standards.

SIP bad status

A general IP card error occurred.

SIP busy everywhere

The participant's endpoints were contacted successfully, but the
participant is busy and does not wish to take the call at this time.

SIP busy here

The participant's endpoint was contacted successfully, but the
participant is currently not willing or able to take additional calls.

SIP capabilities don't match

The remote device capabilities are not compatible with the conference
settings.

SIP card rejected channels

The IP card could not open the media channels.

SIP client error 400

The endpoint sent a SIP Client Error 400 (Bad Request) response.
The request could not be understood due to malformed syntax.

SIP client error 402

The endpoint sent a SIP Client Error 402 (Payment Required)
response.

SIP client error 405

The endpoint sent a SIP Client Error 405 (Method Not Allowed)
response.
The method specified in the Request-Line is understood, but not
allowed for the address identified by the Request-URI.

SIP client error 406

The endpoint sent a SIP Client Error 406 (Not Acceptable) resources.
The remote endpoint cannot accept the call because it does not have
the necessary responses. The resource identified by the request is
only capable of generating response entities that have content
characteristics not acceptable according to the Accept header field
sent in the request.

SIP client error 407

The endpoint sent a SIP Client Error 407 (Proxy Authentication
Required) response.
The client must first authenticate itself with the proxy.

SIP client error 409

The endpoint sent a SIP Client Error 409 (Conflict) response.
The request could not be completed due to a conflict with the current
state of the resource.

SIP client error 411

The endpoint sent a SIP Client Error 411 (Length Required) response.
The server refuses to accept the request without a defined Content
Length.

SIP client error 413

The endpoint sent a SIP Client Error 413 (Request Entity Too Large)
response.
The server is refusing to process a request because the request entity
is larger than the server is willing or able to process.

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Appendix A - Disconnection Causes

Disconnection Cause

Description

SIP client error 414

The endpoint sent a SIP Client Error 414 (Request-URI Too Long)
response.
The server is refusing to service the request because the Request-URI
is longer than the server is willing to interpret.

SIP client error 420

The endpoint sent a SIP Client Error 420 (Bad Extension) response.
The server did not understand the protocol extension specified in a
Require header field.

SIP client error 481

The endpoint sent a SIP Client Error 481 (Call/Transaction Does Not
Exist) response.

SIP client error 482

The endpoint sent a SIP Client Error 482 (Loop Detected) response.

SIP client error 483

The endpoint sent a SIP Client Error 483 (Too Many Hops) response.

SIP client error 484

The endpoint sent a SIP Client Error 484 (Address Incomplete)
response.
The server received a request with a To address or Request-URI that
was incomplete.

SIP client error 485

The endpoint sent a SIP Client Error 485 (Ambiguous) response.
The address provided in the request (Request-URI) was ambiguous.

SIP client error 488

The endpoint sent a SIP Client Error 488 (Not Acceptable Here)
response.

SIP forbidden

The SIP server rejected the request.
The server understood the request, but is refusing to fulfill it.

SIP global failure 603

A SIP Global Failure 603 (Decline) response was returned.
The participant's endpoint was successfully contacted, but the
participant explicitly does not wish to or cannot participate.

SIP global failure 604

A SIP Global Failure 604 (Does Not Exist Anywhere) response was
returned.
The server has authoritative information that the user indicated in the
Request-URI does not exist anywhere.

SIP global failure 606

A SIP Global Failure 606 (Not Acceptable) response was returned.

SIP gone

The requested resource is no longer available at the Server and no
forwarding address is known.

SIP moved permanently

The endpoint moved permanently. The user can no longer be found at
the address in the Request-URI.

SIP moved temporarily

The remote endpoint moved temporarily.

SIP not found

The endpoint was not found.
The server has definitive information that the user does not exist at the
domain specified in the Request-URI.

SIP redirection 300

A SIP Redirection 300 (Multiple Choices) response was returned.

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Appendix A - Disconnection Causes

Disconnection Cause

Description

SIP redirection 305

A SIP Redirection 305 (Use Proxy) response was returned.
The requested resource MUST be accessed through the proxy given
by the Contact field.

SIP redirection 380

A SIP Redirection 380 (Alternative Service) response was returned.
The call was not successful, but alternative services are possible.

SIP remote cancelled call

The endpoint canceled the call.

SIP remote closed call

The endpoint ended the call.

SIP remote stopped responding

The endpoint is not responding.

SIP remote unreachable

The endpoint could not be reached.

SIP request terminated

The endpoint terminated the request.
The request was terminated by a BYE or CANCEL request.

SIP request timeout

The request was timed out.

SIP server error 500

The SIP server sent a SIP Server Error 500 (Server Internal Error)
response.
The server encountered an unexpected condition that prevented it from
fulfilling the request.

SIP server error 501

The SIP server sent a SIP Server Error 501 (Not Implemented)
response.
The server does not support the functionality required to fulfill the
request.

SIP server error 502

The SIP server sent a SIP Server Error 502 (Bad Gateway) response.
The server, while acting as a gateway or proxy, received an invalid
response from the downstream server it accessed in attempting to fulfill
the request.

SIP server error 503

The SIP server sent a SIP Server Error 503 (Service Unavailable)
response.
The server is temporarily unable to process the request due to a
temporary overloading or maintenance of the server.

SIP server error 504

The SIP server sent a SIP Server Error 504 (Server Time-out)
response.
The server did not receive a timely response from an external server it
accessed in attempting to process the request.

SIP server error 505

The SIP server sent a SIP Server Error 505 (Version Not Supported)
response.
The server does not support, or refuses to support, the SIP protocol
version that was used in the request.

SIP temporarily not available

The participant's endpoint was contacted successfully but the
participant is currently unavailable (e.g., not logged in or logged in such
a manner as to preclude communication with the participant).

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Appendix A - Disconnection Causes

Disconnection Cause

Description

SIP remote device did not respond in the
given time frame

The endpoint did not respond in the given time frame.

SIP trans error TCP Invite

A SIP Invite was sent via TCP, but the endpoint was not found.

SIP transport error

Unable to initiate connection with the endpoint.

SIP unauthorized

The request requires user authentication.

SIP unsupported media type

The server is refusing to service the request because the message
body of the request is in a format not supported by the requested
resource for the requested method.

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Appendix B - Active Alarms

Active Alarms
Alarm Code

Alarm Description

A matching activation key is required.
To cancel the upgrade process, reset
the Collaboration Server

The system upgrade requires that a valid activation key be entered. If
none is available, resetting the Collaboration Server will cancel the
upgrade and return the Collaboration Server to the previous version.

A new activation key was loaded.
Reset the system.

A new activation key was loaded:
Reset the MCU.

A new version was installed. Reset the
system.

A new version was installed:
Reset the MCU.

Alarm generated by a Central
Signaling component

A system alert was generated by a component of the Central Signaling.

Alarm generated by an internal
component

A system alert was generated by an internal system component.

Allocation mode was modified
Automatic reset is unavailable in Safe
Mode

The system switches to safe mode if many resets occur during startup. To
prevent additional resets, and allow the system to complete the startup
process the automatic system resets are blocked.

Backup of audit files is required

If the ENABLE_CYCLIC_FILE_SYSTEM_ALARMS is set to YES and a
Cyclic File reaches a file retention time or file storage capacity limit, the
user is alerted that audit files need to be backed up.

Backup of CDR files is required

If the ENABLE_CYCLIC_FILE_SYSTEM_ALARMS is set to YES and a
Cyclic File reaches a file retention time or file storage capacity limit, the
user is alerted that CDR files need to be backed up.

Backup of log files is required

If the ENABLE_CYCLIC_FILE_SYSTEM_ALARMS is set to YES and a
Cyclic File reaches a file retention time or file storage capacity limit, the
user is alerted that log files need to be backed up.

Central signaling component failure

Possible explanations:
• Central signaling component failure; unit type:
[NonComponent\CSMngnt\CSH323\CSSIP]
• Central signaling component failure; unit type: (invalid:
[NonComponent\CSMngnt\CSH323\CSSIP])
• Central signaling component failure - Invalid failure type. Unit id: [id],
Type: [NonComponent\CSMngnt\CSH323\CSSIP], Status:
[Ok\Failed\Recovered]
• Central signaling component failure - Invalid failure type

Central Signaling indicating Faulty
status

Central signaling failure detected in IP Network Service.

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Appendix B - Active Alarms

Active Alarms
Alarm Code

Alarm Description

Central Signaling indicating Recovery
status
Central Signaling startup failure

Central Signaling component is down.

Conference Encryption Error
Configuration of external database did
not complete.

Check the configuration of the external DB.

CPU IPMC software was not updated.

Turn off the MCU and then turn it on.

CPU slot ID not identified

The CPU slot ID required for Ethernet Settings was not provided by the
Shelf Management.

D channel cannot be established
DEBUG mode enabled

Possible explanations:
• System is running in DEBUG mode.
• System DEBUG mode initiated.
In this mode, additional prints are added and Startup and Recovery
Conditions are different then Non Debug Mode.
Change the DEBUG_MODE flag value to NO and reset the Collaboration
Server.

DEBUG mode flags in use

The system is using the DEBUG CFG flags.

DMA not supported by IDE device

Possible explanations:
• DMA (direct memory access) not supported by IDE device:
Incompatible flash card / hard disk being used.
• Flash card / hard drive are not properly connected to the board / one
of the IDE channels is disconnected.
• DMA was manually disabled for testing.

DNS configuration error

Check the DNS configuration.

DNS not configured in IP Network
Service

Configure the DNS in the IP Network Services.

Encryption Server Error. Failed to
generate the encryption key

FIPS 140 test failed while generating the new encryption key.

Error in external database certificate
Error reading MCU time

Failed to read MCU time configuration file ([status]).
Manually configure the MCU Time in the Collaboration Server Web Client
or RMX Manager Manager application.

eUserMsgCode_Cs_EdgeServerDnsF
ailed
eUserMsgCode_Cs_SipTLS_Certificat
eHasExpired

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Appendix B - Active Alarms

Active Alarms
Alarm Code

Alarm Description

eUserMsgCode_Cs_SipTLS_Certificat
eSubjNameIsNotValid_Or_DnsFailed
eUserMsgCode_Cs_SipTLS_Certificat
eWillExpireInLessThanAWeek
eUserMsgCode_Cs_SipTLS_FailedTo
LoadOrVerifyCertificateFiles
eUserMsgCode_Cs_SipTLS_Registrat
ionHandshakeFailure
eUserMsgCode_Cs_SipTLS_Registrat
ionServerNotResponding
Event Mode Conferencing resources
deficiency due to inappropriate license.
Please install a new license
External NTP servers failure

The MCU could not connect to any of the defined NTP server for
synchronization due to the remote server error or network error or
configuration error.
Change the configuration of the NTP server.

Failed to access DNS server

Failed to access DNS server.

Failed to configure the Media card IP
address

Possible reasons for the failure:
• Failure type: [OK Or Not supported.
• Does not exist Or IP failure.
• Duplicate IP Or DHCP failure.
• VLAN failure Or Invalid: [status_Number].

Failed to configure the Users list in
Linux

The authentication process did not start.
Use the Restore to factory Defaults to recover.

Failed to connect to application server

Possible reasons for the failure:
• Failed to connect to application server:
• Failed to establish connection to server, url = [url].

Failed to connect to recording device

The MCU could not connect to the defined recording device due to
configuration error or network error.

Failed to connect to SIP registrar

Cannot establish connection with SIP registrar.

Failed to create Default Profile

Possible reasons for the failure:
• Failed to validate the default Profile.
• Failed to add the default Profile.
Possible action:
• Restore the Collaboration Server configuration from the Backup.
• Use the Non-Comprehensive Restore To Factory Defaults operation.

Failed to initialize system base mode

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Appendix B - Active Alarms

Active Alarms
Alarm Code

Alarm Description

Failed to initialize the file system

Possible reasons for the failure:
• Failed to initialize the file system.
• Failed to initialize the file system and create the CDR index.
Reset the MCU.

Failed to open Users list file

Restore the MCU configuration or re-define the user.

Failed to register with DNS server

Check the DNS configuration.

Failed to subscribe with the OCS,
therefore the A/V Edge Server URI
was not received
Failure in initialization of SNMP agent.
Fallback version is being used

Fallback version is being used. Restore current version.
Version being used: [running version]; Current version: [current version].

Fan Problem Level Critical
Fan Problem Level Major
File error

Possible reasons for the file error:
• XML file does not exist [file name]; Error no: [error number].
• Not authorized to open XML file [file name]; Error no: [error number].
• Unknown problem in opening XML file [file name]; Error no: [error
number].
• Failed to parse XML file [file name].

File system scan failure

File system scan failure: Failed to scan [file system path].
Multiple occurrences may point to a hardware problem.
System is functioning.

File system space shortage

File system space shortage:
Out of file system space in [file system path]; Free space: [free space
percentage]% ([free space] Blocks) - Minimum free space required:
[minimum free space percentage]% ([minimum free space] Blocks).

FIPS 140 failure
FIPS 140 test result not received

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Appendix B - Active Alarms

Active Alarms
Alarm Code

Alarm Description

Gatekeeper failure

Possible reasons for the Gatekeeper failure:
• Failed to register to alternate Gatekeeper.
• Gatekeeper discovery state.
- Check GK IP address (GUI, ping)
• Gatekeeper DNS Host name not found.
• Gatekeeper Registration Timeout.
• Gatekeeper rejected GRQ due to invalid revision.
• Gatekeeper rejected GRQ due to resource unavailability.
• Gatekeeper rejected GRQ due to Terminal Exclusion.
• Gatekeeper rejected GRQ due to unsupported feature.
• Gatekeeper rejected GRQ. Reason 18.
• Gatekeeper rejected RRQ due to Discovery Required.
• Gatekeeper rejected RRQ due to duplicate alias.
- Check duplicate in aliases or in prefixes
• Gatekeeper rejected RRQ due to Generic Data.
• Gatekeeper rejected RRQ due to invalid alias.
• Gatekeeper rejected RRQ due to invalid call signaling address.
• Gatekeeper rejected RRQ due to invalid endpoint ID.
• Gatekeeper rejected RRQ due to invalid RAS address.
• Gatekeeper rejected RRQ due to invalid revision.
• Gatekeeper rejected RRQ due to invalid state.
• Gatekeeper rejected RRQ due to invalid terminal alias.
• Gatekeeper rejected RRQ due to resource unavailability.
• Gatekeeper rejected RRQ due to Security Denial.
• Gatekeeper rejected RRQ due to terminal type.
• Gatekeeper rejected RRQ due to unsupported Additive Registration.
• Gatekeeper rejected RRQ due to unsupported feature.
• Gatekeeper rejected RRQ due to unsupported QOS transport.
• Gatekeeper rejected RRQ due to unsupported transport.
• Gatekeeper rejected RRQ. Full registration required.
• Gatekeeper rejected RRQ. Reason 18.
• Gatekeeper Unregistration State.
• Registration succeeded.
Check the Gatekeeper configuration.

GUI System configuration file is invalid
xml file

The XML format of the system configuration file that contains the user
interface settings is invalid.

Hard disk error

Hard disk not responding.

Hot Backup: Master-Slave
configuration conflict.

Possible reasons:
• When both the MCUs are configured as Master or as Slave
• The slave Collaboration Server is defined with the same IP as the
Master.

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Appendix B - Active Alarms

Active Alarms
Alarm Code

Alarm Description

Hot backup: Network issue
Hot Backup: Paired MCU is
unreachable.
Initialization of ice stack failed
Insufficient resources

The number of resources in the license is higher than the actual system
resources.
Check to make sure sufficient CPU cores are allocated in the Virtual
Machine.

Insufficient UDP Ports

When defining fixed port, the number of defined UDP ports is lower than
the required ports.
Configure additional ports.

Internal System configuration during
startup

System configuration during startup.
Wait until Collaboration Server startup is completed.

Invalid System Configuration
IP addresses of Signaling Host and
Control Unit are the same

IP addresses of Signaling Host and Control Unit are identical.
Assign different IP addresses to the Signaling Host and Control Unit.

IP Network Service added
IP Network Service configuration
modified

IP Network Service was modified.
Reset the MCU.

IP Network Service deleted

IP Network Service was deleted.
Reset the MCU.

IP Network Service not found

IP Service not found in the Network Services list.
Configure the IP Network Service.

IPMC software upgrade in component
IPS 140 test result not received
LDAP TLS: Failed to connect to OSCP
responder
License not found

Polycom®, Inc.

Possible causes:
• The Central Signaling component could not find the IP Services after
startup.
• During Startup, the resources did not get the License required to utilize
their Units.
Possible action:
• Configure IP service if not configured.
• Reset the MCU.
• Change the license

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Appendix B - Active Alarms

Active Alarms
Alarm Code

Alarm Description

Management Network not configured

Configure the Management Network.

Missing Central Signaling configuration

Configure the central signaling.

Missing Central Signaling IP
configuration
MPL startup failure. Authentication not
received.

Authentication was not received from Switch.
Check the switch card.

MPL startup failure. Management
Network configuration not received.

Management Network message was not received.
Check the Switch card.

Network interface is not configured.
New interface need to be chosen
Network traffic capture is on
New certificate for CS need
Collaboration Server reset to take
effect
No default IVR Service in IVR Services
list

No default IVR Service in IVR Services list.
Ensure that one conference IVR Service and one EQ IVR Service are set
as default.

No IP Network Services defined

IP Network Service parameters missing.
Configure the IP Network Service.

No LAN connection
No response from Central Signaling

No connection with central signaling.

No RTM-LAN or RTM-ISDN installed.
One of these cards must be installed in
the RealPresence Collaboration
Server (RMX) 4000
No usable unit for audio controller

No media card is installed, or the media card installed is not functioning.
Install the appropriate media card.

OCS Registration failed
Password expiration warning
Please install a newer version
Port configuration was modified
Power off
Power Problem Level Critical
Power Problem Level Major

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Appendix B - Active Alarms

Active Alarms
Alarm Code

Alarm Description

Product activation failure

Assign a new activation key.

Product Type mismatch. System is
restarting.

The user is alerted to a mismatch between the product type that is stored
in MCU software and the product type received from another system
component.
In such a case the system is automatically restarted.

Received Notification failed
Recording device has disconnected
unexpectedly
Requested changes to the certification
repository were not completed.
Repository must be updated to
implement these changes.
Resource process failed to request the
Meeting Room list during startup.

Without the Meeting Rooms list, the system cannot allocate the
appropriate dial numbers, Conference ID etc. and therefore cannot run
conferences.

Restore Failed

Restoring the system configuration has failed as the system could not
locate the configuration file in the selected path, or could not open the
file.

Restore Succeeded

Restoring the system configuration has succeeded. Reset the MCU.

Restoring Factory Defaults. Default
system settings will be restored once
Reset is completed

Default system settings will be restored once Reset is completed.

Collaboration Server fails to connect to
Active Directory server.
Collaboration Server is uploading the
version file. To cancel the upload and
the upgrade, reset the Collaboration
Server
Collaboration Server user/password
list will be reset
Secured SIP communication failed

Error status (408) received from SIP proxy.

Security mode failed. Certificate has
expired.
Security mode failed. Certificate host
name does not match the
Collaboration Server host name.
Security mode failed. Certificate is
about to expire.

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Appendix B - Active Alarms

Active Alarms
Alarm Code

Alarm Description

Security mode failed. Certificate not
yet valid.
Security mode failed. Error in
certificate file.
Service Request failed
SIP registrations limit reached

SIP registrations limit reached.

SIP TLS: Certificate has expired

The current TLS certificate files have expired and must be replaced with
new files.

SIP TLS: Certificate is about to expire

The current TLS certificate files will expire shortly and will have to be
replaced to ensure the communication with the OCS.

SIP TLS: Certificate subject name is
not valid or DNS failed to resolve this
name

This alarm is displayed if the name of the Collaboration Server in the
certificate file is different from the FQDN name defined in the OCS.

SIP TLS: Failed to load or verify
certificate files

This alarm indicates that the certificate files required for SIP TLS could
not be loaded to the Collaboration Server. Possible causes are:
• Incorrect certificate file name. Only files with the following names can
be loaded to the system: rootCA.pem, pkey.pem, cert.pem and
certPassword.txt
• Wrong certificate file type. Only files of the following types can be
loaded to the system: rootCA.pem, pkey.pem and cert.pem and
certPassword.txt
• The contents of the certificate file does not match the system
parameters

SIP TLS: Registration handshake
failure

This alarm indicates a mismatch between the security protocols of the
OCS and the Collaboration Server, preventing the Registration of the
Collaboration Server to the OCS.

SIP TLS: Registration server not
responding

This alarm is displayed when the Collaboration Server does not receive a
response from the OCS to the registration request in the expected time
frame. Possible causes are:
• The Collaboration Server FQDN name is not defined in the OCS pool,
or is defined incorrectly.
• The time frame for the expected response was too short and it will be
updated with the next data refresh. The alarm may be cleared
automatically the next time the data is refreshed.
• The Collaboration Server FQDN name is not defined in the DNS
server. Ping the DNS using the Collaboration Server FQDN name to
ensure that the Collaboration Server is correctly registered to the DNS.

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Appendix B - Active Alarms

Active Alarms
Alarm Code

Alarm Description

SIP TLS: Registration transport error

This alarm indicates that the communication with the SIP server cannot
be established. Possible causes are:
• Incorrect IP address of the SIP server
• The SIP server listening port is other than the one defined in the
system
• The OCS services are stopped

Software upgrade in component
SSH is enabled
SWITCH not responding

Check the Switch card.

System Cards MPM Plus mode are not
supported in Event mode
System configuration changed. Please
reset the MCU
System Configuration modified

System configuration flags were modified.
Reset the MCU.

System resources of Audio ports
usage has exceeded Port Gauge
threshold
System resources of Video ports
usage has exceeded Port Gauge
threshold
System resources usage has
exceeded Port Gauge threshold
Temperature Level - Critical

Possible explanations:
• Temperature has reached a critical level.

Temperature Level - Major

Possible explanations:
• Temperature has reached a problematic level and requires attention.

The Log file system is disabled
because of high system CPU usage
The MCCF channel is not connected
The software contains patch(es)

The software contains patch(es).

Unable to connect to Exchange Server.
User Name SUPPORT cannot be used
in Enhanced Security Mode
Version upgrade is in progress

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Appendix B - Active Alarms

Active Alarms
Alarm Code

Alarm Description

Voltage problem

Possible reasons for the problem:
• Card voltage problem.
• Voltage problem

Warning: Upgrade started and SAFE
Upgrade protection is turned OFF
Yellow Alarm

Polycom®, Inc.

Problem sending/receiving data from/to network.
Check the cables.

645

Appendix C - CDR Fields, Unformatted
File
The CDR (Call Detail Records) utility is used to retrieve conference information to a file. The CDR utility can
retrieve conference information to a file in both formatted and unformatted formats.
Unformatted CDR files contain multiple records. The first record in each file contains information about the
conference in general, such as the conference name and start time. The remaining records each contain
information about one event that occurred during the conference, such as a participant connecting to the
conference, or a user extending the length of the conference. The first field in each record identifies the
event type, and this is followed by values containing information about the event. The fields are separated
by commas.
Formatted files contain basically the same information as unformatted files, but with the field values
replaced by descriptions. Formatted files are divided into sections, each containing information about one
conference event. The first line in each section is a title describing the type of event, and this is followed by
multiple lines, each containing information about the event in the form of a descriptive field name and value.
The field names and values in the formatted file will appear in the language being used for the
Collaboration Server Web Client user interface at the time when the CDR information is retrieved.
The value of the fields that support Unicode values, such as the info fields, will be stored in the CDR
file in UTF8. The application that reads the CDR file must support Unicode.

The MCU sends the entire CDR file via API or HTTP, and the Collaboration Server or external application
does the processing and sorting. The Collaboration Server ignores events that it does not recognize, that
is, events written in a higher version that do not exist in the current version. Therefore, to enable
compatibility between versions, instead of adding new fields to existing events, new fields are added as
separate events, so as not to affect the events from older versions. This allows users with lower versions to
retrieve CDR files that were created in higher versions.
This appendix describes the fields and values in the unformatted CDR records.
Although the formatted files contain basically the same information, in a few instances a single field in
the unformatted file is converted to multiple lines in the formatted file, and in other cases, multiple
fields in the unformatted file are combined into one line in the formatted file.
In addition, to enable compatibility for applications that were written for the MGC family, the
unformatted file contains fields that were supported by the MGC family, but are not supported by the
Collaboration Server, whereas these fields are omitted from the formatted file.

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Appendix C - CDR Fields, Unformatted File

The Conference Summary Record
The conference summary record (the first record in the unformatted CDR file) contains the following fields
:Conference Summary Record Fields

Field

Description

File Version

The version of the CDR utility that created the file.

Conference Routing
Name

The Routing Name of the conference.

Internal Conference ID

The conference identification number as assigned by the system.

Reserved Start Time

The time the conference was scheduled to start in Greenwich Mean Time (GMT). The
reservation time of a reservation that was started immediately or of an ongoing
conference is the same as the Actual Start Time.

Reserved Duration

The amount of time the conference was scheduled to last.

Actual Start Time

The actual time the conference started in GMT.

Actual Duration

The actual conference duration.

Status

The conference status code as follows:
1 - The conference is an ongoing conference.
2 - The conference was terminated by a user.
3 - The conference ended at the scheduled end time.
4 - The conference ended automatically because no participants joined the conference
for a predefined time period, or all the participants disconnected from the conference and
the conference was empty for a predefined time period.
5 - The conference never started.
6 - The conference could not start due to a problem.
8 - An unknown error occurred.
9 - The conference was terminated by a participant using DTMF codes.
Note: If the conference was terminated by an MCU reset, this field will contain the value
1 (ongoing conference).

File Name

The name of the conference log file.

GMT Offset Sign

Indicates whether the GMT Offset is positive or negative. The possible values are:
0 - Offset is negative. GMT Offset will be subtracted from the GMT Time.
1 - Offset is positive. GMT Offset will be added to the GMT Time.

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Field

Description

GMT Offset

The time zone difference between Greenwich and the Collaboration Server’s physical
location in hours and minutes.
Together with the GMT Offset Sign field the GMT Offset field is used to define the
Collaboration Server local time. For example, if the GMT Offset Sign is 0 and GMT Offset
is 3 hours then the time zone of the Collaboration Server’s physical location is -3, which
will be subtracted from the GMT time to determine the local time. However, if the GMT
Offset Sign is 1 and GMT Offset is 4 hours then the time zone of the Collaboration
Server’s physical location is +4, which will be added to the GMT time to determine the
local time.

File Retrieved

Indicates if the file has been retrieved and saved to a formatted file, as follows:
0 - No
1 - Yes

Event Records
The event records, that is, all records in the unformatted file except the first record, contain standard fields,
such as the event type code and the time stamp, followed by fields that are event specific.
The event fields are separated by commas. Two consecutive commas with nothing between them (,,), or a
comma followed immediately by a semi-colon (,;), indicates an empty field, as in the example below:

Standard Event Record Fields
All event records start with the following fields:
● The CDR event type code. For a list of event type codes and descriptions, refer to CDR Event Types.
● The event date.
● The event time.
● The structure length. This field is required for compatibility purposes, and always contains the value
0.

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Event Types
The table below contains a list of the events that can be logged in the CDR file, and indicates where to find
details of the fields that are specific to that type of event.
The event code identifies the event in the unformatted CDR file, and the event name identifies the
event in the formatted CDR file.

CDR Event Types

Event Code

Event Name

Description

1

CONFERENCE
START

The conference started.
For more information about the fields, see Event Fields for Event 1 CONFERENCE START.
Note: There is one CONFERENCE START event per conference. It is always
the first event in the file, after the conference summary record. It contains
conference details, but not participant details.

2

CONFERENCE
END

The conference ended.
For more information about the fields, see Event Fields for Event 2 CONFERENCE END.
Note: There is one CONFERENCE END event per conference, and it is
always the last event in the file.

7

PARTICIPANT
DISCONNECTED

A participant disconnected from the conference.
For more information about the fields, see Event Fields for Event 7 PARTICIPANT DISCONNECTED.

10

DEFINED
PARTICIPANT

Information about a defined participant, that is, a participant who was added to
the conference before the conference started.
For more information about the fields, see Event Fields for Events 10, 101,
105 - DEFINED PARTICIPANT, USER ADD PARTICIPANT, USER UPDATE
PARTICIPANT.
Note: There is one event for each participant defined before the conference
started.

15

H323 CALL
SETUP

Information about the IP address of the participant.
For more information about the fields, see Event fields for Event 15 - H323
CALL SETUP.

17

H323
PARTICIPANT
CONNECTED

An H.323 participant connected to the conference.
For more information about the fields, see Event Fields for Events 17, 23 H323 PARTICIPANT CONNECTED, SIP PARTICIPANT CONNECTED.

18

NEW
UNDEFINED
PARTICIPANT

A new undefined participant joined the conference.
For more information about the fields, see Event Fields for Event 18 - NEW
UNDEFINED PARTICIPANT.

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Event Code

Event Name

Description

20

BILLING CODE

A billing code was entered by a participant using DTMF codes.
For more information about the fields, see Event Fields for Event 20 - BILLING
CODE.

21

SET
PARTICIPANT
DISPLAY NAME

A user assigned a new name to a participant, or an end point sent its name.
For more information about the fields, see Event Fields for Event 21 - SET
PARTICIPANT DISPLAY NAME.

22

DTMF CODE
FAILURE

An error occurred when a participant entered a DTMF code.
For more information about the fields, see Event Fields for Event 22 - DTMF
CODE FAILURE.

23

SIP
PARTICIPANT
CONNECTED

A SIP participant connected to the conference.
For more information about the fields, see Event Fields for Events 17, 23 H323 PARTICIPANT CONNECTED, SIP PARTICIPANT CONNECTED.

26

RECORDING
LINK

A recording event, such as recording started or recording resumed, occurred.
For more information about the fields, see Event fields for Event 26 RECORDING LINK.

28

SIP PRIVATE
EXTENSIONS

Contains SIP Private Extensions information.
For more information about the fields, see Event Fields for Event 28 - SIP
PRIVATE EXTENSIONS.

30

GATEKEEPER
INFORMATION

Contains the gatekeeper caller ID, which makes it possible to match the CDR
in the gatekeeper and in the MCU.
For more information about the fields, see Event Fields for Event 30 GATEKEEPER INFORMATION.

31

PARTICIPANT
CONNECTION
RATE

Information about the line rate of the participant connection. This event is
added to the CDR file each time the endpoint changes its connection bit rate.
For more information about the fields, see Event fields for Event 31 PARTICIPANT CONNECTION RATE.

33

PARTY CHAIR
UPDATE

Participants connect to the conferences as standard participants and they are
designated as chairpersons either by entering the chairperson password
during the IVR session upon connection, or while participating in the
conference using the appropriate DTM code.
For more information about the fields, see Event fields for Event 33 - PARTY
CHAIR UPDATE.

34

PARTICIPANT
MAXIMUM
USAGE
INFORMATION

This event includes information of the maximum line rate, maximum resolution
and maximum frame rate used by H.323 or SIP participant during the
conference.

35

SVC SIP
PARTICIPANT
CONNECTED

An SVC user connected over SIP.
For more information about the fields, see Event Fields for Event 35 - SVC
SIP PARTICIPANT CONNECTED.

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Event Code

Event Name

Description

100

USER
TERMINATE
CONFERENCE

A user terminated the conference.
For more information about the fields, see Event Fields for Event 100 - USER
TERMINATE CONFERENCE.

101

USER ADD
PARTICIPANT

A user added a participant to the conference during the conference.
For more information about the fields, see Event Fields for Events 10, 101,
105 - DEFINED PARTICIPANT, USER ADD PARTICIPANT, USER UPDATE
PARTICIPANT.

102

USER DELETE
PARTICIPANT

A user deleted a participant from the conference.
For more information about the fields, see Event Fields for Events 102,103,
104 - USER DELETE PARTICIPANT, USER DISCONNECT PARTICIPANT,
USER RECONNECT PARTICIPANT.

103

USER
DISCONNECT
PARTICIPANT

A user disconnected a participant.
For more information about the fields, see Event Fields for Events 102,103,
104 - USER DELETE PARTICIPANT, USER DISCONNECT PARTICIPANT,
USER RECONNECT PARTICIPANT.

104

USER
RECONNECT
PARTICIPANT

A user reconnected a participant who was disconnected from the conference.
For more information about the fields, see Event Fields for Events 102,103,
104 - USER DELETE PARTICIPANT, USER DISCONNECT PARTICIPANT,
USER RECONNECT PARTICIPANT.

105

USER UPDATE
PARTICIPANT

A user updated the properties of a participant during the conference.
For more information about the fields, see Event Fields for Events 10, 101,
105 - DEFINED PARTICIPANT, USER ADD PARTICIPANT, USER UPDATE
PARTICIPANT.

106

USER SET END
TIME

A user modified the conference end time.
For more information about the fields, see Event Fields for Event 106 - USER
SET END TIME.

107

OPERATOR
MOVE PARTY
FROM
CONFERENCE

The participant moved from an Entry Queue to the destination conference or
between conferences.
For more information about the fields, see Event Fields for Events 107 and
109 - OPERATOR MOVE PARTY FROM CONFERENCE and OPERATOR
ATTEND PARTY.

108

OPERATOR
MOVE PARTY
TO
CONFERENCE

The Collaboration Server User moved the participant from an ongoing
conference to another conference.
For more information, see Event Fields for Events 108, 112 - OPERATOR
MOVE PARTY TO CONFERENCE,OPERATOR ATTEND PARTY TO
CONFERENCE.

109

OPERATOR
ATTEND PARTY

The Collaboration Server User moved the participant to the Operator
conference.
For more information, see Event Fields for Events 107 and 109 - OPERATOR
MOVE PARTY FROM CONFERENCE and OPERATOR ATTEND PARTY.

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Event Code

Event Name

Description

111

OPERATOR
BACK TO
CONFERENCE
PARTY

The Collaboration Server User moved the participant back to his Home
(source) conference.
For more information, see Event Fields for Event 111 - OPERATOR BACK TO
CONFERENCE PARTY.

112

OPERATOR
ATTEND PARTY
TO
CONFERENCE

The Collaboration Server User moved the participant from the Operator
conference to another conference.
For more information, see Event Fields for Events 108, 112 - OPERATOR
MOVE PARTY TO CONFERENCE,OPERATOR ATTEND PARTY TO
CONFERENCE.

1001

NEW
UNDEFINED
PARTICIPANT
CONTINUE 1

Additional information about a NEW UNDEFINED PARTICIPANT event.
For more information about the fields, see Event Fields for Event 1001 - NEW
UNDEFINED PARTY CONTINUE 1.

2001

CONFERENCE
START
CONTINUE 1

Additional information about a CONFERENCE START event.
For more information about the fields, see Event Fields for Event 2001 CONFERENCE START CONTINUE 1.

2007

PARTICIPANT
DISCONNECTED
CONTINUE 1

Additional information about a PARTICIPANT DISCONNECTED event.
For more information about the fields, see Event Fields for Event 2007 PARTICIPANT DISCONNECTED CONTINUE 1.

2010

DEFINED
PARTICIPANT
CONTINUE 1

Additional information about a DEFINED PARTICIPANT event.
For more information about the fields, see Event Fields for Events 2010, 2011,
2015 - DEFINED PARTICIPANT CONTINUE 1,USER ADD PARTICIPANT
CONTINUE 1, USER UPDATE PARTICIPANT CONTINUE 1.

2011

RESERVED
PARTICIPANT
CONTINUE PV6
ADDRESS

Additional information about a DEFINED PARTICIPANT event that includes
the IPv6 addressing of the defined participant.
For more details, see Event Fields for Events 2011, 2012, and 2016.

2012

RESERVED
PARTICIPANT
CONTINUE 2

Additional information about a DEFINED PARTICIPANT event.
For more information about the fields, see Event Fields for Event 2011 DEFINED PARTICIPANT CONTINUE 2, Event 2012 - USER ADD
PARTICIPANT CONTINUE 2, Event 2016 - USER UPDATE PARTICIPANT
CONTINUE 2.

2101

USER ADD
PARTICIPANT
CONTINUE 1

Additional information about a USER ADD PARTICIPANT event.
For more information about the fields, see Event Fields for Events 2010, 2011,
2015 - DEFINED PARTICIPANT CONTINUE 1,USER ADD PARTICIPANT
CONTINUE 1, USER UPDATE PARTICIPANT CONTINUE 1.

2102

USER ADD
PARTICIPANT
CONTINUE 2

Additional information about a USER ADD PARTICIPANT event.
For more information about the fields, see Event Fields for Event 2011 DEFINED PARTICIPANT CONTINUE 2, Event 2012 - USER ADD
PARTICIPANT CONTINUE 2, Event 2016 - USER UPDATE PARTICIPANT
CONTINUE 2.

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Event Code

Event Name

Description

2105

USER UPDATE
PARTICIPANT
CONTINUE 1

Additional information about a USER UPDATE PARTICIPANT event.
For more information about the fields, see Event Fields for Events 2010, 2011,
2015 - DEFINED PARTICIPANT CONTINUE 1,USER ADD PARTICIPANT
CONTINUE 1, USER UPDATE PARTICIPANT CONTINUE 1.

2106

USER UPDATE
PARTICIPANT
CONTINUE 2

Additional information about a USER UPDATE PARTICIPANT event.
For more information about the fields, see Event Fields for Event 2011 DEFINED PARTICIPANT CONTINUE 2, Event 2012 - USER ADD
PARTICIPANT CONTINUE 2, Event 2016 - USER UPDATE PARTICIPANT
CONTINUE 2.

3010

PARTICIPANT
INFORMATION

The contents of the participant information fields.
For more information about the fields, see Event Fields for Event 3010 PARTICIPANT INFORMATION.

5001

CONFERENCE
START
CONTINUE 4

Additional information about a CONFERENCE START event.
For more information about the fields, see Event Fields for Event 5001 CONFERENCE START CONTINUE 4.
Note: An additional CONFERENCE START CONTINUE 4 event will be
written to the CDR each time the value of one of the following conference
fields is modified:
• Conference Password
• Chairperson Password
• Info1, Info2 or Info3
• Billing Info
These additional events will only contain the value of the modified field.

6001

CONFERENCE
START
CONTINUE 5

Additional information about a CONFERENCE START event.
For more information about the fields, see Event Fields for Event 6001 CONFERENCE START CONTINUE 5.

11001

CONFERENCE
START
CONTINUE 10

Additional information about a CONFERENCE START event. This event
contains the Display Name.
For more information about the fields, see Event Fields for Event 11001 CONFERENCE START CONTINUE 10.

This list only includes events that are supported by the Collaboration Server. For a list of MGC
Manager events that are not supported by the Collaboration Server, see MGC Manager Events that
are not Supported by the Collaboration Server.

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Event Specific Fields
The following tables describe the fields which are specific to each type of event.
Some fields that were supported by the MGC Manager, are not supported by the Collaboration
Server. In addition, for some fields the Collaboration Server has a fixed value, whereas the MGC
Manager supported multiple values. For more information about the MGC Manager fields and values,
see the MGC Manager User’s Guide Volume II, Appendix A.

Event Fields for Event 1 - CONFERENCE START
Table 1-1
Field

Description

Dial-Out Manually

Indicates whether the conference was a dial-out manually conference or not.
Currently the only value is:
0 - The conference was not a dial-out manually conference, that is, the MCU initiates the
communication with dial-out participants, and the user does not need to connect them
manually.

Auto Terminate

Indicates whether the conference was set to end automatically if no participant joins the
conference for a predefined time period after the conference starts, or if all participants
disconnect from the conference and the conference is empty for a predefined time period.
Possible values are:
0 - The conference was not set to end automatically.
1 - The conference was set to end automatically.

Line Rate

The conference line rate, as follows:
0 - 64 kbps
6 - 384 kbps
12 - 1920 kbps
13 - 128 kbps
15 - 256 kbps
23 - 512 kbps
24 - 768 kbps
26 - 1152 kbps
29 - 1472 kbps
32 - 96 kbps

Line Rate (cont.)

33 - 1024 kbps
34 - 4096 kbps

Restrict Mode

Not supported.
Always contains the value 0.

Audio Algorithm

The audio algorithm.
Currently the only value is:
255 - Auto

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Table 1-1
Field

Description

Video Session

The video session type.
Currently the only value is:
3 - Continuous Presence

Video Format

The video format.
Currently the only value is:
255-Auto

CIF Frame Rate

The CIF frame rate.
Currently the only value is:
255 -Auto

QCIF Frame Rate

The QCIF frame rate:
Currently the only value is:
255 -Auto

LSD Rate

Not supported.
Always contains the value 0.

HSD Rate

Not supported.
Always contains the value 0.

T120 Rate

Not supported.
Always contains the value 0.

Event Fields for Event 2001 - CONFERENCE START CONTINUE 1
Table 1-2
Field

Description

Audio Tones

Not supported.
Always contains the value 0.

Alert Tone

Not supported.
Always contains the value 0.

Talk Hold Time

The minimum time that a speaker has to speak to become the video source.
The value is in units of 0.01 seconds.
Currently the only value is 150, which indicates a talk hold time of 1.5 seconds.

Audio Mix Depth

The maximum number of participants whose audio can be mixed.
Soft MCU: AVC, 4; SVC, 5.

Operator
Conference

Not supported.
Always contains the value 0.

Video Protocol

The video protocol.
Currently the only value is:
255 -Auto

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Table 1-2
Field

Description

Meet Me Per
Conference

Indicates the Meet Me Per Conference setting.
Currently the only value is:
1 - The Meet Me Per Conference option is enabled, and dial-in participants can join the
conference by dialing the dial-in number.

Number of Network
Services

Not supported.
Always contains the value 0.

Chairperson
Password

The chairperson password for the conference.
An empty field “” means that no chairperson password was assigned to the conference.

Chair Mode

Not supported.
Always contains the value 0.

Cascade Mode

The cascading mode.
Currently the only value is:
0 - None

Master Name

Not supported.
This field remains empty.

Minimum Number
of Participants

The number of participants for which the system reserved resources. Additional participants
may join the conference without prior reservation until all the resources are utilized.
Currently the only value is 0.

Allow Undefined
Participants

Indicates whether or not undefined dial-in participants can connect to the conference.
Currently the only value is:
1 - Undefined participants can connect to the conference

Time Before First
Participant Joins

Note: This field is only relevant if the Auto Terminate option is enabled.
Indicates the number of minutes that should elapse from the time the conference starts,
without any participant connecting to the conference, before the conference is automatically
terminated by the MCU.

Time After Last
Participant Quits

Note: This field is only relevant if the Auto Terminate option is enabled.
Indicates the number of minutes that should elapse after the last participant has
disconnected from the conference, before the conference is automatically terminated by the
MCU.

Conference Lock
Flag

Not supported.
Always contains the value 0.

Maximum Number
of Participants

The maximum number of participants that can connect to the conference at one time.
The value 65535 (auto) indicates that as many participants as the MCU’s resources allow
can connect to the conference, up to the maximum possible for the type of conference.

Audio Board ID

Not supported.
Always contains the value 65535.

Audio Unit ID

Not supported.
Always contains the value 65535.

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Table 1-2
Field

Description

Video Board ID

Not supported.
Always contains the value 65535.

Video Unit ID

Not supported.
Always contains the value 65535.

Data Board ID

Not supported.
Always contains the value 65535.

Data Unit ID

Not supported.
Always contains the value 65535.

Message Service
Type

The Message Service type.
Currently the only value is:
3 - IVR

Conference IVR
Service

The name of the IVR Service assigned to the conference.
Note: If the name of the IVR Service contains more than 20 characters, it will be truncated to
20 characters.

Lecture Mode Type

Indicates the type of Lecture Mode, as follows:
0 - None
1 - Lecture Mode
3 - Presentation Mode

Lecturer

Note: This field is only relevant if the Lecture Mode Type is Lecture Mode.
The name of the participant selected as the conference lecturer.

Time Interval

Note: This field is only relevant if Lecturer View Switching is enabled.
The number of seconds a participant is to be displayed in the lecturer window before
switching to the next participant.
Currently the only value is 15.

Lecturer View
Switching

Note: This field is only relevant when Lecture Mode is enabled.
Indicates the lecturer view switching setting, as follows:
0 - Automatic switching between participants is disabled.
1 - Automatic switching between participants is enabled.

Audio Activated

Not supported.
Always contains the value 0.

Lecturer ID

Not supported.
Always contains the value 4294967295.

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Event Fields for Event 5001 - CONFERENCE START CONTINUE 4
Table 1-3
Field

Description

Note: When this event occurs as the result of a change to the value of one of the event fields, the event will only
contain the value of the modified field. All other fields will be empty.
Conference ID

The conference ID.

Conference Password

The conference password.
An empty field “” means that no conference password was assigned to the conference.

Chairperson Password

The chairperson password.
An empty field “” means that no chairperson password was assigned to the conference.

Info1
Info2
Info3

The contents of the conference information fields.
These fields enable users to enter general information for the conference, such as the
company name, and the contact person’s name and telephone number.
The maximum length of each field is 80 characters.

Billing Info

The billing code.

Event Fields for Event 6001 - CONFERENCE START CONTINUE 5
Table 1-4
Field

Description

Encryption

Indicates the conference encryption setting, as follows:
0 - The conference is not encrypted.
1 -The conference is encrypted.

Event Fields for Event 11001 - CONFERENCE START CONTINUE 10
Table 1-5
Field

Description

Display Name

The Display Name of the conference.

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Event Fields for Event 2 - CONFERENCE END
Table 1-6
Field

Description

Conference End
Cause

Indicates the reason for the termination of the conference, as follows:
1 - The conference is an ongoing conference or the conference was terminated by an MCU
reset.
2 - The conference was terminated by a user.
3 - The conference ended at the scheduled end time.
4 - The conference ended automatically because no participants joined the conference for a
predefined time period, or all the participants disconnected from the conference and the
conference was empty for a predefined time period.
5 - The conference never started.
6 - The conference could not start due to a problem.
8 - An unknown error occurred.
9 - The conference was terminated by a participant using DTMF codes.

Event fields for Event 3 - ISDN/PSTN CHANNEL CONNECTED
Event fields for Event 4 - ISDN/PSTN CHANNEL DISCONNECTED
Event fields for Event 5 - ISDN/PSTN PARTICIPANT CONNECTED
Event Fields for Event 7 - PARTICIPANT DISCONNECTED
Table 1-9
Field

Description

Participant Name

The name of the participant.

Participant ID

The identification number assigned to the participant by the MCU.

Call Disconnection
Cause

The disconnection cause. For more information about possible values, see Disconnection
Cause Values.

Q931 Disconnect
Cause

If the disconnection cause is “No Network Connection” or “Participant Hang Up”, then this
field indicates the Q931 disconnect cause.

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Event Fields for Event 2007 - PARTICIPANT DISCONNECTED CONTINUE 1
Table 1-10
Field

Description

Rx Synchronization
Loss

The number of times that the general synchronization of the MCU was lost.

Tx Synchronization
Loss

The number of times that the general synchronization of the participant was lost.

Rx Video
Synchronization
Loss

The number of times that the synchronization of the MCU video unit was lost.

Tx Video
Synchronization
Loss

The number of times that the synchronization of the participant video was lost.

Mux Board ID

Not supported.
Always contains the value 0.

Mux Unit ID

Not supported.
Always contains the value 0.

Audio Codec Board
ID

Not supported.
Always contains the value 0.

Audio Codec Unit
ID

Not supported.
Always contains the value 0.

Audio Bridge Board
ID

Not supported.
Always contains the value 0.

Audio Bridge Unit
ID

Not supported.
Always contains the value 0.

Video Board ID

Not supported.
Always contains the value 0.

Video Unit ID

Not supported.
Always contains the value 0.

T.120 Board ID

Not supported.
Always contains the value 0.

T.120 Unit ID

Not supported.
Always contains the value 0.

T.120 MCS Board
ID

Not supported.
Always contains the value 0.

T.120 MCS Unit ID

Not supported.
Always contains the value 0.

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Table 1-10
Field

Description

H.323 Board ID

Not supported.
Always contains the value 0.

H323 Unit ID

Not supported.
Always contains the value 0.

Event Fields for Events 10, 101, 105 - DEFINED PARTICIPANT, USER ADD PARTICIPANT, USER UPDATE
PARTICIPANT
Table 1-11
Field

Description

User Name

The login name of the user who added the participant to the conference, or updated the
participant properties.

Participant Name

The name of the participant.

Participant ID

The identification number assigned to the participant by the MCU.

Dialing Direction

The dialing direction, as follows:
0 - Dial-out
5 - Dial-in

Bonding Mode

Not supported.
Always contains the value 0.

Number Of
Channels

Not applicable.

Net Channel Width

Not supported.
Always contains the value 0.

Network Service
Name

The name of the Network Service.
An empty field “” indicates the default Network Service.

Restrict

Not supported.
Always contains the value 0.

Audio Only

Indicates the participant’s Audio Only setting, as follows:
0 - The participant is not an Audio Only participant
1 - The participant is an Audio Only participant
255 - Unknown

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Table 1-11
Field

Description

Default Number
Type

Note: This field is only relevant to ISDN/PSTN participants.
The type of telephone number, as follows:
0 - Unknown
1 - International
2 - National
3 - Network specific
4 - Subscriber
6 - Abbreviated
255 - Taken from Network Service, default
Note: For dial-in participants, the only possible value is:
255 - Taken from Network Service

Net Sub-Service
Name

Not supported.
This field remains empty.

Number of
Participant Phone
Numbers

Not applicable.

Number of MCU
Phone Numbers

Not applicable.

Party and MCU
Phone Numbers

Not applicable.

Identification
Method

Note: This field is only relevant to dial-in participants.
The method by which the destination conference is identified, as follows:
1 - Called IP address or alias
2 - Calling IP address or alias

Meet Me Method

Note: This field is only relevant to dial-in participants.
The meet-me per method. Currently the only value is:
3 - Meet-me per participant

Event Fields for Events 2010, 2011, 2015 - DEFINED PARTICIPANT CONTINUE 1,USER ADD PARTICIPANT
CONTINUE 1, USER UPDATE PARTICIPANT CONTINUE 1
Table 1-12
Field

Description

Network Type

The type of network between the participant and the MCU, as follows:
2 - H.323
5 - SIP

H.243 Password

Not supported.
This field remains empty.

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Table 1-12
Field

Description

Chair

Not supported.
Always contains the value 0.

Video Protocol

The video protocol used by the participant, as follows:
1 - H.261
2 - H.263
4 - H.264
255 - Auto

Broadcasting
Volume

The broadcasting volume assigned to the participant.
The value is between 1 (lowest) and 10 (loudest).
Each unit movement increases or decreases the volume by 3 dB.

Undefined
Participant

Indicates whether are not the participant is an undefined participant, as follows:
0 - The participant is not an undefined participant.
2 - The participant is an undefined participant.

Node Type

The node type, as follows:
0 - MCU
1 - Terminal

Bonding Phone
Number

Not applicable.

Video Bit Rate

The video bit rate in units of kilobits per second.
A value of 4294967295 denotes auto, and in this case, the rate is computed by the MCU.

IP Address

The IP address of the participant.
An address of 4294967295 indicates that no IP address was specified for the participant, and
the gatekeeper is used for routing. In all other cases the address overrides the gatekeeper.

Signaling Port

The signaling port used for participant connection.

H.323 Participant
Alias Type/SIP
Participant Address
Type

For H.323 participants, the alias type, as follows:
7 - E164
8 - H.323 ID
13 - Email ID
14 - Participant number
For SIP participants, the address type, as follows:
1 - SIP URI
2 - Tel URL

H.323 Participant
Alias Name/SIP
Participant Address

For H.323 participants:
The participant alias.
The alias may contain up to 512 characters.
For SIP participants:
The participant address.
The address may contain up to 80 characters.

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Event Fields for Event 2011 - DEFINED PARTICIPANT CONTINUE 2, Event 2012 - USER ADD PARTICIPANT
CONTINUE 2, Event 2016 - USER UPDATE PARTICIPANT CONTINUE 2
Table 1-13
Field

Description

Encryption

Indicates the participant’s encryption setting as follows:
0 - The participant is not encrypted.
1 - The participant is encrypted.
2 - Auto. The conference encryption setting is applied to the participant.

Participant Name

The name of the participant.

Participant ID

The identification number assigned to the participant by the MCU.

Event fields for Event 15 - H323 CALL SETUP

Field

Description

Participant Name

The name of the participant.

Participant ID

The identification number assigned to the participant by the MCU.

Connect Initiator

Indicates who initiated the connection, as follows:
0 - MCU
1 - Remote participant
Any other number - Unknown

Min Rate

The minimum line rate used by the participant.
The data in this field should be ignored. For accurate rate information, see CDR event 31.

Max Rate

The maximum line rate achieved by the participant.
The data in this field should be ignored. For accurate rate information, see CDR event 31.

Source Party
Address

The IP address of the calling participant.
A string of up to 255 characters.

Destination Party
Address

The IP address of the called participant.
A string of up to 255 characters.

Endpoint Type

The endpoint type, as follows:
0 - Terminal
1 - Gateway
2 - MCU
3 - Gatekeeper
4 - Undefined

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Event Fields for Events 17, 23 - H323 PARTICIPANT CONNECTED, SIP PARTICIPANT CONNECTED
Table 1-14
Field

Description

Participant Name

The name of the participant.
An empty field “” denotes an unidentified participant or a participant whose name is
unspecified.

Participant ID

The identification number assigned to the participant by the MCU.

Participant Status

The participant status, as follows:
0 -Idle
1 -Connected
2 - Disconnected
3 - Waiting for dial-in
4 – Connecting
5 - Disconnecting
6 - Partially connected. Party has completed H.221 capability exchange
7 - Deleted by a user
8 -Secondary. The participant could not connect the video channels and is connected via
audio only
10 - Connected with problem
11 - Redialing

Capabilities

Not supported.
Always contains the value 0.

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Table 1-14
Field

Description

Remote
Communication
Mode

Not supported.
Always contains the value 0.

Secondary Cause

Note: This field is only relevant if the Participant Status is Secondary.
The cause for the secondary connection (not being able to connect the video channels), as
follows:
0 - Default
11 - The incoming video parameters are not compatible with the conference video
parameters
13 - The conference video settings are not compatible with the endpoint capabilities
14 - The new conference settings are not compatible with the endpoint capabilities
15 - Video stream violation due to incompatible annexes or other discrepancy
16 - Inadequate video resources
17 - When moved to a Transcoding or Video Switching conference, the participant’s video
capabilities are not supported by the video cards
18 - Video connection could not be established
24 - The endpoint closed its video channels
25 - The participant video settings are not compatible with the conference protocol
26 - The endpoint could not re-open the video channel after the conference video mode was
changed
27 - The gatekeeper approved a lower bandwidth than requested
28 - Video connection for the SIP participant is temporarily unavailable
255 - Other

Event Fields for Event 18 - NEW UNDEFINED PARTICIPANT
Table 1-15
Field

Description

Participant Name

The name of the participant.

Participant ID

The identification number assigned to the participant by the MCU.

Dialing Direction

The dialing direction, as follows
0 - Dial-out
5 - Dial-in

Bonding Mode

Not supported.
Always contains the value 0.

Number of
Channels

Not applicable

Net Channel Width

Not supported.
Always contains the value 0.

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Table 1-15
Field

Description

Network Service
Name

The name of the Network Service.
An empty field “” indicates the default Network Service.

Restrict

Not supported.
Always contains the value 0.

Audio Only

Indicates the participant’s Audio Only setting, as follows:
0 - The participant is not an Audio Only participant
1 - The participant is an Audio Only participant
255 - Unknown

Default Number
Type

Not applicable.

Net Sub-Service
Name

Not supported.
This field remains empty.

Number of
Participant Phone
Numbers

Not applicable.

Number of MCU
Phone Numbers

Not applicable.

Party and MCU
Phone Numbers

Not applicable.

Identification
Method

Note: This field is only relevant to dial-in participants.
The method by which the destination conference is identified, as follows:
1 - Called IP address or alias
2 - Calling IP address or alias

Meet Me Method

Note: This field is only relevant to dial-in participants.
The meet-me per method, as follows:
3 - Meet-me per participant

Network Type

The type of network between the participant and the MCU, as follows:
2 - H.323
5 - SIP

H.243 Password

Not supported.
This field remains empty.

Chair

Not supported.
Always contains the value 0.

Video Protocol

The video protocol, as follows:
1 - H.261
2 - H.263
4 - H.264
255 -Auto

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Table 1-15
Field

Description

Broadcasting
Volume

The broadcasting volume assigned to the participant.
The value is between 1 (lowest) and 10 (loudest).
Each unit movement increases or decreases the volume by 3 dB.

Undefined
Participant

Indicates whether are not the participant is an undefined participant, as follows:
0 - The participant is not an undefined participant.
2 - The participant is an undefined participant.

Node Type

The node type, as follows:
0 - MCU
1 - Terminal

Bonding Phone
Number

Note: This field is only relevant to ISDN/PSTN participants.
The phone number for Bonding dial-out calls.
Bonding is a communication protocol that aggregates from two up to thirty 64 Kbps B
channels together, to look like one large bandwidth channel.

Video Bit Rate

The video bit rate in units of kilobits per second.
A value of 4294967295 denotes auto, and in this case, the rate is computed by the MCU.

IP Address

Note: This field is only relevant to IP participants.
The IP address of the participant.
An address of 4294967295 indicates that no IP address was specified for the participant, and
the gatekeeper is used for routing. In all other cases the address overrides the gatekeeper.

Signaling Port

Note: This field is only relevant to IP participants.
The signaling port used for participant connection.
A value of 65535 is ignored by MCU.

H.323 Participant
Alias Type/SIP
Participant Address
Type

For H.323 participants, the alias type, as follows:
7 - E164
8 - H.323 ID
13 - Email ID
14 - Participant number
For SIP participants, the address type, as follows:
1 - SIP URI
2 - Tel URL

H.323 Participant
Alias Name/SIP
Participant Address

For H.323 participants:
The participant alias.
The alias may contain up to 512 characters.
For SIP participants:
The participant address.
The address may contain up to 80 characters.

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Event Fields for Event 1001 - NEW UNDEFINED PARTY CONTINUE 1
Table 1-16
Field

Description

Encryption

Indicates the participant’s encryption setting as follows:
0 - The participant is not encrypted.
1 - The participant is encrypted.
2 - Auto. The conference encryption setting is applied to the participant.

Participant Name

The name of the participant.

Participant ID

The identification number assigned to the participant by the MCU.

Event Fields for Event 20 - BILLING CODE
Table 1-17
Field

Description

Participant Name

The name of the participant who added the billing code.

Participant ID

The identification number, as assigned by the MCU, of the participant who added the billing
code.

Billing Info

The numeric billing code that was added (32 characters).

Event Fields for Event 21 - SET PARTICIPANT DISPLAY NAME
Table 1-18
Field

Description

Participant Name

The original name of the participant, for example, the name automatically assigned to an
undefined participant, such as, “_(000)”.

Participant ID

The identification number assigned to the participant by the MCU.

Display Name

The new name assigned to the participant by the user, or the name sent by the end point.

Event Fields for Event 22 - DTMF CODE FAILURE
Table 1-19
Field

Description

Participant Name

The name of the participant.

Participant ID

The identification number assigned to the participant by the MCU.

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Table 1-19
Field

Description

Incorrect Data

The incorrect DTMF code entered by the participant, or an empty field “” if the participant did
not press any key.

Correct Data

The correct DTMF code, if known.

Failure Type

The type of DTMF failure, as follows:
2 - The participant did not enter the correct conference password.
6 - The participant did not enter the correct chairperson password.
12 - The participant did not enter the correct Conference ID.

Event fields for Event 26 - RECORDING LINK
Table 1-20
Field

Description

Participant Name

The name of the Recording Link participant.

Participant ID

The identification number assigned to the Recording Link participant by the MCU.

Recording
Operation

The type of recording operation, as follows:
0 - Start recording
1 - Stop recording
2 - Pause recording
3 - Resume recording
4 - Recording ended
5 - Recording failed

Initiator

Not supported.

Recording Link
Name

The name of the Recording Link.

Recording Link ID

The Recording Link ID.

Start Recording
Policy

The start recording policy, as follows:
1 - Start recording automatically as soon as the first participant connects to the conference.
2 - Start recording when requested by the conference chairperson via DTMF codes or from
the Collaboration Server Web Client, or when the operator starts recording from the
Collaboration Server Web Client.

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Event Fields for Event 28 - SIP PRIVATE EXTENSIONS
Table 1-21
Field

Description

Participant Name

The name of the participant.

Participant ID

The participant’s identification number as assigned by the system.

Called Participant
ID

The called participant ID.

Asserted Identity

The identity of the user sending a SIP message as it was verified by authentication.

Charging Vector

A collection of charging information.

Preferred Identity

The identity the user sending the SIP message wishes to be used for the P-Asserted-Header
field that the trusted element will insert.

Event Fields for Event 30 - GATEKEEPER INFORMATION
Table 1-22
Field

Description

Participant Name

The name of the participant.

Participant ID

The identification number assigned to the participant by the MCU.

Gatekeeper Caller
ID

The caller ID in the gatekeeper records. This value makes it possible to match the CDR in
the gatekeeper and in the MCU.

Event fields for Event 31 - PARTICIPANT CONNECTION RATE

Field

Description

Participant Name

The participant name.

Participant ID

The identification number assigned to the participant by the MCU.

Participant Current
Rate

The participant line rate in Kbps.

Event Fields for Event 32

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Appendix C - CDR Fields, Unformatted File

Event fields for Event 33 - PARTY CHAIR UPDATE

Field

Description

Participant Name

The participant name.

Participant ID

The identification number assigned to the participant by the MCU.

Chairperson

Possible values:
• True - participant is a chairperson
• False - Participant is not a chairperson participant (is a standard participant)

Event fields for Event 34 - PARTICIPANT MAXIMUM USAGE INFORMATION

Field

Description

Participant Name

The name of the participant.

Participant ID

The identification number assigned to the participant by the MCU.

Maximum Bit Rate

The maximum bit rate used by the participant during the call.

Maximum Resolution

The maximum resolution used by the participant during the call.
Note: The reported resolutions are: CIF, SD, HD720, and HD1080. Other resolutions are
rounded up to the nearest resolution. For example, 2SIF is reported as SD resolution.

Maximum Frame Rate

The maximum frame rate used by the participant during the call.

Participant Address

For H.323 participants, the participant alias. The alias may contain up to 512 characters.
For SIP participants, the participant address. The address may contain up to 80
characters.

Event Fields for Event 35 - SVC SIP PARTICIPANT CONNECTED

Field

Description

Participant Name

The name of the participant.
An empty field "" denotes an unidentified participant or a participant whose name is
unspecified

Participant ID

The identification number assigned to the participant by the MCU.

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Appendix C - CDR Fields, Unformatted File

Field

Description

Participant Status

The participant status, as follows:
0 - Idle
1 - Connected
2 - Disconnected
3 - Waiting for dial-in
4 - Connecting
5 - Disconnecting
6 - Partially connected. Party has completed H.221 capability exchange
7 - Deleted by a user
8 - Secondary. The participant could not connect the video channels and is connected via
audio only
10 - Connected with problem
11 - Redialing

Receive line rate

Negotiated reception line rate

Transmit line rate

Negotiated transmission line rate

Uplink Video
Capabilities

a.Number of uplink streams
b.Video stream (multiple streams)
i.Resolution width
ii.resolution height
iii.max frame rate
iv.max line rate

Audio Codec

SAC, Other

Secondary Cause

Event Fields for Event 100 - USER TERMINATE CONFERENCE
Table 1-24
Field

Description

Terminated By

The login name of the user who terminated the conference.

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Appendix C - CDR Fields, Unformatted File

Event Fields for Events 102,103, 104 - USER DELETE PARTICIPANT, USER DISCONNECT PARTICIPANT, USER
RECONNECT PARTICIPANT
Table 1-25
Field

Description

User Name

The login name of the user who reconnected the participant to the conference, or
disconnected or deleted the participant from the conference.

Participant Name

The name of the participant reconnected to the conference, or disconnected or deleted from
the conference.

Participant ID

The identification number assigned to the participant by the MCU.

Event Fields for Event 106 - USER SET END TIME
Table 1-26
Field

Description

New End Time

The new conference end time set by the user, in GMT.

User Name

The login name of the user who changed the conference end time.

Event Fields for Events 107 and 109 - OPERATOR MOVE PARTY FROM CONFERENCE and OPERATOR
ATTEND PARTY
Table 1-27
Field

Description

Operator Name

The login name of the user who moved the participant.

Party Name

The name of the participant who was moved.

Party ID

The identification number of the participant who was moved, as assigned by the MCU.

Destination Conf
Name

The name of the conference to which the participant was moved.

Destination Conf ID

The identification number of the conference to which the participant was moved.

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Appendix C - CDR Fields, Unformatted File

Event Fields for Events 108, 112 - OPERATOR MOVE PARTY TO CONFERENCE,OPERATOR ATTEND PARTY
TO CONFERENCE

Field

Description

Operator Name

The login name of the operator who moved the participant to the conference.

Source Conf Name

The name of the source conference.

Source Conf ID

The identification number of the source conference, as assigned by the MCU.

Party Name

The name of the participant who was moved.

Party ID

The identification number assigned to the participant by the MCU.

Connection Type

The connection type, as follows:
0 - Dial-out
5 - Dial-in

Bonding Mode

Not applicable.

Number Of
Channels

Note: This field is only relevant to ISDN/PSTN participants.
The number of channels, as follows:
255 - Auto
Otherwise, in range of 1 - 30

Net Channel Width

The bandwidth of each channel.
This value is always 0, which represents a bandwidth of 1B, which is the only bandwidth that
is currently supported.

Net Service Name

The name of the Network Service.
An empty field “” indicates the default Network Service.

Restrict

Indicates whether or not the line is restricted, as follows:
27 - Restricted line
28 - Non restricted line
255 - Unknown or not relevant

Voice Mode

Indicates whether or not the participant is an Audio Only participant, as follows:
0 - The participant is not an Audio Only participant
1 - The participant is an Audio Only participant
255 - Unknown

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Appendix C - CDR Fields, Unformatted File

Field

Description

Number Type

Note: This field is only relevant to dial-out, ISDN/PSTN participants.
The type of telephone number, as follows:
0 - Unknown
1 - International
2 - National
3 - Network specific
4 - Subscriber
6 - Abbreviated
255 - Taken from Network Service, default

Net SubService
Name

Note: This field is only relevant to dial-out, ISDN/PSTN participants.
The network sub-service name.
An empty field “” means that MCU selects the default sub-service.

Number of Party
Phone Numbers

Note: This field is only relevant to ISDN/PSTN participants.
The number of participant phone numbers.
In a dial-in connection, the participant phone number is the CLI (Calling Line Identification)
as identified by the MCU.
In a dial-out connection, participant phone numbers are the phone numbers dialed by the
MCU for each participant channel.

Number of MCU
Phone Numbers

Note: This field is only relevant to ISDN/PSTN participants.
The number of MCU phone numbers.
In a dial-in connection, the MCU phone number is the number dialed by the participant to
connect to the MCU.
In a dial-out connection, the MCU phone number is the MCU (CLI) number as seen by the
participant.

Party and MCU
Phone Numbers

Note: This field is only relevant to ISDN/PSTN participants.
The participant phone numbers are listed first, followed by the MCU phone numbers.

Ident. Method

Note: This field is only relevant to dial-in participants.
The method by which the destination conference is identified, as follows:
0 - Password
1 - Called phone number, or IP address, or alias
2 - Calling phone number, or IP address, or alias

Meet Method

Note: This field is only relevant to dial-in participants.
The meet-me per method, as follows:
1 - Meet-me per MCU-Conference
3 - Meet-me per participant
4 - Meet-me per channel

Net Interface Type

The type of network interface between the participant and the MCU, as follows:
0 - ISDN
2 - H.323
5 - SIP

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Field

Description

H243 Password

The H.243 password, or an empty field “” if there is no password.

Chair

Not supported.
Always contains the value 0.

Video Protocol

The video protocol, as follows:
1 - H.261
2 - H.263
3 - H.264*
4 - H.264
255 -Auto

Audio Volume

The broadcasting volume assigned to the participant.
The value is between 1 (lowest) and 10 (loudest).

Undefined Type

The participant type, as follows:
0 - Defined participant. (The value in the formatted text file is “default”.)
2 - Undefined participant. (The value in the formatted text file is “Unreserved participant “.)

Node Type

The node type, as follows:
0 - MCU
1 - Terminal

Bonding Phone
Number

Note: This field is only relevant to ISDN/PSTN participants.
The phone number for Bonding dial-out calls.

Video Rate

Note: This field is only relevant to IP participants.
The video rate in units of kilobits per second.
A value of 4294967295 denotes auto, and in this case, the rate is computed by the MCU.

IP Address

Note: This field is only relevant to IP participants.
The IP address of the participant.
An address of 4294967295 indicates that no IP address was specified for the participant,
and the gatekeeper is used for routing. In all other cases the address overrides the
gatekeeper.

Call Signaling Port

Note: This field is only relevant to IP participants.
The signaling port used for participant connection.
A value of 65535 is ignored by MCU.

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Appendix C - CDR Fields, Unformatted File

Field

Description

H.323 Party Alias
Type/SIP Party
Address Type

Note: This field is only relevant to IP participants.
For H.323 participants, the alias type, as follows:
7 - E164
8 - H.323 ID
11 - URL ID alias type
12 - Transport ID
13 - Email ID
14 - Participant number
For SIP participants, the address type, as follows:
1 - SIP URI
2 - Tel URL

H.323 Party
Alias/SIP Party
Address

Note: This field is only relevant to IP participants.
For H.323 participants, the participant alias. The alias may contain up to 512 characters.
For SIP participants, the participant address. The address may contain up to 80 characters.

Event Fields for Event 111 - OPERATOR BACK TO CONFERENCE PARTY

Field

Description

Operator Name

The login name of the operator moving the participant back to the conference.

Party Name

The name of the participant being moved.

Party ID

The identification number, as assigned by the MCU, of the participant being moved.

Event Fields for Events 2011, 2012, and 2016
Table 1-28
Field

Description

IP V6

IPv6 address of the participant’s endpoint.

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Event Fields for Event 3010 - PARTICIPANT INFORMATION

Field

Description

Info1
Info2
Info3
Info4

The participant information fields.
These fields enable users to enter general information about the participant, such as the
participant’s e-mail address.
The maximum length of each field is 80 characters.

VIP

Not supported.
Always contains the value 0.

Disconnection Cause Values
For an explanation of the disconnection causes, see Appendix A: Appendix A - Disconnection Causes.

Disconnection Cause Values

Value

Call Disconnection Cause

0

Unknown

1

Participant hung up

2

Disconnected by User

5

Resources deficiency

6

Password failure

20

H323 call close. No port left for audio

21

H323 call close. No port left for video

22

H323 call close. No port left for FECC

23

H323 call close. No control port left

25

H323 call close. No port left for video content

51

A common key exchange algorithm could not be established between the MCU and the remote device

53

Remote device did not open the encryption signaling channel

59

The remote devices’ selected encryption algorithm does not match the local selected encryption algorithm

141

Called party not registered

145

Caller not registered

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Value

Call Disconnection Cause

152

H323 call close. ARQ timeout

153

H323 call close. DRQ timeout

154

H323 call close. Alt Gatekeeper failure

191

H323 call close. Remote busy

192

H323 call close. Normal

193

H323 call close. Remote reject

194

H323 call close. Remote unreachable

195

H323 call close. Unknown reason

198

H323 call close. Small bandwidth

199

H323 call close. Gatekeeper failure

200

H323 call close. Gatekeeper reject ARQ

201

H323 call close. No port left

202

H323 call close. Gatekeeper DRQ

203

H323 call close. No destination IP value

204

H323 call close. Remote has not sent capability

205

H323 call close. Audio channels not open

207

H323 call close. Bad remote cap

208

H323 call close. Capabilities not accepted by remote

209

H323 failure

210

H323 call close. Remote stop responding

213

H323 call close. Master slave problem

251

SIP timer popped out

252

SIP card rejected channels

253

SIP capabilities don’t match

254

SIP remote closed call

255

SIP remote cancelled call

256

SIP bad status

257

SIP remote stopped responding

258

SIP remote unreachable

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Value

Call Disconnection Cause

259

SIP transport error

260

SIP bad name

261

SIP trans error TCP invite

300

SIP redirection 300

301

SIP moved permanently

302

SIP moved temporarily

305

SIP redirection 305

380

SIP redirection 380

400

SIP client error 400

401

SIP unauthorized

402

SIP client error 402

403

SIP forbidden

404

SIP not found

405

SIP client error 405

406

SIP client error 406

407

SIP client error 407

408

SIP request timeout

409

SIP client error 409

410

SIP gone

411

SIP client error 411

413

SIP client error 413

414

SIP client error 414

415

SIP unsupported media type

420

SIP client error 420

480

SIP temporarily not available

481

SIP client error 481

482

SIP client error 482

483

SIP client error 483

484

SIP client error 484

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Appendix C - CDR Fields, Unformatted File

Value

Call Disconnection Cause

485

SIP client error 485

486

SIP busy here

487

SIP request terminated

488

SIP client error 488

500

SIP server error 500

501

SIP server error 501

502

SIP server error 502

503

SIP server error 503

504

SIP server error 504

505

SIP server error 505

600

SIP busy everywhere

603

SIP global failure 603

604

SIP global failure 604

606

SIP global failure 606

MGC Manager Events that are not Supported by the
Collaboration Server
The following MGC Manager events are not supported by the Collaboration Server:
For details of these events see the MGC Manager User’s Guide Volume II, Appendix A.

● Event 8 - REMOTE COM MODE
● Event 11 - ATM CHANNEL CONNECTED
● Event 12 - ATM CHANNEL DISCONNECTED
● Event 13 - MPI CHANNEL CONNECTED
● Event 14 - MPI CHANNEL DISCONNECTED
● Event 15 - H323 CALL SETUP
● Event 16 - H323 CLEAR INDICATION
● Event 24 - SIP CALL SETUP
● Event 25 - SIP CLEAR INDICATION
● Event 27 - RECORDING SYSTEM LINK
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● Event 110 - OPERATOR ON HOLD PARTY
● Event 113 - CONFERENCE REMARKS
● Event 2108 - OPERATOR MOVE PARTY TO CONFERENCE CONTINUE 1
● Event 3001 - CONFERENCE START CONTINUE 2
● Event 3108 - OPERATOR MOVE PARTY TO CONFERENCE CONTINUE 2
● Event 4001 - CONFERENCE START CONTINUE 3
● Event 4108 - OPERATOR MOVE PARTY TO CONFERENCE CONTINUE 3

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Appendix D - Ad Hoc Conferencing and
External Database Authentication
The Polycom® RealPresence® Collaboration Server 800s and Polycom® RealPresence® Collaboration
Server Virtual Edition Ad Hoc conferencing feature enables participants to start ongoing conferences
on-the-fly, without prior definition when dialing an Ad Hoc-enabled Entry Queue. The created conference
parameters are taken from the Profile assigned to the Ad Hoc-enabled Entry Queue.
Ad Hoc conferencing is available in two the following modes:
● Ad Hoc Conferencing without Authentication
Any participant can dial into an Entry Queue and initiate a new conference if the conference does not
exist. This mode is usually used for the organization’s internal Ad Hoc conferencing.
● Ad Hoc Conferencing with External Database Authentication
In this mode, the participant’s right to start a new conference is validated against a database.
The external database application can also be used to validate the participant’s right to join an ongoing
conference. Conference access authentication can be:
● Part of the Ad Hoc conferencing flow where the participants must be authorized before they can enter
the conference created in the Ad Hoc flow.
● Independent of Ad Hoc conferencing where conference access is validated for all conferences
running on the MCU regardless of the method in which the conference was started.

Ad Hoc Conferencing without Authentication
A participant dials in to an Ad Hoc-enabled Entry Queue and starts a new conference based on the Profile
assigned to the Entry Queue. In this configuration, any participant connecting to the Entry Queue can start
a new conference, and no security mechanism is applied. This mode is usually used in organizations where
Ad Hoc conferences are started from within the network and without security breach.

A conference is started using one of the following method:
1 The participant dials in to the Ad Hoc-enabled Entry Queue.
2 The Conference ID is requested by the system.
3 The participant inputs a Conference ID via his/her endpoint remote control using DTMF codes.
4 The MCU checks whether a conference with the same Conference ID is running on the MCU. If
there is such a conference, the participant is moved to that conference. If there is no ongoing
conference with that Conference ID, the system creates a new conference, based on the Profile
assigned to the Entry Queue, and connects this participant as the conference chairperson.

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Ad Hoc Conference Initiation without Authentication

To enable this workflow, the following components must be defined in the system:
● An Entry Queue IVR Service with the appropriate audio file requesting the Conference ID
● An Ad Hoc-enabled Entry Queue with an assigned Profile

Ad Hoc Conferencing with Authentication
The MCU can work with an external database application to validate the participant’s right to start a new
conference. The external database contains a list of participants, with their assigned parameters. The
conference ID entered by the participant is compared against the database. If the system finds a match, the
participant is granted the permission to start a new conference.
To work with an external database application to validate the participant’s right to start a new conference,
the Entry Queue IVR Service must be configured to use the external database application for authentication.
In the external database application, you must define all participants (users) with rights to start a new
conference using Ad Hoc conferencing. For each user defined in the database, you enter the conference
ID, Conference Password (optional) and Chairperson Password (when applicable), billing code, Conference
general information (corresponding to the User Defined 1 field in the Profile properties) and user’s PIN code.
The same user definitions can be used for conference access authentication, that is, to determine who can
join the conference as a participant and who as a chairperson.

Entry Queue Level - Conference Initiation Validation with an External
Database Application
Starting a new conference with external database application validation entails the following steps:

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Conference Initiation Validation with External Database Application

IP Endpoint

MCU

92
51
00
0
IP Endpoint

MCU Prefix in
Gatekeeper - 925

925DefaultEQ

Network

1001
1200

On Going
Conference
with this ID?

1300

5
92

00
10

Name: DefaultEQ
Numeric ID: 1000

IP Endpoint
New Conference
ID: 1300

1 The participant dials in to an Ad Hoc-enabled Entry Queue.
2 The participant is requested to enter the Conference ID.
3 The participant enters the conference ID via his/her endpoint remote control using DTMF codes. If
there is an ongoing conference with this Conference ID, the participant is moved to that conference
where another authentication process can occur, depending on the IVR Service configuration.
4 If there is no ongoing conference with that Conference ID, the MCU verifies the Conference ID with
the database application that compares it against its database. If the database application finds a
match, the external database application sends a response back to the MCU, granting the
participant the right to start a new ongoing conference.
If this Conference ID is not registered in the database, the conference cannot be started and this
participant is disconnected from the Entry Queue.
5 The external database contains a list of participants (users), with their assigned parameters. Once a
participant is identified in the database (according to the conference ID), his/her parameters (as
defined in the database) can be sent to the MCU in the same response granting the participant the
right to start a new ongoing conference. These parameters are:
 Conference Name
 Conference Billing code
 Conference Password
 Chairperson Password
 Conference Information, such as the contact person name. These fields correspond to Info 1, 2
and 3 fields in the Conference Properties - Information dialog box.
 Maximum number of participants allowed for the conference

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 Conference Owner
These parameters can also be defined in the conference Profile. In such a case, parameters sent
from the database overwrite the parameters defined in the Profile. If these parameters are not sent
from the external database to the MCU, they will be taken from the Profile.
6 A new conference is started based on the Profile assigned to the Entry Queue.
7 The participant is moved to the conference.
If no password request is configured in the Conference IVR Service assigned to the conference, the
participant that initiated the conference is directly connected to the conference, as its chairperson.
If the Conference IVR Service assigned to the conference is configured to prompt for the conference
password and chairperson password, without external database authentication, the participant has
to enter these passwords in order to join the conference.
To enable this workflow, the following components must be defined in the system:
● A Conference IVR Service with the appropriate prompts. If conference access is also validated with
the external database application it must be configured to access the external database for
authentication.
● An Entry Queue IVR Service configured with the appropriate audio prompts requesting the
Conference ID and configured to access the external database for authentication.
● Create a Profile with the appropriate conference parameters and the appropriate Conference IVR
Service assigned to it.
● An Ad Hoc-enabled Entry Queue with the appropriate Entry Queue IVR Service and Conference
Profile assigned to it.
● An external database application with a database containing Conference IDs associated with
participants and their relevant properties.
● Define the flags required to access the external database in System Configuration.
For more information, see MCU Configuration to Communicate with an External Database
Application.

Conference Access with External Database
Authentication
The MCU can work with an external database application to validate the participant’s right to join an existing
conference. The external database contains a list of participants, with their assigned parameters. The
conference password or chairperson password entered by the participant is compared against the
database. If the system finds a match, the participant is granted the permission to access the conference.
To work with an external database application to validate the participant’s right to join the conference, the
Conference IVR Service must be configured to use the external database application for authentication.
Conference access authentication can be performed as:
● Part of the Ad Hoc conferencing flow where the participants must be authorized before they can enter
the conference created in the Ad Hoc flow
● Independent of Ad Hoc conferencing where conference access is validated for all conferences
running on the MCU regardless of the method in which the conference was started.
Conference access authentication can be implemented for all participants joining the conference or for
chairpersons only.

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Conference Access Validation - All Participants (Always)
Once the conference is created either via an Ad Hoc Entry Queue, or a standard ongoing conference, the
right to join the conference is authenticated with the external database application for all participants
connecting to the conference.
Conference Access - Conference Password validation with External Database Application
MCU
IP Endpoint

92
51

Correc t
Password?

00
0

MCU Prefix in
Gatekeeper - 925

IP Endpoint

925DefaultEQ

1001
1001

Network

On Going
Conference
with this ID?

1300

51
92

00

0

Name: DefaultEQ
Numeric ID: 1000

IP Endpoint

New Conference
ID: 1300

Joining the conference entails the following steps:
● When the conference is started (either in the Ad Hoc flow or in the standard method), all participants
connecting to the conference are moved to the Conference IVR queue where they are prompted for
the conference password.
● When the participant enters the conference password or his/her personal password, it is sent to the
external database application for validation.
● If there is a match, the participant is granted the right to join the conference. In addition, the external
database application sends to the MCU the following parameters:
 Participant name (display name)
 Whether or not the participant is the conference chairperson
 Participant Information, such as the participant E-mail. These fields correspond to Info 1, 2, 3 and
4 fields in the Participant Properties - Information dialog box.
If there is no match (i.e. the conference or personal password are not defined in the database), the
request to access the conference is rejected and the participant is disconnected from the MCU.
● If the Conference IVR Service is configured to prompt for the chairperson identifier and password,
the participant is requested to enter the chairperson identifier.
 If no identifier is entered, the participant connects as a standard, undefined participant.
● If the chairperson identifier is entered, the participant is requested to enter the chairperson password.
In this flow, the chairperson password is not validated with the external database application, only
with the MCU.
 If the correct chairperson password is entered, the participant is connected to the conference as
its chairperson.

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 If the wrong password is entered, he/she is disconnected from the conference.
To enable conference access validation for all participants the following conferencing components are
required:
● The external database must hold the conference password or the participant personal password/PIN
code or the participant’s Alias.
● The Conference IVR Service assigned to the conference (defined in the Profile) must be configured
to authenticate the participant’s right to access the conference with the external database application
for all requests. In addition it must be configured to prompt for the Conference Password.

Conference Access Validation - Chairperson Only (Upon Request)
An alternative validation method at the conference level is checking only the chairperson password with the
external database application. All other participants can be checked only with the MCU (if the Conference
IVR Service is configured to prompt for the conference password) or not checked at all (if the Conference
IVR Service is configured to prompt only for the chairperson password).
Conference Access - Chairperson Password validation with external database application

MCU
IP Endpoint
Chairperson
Password?

92
51
00
0

MCU Prefix in
Gatekeeper - 925

IP Endpoint

925DefaultEQ

Network

Enters the
c onferenc e as
Chairperson

1001
1001

Enters the
c onferenc e as
standard
partic ipant

On Going
Conference
with this ID?

1300

92

51

0
00

Name: DefaultEQ
Numeric ID: 1000

IP Endpoint

New Conference
ID: 1300

Joining the conference entails the following steps:
● When the conference is started (either in the Ad Hoc flow or in the standard method), all participants
connecting to the conference are moved to the conference IVR queue where they are prompted for
the conference password.
● If the Conference IVR Service is configured to prompt for the Conference password, the participant
is requested to enter the conference password. In this flow, the conference password is not validated
with the external database application, only with the MCU.
 If the wrong password is entered, he/she is disconnected from the conference.
● If the correct conference password is entered, the participant is prompted to enter the chairperson
identifier key.
 If no identifier is entered, the participant is connected to the conference as a standard participant.
● If the chairperson identifier is entered, the participant is prompted to enter the chairperson password.
● When the participant enters the chairperson password or his/her personal password, it is sent to the
external database application for validation.
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 If the password is incorrect the participant is disconnected from the MCU.
● If there is a match, the participant is granted the right to join the conference as chairperson. In
addition, the external database application sends to the MCU the following parameters:
 Participant name (display name)
 Participant Information, such as the participant E-mail. These fields correspond to Info 1, 2, 3 and
4 fields in the Participant Properties - Information dialog box.
To enable conference access validation for all participants the following conferencing components are
required:
● The external database must hold the Chairperson Password or the participant’s Alias.
● The Conference IVR Service assigned to the conference (defined in the Profile) must be configured
to check the external database for the Chairperson password only when the participant enters the
chairperson identifier key (either pound or star). In addition, it must be configured to prompt for the
chairperson identifier key and password.

System Settings for Ad Hoc Conferencing and External
Database Authentication
Ad Hoc Settings
Before a participant can initiate an Ad Hoc conference (with or without authentication), the following
components must be defined:
● Profiles
Defines the conference parameters for the conferences that will be initiated from the Ad Hoc-enabled
Entry Queue. For more details, see Using Conference Profiles.
● Entry Queue IVR Service with Conference ID Request Enabled
The Entry Queue Service is used to route participants to their destination conferences, or create a
new conference with this ID. For details, see IVR Services.
In Ad Hoc conferencing, the Conference ID is used to check whether the destination conference is
already running on the MCU and if not, to start a new conference using this ID.
● Ad Hoc - enabled Entry Queue
Ad Hoc conferencing must be enabled in the Entry Queue and a Profile must be assigned to the Entry
Queue. In addition, an Entry Queue IVR Service supporting conference ID request. For details, see
Entry Queues.

Authentication Settings
● MCU Configuration
Usage of an external database application for authentication (for starting new conferences or joining
ongoing conferences) is configured for the MCU in the System Configuration. For details, see MCU
Configuration to Communicate with an External Database Application.

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● Entry Queue IVR Service with Conference Initiation Authentication Enabled
Set the Entry Queue IVR Service to send authentication requests to the external database application
to verify the participant’s right to start a new conference according to the Conference ID entered by
the participant. For details, see Enabling External Database Validation for Starting New Ongoing
Conferences.
● Conference IVR Service with Conference Access Authentication Enabled
Set the Conference IVR Service to send authentication requests to the external database application
to verify the participant’s right to connect to the conference as a standard participant or as a
chairperson. For details, see Enabling External Database Validation for Conferences Access.
● External Database Application Settings
The external database contains a list of participants (users), with their assigned parameters. These
parameters are:
 Conference Name
 Conference Billing code
 Conference Password
 Chairperson Password
 Conference Information, such as the contact person name. These fields correspond to Info 1, 2
and 3 fields in the Conference Properties - Information dialog box.
 Maximum number of participants allowed for the conference
 Conference Owner
 Participant name (display name)
 Participant Information, such as the participant E-mail. These fields correspond to Info 1, 2, 3 and
4 fields in the Participant Properties - Information dialog box.

MCU Configuration to Communicate with an External Database Application
To enable the communication with the external database application, several flags must be set in the
System Configuration.

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To set the System Configuration flags:
1 On the Setup menu, click System Configuration.
The System Flags dialog box opens.

2 Modify the values of the following flags:
Flag Values for Accessing External Database Application
Flag

Description and Value

ENABLE_EXTERNAL_DB_ACCESS

The flag that enables the use of the external database application.

EXTERNAL_DB_IP

The IP address of the external database application server. default
IP: 0.0.0.0.

EXTERNAL_DB_PORT

The port number used by the MCU to access the external
application server.
Default Port = 80.

EXTERNAL_DB_LOGIN

The user name defined in the external database application for the
MCU.

EXTERNAL_DB_PASSWORD

The password associated with the user name defined for the MCU
in the external database application.

EXTERNAL_DB_DIRECTORY

The URL of the external database application.

3 Click OK.
4 Reset the MCU for flag changes to take effect.

Enabling External Database Validation for Starting New Ongoing Conferences
The validation of the participant’s right to start a new conference with an external database application is
configured in the Entry Queue IVR Service - Global dialog box.

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● Set the External Server Authentication field to Numeric ID.

Enabling External Database Validation for Conferences Access
The validation of the participant’s right to join an ongoing conference with an external database application
is configured in the Conference IVR Service - Global dialog box.
You can set the system to validate all the participants joining the conference or just the chairperson.
● Set the External Server Authentication field to:
 Always - to validate the participant’s right to join an ongoing conference for all participants

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▪

Upon Request - to validate the participant’s right to join an ongoing conference as chairperson

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Appendix E - Participant Properties
Advanced Channel Information
The following appendix details the properties connected with information about audio and video parameters,
as well as, problems with the network which can affect the audio and video quality.
Participant Properties - Channel Status Advanced Parameters
Field

Description

Media Info
Algorithm

Indicates the audio or video algorithm and protocol.

Frame per packet
(audio only)

The number of audio frames per packet that are transferred
between the MCU and the endpoint. If the actual Frame per
Packets are higher than Frame per Packets declared during the
capabilities exchange, a Faulty flag is displayed.

Resolution

Indicates the video resolution in use. If the actual resolution is
higher than resolution declared in the capabilities exchange, the
Faulty flag is displayed. For example, if the declared resolution is
CIF and the actual resolution is 4CIF, the Faulty flag is displayed.

(video only)

Frame Rate
(video only)

The number of video frames per second that are transferred
between the MCU and the endpoint.

Annexes (video only)

Indicates the H.263 annexes in use at the time of the last RTCP
report. If the actual annexes used are other than the declared
annexes in the capabilities exchange, the Faulty flag is displayed.

Channel Index

For Polycom Internal use only.

RTP Statistics
Actual loss

The number of missing packets counted by the IP card as reported
in the last RTP Statistics report. If a packet that was considered lost
arrives later, it is deducted from the packet loss count. Packet loss
is displayed with the following details:
•
•
•
•

•

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Accumulated N - number of lost packets accumulated since the
channel opened.
Accumulated % - percentage of lost packets out of the total number
of packets transmitted since the channel opened.
Interval N - number of packets lost in the last RTP report interval
(default interval is 5 minutes).
Interval % - percentage of lost packets out of the total number of
packets transmitted in the last RTP report interval (default interval is 5
minutes).
Peak - the highest number of lost packets in a report interval from the
beginning of the channel's life span.

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Field

Description

Out of Order

The number of packets arriving out of order. The
following details are displayed:
•
•
•
•

•
Fragmented

Accumulated N - total number of packets that arrived out of order
since the channel opened.
Accumulated % - percentage of packets that arrived out of order out
of the total number of packets transmitted since the channel opened.
Interval N - number of packets that arrived out of order in the last RTP
report interval (default interval is 5 minutes).
Interval % - percentage of packets that arrived out of order out of the
total number of packets transmitted in the last RTP report interval
(default interval is 5 minutes).
Peak - the highest number of packets that arrived out of order in a
report interval from the beginning of the channel's life span.

Indicates the number of packets that arrived to the IP card
fragmented (i.e., a single packet broken by the network into
multiple packets). This value can indicate the delay and reordering
of fragmented packets that require additional processing, but is not
considered a fault.
The Fragmented information is displayed with the following details:
•
•
•
•

•

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Accumulated N - total number of packets that were fragmented since
the channel opened.
Accumulated % - percentage of fragmented packets out of the total
number of packets transmitted since the channel opened.
Interval N - number of fragmented packets received in the last RTP
report interval (default interval is 5 minutes).
Interval % - percentage of fragmented packets out of the total number
of packets transmitted in the last RTP report interval (default interval is
5 minutes).
Peak - the highest number of fragmented packets in a report interval
from the beginning of the channel's life span.

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Appendix G - Configuring Direct
Connections to the Collaboration Server
Direct connection to the Collaboration Server is necessary if you want to:
● Modify the Collaboration Server’s Factory Default Management Network settings without using the
USB memory stick.
● Connect to the Collaboration Server’s Alternate Management Network for support purposes.
● Connect to the Collaboration Server via a modem.

Management Network (Primary)
If you do not want to use the USB memory stick method of modifying the Collaboration Server’s
Management Network parameters, it is necessary to establish a direct connection between a workstation
and the Collaboration Server.

Configuring the Workstation
The following procedures show how to modify the workstation’s networking parameters using the Windows
New Connection Wizard.
For non-Windows operating systems an equivalent procedure must be performed by the system
administrator.
Before connecting directly, you must modify the IP Address, Subnet Mask and Default Gateway settings of
the workstation to be compatible with either the Collaboration Server’s Default Management Network or
Alternate Management Network.

To modify the workstation’s IP addresses:
1 On the Windows Start menu, select Settings > Network Connections.
2 In the Network Connections window, double-click the Local Area Connection that has Connected
status.

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In the Local Area Connection Status dialog box, click the Properties button.

3 In the Local Area Connection Properties dialog box, select Internet Protocol [TCP/IP] >
Properties.

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4 In the Internet Protocol (TCP/IP) Properties dialog box, select Use the following IP address.

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5 Enter the IP address, Subnet mask and Default gateway for the workstation.

The workstation’s IP address should be in the same network neighborhood as the Collaboration
Server’s Control Unit IP address.
Example: IP address – near 192.168.1.nn
None of the reserved IP addresses listed in Reserved IP Addresses should be used for the IP
Address.

The Subnet mask and Default gateway addresses should be the same as those for the Collaboration
Server’s Management Network.
The addresses needed for connection to either the Collaboration Server’s Default Management
Network or Alternate Management Network are listed in the table below.
For more information about connecting to the Alternate Management Network, see Appendix G Configuring Direct Connections to the Collaboration Server.
Reserved IP Addresses
IP Address

Network Entity

Management
Network
(Factory Default)

Alternate Network

Control Unit IP Address

192.168.1.254

169.254.192.10

Control Unit Subnet Mask

255.255.255.0

255.255.240.0

Default Router IP Address

192.168.1.1

169.254.192.1

Shelf Management IP Address

192.168.1.252

169.254.192.16

Shelf Management Subnet Mask

255.255.255.0

255.255.240.0

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IP Address

Network Entity

Management
Network
(Factory Default)

Alternate Network

Shelf Management Default Gateway

192.168.1.1

169.254.192.1

6 Click the OK button.

Connecting to the Management Network
To connect directly to the Collaboration Server:
1 Connect one LAN cable between the PC and LAN 1 on the Collaboration Server’s back
panel.Connect the power cable and power the Collaboration Server On.
2 Start the Collaboration Server Web Client application on the workstation, by entering the factory
setting Management IP address in the browser’s address line and pressing Enter.
3 In the Collaboration Server Web Client Login screen, enter the default Username (POLYCOM) and
Password (POLYCOM) and click the Login button.
The Fast Configuration Wizard starts.

If no USB memory stick is detected and either: this is the First Time Power-up or the Default IP
Service has been deleted and the Collaboration Server has been reset, the following dialog box is
displayed:
For more information about First-time Power-up and the Fast Configuration Wizard see the
Polycom® RealPresence Collaboration Server 800s / Virtual Edition Getting Started Guide,
Procedure 1: First-time Power-up.
4 Enter the following parameters using the information supplied by your network administrator:
 Control Unit IP Address

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 Shelf Management IP Address
 Control Unit Subnet Mask
 Default Router IP Address
5 Click the Save & Close button.
The system prompts you to sign in with the new Control Unit IP Address.

6 Disconnect the LAN cable between the workstation and the LAN 1Port on the Collaboration Server’s
back panel.
7 Connect LAN 1Port on the Collaboration Server’s back panel to the local network using a LAN
cable.
8 Enter the new Control Unit IP Address in the browser’s address line, using a workstation on the local
network, and press Enter to start the Collaboration Server Web Client application.
9 In the Collaboration Server Web Client Login screen, enter the default Username (POLYCOM) and
Password (POLYCOM) and click the Login button.

Connecting to the Collaboration Server via Modem
Remote access to the Collaboration Server’s Alternate Management Network is supported via an external
PSTN <=> IP modem.
To connect via modem to the Alternate Management Network the following procedures must be performed:
1 Procedure 1: Install the RMX Manager – the web client enables direct access to the Collaboration
Server for support purposes.
2 Procedure 2: Configure the modem – by assigning it an IP address on a specific subnet in the
Alternate Management Network.
3 Procedure 3: Create a dial-up connection – using the Windows New Connection Wizard.
4 Procedure 4: Connect to the Collaboration Server – via the RMX Manager.

Procedure 1: Install the RMX Manager
Before installing the RMX Manager, verify that you have at least 150Mb of free space on your workstation.
For more information see Installing the RMX Manager Application.

Procedure 2: Configure the Modem
Configure the modem as follows:
● IP address – near 169.254.192.nn

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● Subnet Mask – 255.255.240.0
None of the reserved IP addresses listed in Table 5-127 on page 5-10 should be used for the IP
Address.

Procedure 3: Create a Dial-up Connection
To create a dial-up connection:
This procedure is performed once. Only the Dial field in the Connect applet (see step 10 on Click the Dial
button to establish a connection to LAN 3 Port via the modem.) is modified for connection to different
modems.
1 In Windows, navigate via the Control Panel to the Network Connections applet and select Create a
new connection.
2 When the New Connection Wizard is displayed, click the Next button.

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3 In the Network Connection Type box, select Connect to the Internet and click the Next button.

4 In the Getting Ready box, select Set up my connection manually and click the Next button.

5 In the Internet Connection box, select Connect using dial-up modem and click the Next button.

6 In the Connection Name box, enter a Name for the modem connection (e.g. Modem Connection)
and click the Next button.

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7 In the Phone Number to Dial box, enter the Phone Number for the modem and click the Next
button.

8 In the Connection Availability box, select Anyone’s use and click the Next button.

9 In the Internet Account Information box, complete the Username, Password and Confirm Password
fields and click the Next button.

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10 The Connection applet is displayed with the field values filled in as specified by the New Connection
Wizard.

11 Click the Dial button to establish a connection to LAN 3 Port via the modem.
The Windows – Network Connections applet displays Connected status for the new connection.

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Procedure 4: Connect to the Collaboration Server
To Connect using the RMX Manager:
To use the browser:
● In the browser’s command line, enter http:///RmxManager.html
and press Enter.

To use the Windows Start menu:
1 Click Start.
a If the RMX Manager is displayed in the recently used programs list, click RMX Manager in the list
to start the application.
or
b Click All Programs.
The All Programs list is displayed.
c Select Polycom and then select RMX Manager.

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The RMX Manager – Welcome screen is displayed.

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Appendix H - Integration Into Microsoft
Environments
Integration into Microsoft environment (using Lync endpoints) is supported in AVC CP Conferencing
Mode only.

Overview
The Polycom® Visual Communications offers high quality video and audio multipoint conferencing by
integrating the Polycom network devices and endpoints into Microsoft® platforms. The Polycom®
RealPresence® Collaboration Server (Collaboration Server) system can be integrated into the following
Microsoft environments:
● Office Communications Server 2007 environment (Microsoft R2, Wave 13)
● Lync Server 2010 environment (Microsoft Wave 14).
Point-to-point and multipoint audio and video meetings can be initiated from Office Communicator/ Lync
client, Windows Messenger and Polycom video endpoints (HDX and VSX) when the environment
components are installed and configured.
Multipoint calls are enabled when the Collaboration Server is installed in the Microsoft environment and is
configured for unified communications. Routing to conferences can be performed by the Office
Communications Server/Lync Server either by:
● Matched URI dialing - using the SIP URI address.(both Office Communications Server and Lync
Server)
● Numerical dialing - enables a common dialing plan for Meeting Rooms across Office
Communications Server and H.323 infrastructures (not available in Lync server environment).
Only TLS connections to the Collaboration Server will work, TCP connections will not work.
The Collaboration Server does not support working with multiple Edge servers.

TLS certificates can be generated using the following methods: CSR, PFX and PEM; each giving different
options for Encryption Key length. The table below lists the SIP TLS Encryption Key length support for the
various system components.
SIP TLS - Encryption Key Support by System Component
System Component

Key Generation Method

Key Length (bits)

Key generated by

SIP Signaling

CSR

2048

Collaboration Server

PFX / PEM

1024 or 2048

User

CSR

2048

Collaboration Server

Management
LDAP

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Conferencing Entities Presence
Conferencing entities (Meeting Rooms, Entry Queues and SIP Factories) can be registered with the SIP
server (Office Communication Server or Lync server) enabling the addition of these conferencing entities to
the buddy list while displaying their presence (availability status: Available, Offline, or Busy). Office
Communication Server client or Lync Server client users can connect to conferencing entities directly from
the buddy list.
The configuration of the environment to enable Presence, is usually done once the basic configuration is
completed.
For more details, see Adding Presence to Conferencing Entities in the Buddy List.

Multiple Networks
A more complex configuration, in which two Microsoft SIP servers are used (one Lync Server and one Office
Communications Server) is also supported using the Collaboration Server Multiple Networks configuration.
In this configuration, each Microsoft SIP Server is defined in a Network Service of its own (in this case two
IP Network Services are defined). A DNS server can be specified for each IP Network Service and for the
RMX Management Network Service.
Collaboration Server Multiple Networks Topology

Guidelines
● If ICE initialization fails in a Network Service:
 The Network Service remains functional but without ICE capability.
 ICE capability on media cards that share the same Network Service also remain functional but
without ICE capability.
 Other Network Services with ICE capability on other media cards are unaffected.
● A DNS server can be specified for each IP Network Service and for the Collaboration Server
Management Network Service.

Interactive Connectivity Establishment (ICE)
Interactive Connectivity Establishment (ICE) provides a structure/protocol to unify the various NAT
Traversal techniques that are used to cross firewalls.
It enables SIP based endpoints to connect while traversing a variety of firewalls that may exist between the
calling endpoint (local) and the MCU or called endpoint (remote). It is the only way for remote Microsoft
Office Communicator/Lync users to call into the enterprise without a VPN.

ICE Guidelines
● Collaboration Server ICE implementation complies with Microsoft ICE implementation.
● ICE is available only in IPv4 environment.
● ICE can be implemented in an environment that includes a STUN server and a Relay server (for
example, Microsoft AV Edge server).
● The firewall must be UDP enabled.
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When ICE over UDP is blocked in the firewall UDP port, the ICE connection through the TCP
protocol is automatically used instead of UDP for fallback.

● The Collaboration Server must have a unique account in the Active Directory and must be registered
with the Office Communications/Lync server.
● ICE is supported with Collaboration Server Multiple Networks.
● Ensure that the Collaboration Server system SIP signaling domain has been allowed on the Lync
Server edge server to which you are federating (if your deployment does not include a DMA system).
● Content sharing (BFCP protocol) is not supported in ICE environment.

Connecting to the Collaboration Server in ICE Environment
The dialing methods that can be used by an endpoint to connect to another endpoint depends on the ICE
environment: Local, Remote or Federation.
ICE Environmen

I

Local connection - a connection between the Collaboration Server and endpoints that reside within the
same organization. For example, an endpoint in Zone A calls the Collaboration Server in Zone A.
Branch Office - a connection between an endpoint that is behind a firewall and the Collaboration Server that
reside in the same zone. The user in the Branch Office can also place and receive calls from other
enterprises and remote users. For example, Enterprise A also contains a branch office, which in this
example is a Polycom HDX user who is behind more than one firewall.
Remote - a connection between Collaboration Server that resides within the organization and an endpoint
that resides outside of the organization (on a WAN). For example, an endpoint on the internet that calls the
Collaboration Server in Zone A. In such a case, the call has to traverse at least one firewall.
Federation - a connection between Collaboration Server that resides within one organization and an
endpoint that resides within another organization. For example, an endpoint in Zone A calls the
Collaboration Server in Zone B. The call has to traverse two or more firewalls.

Dialing Methods
The ICE protocol enables remote and federation connections using the registered user name for dialing.
The endpoint connects to the Collaboration Server by entering the Collaboration Server registered user
name in the following format:
[Collaboration Server registered user name]@[OCS/Lync server domain name]
For example:
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rmx111@ilsnd.vsg.local
The call reaches the Transit Entry Queue of the Collaboration Server and via IVR is routed to the destination
conference.
This method is added to the local connections and Matched URI and Numerical Dialing methods available
in Microsoft Office Communication environment and the Numerical Dialing method available in the Lync
server environment.
The following table summarizes the dialing methods and its availability in the various configurations.
Available dialing methods per Connection Type
Matched URI Routing

Numerical Dialing

Registered User Name

Local







Branch office

*





Remote

*





Federation

*





* To enable the Matched URI dialing in the federated environment to be able to connect to the Collaboration
Server SIP signaling domain, you must also configure the Office Communications Server/Lync Server.
When federating an Office Communications Server/Lync Edge server with another Office Communications
Server/Lync server environment, you need to include the FQDN of the Office Communications Server/Lync
Edge server as well as the SIP signaling domain for federated environment. The SIP signaling domain is
the FQDN of the Polycom DMA system or a Polycom Collaboration Server system (when your deployment
does not include a DMA system).
For example, if company B wants to set up federation with company A and receive and send SIP calls that
will be handled by the Polycom SIP signaling domain in company A, you need to add the FQDN of the
company A Office Communications Server domain as well as the SIP signaling domain of company A to the
list of internal SIP Server domains supported by the company B Office Communications Server/Lync Server
environment.

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Integrating the Collaboration Server into the Microsoft
Office Communications Server Environment
When the Collaboration Server is integrated into the Office Communications Server environment, calls to
conferences running on the Collaboration Server can be routed using Matched URI dialing and/or Numerical
dialing.
Both routing methods (numerical dialing and Matched URI dialing) can be enabled simultaneously in the
Office Communications Server and the Collaboration Server system or you can enable one of these
methods, depending on your environment requirements.
In both methods, the Collaboration Server configuration is the same.

Setting the Matched URI Dialing Method
To enable the Matched URI dialing method the following tasks have to be completed:

Office Communications Server side:
1 Set the Static Route & Trusted Host for Collaboration Server in the Office Communications Server.
2 Optional if Load Balancer Server is present. Set the Static Route & Trusted Host for
Collaboration Server in the Load Balancer server.

Collaboration Server side:
The following tasks are detailed in Configuring the Collaboration Server for Microsoft Integration.
1 Modify the Management Network Service to include the DNS server and set the Transport Type to
TLS.
2 Create the security certificate (using one of the two available methods)
3 Define a SIP Network Service in the Collaboration Server and install the TLS certificate.
4 Modify and add the required system flags in the Collaboration Server System Configuration.
5 Optional. Defining additional Entry Queues and Meeting Rooms in the Collaboration Server
environment. For more information see Defining a New Entry Queue and Creating a New Meeting
Room.
For a detailed description of the configuration of the Polycom conferencing components for the integration
in Microsoft Office Communications Server 2007 see the Polycom® HDX and Collaboration Server™
Systems Integration with Microsoft Office Communications Server 2007 Deployment Guide.
In an ICE environment, to enable the Matched URI dialing in the federated environment to be able to
connect to the Collaboration Server SIP signaling domain, you must also configure the Office
Communications Server. When federating an Office Communications Server edge server with another
Office Communications Server environment, you need to include the FQDN of the Office Communications
Server edge server as well as the SIP signaling domain for federated environment. The SIP signaling
domain is the FQDN of the Polycom DMA system or a Polycom Collaboration Server system (when your
deployment does not include a DMA system).
Note: The RMX does not support working with multiple edge servers.

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For example, if company B wants to set up federation with company A and receive and send SIP calls that
will be handled by the Polycom SIP signaling domain in company A, you need to add the FQDN of the
company A Office Communications Server domain as well as the SIP signaling domain of company A to the
list of Internal SIP Server domains supported by the company B Office Communications Server
environment.
For more information, see the Microsoft documentation and the Visual Communications Deployment
Administration Guide.

Configuring the Office Communications Server for Collaboration Server Systems
To be able to work with the Office Communications Server, the Collaboration Server unit must be configured
as a Trusted Host in the OCS. This is done by defining the IP address of the signaling host of each
Collaboration Server unit as Trusted Host.
Meeting Rooms are usually not registered to the OCS, and Static Routes are used instead. Setting Static
Routes in the OCS enables SIP entities / UAs to connect to conferences without explicit registration of
conferences with the OCS.
Routing is performed by the OCS based on the comparison between the received URI and the provisioned
static route pattern. If a match is found, the request is forwarded to the next hop according to the defined
hop’s address.
This is the recommended working method. It alleviates the need to create a user account in the OCS for
each Meeting Room and Entry Queue. This also allows users to join ongoing conferences hosted on the
MCU without registering all these conferences with OCS.
Entry Queues can also be for Ad-hoc conferencing enabling Office Communicator clients to dial to the Entry
Queue and create a new ongoing conference using DTMF codes to enter the target conference ID. In such
a case, other OC users will have to use that ID to join the newly created conference.

Setting the Trusted Host for Collaboration Server in the Office Communications
Server
To set the Collaboration Server as trusted in OCS:
1 Open the OCS Management application.

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2 Expand the Enterprise Pools list.

3 Right-click the server pool icon, click Properties > Front End Properties.

The Pool Front End Properties dialog box opens.

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4 Click the Host Authorization tab.

5 Click the Add button to add the Collaboration Server as trusted host.
The Add Authorized Host dialog box opens.

6 In the Add Authorized Host dialog box, enter the Collaboration Server FQDN name as defined in the
DNS and will be used in the Static Routes definition.
7 In the Settings section, select the Throttle as Server and Treat As Authenticated check boxes.
8 Click OK.
The defined Collaboration Server is displayed in the trusted servers list in the server Front End
Properties—Host Authorization dialog box.
If routing between the Collaboration Server and the OCS using Static Routes is required, do not
close this dialog box, and continue with the following procedure. If you do not want to define Static
Routes, click OK to close this dialog box.

Setting the Static Route for Collaboration Server in the OCS
To add Collaboration Server to the Routing Roles:
1 In the Front End Properties dialog box, click the Routing tab.

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2 Click the Add button.

The Add Static Routes dialog box opens.
3 In the Matching URI section, enter the Domain name for the Collaboration Server. Any domain
name can be used.
4 In the Next hop section enter the Collaboration Server FQDN name as defined in the DNS and is
used in the Host Authorization definition.

5 In the Transport field, select TLS to enable the dial-out from conferences to SIP endpoints.
6 Click OK.
The new Route is added to the list of routes in the Front End Properties—Routes dialog box.
7 Click OK.

Setting the Static Route & Trusted Host for Collaboration Server in the Load
Balancer Server (Optional)
If your network includes a Load Balancer server, the Collaboration Server unit must be configured as a
trusted host in the Load Balancer server in the same way it is configured in the OCS. In addition, Static
Routes must also be defined in the Load Balancer server in the same way it is configured in the OCS,
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however, the Load Balancer should be pointed to the OCS pool and not to the Collaboration Server directly.
This configuration procedure is done in addition to the configuration in the OCS.

To set the Collaboration Server as trusted and define Static routes in the Load Balancer
Server:
1 Open the OCS Management application.
2 Expand the Enterprise Pools list.
3 Right-click the Load icon, click Properties > Front End Properties.

The Load Front End Properties dialog box opens.
The definition procedure is the same as for setting the Collaboration Server as trusted and define
Static routes in the OCS.
For details, see Setting the Trusted Host for Collaboration Server in the Office Communications
Server.

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Make sure that when defining the Static Route it is pointing to the OCS pool and not to the
Collaboration Server directly.

Configuring the Collaboration Server System
The required tasks are detailed in Configuring the Collaboration Server for Microsoft Integration.

Dialing to an Entry Queue, Meeting Room or Conference Using the Matched URI
Method
Once the Collaboration Server is configured for integration in the OCS environment, the preferred dialing
mode to the conferencing entities such as Meeting Rooms, conferences and Entry Queues is direct dial in
using the domain name defined in the OCS Static Routes. This eliminates the need to register the
conferencing entities with the SIP server and to define a separate user for each conferencing entity in the
Active Directory.
In such a case, after the first dial in, the conferencing entity will appear in the OC client directory for future
use.

To dial in directly to a conference or Entry Queue:
Enter the conferencing entity SIP URI in the format:
conferencing entity routing name@domain name
The domain name is identical to the domain name defined in the OCS Static Routes.
For example, if the domain name defined in the OCS static routes is lcs2007.polycom.com and the Routing
Name of the Meeting Room is 4567, the participant enters 4567@lcs2007.polycom.com.
Another dialing method is to register the Entry Queues with the SIP Server and create a user for each Entry
Queue in the Active Directory. In such a case, OC clients can select the Entry Queue from the Contacts list
and dial to the Entry Queue.

Setting the Numerical Dialing Method
The Collaboration Server can be configured as a Voice Gateway in the OCS environment, enabling dialing
in to meeting rooms using numbers instead of or in addition to SIP URI addresses which are long strings.
In such configuration, HDX or MOC users dial a number rather than a full SIP URI, simplifying the dialing,
which is especially beneficial with the HDX remote control.
Such configuration also enables a common dialing plan for meeting rooms across OCS and H.323
infrastructures. In an integrated environment that also includes Microsoft Exchange Server and Polycom
Conferencing Add-in for Microsoft Outlook, a single number can be inserted into a calendar invitation and it
will be valid for OC client endpoints and H.323 endpoints.
This dialing method can be configured in parallel to the matching URI dialing method (using Static Routes).

Setting the Numerical Dialing for Collaboration Server Meeting Rooms
The following processes are required to set up the numerical dialing for the Collaboration Server Meeting
Rooms in the OCS infrastructure:

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OCS side:
● Configuring the Collaboration Server as a Routable Gateway - The Collaboration Server (or DMA)
must be set as a trusted voice gateway in the OCS infrastructure. This does not restrict Collaboration
Server to just voice operation, rather it means that the Collaboration Server (or DMA) can be set as
a destination for a voice route using the OCS management console.
Setting the Collaboration Server as a trusted voice gateway also enables it to be used as a trusted
gateway for static routes using URI matching.
● Establishing a Voice Route to the Collaboration Server “Voice” Gateway - The Voice Route to the
Collaboration Server (or DMA) must be configured in the OCS infrastructure.
If the Collaboration Server was previously defined as a Trusted Host for matching URI dialing
method, this definition must be removed before configuring the Collaboration Server as a voice
gateway. It will be defined as trusted host as part of the voice gateway configuration. For more
details, see Optional. Removing the Collaboration Server from the Host Authorization List.

● Configure Office Communicator Users for Enterprise Voice.

Collaboration Server side:
The following tasks are detailed in Configuring the Collaboration Server for Microsoft Integration.
1 Modify the Management Network Service to include the DNS server and set the Transport Type to
TLS.
2 Create the security certificate (using one of the two available methods)
3 Define a SIP Network Service in the Collaboration Server and install the TLS certificate.
4 Modify and add the required system flags in the Collaboration Server System Configuration.
5 Optional. Defining additional Entry Queues and Meeting Rooms in the Collaboration Server
environment. For details see Meeting Rooms and Entry Queues.
For a detailed description of the configuration of the Polycom conferencing components for the integration
in Microsoft Office Communications Server 2007 see the Polycom® HDX and Collaboration Server™
Systems Integration with Microsoft Office Communications Server 2007 Deployment Guide.

Optional. Removing the Collaboration Server from the Host Authorization List
If you have defined the Collaboration Server as Trusted Host to enable dialing using the Static Routes and
you want to use numerical dialing in addition or instead of SIP URI dialing, you need to remove the current
definition of the Collaboration Server and redefine it as a voice gateway.

To remove the definition of the Collaboration Server as trusted host from the Front End
Properties:
1 In the OCS application, display the Front End Properties (right-click the Front End and select
Properties).
2 Click the Host Authorization tab.
3 In the Trusted Hosts list, click the Collaboration Server entry and then click the Remove button.

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4 Click OK.

Configuring the Collaboration Server as a Routable Gateway
The Collaboration Server must be set as a routable voice gateway in the Office Communications Server
infrastructure. This does not restrict the Collaboration Server to just voice operation, rather it means that the
Collaboration Server can be set as a destination for a voice route in the Office Communications Server
infrastructure.
The Office Communications Server infrastructure uses the WMI class
MSFT_SIPTrustedAddInServiceSetting to store information for each voice gateway in the
infrastructure. Typically, these gateways are Office Communications Server Mediation Servers, but in this
case, the Collaboration Server is set as a voice gateway by creating a new instance of the class
MSFT_SIPTrustedAddInServiceSetting.
Polycom recommends using the Office Communications Server 2007 R2 Resource Kit Tools to accomplish
this.

To set up the Collaboration Server/DMA as a Voice Gateway:
1 Download and install the Office Communications Server 2007 R2 Resource Kit Tools from the
following URL:
http://www.microsoft.com/downloads/details.aspx?familyid=9E79A236-C0DF-4A72-ABA6-9A9602A93ED
0&displaylang=en
2 Open a command prompt and navigate to where you installed the resource kit. For example,
C:\Program Files\Microsoft Office Communications Server 2007 R2\ResKit\.

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3 Run the following command:
cscript OCSTrustEntry.vbs /action:add /type:trustedservice /fqdn: /service:MediationServer /port:5061 /version:4 /routable:TRUE
Where  is the FQDN of your Collaboration Server system. The script automatically
generates the GUID discover the proper Active Directory container to store the object.

Your Collaboration Server system is now established as a trusted gateway by all Office
Communications Server pools in the domain. It is displayed in the list of voice gateways when you
establish a voice route.

Establishing a Voice Route to the Collaboration Server “Voice” Gateway
The OCS infrastructure enables you to establish a voice route to a voice gateway. Typically, this means that
all SIP INVITEs to phone numbers which match a particular pattern will be routed to a specific gateway. In
this example, all INVITEs to numbers which start with “11” will be routed to Collaboration Server11 (DNS
name rmx11.r13.vsg.local2).

To establish the voice route:
1 Open the OCS R2 management Console and right click on Forest and then click Properties >
Voice Properties.

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The Office Communications Server Voice Properties dialog box opens.
2 Click the Routes tab.
Office Communications Server Voice Properties - Routes dialog box opens.
3 Click the Add button.

The Add Route dialog box opens.
4 In the Name field, enter a name that will identify this voice route.
5 Optional. In the Description field, enter a description.

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6 In the Target Regular Expression field enter ^ and the MCU prefix as defined in the gatekeeper. This
prefix is also defined in the Collaboration Server IP Network Service.

For example, if 11 is the Collaboration Server prefix defined in gatekeeper, enter ^11. The circumflex
expression "^11" causes this route to be applied to all numbers starting with "11".
If you have not defined such a prefix in the IP Network Service in the Collaboration Server
configuration, you can add it later, using value entered here.

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7 In the Gateways - Addresses box, click the Add button.
The Add Route Gateway dialog box opens.

8 Select the Collaboration Server gateway address that was set up in Configuring the Collaboration
Server as a Routable Gateway that is displayed in the drop down list of gateways.
9 Click OK to save the address and return to the Add Route dialog box.
10 In the Phone Usage box, click the Configure button.
The Configure Phone Usage Records dialog box opens.
11 In the Available box, click Default Usage and then click the > button.

The Default Usage option is displayed in the Configured box.
12 Click OK.
13 In the Add Route dialog box, click OK to save the new route.

Configuring Office Communicator Users for Enterprise Voice
Each of the endpoints in the OCS environment must be set to use the voice route.
The setting is done in the Office Communications Server management console for all required users
(endpoints) simultaneously or in the Active Directory for each of the Users (endpoints).

To Configure Office Communicator Users for Enterprise Voice in the Office
Communications Server management console:
1 Navigate to Start > All Programs > Administrative Tools > Office Communications Server 2007
R2 to open the Office Communications Server management console.
2 Expand the Enterprise pool or Standard Edition server node where your users reside.
3 Expand the pool or server where your users reside, and then click the Users node.

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4 In the right pane, right-click one or more users whom you want to configure, and then select
Configure users.

The Welcome to the Configure Users Wizard opens.
5 On the Welcome to the Configure Users Wizard dialog box, click Next.
6 On the Configure User Settings dialog box, click Next.
7 On the Configure Meeting Settings dialog box, click Next.
8 On the Configure User Settings specify meeting policy dialog box, click Next.
9

On the Configure Enterprise Voice dialog box, select Change Enterprise Voice Settings for
selected users, and then click Enable Enterprise Voice.

10 Click Next.

11 On the Configure Enterprise Voice Settings and Location Profile dialog box, select Change
Enterprise Voice Policy for selected users.

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12 Select an Enterprise Voice policy from the list.

13 Select Change location profile for selected users.
14 Select a location profile from the list, and then click Next.
15 On the Ready to Configure Users dialog box, review the settings, and then click Next.

16 On the Configure Operation Status dialog box, verify that the operation succeeded, and then click
Finish.

To Configure Office Communicator Users for Enterprise Voice in the in the Active
Directory:
1 Open the Active Directory and navigate to the endpoint whose properties require changing.

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2 Right-click the endpoint and select Properties.
The Properties dialog box opens.
3 Click the Communications tab.

4 Click the Telephony Settings - Configure button.
The Telephony Options dialog box opens.
5 Select the Enable Enterprise Voice option.

6 Click OK to return to the Properties - Communications dialog box.
7 Click OK.

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Starting a Conferencing Call from the MOC
1 In the Office Communicator application, enter the number to dial, for example, 113344. This number
is composed of the Collaboration Server Prefix in the Gatekeeper (for example, 11) and the Meeting
Room ID, as defined on the Collaboration Server (for example, 3344).

2 Click Call, and then click Other.
The call is routed to the Meeting Room on the Collaboration Server, and the caller that initiated the
call connects as the conference chairperson.
3 The MOC User can then add video to the call, by selecting Add Video in the Office Communicator
window.

Setting Simultaneous Numerical Dialing and Matched URI Routing
You can simultaneously set up an Collaboration Server for both numerical and Matched URI dialing. If you
want to do this, follow these instructions:
1 Set the Collaboration Server as a trusted service (MediationServer) and a voice gateway using the
instructions in Setting the Numerical Dialing Method.
2 Set up a matching URI route to the Collaboration Server/DMA by right-clicking the OCS Pool,
selecting Properties > Front End Properties > Routing Tab and follow the instructions in Setting
the Static Route for Collaboration Server in the OCS.
•

•

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When defining both routing methods, you cannot add an Collaboration Server as an Authorized
Host using the Front End Properties > Host Authorization tab. There can only be one trusted
service entry for the Collaboration Server even though there are two different routes to the
Collaboration Server (i.e., Matched URI and numerical dialing). If the Matched URI routing
method was previously defined and the Collaboration Server was set as trusted host, and you are
adding the numerical dialing method, you have to remove the Collaboration Server from the
Trusted Hosts list. For more details, see Optional. Removing the Collaboration Server from the
Host Authorization List.
Only TLS connections to the Collaboration Server will work, TCP connections will not work.

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PFX Method - Creating the Security (TLS) Certificate in the OCS and
Exporting the Certificate to the Collaboration Server Workstation
If you are using the PFX method to create and send the security certificate to the Collaboration Server,
certificate files rootCA.pem, pkey.pem and cert.pem must be sent to the Collaboration Server unit. These
files can be created and sent to the Collaboration Server in two methods:
●

The files rootCA.pem, pkey.pem and cert.pem are provided by a Certificate Authority and are sent
independently or together with a password file to the Collaboration Server. This is the recommended
method.

● Alternatively, the TLS certificate files are created internally in the OCS and exported to the
Collaboration Server workstation from where the files can be downloaded to the Collaboration Server.
If the certificate is created internally by the OCS, one *.pfx file is created. In addition, a text file
containing the password that was used during the creation of the *.pfx file is manually created. Both
files can then be sent from the Collaboration Server workstation to the Collaboration Server unit.
When the files are sent to the Collaboration Server, the *.pfx file is converted into three certificate
files: rootCA.pem, pkey.pem and cert.pem.
Sometimes, the system fails to read the *.pfx file and the conversion process fails. Resending *.pfx
file again and then resetting the system may resolve the problem.
The following procedure describes how to create the *.PFX file in the OCS and export it so it can be sent to
the Certificate Authority or to the Collaboration Server.

To create the TLS certificate in the Office Communications Server:
1 In the OCS Enterprise Pools tree, expand the Pools list and the server pool list.
2 Right-click the pool Front End entity, and click Certificate.

The Office Communicator Server Wizard Welcome window is displayed.
3 Click Next.
The Available Certificate Tasks window is displayed.

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4 Select Create a New Certificate and click Next.

The Delayed or Immediate Request window is displayed.
5 Select Send the Request immediately to an online certificate authority and click Next.

The Name and Security Settings window is displayed.
6 In the Name field, select the Collaboration Server name you entered in the FQDN field when
defining the trusted host or as defined in the DNS server.

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7 Select the Mark cert as exportable check box.

8 Click Next.
The Organization Information window is displayed.
9 Enter the name of the Organization and the Organization Unit and click Next.

Your Server’s Subject Name window is displayed.
10 In the Subject name field, select the FQDN name of the Collaboration Server from the list or enter its
name.

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11 Keep the default selection in the Subject alternate name field and click Next.

12 If an error message is displayed, click Yes to continue.
The Geographical Information window is displayed.
13 Enter the geographical information as required and click Next.

The Choose a Certification Authority window is displayed.

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14 Ensure that the Select a certificate authority from the list detected in your environment option
is selected and that the local OCS front end entity is selected.

15 Click Next.
The Request Summary window is displayed.
16 Click Next to confirm the listed parameters and create the requested certificate.

The Assign Certificate Task window is displayed.

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17 Select Assign certificate later and click Next (MS R2).

The Certificate Wizard Completed window is displayed (MS R2).
18 Click Finish (MS R2).

Retrieving the Certificate from the OCS to be sent to the Collaboration Server
Workstation
1 In the OCS Enterprise Pools tree, expand the Pools list and the Server Pool list.
2 Right-click the pool Front End entity, and select Certificate.
The Available Certificate Tasks window is displayed.
3 Select Export a certificate to a *.pfx file and click Next.

The Available Certificates window is displayed.

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4 Select the certificate Subject Name of the Collaboration Server and click Next.

The Export Certificate window is displayed.
5 Enter the path and file name of the certificate file to be exported or click the Browse button to select
the path from the list.
The new file type must be *.pfx and its name must include the .pfx extension.

6 Select the Include all certificates in the certification path if possible check box and then click
Next.
The Export Certificate Password window is displayed.
7 If required, enter any password. For example, Polycom.

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8 Write down this password as you will have to manually create a password file in which this password
will be displayed.

Click Next.
The Certificate Wizard Completed window is displayed.
9 Click Finish.
The created *.pfx file is added in the selected folder.

Optional. Creating the Certificate Password File (certPassword.txt)
If you have used a password when creating the certificate file (*.pfx), you must create a certPassword.txt
file. This file will be sent to the Collaboration Server together with the *.pfx file.

To create the certPassword.txt file:
1 Using a text editor application, create a new file.
2 Type the password as you have entered when creating the certificate file. For example, enter
Polycom.
3 Save the file naming it certPassword.txt (file name must be exactly as show, the Collaboration
Server is case sensitive).

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Supporting Remote and Federated Users in Office Communications
Server ICE Environment
To enable the remote and Federation connections the following operations must be performed:
● Create an Active Directory account for the Collaboration Server that will be used for registering and
operating in the MS ICE environment
● Enable the Collaboration Server User Account for Office Communication Server
● Configure the Collaboration Server for ICE dialing for more details, see Configuring the Collaboration
Server for Federated (ICE) Dialing.
To place federated calls between Domain A and Domain B in ICE environment sub domains must be
federated to the main domain or the Collaboration Server system must be installed on a main
domain.

The Collaboration Server can also be set for Matched URI Routing and/or Numerical Dialing to Meeting
Rooms. For more details, see Setting the Matched URI Dialing Method and Setting the Numerical Dialing
Method.

Creating an Active Directory Account for the Collaboration Server
The User account created for the Collaboration Server is used for registration in the Office Communication
Server and to automatically synchronize with the STUN and relay (Edge) servers.

To add the Collaboration Server user to the Active Directory:
1 Go to Start > Run and enter dsa.msc to open the Active Directory Users and Computers console
2 In the console tree, select Users > New > User.

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3 In the New User wizard, define the following parameters:

Active Directory - New User Parameters for the Collaboration Server
Field

Description

First Name

Enter the name for the Collaboration Server user. This name will be used in the
configuration of the ICE environment in the Collaboration Server.

Full Name

Enter the same name as entered in the First Name field.

User Login Name

Enter the same name as entered in the First Name field and select from the
drop down list the domain name for this user. It is the domain name defined for
the Office Communication Server.

4 Click Next.
5 Enter the password that complies with the Active Directory conventions and confirm the password.

6 Select the options: User cannot change password and Password never expires.
Clear the other options.

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7 Click Next.
The system displays summary information.

8 Click Finish.
The new User is added to the Active Directory Users list.

Enabling the Collaboration Server User Account for Office Communication Server
The new Collaboration Server user must be enabled for registration with the Office Communications Server.

To enable the Collaboration Server User Account for Office Communication Server:
1 In the Active Directory Users and Computers window, right-click the Collaboration Server user and
then click Properties.

2 In the Properties dialog box, click the Communications tab.

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3 In the Sign in name field, enter the Collaboration Server user name in the format SIP:rmx user
name (for example sip:rmx1234) and select the domain name (for example, ilsnd.vsg.local) as
entered in the New User dialog box.

4 Select the Server or Pool from the list.
5 Click Apply and then OK.

Configure the Collaboration Server for ICE dialing
For details, see Configuring the Collaboration Server for Federated (ICE) Dialing.

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Registering the Collaboration Server as a Trusted
Application for Lync 2010/2013
The following procedures are mandatory to register the Collaboration Server to Lync 2010/2013.
In the following procedures Domain Names, Server Names and Collaboration Server Names are
examples for syntax purposes only and must be adapted to local network requirements.

Configure the Collaboration Server FQDN in the DNS
Perform the following steps on the Lync Server to configure the Collaboration Server FQDN in the DNS.

To Configure the Collaboration Server FQDN in the DNS:
1 Connect to the DNS Server.
2 Open the DNS Manager.

3 Navigate to and expand the Forward Lookup Zones folder.

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4 Right click on the Zone for your Domain.

5 Select New Host (A or AAAA).

6 Enter the Collaboration Server Name and the Signaling IP address.
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7 Click Add Host.

Configure Collaboration Server Static Route and Trusted Application
Perform the following steps on the Lync Server to configure the Collaboration Server Static Route and
Trusted Application.

To Configure the Collaboration Server Static Route and Trusted Application:
1 Connect to Lync Front End server
2 In the Start Menu Search field, enter Lync and click the Search button.

3 Select the Lync Server Management Shell.

4 Add a Static Route:
a Type: $route = New-CsStaticRoute -TLSRoute -destination "" -port 5061
-matchuri "" -usedefaultcert $true and press Enter.
 is the Collaboration Server name.
b Type: Set-CsStaticRoutingConfiguration -identity global -route @{Add=$route} and press
Enter
(To check the Static Route configuration enter the following command:
Get-CsStaticRoutingConfiguration and press Enter.)
5 Create a Trusted Application Pool and a Trusted Application:

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a Type: New-CsTrustedApplicationPool -Identity  -Registrar
Registrar: -site 1 -ComputerFqdn 
-ThrottleAsServer $true -TreatAsAuthenticated $true and press Enter.
b Enter YES.
c Type: New-CsTrustedApplication -ApplicationId  -TrustedApplicationPoolFqdn
 -Port 5061 and press Enter.
ApplicationId is the name of the application. This must be a string that is unique within the pool
that is specified in the TrustedApplicationPoolFqdn parameter.
TrustedApplicationPoolFqdn is the FQDN of the Trusted Application Pool in which the
application will reside.
6 Add a Trusted Application Endpoint by typing: New-CsTrustedApplicationEndpoint
-sipaddress sip:@ -ApplicationId  -TrustedApplicationPoolFqdn
 and pressing Enter.
7 Enable the changes, by typing: Enable-CsTopology.
8 Get the GRUU (Globally Routable User-Agent URI):
a Type: Get-CsTrustedApplication -Filter "*" | fl ServiceGruu and press Enter.
The Lync Server will reply with a string similar to the following:
ServiceGruu:sip:rmx23772.reg13.ent@reg13.ent;gruu;opaque=srvr:
rmx23772.reg13.ent:0I2d_trGDFSQv4FntDXH-wAA
b Select, Copy, and transfer the ServiceGruu string into the workstation that is connected to the
Collaboration Server. The SIP prefix (sip:) can be omitted from the copied string.
9 Create and export a Certificate for the Collaboration Server:
a Type: Request-CsCertificate -New -Type Default -KeyAlg RSA -CA  -City PT
-State Israel -ComputerFqdn  -Country IL -DomainName
 -FriendlyName  -Organization 'Polycom '
-PrivateKeyExportable $true and press Enter.
b In the Start Menu, select Administrative Tools-> Internet Information Services (IIS) Manager.

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c Click Server Certificates.

d In the Certificate List, locate the Collaboration Server’s certificate.

e Right click the Collaboration Server’s certificate.

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f

Click Export.

g In the Export to field, enter the file location for exported certificate.
h In the Password and Confirm Password fields, enter the password.
i

Click OK.

Configure the Collaboration Server for Lync 2010/2013
Perform the following steps on the Collaboration Server to configure it for Lync 2010/2013.

To Configure the Collaboration Server for Lync 2010/2013:
1 Open the IP Network Management Services list.

2 Double click Management Network.

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3 Click the DNS tab.

4 In the MCU Host Name field, enter the name of the DNS (as listed in the DNS Manager).
5 In the Local Domain Name field, enter the Domain Name (as listed in the DNS Manager).
6 In the DNS Servers Primary Server field, enter the DNS IP Address (as listed in the DNS Manager).
7 Click OK
8 Restart the Collaboration Server.
9 Login again and open the IP Network Management Services list.
10 Double click IP Network Service.

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11 Click the DNS tab.

12 In the IP Network Type drop down menu, select H.323 &SIP.
13 In the Service Name (FQDN) field, enter the DNS Name (as listed in the DNS Manager).
This should not be the FQDN.
14 In the DNS drop down menu, select Specify.
15 In the Local Domain Name field, enter the Domain Name (as listed in the DNS Manager).
16 In the DNS Server Address field, enter the IP Address of the DNS server (as listed in the DNS
Manager).

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17 Click the SIP Servers tab.

18 In the IP Network Type drop down menu, select H.323 &SIP.
19 In the SIP Server drop down menu, select Specify.
20 In the SIP Server Type drop down menu, select Microsoft.
21 In the Transport Type drop down menu, select TLS.
22 In the SIP Servers and Outbound Proxy Servers Parameter tables:
 Set the Server IP Address to the DNS Pool name (as listed in the DNS Manager).
 Set the Server Domain to the Domain Name (as listed in the DNS Manager).
 Set Port to 5061
23 Click the SIP Advanced tab.

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24 In the Server User Name field, enter the Collaboration Server’s SIP Address.
This is the first segment of the ServiceGruu string with the an “@” replacing the “.”
For example if the ServiceGruu string was:
rmx23772.reg13.ent@reg13.ent;gruu;opaque=srvr:rmx23772.reg13.ent:0I2d_trGDFSQv4FntDXHwAA,
the Server User Name would be rmx23772@reg13.
25 Add the SIP_CONTACT_OVERRIDE_STR system flag with the ServiceGruu string as its value.
a On the workstation that is connected to the Collaboration Server, on the RMX Menu, click Setup
> System Configuration.
The System Flags dialog opens.

b In the System Flags dialog, click the New Flag button.
The New Flag dialog box is displayed.

c In the New Flag field, enter SIP_CONTACT_OVERRIDE_STR.
d Locate, Copy and Paste the ServiceGruu string, that was transferred to the workstation in Step
8b of the Configure the Collaboration Server Static Route and Trusted Application above,
into the Value field.
e Click OK to close the New Flag dialog.
26 Add the LIMIT_CIF_SD_PORTS_PER_MPMX_CARD System Flag and set its value to YES.
Setting this flag’s value to YES will limit the number of CIF and SD ports (including non Lync clients)
to 45 per media card. The flag value and must be set to YES when ICE is active because ICE ports
consume more system resources. HD participants are not affected.

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27 Repeat Steps 25b - 25e above with:
 New Flag field: LIMIT_CIF_SD_PORTS_PER_MPMX_CARD.
 Value field: YES.
28 Reset the Collaboration Server for the new flag settings to take effect.
a Click OK to close the New Flag dialog.
b Click OK to close the System Flags dialog.
c Reset the Collaboration Server.
29 After the rest has completed, Login to the Collaboration Server again.

Import and install the Certificate on the Collaboration Server
The Certificate created in Step 9 of the Configure Collaboration Server Static Route and Trusted
Application procedure above must be imported an installed on the Collaboration Server.

To import and install a certificate:
1 In the Collaboration Server menu, click Setup > RMX Secured Communications > Certificate
Repository.
2 Click the Personal Certificates tab.

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3 Click Add.

4 In the Network Service Name drop down menu, select IP Network Service.
5 In the Certificate Method drop down menu, select PEM/PFX.
6 In the Password field, enter the password entered in Step 9h of the Configure Collaboration
Server Static Route and Trusted Application procedure.
7 Click Send Certificate.

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8 Click Browse, and then navigate to the folder where the certificate was saved.

9 Select the certificate and click Open.
The certificate is installed and a Confirmation Message is displayed.

10 Click OK in the confirmation message.
A System Reset confirmation message is displayed.
11 If you are ready to reset the system, click YES.
The System Reset completes the Registration of the Collaboration Server to Lync 2010/2013.

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Collaboration Server System Flag Configuration
Enabling the Microsoft Environment
The Collaboration Server can be installed in Microsoft R2 environments. To adjust the Collaboration Server
behavior to the Microsoft environment in each release, system flags must be set.

To configure the system flags on the Polycom Collaboration Server system:
1 On the Collaboration Server menu, click Setup > System Configuration.
The System Flags - MCMS_PARAMETERS_USER dialog box opens.
2 Scroll to the flag MS_ENVIRONMENT and click it.
The Edit Flag dialog box is displayed.
3 In the Value field, enter YES to set the Collaboration Server SIP environment to Microsoft solution.
Collaboration Server set to MS_ENVIRONMENT=YES supports SIP over TLS only and not over
TCP.

4 Click OK to complete the flag definition.
5 When prompted, click Yes to reset the MCU and implement the changes to the system
configuration. After system reset the Collaboration Server can register to the OCS server and make
SIP calls.
Sometimes the system fails to read the *.pfx file and the conversion process fails, which is indicated
by the active alarm “SIP TLS: Registration server not responding” and/or “SIP TLS: Registration
handshake failure”. Sending *.pfx file again, as described in this procedure and then resetting the
system may resolve the problem.

In some configurations, the following flags may require modifications when MS_ENVIRONMENT flag is set
to YES:

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Additional MS Environment Flags in the Collaboration Server MCMS_PARAMETERS_USER Tab
Flag Name

Value and Description

SIP_FREE_VIDEO_RESOURCES

Default value in Microsoft environment: NO.
When set to NO, video resources that were allocated to participants
remain allocated to the participants as long as they are connected to
the conference even if the call was changed to audio only. The system
does not allocate the resources to other participants ensuring that the
participants have the appropriate resources in case they want to return
to the video call.
The system allocates the resources according to the participant’s
endpoint capabilities, with a minimum of one CIF video resource.
When this flag is set to YES, video ports are dynamically allocated or
released according to the in the endpoint capabilities. For example,
when an audio Only call is escalated to Video and vice versa or when
the resolution is changed.

SIP_FAST_UPDATE_INTERVAL_ENV

Default setting is 0 to prevent the Collaboration Server from
automatically sending an Intra request to all SIP endpoints.
Enter n (where n is any number of seconds other than 0) to let the
Collaboration Server automatically send an Intra request to all SIP
endpoints every n seconds.
It is recommended to set the flag to 0 and modify the frequency in
which the request is sent at the endpoint level (as defined in the next
flag).

Setting the audio protocol for the Microsoft Client running on a single core PC
By default, Microsoft Office Communicator R2 or Lync Clients are connected to conferences using the
G.722.1 audio algorithm. However, when these clients are hosted on single processor workstations, they
may experience audio quality problems when this algorithm is used.
The System Flag FORCE_AUDIO_CODEC_FOR_MS_SINGLE_CORE is used to force the use of a
specific Audio algorithm such as G.711 when a Microsoft Office Communicator R2 or Lync Client is detected
as being hosted on a single core processor.
This flag can be set to:
● AUTO – No forcing occurs and the Collaboration Server negotiates a full set of Audio algorithm during
capabilities exchange.
● G711A/U or G722 – Set this flag value according to the hosting workstation capabilities. If the
Collaboration Server detects single core host during capabilities exchange it will assign a G.711 or
G.722 Audio algorithm according to the flag value.
Possible values: AUTO, G711A, G711U, G722
Default: G711A

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Microsoft RTV Video Protocol Support in CP
Conferences
Microsoft RTV (Real Time Video) protocol provides high quality video conferencing capability to Microsoft
OC (Office Communicator) Client endpoints at resolutions up to HD720p30. Interoperability between
Polycom HDX and OCS endpoints is improved.

Guidelines
● The RTV protocol is supported:
 In SIP networking environments only
 In CP mode only
● OCS (Wave 13) and Lync Server (Wave 14) clients are supported.
● RTV is supported in Basic Cascade mode.
● RTV is the default protocol for OCS endpoints and Lync Server clients connecting to a conference.
● RTV participants are supported in recorded conferences.
● RTV participant encryption is supported using the SRTP protocol.
● Video Preview is not supported for RTV endpoints.
● Custom Slides in IVR Services are not supported for RTV endpoints.
● HD720p30 resolution is supported at bit rates greater than 600 kbps. The following table summarizes
the resolutions supported at the various bit rates.
RTV - Resolution by Bit Rate
Resolution

Bitrate

QCIF

Bitrate <180kbps

CIF30

180kbps < Bitrate < 250kbps

VGA (SD30)
HD720p30

250kbps < Bitrate < 600kbps *
600kbps < Bitrate *

* Dependant on the PC’s capability
● System Resource usage is the same as for the H.264 protocol. The table below summarizes System
Resource usage for each of the supported resolutions.
RTV - Resolution by Resolution
Resolution

HD Video Resources Used

QCIF / CIF30

0.5

VGA (SD30) / W4CIF

0.5

HD720p30

1

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Participant Settings
When defining a new participant or modifying an existing participant, select SIP as the participant’s
networking environment Type in the New Participant or Participant Properties - General tab.

The participants Video Protocol in the New Participant or Participant Properties - Advanced tab should be left
at (or set to) its default value: Auto.
The Auto setting allows the video protocol to be negotiated according to the endpoint’s capabilities:
● OCS endpoints and Lync Server clients connect to the conference using the RTV protocol.
● Other endpoints negotiate the video protocol in the following sequence: H.264, followed by RTV,
followed by H.263 and finally H.261.

Protocol Forcing
Selecting H.264, RTV, H.263 or H.261 as the Video Protocol results in endpoints that do not support the
selected Video Protocol connecting as Secondary (audio only).

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Monitoring RTV
RTV information is displayed in all three panes of the Participant Properties - SDP tab.

Controlling Resource Allocations for Lync Clients Using RTV Video
Protocol
The number of resources used by the system to connect a Lync client with RTV is determined according to
the conference line rate and the Maximum video resolution set in the Conference Profile.
system flag MAX_RTV_RESOLUTION enables you to override the Collaboration Server resolution
selection and limit it to a lower resolution. Resource usage can then be minimized the 1 or 1.5 video
resources per call instead of 3 resources, depending on the selected resolution.
Possible flag values are: AUTO (default), QCIF, CIF, VGA or HD720.
For example, if the flag is set to VGA, conference line rate is 1024Kbps, and the Profile Maximum Resolution
is set to Auto, the system will limit the Lync RTV client to a resolution of VGA instead of HD720p and will
consume only 1.5 video resources instead of 3 resources.

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When set to AUTO (default), the system uses the default resolution matrix based on the conference line
rate.
To change the default flag setting, add the MAX_RTV_RESOLUTION flag to the System Configuration flags
and set its value. For information, see the RealPresence Collaboration Server 800s Administrator’s
Guide, the RealPresence Collaboration Server 800s Administrator’s Guide, Modifying System
Flags.
The following table summarizes the Collaboration Server resources allocated to a Lync Client based on the
MAX_RTV_RESOLUTION flag setting, the connection line rate and the video resolution.
Selected video resolution based on flag setting and conference line rate and core processorThe following table

describes the number of allocated video resources for each video resolution when using the RTV protocol.
Allocated video resolutions per video resolution

Threshold HD Flag Settings using the RTV Video Protocol
The system flag MAX_ALLOWED_RTV_HD_FRAME_RATE defines the threshold Frame Rate (fps) in
which RTV Video Protocol initiates HD resolutions.
Flag values are as follows:
● Default: 0 (fps) - Implements any Frame Rate based on Lync RTV Client capabilities
● Range: 0-30 (fps)
For example, when the flag is set to 15 and the Lync RTV Client declares HD 720P at 10fps, because the
endpoint’s frame rate (fps) of 10 is less than flag setting of 15, then the endpoint’s video will open VGA and
not HD.
In another example, when the flag is set to a frame rate of 10 and the Lync RTV Client declares HD 720P
at 13fps, because the endpoint’s frame rate (fps) of 13 is greater than flag setting of 10, then the endpoint’s
video will open HD and not VGA

Sharing Content via the Polycom CSS Plug-in for Lync
Clients
From version 8.1, Polycom CSS (Content Sharing Suite) Plug-in for Lync clients allows Lync clients to
receive and send Content on a separate channel, without having to use the video channel. Content is
transmitted using SIP BFCP.
When Lync clients connect, each endpoint is represented twice in the RMX Manager or Collaboration Server
Web Client. One connection represents the actual Lync client, while the second connection represents the
content channel via the Polycom plug-in.
The name of the plug-in “participant” is derived from the name of the Lync client with the suffix “_cssplugin”.
When a Lync client connects to a conference, the plug-in connects automatically, regardless of whether the
Lync client dials into a conference or is called from the MCU.

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Guidelines
● The maximum resolution for content sharing via the Polycom CSS plug-in is HD720p5.
● The Polycom CSS plug-in supports H.263 and H.264 video protocols for content sharing.
● SVC-enabled endpoints use the AVC (H.264) protocol for sharing content.
● Content can be shared between different types of endpoints, using different network protocols
(H.323, SIP and ISDN/PSTN).
● TIP content is not supported.
● Lync 2013 is supported.
● ICE is not supported.

Configuring the MCU for Content Sharing via the Polycom CSS Plug-in
You can configure the MCU for content sharing via the Polycom CSS plug-in by setting the following
parameters:
● Setting the BLOCK_CONTENT_LEGACY_FOR_LYNC system flag
● Setting the Content parameters in the conference Profile

Setting the System Flag
By configuring the system flag BLOCK_CONTENT_LEGACY_FOR_LYNC you control the system
behavior in an environment where some Lync clients use the Polycom CSS plug-in and some do not. This
flag must be manually added to the system configuration to change its value.
● When set to NO (default), Content is sent to all Lync clients over the video channel, including those
with the Polycom CSS plug-in installed, even when the Send Content to Legacy Endpoints is
disabled.
Other, non-Lync legacy endpoints will not be affected by this flag and will receive content according
to the Send Content to Legacy Endpoints settings in the conference Profile.
● When set to YES, Content is not sent to Lync clients over the video channel including those with the
Polycom CSS plug-in installed, even when the Send Content to Legacy Endpoints is enabled.
Other, non-Lync legacy endpoints will not be affected by this flag and will receive content according
to the Send Content to Legacy Endpoints settings in the conference Profile.

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Conference Profile Settings
Content is shared in a video switching mode. Therefore, when a Lync client connects to the conference via
the Polycom CSS plug-in, the content resolution will be adjusted to the maximum content rate possible by
the Lync client, up to a maximum of 720p 5fps in all line rates, even if you select a higher content rate and
resolution in the Conference Profile.

Monitoring the Participant connection
● Under properties of the participant representing the CSS plug-in, Channel Status, audio channels are
shown, but audio is not used in the plug-in. The information can be ignored.

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● The properties of the Lync client are those of a video participant. However, the Content channel will
show 0 as there is no content channel.

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Adding Presence to Conferencing Entities in the
Buddy List
Registration of conferencing entities (Meeting Rooms, Entry Queues and SIP Factories) with the SIP server
adds these conferencing entities to the buddy list with their presence. It enables the Office Communication
Server client or LYNC Server client users to see the availability status (Available, Offline, or Busy) of these
conferencing entities and connect to them directly from the buddy list.

Guidelines
● Registration with Presence of up to 100 conferencing entities to a single SIP Server is supported.
When this number is exceeded, the additional conferencing entity may appear to be successfully
registered but the presence status will be shown as 'Offline' in Lync for any entities beyond the limit.
● Lync endpoints cannot connect to conferencing entities that their presence is “offline”.
● The Conferencing Entity: Meeting Room, Entry Queue, SIP Factory (Routing Name) has to be added
to the Active Directory as a User.
Make sure that a unique name is assigned to the conferencing entity and it is not already used for
another user account in the Active Directory.
● The conferencing entity name must not include any upper case letters or special characters: @ # $
% ^ & * ( ) _ - = + | } { : " \ ] [ ; / ? > < , . (space) ~.
● When the MCU system is shutting down while a Meeting Room is still active, as indicated by its
presence, the status remains active for 10 minutes during which Lync endpoints cannot connect to
the Meeting Room. After 10 minutes, the Meeting Room Status changes to “offline”.
● Registration of the conferencing entity is defined in the Conference Profile (and not in the IP Network
Service), enabling you to choose the conferencing entity to register.
● In Multiple Networks configuration, an IP Network Service that is enabled for registration in a
Conference Profile cannot be deleted.

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Enabling the Registration of the Conferencing Entities
The creation of the various conferencing entities is described in the following chapters:
● Meeting Rooms
● Entry Queues, Ad Hoc Conferences and SIP Factories
Registration with presence of conferencing entities with the SIP Server is enabled by performing the
following processes:
● Creating an Active Directory Account for the Conferencing Entity.
● Enabling the Conferencing Entity User Account for Office Communication Server or Lync Server
● Defining the Microsoft SIP Server in the IP Network Service
● Enabling Registration in the Conference Profile

Creating an Active Directory Account for the Conferencing Entity
The User account created for the Conferencing entity is used for registration with the Office Communication
Server or Lync server and to automatically synchronize with the STUN and relay (Edge) servers.

To add the conferencing entity user to the Active Directory:
1 Go to Start > Run and enter dsa.msc to open the Active Directory Users and Computers console.
2 In the console tree, select Users > New > User.

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3 In the New User wizard, define the following parameters:

Active Directory - New User Parameters for the Collaboration Server
Field

Description

First Name

Enter the name of the conferencing entity user. This name will appear in the
buddy list of the Office Communication Server or Lync server. For example,
vmr10.
Notes:
• This name must be the identical to the Routing Name assigned to the
conferencing entity in the Collaboration Server system. It must also be the
User Login Name in the Active Directory.
• The name can include only lower case characters and/or numbers.

Full Name

Enter the same name as entered in the First Name field.

User Login Name

Enter the same name as entered in the First Name field and select from the
drop down list the domain name for this user. It is the domain name defined for
the Office Communication Server or Lync server.

4 Click Next.
5 Enter the password that complies with the Active Directory conventions and confirm the password.
6 Select the options: User cannot change password and Password never expires.
Clear the other options.
7 Click Next.
The system displays summary information.
8 Click Finish.
The new User is added to the Active Directory Users list.
9 Repeat for each Collaboration Server conferencing entity.

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Enabling the Conferencing Entity User Account for Office Communication Server
or Lync Server
The new Conferencing Entity user must be enabled for registration with the Office Communications Server
or Lync Server.

To enable the Conferencing Entity User Account for Office Communication Server:
1 In the Active Directory Users and Computers window, right-click the conferencing entity user and
then click Properties.
2 In the Properties dialog box, click the Communications tab.
3 In the Sign in name field, enter the conferencing entity user name in the format SIP:conferencing
entity user name (for example sip:vm10) and select the domain name (for example, lab.vsg.local)
as entered in the New User dialog box.
4 Select the Server or Pool from the list.
5 Click Apply and then OK.

To enable the Conferencing Entity User Account for Lync Server:
1 On the computer running the Lync Server 2010, go to Start->All Programs->Microsoft Lync
Server 2010>Lync Server Control Panel.
Windows Security window opens.
2 Enter your User name and Password as configured in the Lync Server and click OK.
The Microsoft Lync Server 2010 Control Panel window opens.
3 Click the Users tab.
4 In the User Search pane, click the Enable Users heading.
The New Lync Server User pane opens.
5 Click the Add button.
The Select from Active Directory dialog box opens.
6 Enter the conferencing entity user name as defined in the Active Directory, and then click the Find
button.
The requested user is listed in the Select From Active Directory dialog box.
7 Select the listed user (conferencing entity user) and click OK.
The selected user is displayed in the New Lync Server User pane.

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8 Select the following parameters:



In Assign users to a pool field, select the required pool.

 In the Generate user SIP URI, define the SIP URI of the conferencing entity using one of the
following methods:
 Select the Specify a SIP URI option and enter the conferencing entity user portion of SIP URI
defined in the active directory. This SIP URI must match the conferencing entity Routing Name
configured in Collaboration Server. For example, for the meeting room account
sip:vmr10@wave4.eng, use only the vmr10 portion of the address.
or
 Select the Use the user principal name (UPN) option.
9 Click the Enable button.
The selected user is displayed as enabled in the User Search pane.

Defining the Microsoft SIP Server in the IP Network Service
To enable the registration of the conferencing entities the SIP Server Type must be set to Microsoft and the
Office Communication Server or Lync Server properties in the IP Network Service - SIP Servers dialog box.
For more details, see Configuring the Collaboration Server IP Network Service.

Enabling Registration in the Conference Profile
Registration of conferencing entities such as ongoing conferences, Meeting Rooms, Entry Queues, SIP
Factories and Gateway Sessions with SIP servers is done per conferencing entity. This allows better control
on the number of entities that register with each SIP server.

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Selective registration is enabled by assigning a conference Profile in which registration is enabled to the
conferencing entities that require registration. Assigning a conference Profile in which registration is
disabled (registration check box is cleared) to conferencing entities will prevent them from registering. By
default, Registration is disabled in the Conference Profile, and must be enabled in Profiles assigned to
conferencing entities that require registration.

Profile Properties - Network Services
Parameter

Description

IP Network Services:
Service Name

This column lists all the defined Network Services, one or several depending on the
system configuration (single Network or Multiple Networks).

SIP Registration

To register the conferencing entity to which this profile is assigned, with the SIP Server
defined for that Network Service, click the SIP Registration check box of that Network
Service.

Accept Calls

To prevent dial in participants from connecting to a conferencing entity when connecting
via a certain Network Service, clear the Accept Calls check box of that Network Service.

Verifying the Collaboration Server Conferencing Entity Routing Name and Profile
Collaboration Server conferencing entity can be dialed directly from the buddy list of the Office
Communications client or the Lync client if its routing name matches the user name of Active Directory
account you created and Registration is enabled in the Conference Profile assigned to it.

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● To ensure that the Collaboration Server meeting room or conferencing entity is properly configured
for registration the following parameters must be defined:
 The user name on the conferencing entity in Active Directory account must be identical to its
Routing Name on the Collaboration Server.
For example, if the SIP URI in the Active Directory is sip:vmr10@wave4.eng, it must be defined
as vmr10 in the Routing Name field of that Collaboration Server conferencing entity.

 In the Profile field, make sure that a conference Profile that has been enabled for SIP registration
is selected.

Monitoring the Registration Status of a Conferencing Entity in the
Collaboration Server Web Client or RMX Manager Application
The Status of the SIP registration can be viewed in the appropriate conferencing Entity list or when
displaying its properties.

Conferencing Entity List
The list of conferencing entity includes an additional column - SIP Registration, which indicates the status
of its registration with the SIP server. The following statuses are displayed:
● Not configured - Registration with the SIP Server was not enabled in the Conference Profile
assigned to this conferencing Entity. In Multiple Networks configuration, If one service is not
configured while others are configured and registered, the status reflects the registration with the
configured Network Services. The registration status with each SIP Server can be viewed in the
Properties - Network Services dialog box of each conferencing entity.
When SIP registration is not enabled in the conference profile, the Collaboration Server's registering
to SIP Servers will each register with an URL derived from its own signaling address. This unique
URL replaces the non-unique URL, dummy_tester, used in previous versions.
● Failed - Registration with the SIP Server failed.
This may be due to incorrect definition of the SIP server in the IP Network Service, or the SIP server
may be down, or any other reason the affects the connection between the Collaboration Server or the
SIP Server to the network.
● Registered - the conferencing entity is registered with the SIP Server.
● Partially Registered - This status is available only in Multiple Networks configuration, when the
conferencing entity failed to register to all the required Network Services (if more than one Network
Service was selected for Registration). The registration status with each SIP Server can be viewed
in the Properties - Network Services dialog box of each conferencing entity.

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Ongoing Conferences list - SIP Registration

Meeting Rooms list - SIP Registration

Entry Queues list - SIP Registration

SIP Factories list - SIP Registration

Conferencing Entity Properties
Registration status is reflected in the Properties - Network Services dialog box:

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Ongoing conference Properties - Network Services - SIP Registration

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Meeting Room Properties - Network Services - SIP Registration

Entry Queue Properties - Network Services - SIP Registration

Collaboration Server Configuration for CAC
Implementation
CAC is enabled by manually adding the flags to the system Configuration and setting their values as follows:
● To enable the Call Admission Control implementation in the Collaboration Server:
 CAC_ENABLE=YES
● In addition, to ensure that endpoints such as HDX remain connected to the conference for its duration
when the Collaboration Server is configured with FQDN address and the Lync server is working with
load balancing and holds more than one address, the following two flags must be manually added
and set to:
 MS_KEEP_ALIVE_ENABLE = YES
Note: Since the keep alive is only required when the Lync server is working with load balancing
and holds more than one address, the default value is NO.
 SIP_TCP_PORT_ADDR_STRATEGY = 1 (default setting)

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● When Call Admission Control is enabled in the local network, by default the local the ICE channel is
closed after applying CAC bandwidth management.
To change and preserve the ICE channel open throughout the call:
 PRESERVE_ICE_CHANNEL_IN_CASE_OF_LOCAL_MODE=YES.

Conferencing Behavior in CP Conferences
In Continuous Presence conference, Lync clients connect with any allocated bandwidth.

Monitoring Participant Connections
Activation of the Call Admission Control for a call can be viewed in the Participant Properties - Call
Admission Control dialog box.

This information applies only to dial-in participants.
The following information is available:
Participant Properties - Call Admission Control Parameters
Field

Description

Requested Bandwidth

Indicates the bandwidth requested by the Lync client (usually the line rate set
for the conference).
NA - indicates that Call Admission Control is disabled.

Allocated Bandwidth

The actual bandwidth allocated by the Lync Policy Server.
NA - indicates that Call Admission Control is disabled.

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Connecting a Collaboration Server Meeting Room to a
Microsoft AV-MCU Conference
Microsoft Lync users can connect an Collaboration Server Meeting Room to a conference running on the
Microsoft A/V MCU. This allows Collaboration Server Lync users to connect with a conference in progress
on the A/V MCU and be an active participant in the conference.
The connection to the A/V MCU uses the same configuration as in a cascading conference between multiple
Collaboration Server MCUs, with the only difference that both the MCU and the AV-MCU should reside on
the same network subnet, so that no firewall or other barriers will trigger the NAT Traversal functionality.

To connect to an A/V MCU conference:
1 From the Menu bar, click Meet Now to create an ad-hoc conference.
The Group Conversation dialog box is displayed.

2 From the Contacts List on Lync, drag a Virtual Meeting Room (VMR) into the Group Conversation
list.
After the Virtual Meeting Room is connected on Lync, an invitation is sent from the A/V MCU to the
Collaboration Server using the Centralized Conference Control Protocol (CCCP). The Collaboration Server
responds and triggers a standard SIP invite sent from the A/V MCU to the Collaboration Server.
Multiple participants can now connect to both the Collaboration Server Meeting Room and the A/V MCU,
and participate in a cascaded conference.
When a conference begins with Audio Only, a Lync user cannot add video to the conference after
the VMR is connected to the conference. The conference will remain as Audio Only.

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Configuring the Collaboration Server for Federated
(ICE) Dialing
The Collaboration Server Default IP Network Service must be configured to work with the Office
Communication Server/Lync Server as the SIP Server and the Collaboration Server user defined in the
Active Directory must also be defined in the Collaboration Server ICE environment parameters to enable
remote dialing in a federated (ICE) environment, .
The procedure described here assumes that the Collaboration Server is configured to work in
Microsoft environment as described in Configuring the Collaboration Server for Microsoft
Integration.

To configure the Collaboration Server for ICE Dialing:
1 In the RealPresence Collaboration Server Web browser, in the RealPresence Collaboration Server
RMX Management pane, expand the Rarely Used list and click IP Network Services.
2 In the IP Network Services pane, double-click the Default IP Service entry.
The Default IP Service - Networking IP dialog box opens.
3 Click the SIP Servers tab.

4 Make sure that the SIP Server is set to Specify.
5 Make sure that the SIP Server Type is set to Microsoft.
6 Make sure that the IP address of the Office Communications Server 2007 or Lync Server 2010 is
specified and the Server Domain Name is the same as defined in the OCS/Lync Server and in the
Management Network for the DNS.

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7 Click the SIP Advanced tab.

8 In the ICE Environment field, select MS (for Microsoft ICE implementation) to enable the ICE
integration.
9 In the Server User Name field, enter the Collaboration Server User name as defined in the Active
Directory. For example, enter rmx111.
This field is disabled if the ICE Environment field is set to None.
10 Multiple Lync Registrations can be configured on the Collaboration Server. For more information see
Multiple Lync Registrations on the Collaboration Server.
11 Optional if the Fixed Ports options was selected previously.
Click the Ports tab to modify the number of UDP Ports allocated to the calls to accommodate the
number of ports required for ICE dialing.
12 In the UDP Port Range, modify the number of UDP ports by enter the first and last port numbers in
the range. When ICE environment is enabled, the number of ports defined in the range should be
2048.
13 Click OK.
The Collaboration Server will register with the OCS/Lync Server enabling automatic retrieval of the
STUN server and Relay server parameters for ICE dialing.
These parameters can be viewed in the Signaling Monitor - ICE Servers dialog box.

Monitoring the Connection to the STUN and Relay Servers in the ICE
Environment
1 In the Collaboration Server Web browser, in the Collaboration Server Management pane, click
Signaling Monitor.
2 In the Signaling Monitor pane, click the IP Network Service entry.
3 Click the ICE Servers tab.
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The system lists the ICE servers to which it is connected.It addition, the system indicates the status
of the firewall detection in the Collaboration Server.

Monitoring the Participant Connection in ICE Environment
For each participant in the conference running in ICE environment, you can view the local and the external
IP addresses and the type of connection between the Collaboration Server and the participant (remote).
The ICE information is displayed only for the media channels and not the signaling channel.

To view the channel properties of the participant:
1 In the participants pane, double-click the participant entry or right-click the participant entry and then
click Properties.
2 Click the Channel Status - Advanced tab.

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The following connection information is displayed:
Participant Properties - ICE Connection Parameters
Field

Description

Collaboration Server IP
Address

The local IP address and port (in the format IP address:Port) of the
Collaboration Server.

Participant IP Address

The local IP address and port (in the format IP address:Port) of the endpoint.

ICE Collaboration Server
IP Address

The IP address and the Port number of the Collaboration Server used to pass
through the media. This information changes according to the ICE connection
type:
• When ICE connection type is local, it is identical to the IP address:Port
displayed in the Collaboration Server IP Address.
• When ICE connection type is relay, the system displays the IP address and
port number of the relay server used to pass the media from the Collaboration
Server to the participant.
• When ICE connection type is firewall, the system displays the public IP
address and port of the Collaboration Server as seen outside the private
network.

ICE Participant IP
Address

The IP address and the Port number of the endpoint used to pass through the
media. This information changes according to the ICE connection type:
• When ICE connection type is local, it is identical to the IP address:Port
displayed in the Participant IP Address.
• When ICE connection type is relay, the system displays the IP address and
port number of the relay server used to pass the media from the participant
to the Collaboration Server.
• When ICE connection type is firewall, the system displays the public IP
address and port of the endpoint as seen outside the private network.

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Field

Description

ICE Connection Type

Indicates the type of connection between the Collaboration Server and the
participant in the ICE environment:
• Local (or Host) - The endpoint (Remote) is on the same network as the
Collaboration Server and the media connection is direct, using local
addresses.
• Relay - Media between the Collaboration Server and the participant passes
through a media relay server.
• Firewall - Media connection between the Collaboration Server and the
participant is done using their external IP addresses (the IP addresses as
seen outside of the local network).

For a detailed description of ICE Active alarms, see ICE Active Alarms.

Active Alarms and Troubleshooting
Active Alarms
The following active alarms may be displayed in the Collaboration Server System Alerts pane when the
Collaboration Server is configured for integration in the OCS environment:
Active Alarms
Alarm Code

Alarm Description

SIP TLS: Failed to load or verify
certificate files

This alarm indicates that the certificate files required for SIP TLS could not
be loaded to the Collaboration Server. Possible causes are:
• Incorrect certificate file name. Only files with the following names can be
loaded to the system: rootCA.pem, pkey.pem, cert.pem and
certPassword.txt
• Wrong certificate file type. Only files of the following types can be loaded
to the system: rootCA.pem, pkey.pem and cert.pem and
certPassword.txt
• The contents of the certificate file does not match the system
parameters

SIP TLS: Registration transport error

This alarm indicates that the communication with the SIP server cannot be
established. Possible causes are:
• Incorrect IP address of the SIP server
• The SIP server listening port is other than the one defined in the system
• The OCS services are stopped
Note:
Sometimes this alarm may be activated without real cause. Resetting the
MCU may clear the alarm.

SIP TLS: Registration handshake
failure

This alarm indicates a mismatch between the security protocols of the
OCS and the Collaboration Server, preventing the Registration of the
Collaboration Server to the OCS.

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Alarm Code

Alarm Description

SIP TLS: Registration server not
responding

This alarm is displayed when the Collaboration Server does not receive a
response from the OCS to the registration request in the expected time
frame. Possible causes are:
• The Collaboration Server FQDN name is not defined in the OCS pool,
or is defined incorrectly.
• The time frame for the expected response was too short and it will be
updated with the next data refresh. The alarm may be cleared
automatically the next time the data is refreshed.
Alternatively, the OCS Pool Service can be stopped and restarted to
refresh the data.
• The Collaboration Server FQDN name is not defined in the DNS server.
Ping the DNS using the Collaboration Server FQDN name to ensure
that the Collaboration Server is correctly registered to the DNS.

SIP TLS: Certificate has expired

The current TLS certificate files have expired and must be replaced with
new files.

SIP TLS: Certificate is about to expire

The current TLS certificate files will expire shortly and will have to be
replaced to ensure the communication with the OCS.

SIP TLS: Certificate subject name is
not valid or DNS failed to resolve this
name

This alarm is displayed if the name of the Collaboration Server in the
certificate file is different from the FQDN name defined in the OCS.
Note:
Occasionally this alarm may be activated without real cause. Resetting the
MCU may clear the alarm.

ICE Active Alarms
When ICE environment is enabled in the Collaboration Server, failure to communicate with a required
component triggers the display of an Active Alarm in the System Alerts pane.

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The following table lists these active alarms:
ICE Environment - Collaboration Server Active Alarms
Active Alarm

Phase

Alarm Displayed When

Troubleshooting

ICE failure: Failed to
register with OCS.
Check the
Collaboration Server
Server Name.

Registration

The Collaboration Server did
not receive a confirmation
response from the OCS to the
Registration request.

•

•

ICE failure: Failed to
subscribe with the
OCS, therefore the A/V
Edge Server URI was
not received.

Subscribe

The Collaboration Server did
not receive a confirmation
response from the OCS to the
Subscription request.
The Subscription is required
for obtaining the A/V Edge
Server URI which is followed
by the notify message
containing the credentials).

ICE failure: The Notify
message containing the
A/V Edge Server URI
was not received

Notify

The Notify message
containing the A/V Edge
Server URI was not received
by the Collaboration Server.

ICE failure: Received
Notification does not
contain URI.

Notify

The notify message that was
sent from the A/V Edge Server
does not contain the A/V Edge
server URI.

ICE failure: No
response from the A/V
Edge Server to the
Collaboration Server
Service Request

Service

The Collaboration Server did
not receive a confirmation
response from the A/V Edge
Server to the Service request.

ICE failure: Received
Service message does
not contain the
Credentials.

Service

The Service message
response does not contain the
Credentials.

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Check that the Collaboration
Server Name in IP Network
Service - SIP Advanced is
identical to the User name
defined for the Collaboration
Server in the OCS Active
Directory.
Make sure that the Collaboration
Server user is defined in the OCS
Active Directory.

Verify the A/V Edge server is
configured in the OCS.

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Active Alarm

Phase

Alarm Displayed When

ICE failure:
A/V Edge server URI
cannot be resolved

Service

The Collaboration Server
failed to resolve The remote
address of the Edge server
URI.

ICE failure: Service
credential denied. A/V
Edge server credentials
rejected by the OCS.

Service

This alarm indicates that the
OCS does not configure with
the.
Generated by the ICE stack.

Troubleshooting

Troubleshooting
● At the end of the installation and configuration process, to test the solution and the integration with
the OCS, create an ongoing conference with two participants, one dial-in and one dial-out and
connect them to the conference.
● If the active Alarm “SIP TLS: Registration server not responding” is displayed, stop and restart the
OCS Pool Service.
● If the communication between the OCS and the Collaboration Server cannot be established, one of
the possible causes can be that the Collaboration Server FQDN name is defined differently in the
DNS, OCS and Collaboration Server. The name must be defined identically in all three devices, and
defined as type A in the DNS. The definition of the Collaboration Server FQDN name in the DNS can
be tested by pinging it and receiving the Collaboration Server signaling IP from the DNS in return.
● The communication between the OCS and the Collaboration Server can be checked in the Logger
files:
 SIP 401/407 reject messages indicate that the Collaboration Server is not configured as Trusted
in the OCS and must be configured accordingly.
 SIP 404 reject indication indicates that the connection to the OCS was established successfully.

Known Issues
● Selecting Pause my Video in OC client causes the call to downgrade to audio only call if the call was
not in Audio Only mode at all (the call was started as a video call).
If the call is started as an audio only call and video is added to it, or if the call was started as video
call and during the call it was changed to Audio Only and back to video, selecting Pause my Video
will suspend it as required.
● Rarely, the OC client disconnects after 15 minutes. The OC client can be reconnected using the same
dialing method in which they were previously connected (dial-in or dial-out).
● Rarely, all SIP endpoints disconnect at the same time. The SIP endpoint can be reconnected using
the same dialing method in which they were previously connected (dial-in or dial-out).

Polycom Solution Support
Polycom Implementation and Maintenance services provide support for Polycom solution components only.
Additional services for supported third-party Unified Communications (UC) environments integrated with
Polycom solutions are available from Polycom Global Services and its certified Partners. These additional

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services will help customers successfully design, deploy, optimize and manage Polycom visual
communications within their UC environments.
Professional Services for Microsoft Integration is mandatory for Polycom Conferencing for Microsoft Outlook
and Microsoft Office Communications Server integrations. For additional information and details please see
http://www.polycom.com/services/professional_services/index.html or contact your local Polycom
representative.

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Lync 2013 SVC Connectivity to Polycom MCU
The Microsoft H.264 SVC codec replaces the H.263 codec previously used with Lync 2013 clients. Although
similar Polycom’s standards-based H.264 and SVC implementation, it is proprietary to Microsoft, enabling
video calls between Lync 2013 clients (endpoints) and Polycom endpoints to be established.
The Collaboration Server considers Lync 2010 and H.264 SVC Lync 2013 clients to be AVC endpoints. The
administrator must set the Conferencing Mode in the Conference Profile to CP (Continuous Presence) to
enable H.264 SVC Lync clients to connect to the conference.

Deployment Architectures
Two Deployment Architectures are presented as examples. Both require that a Polycom RealPresence
Distributed Media Application (DMA) System 7000 be configured as part of the solution.
● Deployment Architecture 1 - Collaboration Server Hosted (Direct)
Lync 2013 clients connect to a conference hosted on Polycom RealPresence Collaboration Server.
● Deployment Architecture 2 - MS AV MCU Cascade
Lync 2013 clients connect to a conference on Microsoft AVMCU which is connected to a Polycom
RealPresence Collaboration Server.
The following table summarizes current and legacy (non DMA) conferencing modes within the deployment
architectures:
Conference Modes by Deployment Architecture
Deployment Architecture 2

Deployment Architecture 1
Conference
Mode

AVC Only

Mixed Mode
SVC Only

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Direct Dial In/Out (No AV
MCU cascade)
With or without DMA

AV MCU Cascade Call (Indirect Dial In)
Non DMA
(Backward Compatibility)

With DMA

Supported.

Supported.

Supported.

Dial Out out from DMA is not
supported.
For backward compatibility,
Dial out from the RMX Web
Client or RMX Manager can
be used.
When using backward
compatibility mode for Dial
out, H.264 SVC Lync 2013
clients are not be supported.
If the Video Protocol field in
the Participant dialog Advanced tab is set to Auto,
RTV protocol is used.

Simulcast transmitted.
Multiple participants received.

One transmitted RTV stream.
One received RTV stream.

Supported

Not Supported
Not Supported

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Backward compatibility to Lync 2010
All Lync 2013 functionality can be disabled by adding the BLOCK_NEW_LYNC2013_FUNCTIONALITY
System Flag and setting its value to YES.
The flag’s default value is NO, and when set to yes, all Lync 2013 new functionality is disabled. All Lync
2013 clients, whether connected Directly or by MS AV MCU cascade, will connect using the RTV codec, not
the MS SVC codec.

Video Resource Requirements and Implications
Lync 2013 SVC clients may not all connect to a VMR with the same stream layout. They are therefore
considered H.264 AVC participants and transcoding resources are allocated to them as summarized in the
following table.
Bandwidth and Resource Consumption by Video Codec
Video Codec

Resolution and Aspect
Ratio

Maximum Video
Payload Bit rate
(Kbps)

Minimum Video
Payload Bit rate
(Kbps)

Resources

H.264

320x180 (16:9)
Coded as 320x192
212x160 (4:3)

250

15

CIF

H.264/RTVideo

424x240 (16:9)
Coded as 432x240
320x240 (4:3)

350

100

CIF

H.264

480x270 (16:9)
Coded as 480x272
424x320 (4:3)
Coded as 432x320

450

200

SD

H.264/RTVideo

640x360 (16:9)
Coded as 640:368
640x480 (4:3)

800

300

SD

H.264

848x480 (16:9)

1500

400

SD

H.264

960x540 (16:9)
Coded as 960:544

2000

500

SD

H.264/RTV

1280x720 (16:9)

2500

700

HD720p30

H.264

1920x1080 (16:9)

4000

1500

HD1080p30

Support for HD1080p Resolution
The Collaboration Server Hosted deployment supports HD1080p30 video resolution symmetrically for direct
calls.

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The MS AV MCU Cascade deployment supports HD1080p30 video resolution only if Video Optimized mode
is selected and according to the settings of the LYNC_AVMCU_1080p30_ENCODE _RESOLUTION
System Flag:
NO (Default) Video streams sent to and received from the MS AV MCU are HD720p30, SD, and CIF.
YES Video streams sent to the MS AV MCU are HD1080p30, SD, CIF. Video streams received from
the MS AV MCU are 720p30,SD, and CIF.

Limit Maximum Resolution for MS SVC Using a System flag
The MAX_MS_SVC_RESOLUTION System Flag can be used to minimizing the resource usage by
overriding the default resolution selection and limiting it to a lower resolution.
Range: AUTO, CIF, VGA, HD720, HD1080
Default: AUTO
The MAX_MS_SVC_RESOLUTION System Flag operates independently from the
MAX_RTV_RESOLUTION System Flag allowing differing maximum resolutions to be selected for the MS
SVC and RTV protocols.
If you want to modify System Flag values, the flags must be added to the System Configuration file. For
more information see: Modifying System Flags and Controlling Resource Allocations for Lync Clients
Using RTV Video Protocol in the Collaboration Server (RMX) Administrator’s Guide.

ICE Configuration
The Collaboration Server should be configured with ICE regardless of whether an EDGE Server is part of
the configuration or not.

Federation Configuration
A secondary SIP domain can be added to the Lync 2013 environment. It must be configured as Federated.

For more information see: Appendix H - Integration Into Microsoft Environments in the Collaboration
Server (RMX) Administrator’s Guide.

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System Flags for Cropping Control
Cropping occurs when the video source (endpoint) aspect ratio is different from the video cell aspect ratio
in the Polycom Video Layout.
For all endpoints other than ITP endpoints and panoramic cells the Collaboration Server calculates the
mismatch percentage between the source video aspect ratio and Polycom video layout cell aspect ratio. The
mismatch percentage is used to determine whether cropping or striping will be applied to the video cell in
the Polycom video layout.
For non-panoramic layouts, cropping and striping can be controlled by adding the
CROPPING_PERCENTAGE_THRESHOLD_GENERAL System Flag and setting its value accordingly.
For panoramic layouts, cropping and striping can be controlled by adding the
CROPPING_PERCENTAGE_THRESHOLD_PANORAMIC System Flag and setting its value accordingly.
For both System Flags:
Range: -1-100.
Default: -1
If the calculation result is less than or equal to the value of the
CROPPING_PERCENTAGE_THRESHOLD_GENERAL System Flag, cropping will be applied.
If the calculation result is greater than the value of the
CROPPING_PERCENTAGE_THRESHOLD_GENERAL System Flag, striping will be applied.
If the CROPPING_PERCENTAGE_THRESHOLD_GENERAL System Flag value is set to 0, cropping will
not be applied.
If the CROPPING_PERCENTAGE_THRESHOLD_GENERAL System Flag value is set to -1 cropping will
always be applied.
If the CROPPING_PERCENTAGE_THRESHOLD_GENERAL System Flag value is set to 100, always
apply cropping with the exception of mobile aspect ratios. Mobile aspect ratios are those with a larger vertial
aspect than horizontal for example 3:4.
Example: An iPhone Lync client sends 288x352 (3:4) aspect ratio. Cropping this resolution to 16:9 would
require the cropping of 190 pixels on the vertical aspect. The mismatch calculation would yield 190/352 ~54.
If the applicable System Flag's value is set to >= 54, striping, rather than cropping, is applied.
For more information see, Modifying System Flags in the Collaboration Server (RMX) Administrator’s Guide.

Sharing Content during a Conference
Using Polycom CSS (Content Sharing Suite) plug in, Lync 2013 clients are able to share content in Polycom
Content sessions in both Collaboration Server Hosted calls and Cascaded MS AV MCU calls, with both Lync
2010 & 2013 clients.
Content Sharing behavior is summarized in the following tables:
Content Sharing by Lync Version and Deployment Architecture
Lync Version

Collaboration Server Hosted

Cascaded MS AV MCU

Lync 2010

Supported, as in previous versions.

Not supported

Lync 2013

Supported.

Supported

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Content Sharing Behavior by Lync Connection and MCU Type
Connection Type

Content Sharing Behavior

Point to point: Lync client to
Lync client.

Microsoft Content is used for the entire session.

Lync calls MS AV MCU
conference with no
Collaboration Server cascade
participant.

Microsoft content is used until Collaboration Server will joins, from which time it
will switch to Polycom content

Lync calls VMR directly

Polycom Content is used for the entire session.

Collaboration Server joins an
MS AV MCU conference
before Content sharing is
initiated.

Polycom Content is used until Collaboration Server leaves the conference.

Collaboration Server joins an
MS AV MCU conference while
Content is being shared.

When the CSS plug in of the Content speaker detects that Microsoft Content is
being sent by the Lync client, it automatically stops the Microsoft Content and
switches to Polycom Content and starts sending Polycom Content.
Polycom Content is used until the Collaboration Server leaves the MS AV MCU
conference.

CSS Behavior by Lync Content Type
Lync Content Type

CSS Behavior

Desktop Sharing

Send the desktop In cases where the is more than one monitor, the Lync client asks
which monitor to use and the CSS will comply.

Program Sharing

Only the application is sent.

Power Point Sharing

Not supported. CSS should not send the Power Point to the Collaboration Server but
the Lync clients will be able to send/receive the Power Point. CSS issues a notification
in the Lync Content presenter device stating that the Power Point cannot be shared
with non Lync devices.

Whiteboard Sharing

Not supported. CSS should not send the whiteboard to the Collaboration Server but the
Lync clients will be able to send/receive the whiteboard. CSS issues a notification in the
Lync Content presenter device stating that the whiteboard cannot be shared with non
Lync devices.

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Cisco TIP Support
Polycom’s solution that allows the Collaboration Server to natively inter-operate with Cisco TelePresence
Systems using Cisco TIP protocol is supported.
MLA (Multipoint Layout Application) is required for managing Cisco TelePresence layouts (whether Polycom
ITP Systems are deployed or not. MLA is a Windows® application that allows conference administrators to
configure and control video layouts for multipoint calls involving Polycom Immersive Telepresence (ITP)
systems. For more information see the Polycom® Multipoint Layout Application (MLA) User’s Guide for Use
with Polycom Telepresence Solutions.
System behavior can be controlled by adding the MS_AV_MCU_ MONITORING System Flag and setting
its value accordingly as summarized in the following table.
System Behavior by MS_AV_MCU_MONITORING System Flag Value and MLA Mode
MS_AV_MCU_MONITORING=
MAIN_AND_IN_SLAVE
(Default)
NO
(Not recommended)

MLA Mode
Room Switch

Sees the AV MCU current speaker.

CP Layout

Sees all connected Lync Clients in
the layout.

Room Switch
CP Layout
Room Switch

YES

Collaboration Server Side

CP Layout
(Not
recommended)

Sees a Lync Client which may or
may not be the current speaker.
Sees the MS AV MCU current
speaker.

MS AV MCU Side

Sees the Collaboration
Server hosted current
speaker in a 1x1
layout.

Sees cascaded slave MS AV
MCUs as empty cells in the layout.

For more information seeCollaboration With Cisco’s Telepresence Interoperability Protocol (TIP)and
Collaboration with Microsoft and Ciscoin the Collaboration Server (RMX) Administrator's Guide.

Lync 2013 Participant monitoring
Lync Clients connected to a conference using the Collaboration Server Hosted architecture will experience
normal monitoring, with the addition of the MS SVC codec.
Lync Clients connected to a conference using a MS AV MCU Cascade link will be monitored as a single
participant. In the Conference list the MS AV MCU is listed as Lync AV MCU_x, where x is an incrementing
number, should multiple be multiple conferences connected using MS AV MCU Cascade links.
For more information see Microsoft RTV Video Protocol Support in CP Conferencesin the Collaboration
Server (RMX) Administrator's Guide.

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Monitoring Participant Properties - Channel Status Tab
Two Channel parameters for each MS AV MCU Cascade link are displayed: Video in and Video out.
For both Video in and Video out, the Bit Rate and Packet Loss parameters are displayed as aggregate
values. For all other Channel parameters (Jitter, Latency, etc.) the highest values ar displayed for each video
stream.

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Monitoring Participant Properties - Channel Status - Advanced Tab
Media Info of each media stream sent by the MS AV MCU is displayed:
Stream name: The Display Name of the Lync client.
Algorithms: H.264 or RTV.
Resolution: CIF, SD, VGA, HD720 etc.
Frame Rate: 7.5, 15, 30 etc.
Annexes: Used for H.263 only
RTP Statistics are aggregated and are not detailed per stream.

For more information see Participant Level Monitoringin the Collaboration Server (RMX) Administrator’s
Guide.

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Deployment Architecture 1 - Collaboration Server
Hosted
Lync 2013 clients connect to a conference hosted on a Polycom Collaboration Server.

● H.264 SVC Lync clients are connected using UCConfig Mode 1:
 The SVC Codec’s Temporal Scaling capability is used to send one video stream to and from the
Collaboration Server for each resolution at multiple frame rates.
 H.264 SVC uses H.264 SEI messages to send stream layout information rather than SDP
messages.
 There is one audio stream per direction.
● Lync clients place calls to a Virtual Meeting Room provisioned on the DMA, for example,
1234@dma.example.com
● The Collaboration Server can connect Lync 2013 participants to either mixed AVC/SVC or to AVC
only conferences.
● Lync 2013 clients dialing to a VMR, where the type of the conference is SVC/AVC mixed and AVC
CP only have their video decoded. The Collaboration Server sends encoded video to Lync 2013
participants.

Look and Feel
All participants experience a Microsoft Point to Point conference with Polycom video layouts.
Only Classic Skin is supported. For more information about Skins, see Defining AVC-Based Conference
Profiles in the Collaboration Server (RMX) Administrator’s Guide.

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Deployment Architecture 2 - MS AV MCU Cascade
Cascaded VMR Participants (Lync 2013 clients) connect to a conference on a Microsoft AV MCU which is
cascaded with a Collaboration Server.

In this deployment architecture, participants connecting to the conference by way of the Collaboration
Server are referred to as Cascaded VMR Participants.
● Lync clients place calls to a Virtual Meeting Room provisioned on the DMA, for example,
1234@dma.example.com
● The Cascaded VMR connects as a Lync client to the AV MCU.
 The Collaboration Server utilizes the SVC Codec’s Temporal Scaling capability to send up to three
simulcast video streams to the AV MCU, each at multiple frame rates.
 The Collaboration Server receives media from up to five different Lync clients from the AV MCU.
● Lync clients experience a Microsoft Point to Point conference with Cascaded VMR Participants
appearing as Lync clients.
● Cascaded VMR Participants experience a conference with Polycom video layouts.
● Lync clients will see the active speaker from the
● Cascaded VMR while Cascaded VMR Participants will see up to 5 Lync clients in addition to other
participants.

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● Lync 2013 clients, connected by means of the Lync AV MCU, in point to point calls, can connect to
Collaboration Server VMR participants by escalating the Lync call to a multipoint conference,
including the Collaboration Server VMR meeting room and its participants.
 The Buddy List can be used to select participants followed by, right-clicking and selecting Start a
Video Call.
 An ad-hoc (Meet Now) Lync conference can be started; a drag-and-drop operation can then be
used in the Buddy List to add a Cascaded VMR to the conference.
● Cascaded VMR Participants, in point to point calls, can connect to Lync AV MCU participants by
escalating the call to a multipoint Lync Conference.
 A re-INVITE is issued to escalate the conference from point-to-point to multipoint.
● A re-INVITE can be issued from an ongoing Audio conference to escalate it, enabling connected
participants to start sending video.
● Conferences hosted on a Collaboration Server can connect in cascade to only one AV MCU hosted
conference.
● The Collaboration Server can host multiple conferences, each connected in cascade to a different AV
MCU hosted conference. Conferences, connected in cascade to an AV MCU hosted conference
cannot be connected in cascade to other Collaboration Servers.

Look and Feel for Lync clients and Group Series Endpoints
Lync clients and Group Series endpoints with native Lync capability connect to the AV MCU directly and
experience a Lync look and feel conference and can see all Lync clients; up to 5 simultaneously in Gallery
View.

If the active speaker is a Cascaded VMR Participant, the participant is seen by Lync Clients in a video
window in the Gallery View.
By default, Cascaded VMR Participants are forced to a 1x1 layout by the default setting of the
FORCE_1X1_LAYOUT_ON_CASCADED_LINK_CONNECTION System Flag. The flag’s default setting is
YES, which prevents a VMR layout being displayed within a Gallery View video window. If required,
alternative Personal Layouts can be forced only after this System Flag has been added and its value set to
NO.
For more information see, Modifying System Flags in the Collaboration Server (RMX) Administrator’s Guide.

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Look and Feel for Legacy Endpoints
Legacy endpoints will connect to the Collaboration Server and will be able to see all Lync clients; up to 5
simultaneously in Polycom video layouts. Only Classic Skin is supported. For more information about Skins,
see Defining AVC-Based Conference Profiles in the Collaboration Server (RMX) Administrator’s Guide.

Video Resource Requirement Selection in MS AV MCU Cascade
Collaboration Server resource usage in MS AV MCU Cascade can be configured in the conference
Advanced tab of the Profile dialog by selecting either Resource Optimized or Video Optimized.
Resource Optimized
The Collaboration Server’s Resolution Configuration menu, resolutions of up to HD540p30 (SD30) are
supported, depending on the conference’s profile setting.
Video Optimized
The Collaboration Server’s Resolution Configuration menu, resolutions of up to HD720p30 are supported,
depending on the conference’s profile setting.

If the Collaboration Server has insufficient resources, endpoints will be connected at the lowest resolutions
possible: CIF or SD. If the Collaboration Server has no available resources, endpoints will not be connected.
Calls that are initially connected as Audio Only will only have video resources allocated to them if they are
escalated to video calls.

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Video Forcing and Changing Layout in MS AV MCU Cascade
MS AV MCU Cascade behaves in the same manner as Collaboration Server to Collaboration Server
Cascading.
The Conference Layout as well as the Personal Layouts of participants can be changed. Participants can
be forced to appear in specific video cells of the layouts.
If Lync 2013 video streams are to be included in a Polycom Conference Layout, the Collaboration Server
will remove these streams from the layout sent over the AV MCU cascade link.
If Lync 2013 video streams are to be included in a Personal Conference Layout, the Collaboration Server
will not remove these streams from the layout to be sent over the AV MCU cascade link.

Handle Low Bit Rate Calls From the AV MCU
If the Collaboration Server, or a Group Series endpoint is connected to the AV MCU at a bit rate of 256kbps,
the AV MCU transmits only one video stream even if it receives multiple video source requests.
At bit rates lower than256kbps (128kbps and 192kbps) the AV MCU does not transmit video.
This limitation can be controlled using the DISABLE_LYNC_AV_MCU_128_AND_192_KBPS System
Flag. The flag must be manually added to the System Configuration and its value modified as required:
NO (Default)—The Collaboration Server sends 128kbps or 192kbps (according to the call rate) but
will receive 256kbps for each incoming video stream.
YES—The Collaboration Server will not send or receive video from the Lync AV MCU. The
connection is audio only.
For more information see, Modifying System Flags in the Collaboration Server (RMX) Administrator’s Guide.

Remove Empty Cells From the Video Layout
Empty cells in the Video Layout can occur as result of the following causes:
Case 1 — A camera connected a PC that is hosting a Lync client is switched off, the cell in which the Lync
client was displayed remains in the video layout and is empty.
Case 2 — A Lync 2013 Client is connected using a CIF port at a bit rate that exceeds 192kbps.

Case 1
The empty cell can be removed from the video layout by adding the
REMOVE_EP_FROM_LAYOUT_ON_NO_VIDEO_TIMER System Flag and setting its value as required.
Range:

0 – 19 (seconds): The feature is disabled.
20 – 300 (seconds): The feature is enabled.

Default:

20

When enabled (flag value 20 - 300), the endpoint is removed when the empty cell is detected, and the cell
is used for another participant if:
● No video RTP messages are received from the EP for the defined timer value in addition to one of
the following timers, depending on the call type:
 DETECT_SIP_EP_DISCONNECT_TIMER
 DETECT_H323_EP_DISCONNECT_TIMER

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● Either the PRESERVE_PARTY_CELL_ON_FORCE_LAYOUT System Flag =NO
or
The endpoint is not forced in the layout.
In Lync environments that do not include ICE, the empty cell will remain in the layout for direct Lync
calls.

Case 2
The RTV_MAX_BIT_RATE_FOR_FORCE_CIF_PARTICIPANT System Flag has been added to the
system with a default value of 192 (kbps). This ensures that the Lync Client sends the correct resolution and
that its cell in the Video Layout is displayed correctly.

Configuring the Collaboration Server as a Trusted
Application for Lync 2013
Registering the Collaboration Server as a Trusted
Application for Lync 2010/2013
The following procedures are mandatory to register the Collaboration Server to Lync 2010/2013.
In the following procedures Domain Names, Server Names and Collaboration Server Names are
examples for syntax purposes only and must be adapted to local network requirements.

Configure the Collaboration Server FQDN in the DNS
Perform the following steps on the Lync Server to configure the Collaboration Server FQDN in the DNS.

To Configure the Collaboration Server FQDN in the DNS:
1 Connect to the DNS Server.
2 Open the DNS Manager.

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3 Navigate to and expand the Forward Lookup Zones folder.

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4 Right click on the Zone for your Domain.

5 Select New Host (A or AAAA).

6 Enter the Collaboration Server Name and the Signaling IP address.
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7 Click Add Host.

Configure Collaboration Server Static Route and Trusted Application
Perform the following steps on the Lync Server to configure the Collaboration Server Static Route and
Trusted Application.

To Configure the Collaboration Server Static Route and Trusted Application:
1 Connect to Lync Front End server
2 In the Start Menu Search field, enter Lync and click the Search button.

3 Select the Lync Server Management Shell.

4 Add a Static Route:
a Type: $route = New-CsStaticRoute -TLSRoute -destination "" -port 5061
-matchuri "" -usedefaultcert $true and press Enter.
 is the Collaboration Server name.
b Type: Set-CsStaticRoutingConfiguration -identity global -route @{Add=$route} and press
Enter
(To check the Static Route configuration enter the following command:
Get-CsStaticRoutingConfiguration and press Enter.)
5 Create a Trusted Application Pool and a Trusted Application:

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a Type: New-CsTrustedApplicationPool -Identity  -Registrar
Registrar: -site 1 -ComputerFqdn 
-ThrottleAsServer $true -TreatAsAuthenticated $true and press Enter.
b Enter YES.
c Type: New-CsTrustedApplication -ApplicationId  -TrustedApplicationPoolFqdn
 -Port 5061 and press Enter.
ApplicationId is the name of the application. This must be a string that is unique within the pool
that is specified in the TrustedApplicationPoolFqdn parameter.
TrustedApplicationPoolFqdn is the FQDN of the Trusted Application Pool in which the
application will reside.
6 Add a Trusted Application Endpoint by typing: New-CsTrustedApplicationEndpoint
-sipaddress sip:@ -ApplicationId  -TrustedApplicationPoolFqdn
 and pressing Enter.
7 Enable the changes, by typing: Enable-CsTopology.
8 Get the GRUU (Globally Routable User-Agent URI):
a Type: Get-CsTrustedApplication -Filter "*" | fl ServiceGruu and press Enter.
The Lync Server will reply with a string similar to the following:
ServiceGruu:sip:rmx23772.reg13.ent@reg13.ent;gruu;opaque=srvr:
rmx23772.reg13.ent:0I2d_trGDFSQv4FntDXH-wAA
b Select, Copy, and transfer the ServiceGruu string into the workstation that is connected to the
Collaboration Server. The SIP prefix (sip:) must be omitted from the copied string.
9 Create and export a Certificate for the Collaboration Server:
a Type: Request-CsCertificate -New -Type Default -KeyAlg RSA -CA  -City PT
-State Israel -ComputerFqdn  -Country IL -DomainName
 -FriendlyName  -Organization 'Polycom '
-PrivateKeyExportable $true and press Enter.
b In the Start Menu, select Administrative Tools-> Internet Information Services (IIS) Manager.

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c Click Server Certificates.

d In the Certificate List, locate the Collaboration Server’s certificate.

e Right click the Collaboration Server’s certificate.

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f

Click Export.

g In the Export to field, enter the file location for exported certificate.
h In the Password and Confirm Password fields, enter the password.
i

Click OK.

Configure the Collaboration Server for Lync 2010/2013
Perform the following steps on the Collaboration Server to configure it for Lync 2010/2013.

To Configure the Collaboration Server for Lync 2010/2013:
1 Open the IP Network Management Services list.

2 Double click Management Network.

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3 Click the DNS tab.

4 In the MCU Host Name field, enter the name of the DNS (as listed in the DNS Manager).
5 In the Local Domain Name field, enter the Domain Name (as listed in the DNS Manager).
6 In the DNS Servers Primary Server field, enter the DNS IP Address (as listed in the DNS Manager).
7 Click OK
8 Restart the Collaboration Server.
9 Login again and open the IP Network Management Services list.
10 Double click IP Network Service.

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11 Click the DNS tab.

12 In the IP Network Type drop down menu, select H.323 &SIP.
13 In the Service Name (FQDN) field, enter the DNS Name (as listed in the DNS Manager).
This should not be the FQDN.
14 In the DNS drop down menu, select Specify.
15 In the Local Domain Name field, enter the Domain Name (as listed in the DNS Manager).
16 In the DNS Server Address field, enter the IP Address of the DNS server (as listed in the DNS
Manager).

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17 Click the SIP Servers tab.

18 In the IP Network Type drop down menu, select H.323 &SIP.
19 In the SIP Server drop down menu, select Specify.
20 In the SIP Server Type drop down menu, select Microsoft.
21 In the Transport Type drop down menu, select TLS.
22 In the SIP Servers and Outbound Proxy Servers Parameter tables:
 Set the Server IP Address to the DNS Pool name (as listed in the DNS Manager).
 Set the Server Domain to the Domain Name (as listed in the DNS Manager).
 Set Port to 5061
23 Click the SIP Advanced tab.

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24 In the Server User Name field, enter the Collaboration Server’s SIP Address.
This is the first segment of the ServiceGruu string with the an “@” replacing the “.”
For example if the ServiceGruu string was:
rmx23772.reg13.ent@reg13.ent;gruu;opaque=srvr:rmx23772.reg13.ent:0I2d_trGDFSQv4FntDXHwAA,
the Server User Name would be rmx23772@reg13.ent
25 Add the SIP_CONTACT_OVERRIDE_STR system flag with the ServiceGruu string as its value.
a On the workstation that is connected to the Collaboration Server, on the RMX Menu, click Setup
> System Configuration.
The System Flags dialog opens.

b In the System Flags dialog, click the New Flag button.
The New Flag dialog box is displayed.

c In the New Flag field, enter SIP_CONTACT_OVERRIDE_STR.
d Locate, Copy and Paste the ServiceGruu string, that was transferred to the workstation in Step
8b of the Configure the Collaboration Server Static Route and Trusted Application above,
into the Value field.
e Click OK to close the New Flag dialog.
26 Add the LIMIT_CIF_SD_PORTS_PER_MPMX_CARD System Flag and set its value to YES.
Setting this flag’s value to YES will limit the number of CIF and SD ports (including non Lync clients)
to 45 per media card. The flag value and must be set to YES when ICE is active because ICE ports
consume more system resources. HD participants are not affected.

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27 Repeat Steps 25b - 25e above with:
 New Flag field: LIMIT_CIF_SD_PORTS_PER_MPMX_CARD.
 Value field: YES.
28 Reset the Collaboration Server for the new flag settings to take effect.
a Click OK to close the New Flag dialog.
b Click OK to close the System Flags dialog.
c Reset the Collaboration Server.
29 After the rest has completed, Login to the Collaboration Server again.

Import and install the Certificate on the Collaboration Server
The Certificate created in Step 9 of the Configure Collaboration Server Static Route and Trusted
Application procedure above must be imported an installed on the Collaboration Server.

To import and install a certificate:
1 In the Collaboration Server menu, click Setup > RMX Secured Communications > Certificate
Repository.
2 Click the Personal Certificates tab.

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3 Click Add.

4 In the Network Service Name drop down menu, select IP Network Service.
5 In the Certificate Method drop down menu, select PEM/PFX.
6 In the Password field, enter the password entered in Step 9h of the Configure Collaboration
Server Static Route and Trusted Application procedure.
7 Click Send Certificate.

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8 Click Browse, and then navigate to the folder where the certificate was saved.

9 Select the certificate and click Open.
The certificate is installed and a Confirmation Message is displayed.

10 Click OK in the confirmation message.
A System Reset confirmation message is displayed.
11 If you are ready to reset the system, click YES.
The System Reset completes the Registration of the Collaboration Server to Lync 2010/2013.

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Collaboration Network (POCN)
Working in the Open Collaboration Server and TIP protocol are supported in AVC Conferencing Mode
only.

Collaboration With Cisco’s Telepresence
Interoperability Protocol (TIP)
TIP is a proprietary protocol created by Cisco for deployment in Cisco TelePresence systems (CTS). Since
TIP is not compatible with standard video communication systems, interoperability between Cisco and other
vendors’ Telepresence systems was initially impossible.
Gateways were developed to provide interoperability but were subject to the inherent problems of additional
latency (delay) in connections and low video quality resulting from the reformatting of video and audio
content.
Polycom’s solution is to allow the Collaboration Server to natively inter-operate with Cisco TelePresence
Systems, ensuring optimum quality multi-screen, multipoint calls between:
● Polycom Immersive Telepresence Systems (ITP) Version 3.1.1:
 RPX 200
 RPX 400
 OTX 300
(At Telepresence Licence is required on the Collaboration Server.)
● Polycom video conferencing endpoints
 Standalone HDX
 Polycom Group Series 300/500
● Microsoft
 MS Lync (using MS-ICE)
 RTV 720p
● Cisco TelePresence® System (CTS) Versions 1.10Collaboration ServerCollaboration
ServerCollaboration Server
 CTS 1300
 CTS 3010
Conferences hosted on the Collaboration Server can include a mix of existing end points (that do not
support TIP) and CTS endpoints.
TIP-enabled endpoints must support TIP Version 7 or higher. Calls from endpoints supporting older versions
of TIP will be rejected.

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Deployment Architectures
The following multipoint topologies are given as examples. Actual deployments will depend on user
requirements and available infrastructure:
● Single company with Polycom and Cisco Infrastructure
 CTS and Polycom Telepresence Rooms in a corporate environment.
● Company to company via Service Provider
 Model 1: Mixed Polycom and Cisco infrastructure at one of the companies, Cisco only
infrastructure at the other.
 Model 2: Polycom only infrastructure at one of the companies, Cisco only infrastructure at the
other.

Single Company Model - Polycom and Cisco Infrastructure
The deployment architecture in Single company with Polycom and Cisco Infrastructure - Polycom endpoints
using SIP shows a company that has a mixture of Polycom and Cisco endpoints, room systems and
telephony equipment that needs to enable multipoint calls between all its video and audio endpoints using
the Collaboration Server as the conference bridge.
As shown in Single company with Polycom and Cisco Infrastructure - Polycom endpoints using SIP, prior to
Version 8.1.1, Cisco Telepresence endpoints could connect to conferences using the TIP protocol, Polycom
endpoints connected to the same conferences using SIP protocol.

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Single company with Polycom and Cisco Infrastructure - Polycom endpoints using SIP

Polycom endpoints can also connect to Entry Queues, Meeting Rooms and conferences using all protocols,
including TIP and SIP.
The following table lists components and versions of the Collaboration Server and Cisco Telepresence
Systems (CTS) Integration Solution Architecture.
Solution Architecture Components
Component

Version

Description

CUCM

8.5.1, 8.6.2

Cisco Unified Communication Manager:
CUCM must be configured to:
• Route calls to DMA (if present).
• Route all H.323 calls to the gatekeeper, which can be either CMA or
IOS.

IOS

15.1T

Cisco Internetwork Operating System - Gatekeeper

Endpoints (CTS)

1.7.2 (ATT),
1.8.1

Telephony, desktop and room systems.
• CTS endpoints must register to CUCM.

CISCO Equipment

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Component

Version

Description

Cisco Unified Video
Conferencing 5230

7.2

MCU.

Cisco Unified
Presence

8.5, 8.6

Network-based Presence and Instant Messaging.

Cisco Unified
Contact Center
Express

8.0, 8.5

Call distributor (ACD), interactive voice
response (IVR) and computer telephony integration (CTI).

Cisco IP
Communicator

7.0,8.6

Windows PC-based softphone application.

Cisco Unified
Personal
Communicator

8.5(2),8.5(5)

Web client for Presence and Instant Messaging.

Cisco Unified Video
Advantage

2.2(2)

Video telephony functionality for Cisco Unified IP phones.

Cisco Unified IP
Phones 7960, 7961,
7962, 7965, 7975

CUCM 8.5.1 /
CUCM 8.6.1
compatible

IP Phones.

Cisco Unified IP
Phones 9971

CUCM 8.5 /
CUCM 8.6(2)
compatible

CTMS

1.7.3, 1.8.2

Cisco TelePresence Multipoint Switch.

Cisco Unified Border
Element

15.1T

SBC - Voice and video connectivity from enterprise IP network to
Service Provider SIP trunks.

Telepresence Server

2.2(1.54)

Telepresence Server.

VCS

X7.1

Video Communication Server / Session Manager.

4.0

Polycom Distributed Media Application
• DMA is an optional component but is essential if Content sharing is
to be enabled.
• All SIP endpoints register to DMA as a SIP Proxy.
• DMA should be configured to route SIP calls (with CTS destination)
to CUCM. If DMA is not present in the solution architecture, SIP
endpoints must register to CUCM as gatekeeper.
• DMA must be configured with a VMR (Virtual Meeting Room).
Incoming calls are then routed to the Collaboration Server.

Polycom Equipment
DMA 7000

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Appendix I - Polycom Open Collaboration Network (POCN)

Component

Version

Description

Collaboration Server

7.6 and higher

MCU:
• Functions as the network bridge for multipoint calls between H.323,
SIP and TIP endpoints.
• The Collaboration Server can be interfaced to CUCM using a SIP
trunk, enabling CTS to join multipoint calls on Collaboration Server.
Signaling goes through the CUCM while the media in TIP format
goes directly between the CTS and Collaboration Server.
• The Collaboration Server must be configured to route outbound SIP
calls to DMA.
• The H.323 Network Service of the Collaboration Server should
register it's dial prefix with the CMA gatekeeper.
• When DMA is not used an Ad-hoc Entry Queue, designated as
Transit Entry Queue, must be pre-defined on the Collaboration
Server.

MLA

3.0.3

Multipoint Layout Application
Required for managing multi-screen endpoint layouts for Cisco CTS
3XXX, Polycom TPX, RPX or OTX systems.

CMA

5.5

Polycom Converged Management Application - Gatekeeper
• The gatekeeper must route calls to Collaboration Server based on
the Collaboration Server prefix registration on the gatekeeper.

Endpoints

Polycom®, Inc.

Telephony, desktop and room systems.
• H.323 endpoints must register to the CMA or IOS gatekeeper.
• Polycom SIP endpoints must register to DMA as SIP Proxy when
DMA is used.
• H.323 endpoints must register to the CMA or IOS gatekeeper.

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Appendix I - Polycom Open Collaboration Network (POCN)

Call Flows
Multipoint call with DMA
In this example:
● Collaboration Server prefix in the gatekeeper: 72
● Virtual Meeting Room in DMA: 725123
● DMA Meeting Number: Generated by DMA

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Appendix I - Polycom Open Collaboration Network (POCN)

Multipoint call without DMA
In this example:
● Collaboration Server prefix in the gatekeeper: 72
● CUCM: According to its Dial Plan forwards calls with prefix 72 to the Collaboration Server

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Appendix I - Polycom Open Collaboration Network (POCN)

Company to Company Models Using a Service Provider
Using this topology, both companies connect to a Service Provider via a Cisco Session Border Controller
(SBC). The Service Provider functions as a B2B Telepresence Exchange, enabling multipoint calls between
the two companies and their respective video and audio endpoints using the Collaboration Server as the
conference bridge.
The SBC functions as a firewall that the Service Provider can configure according to Trust Relationships
between two or several companies. By using this method, companies do not have to open their corporate
firewalls and administer connectivity with the many companies they may need to communicate with.
Two topology models are discussed:
● Model 1:
 Company A has a Polycom only environment.
 Company B has a Cisco only Environment.
● Model 2:
 Company A has a mixed Polycom and Cisco environment.
 Company B has a Cisco only Environment.

Model 1
The deployment architecture in Call Flow shows two companies: Company A and Company B.
Company A - has deployed a Polycom solution including:
● DMA
● Collaboration Server
● MLA
● CMA Gatekeeper
● Polycom telephony and desktop endpoints.
The roles of the Polycom components are described in the Polycom Equipment section of the Solution
Architecture Components table.
Company B - has deployed a Cisco solution including:
● CTS 1000
● CTS 3000

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Appendix I - Polycom Open Collaboration Network (POCN)

● Cisco telephony and desktop endpoints
Company to Company via Service Provider - Model 1

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Appendix I - Polycom Open Collaboration Network (POCN)

Call Flow
Multipoint call via Service Provider - Model 1
In this example:
● Collaboration Server prefix in the gatekeeper: 72
● Virtual Meeting Room in DMA: 725123
● DMA Meeting Number Generated by DMA

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Appendix I - Polycom Open Collaboration Network (POCN)

Multipoint call via Service Provider - Model 2
The deployment architecture in The deployment architecture includes: shows two companies: Company A
and Company B.
Company A - has the same deployment architecture as shown in Single Company Model - Polycom and
Cisco Infrastructure.
Company B - has deployed a Cisco solution including:
● CTS 1000
● CTS 3000
● Cisco telephony endpoints.

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Appendix I - Polycom Open Collaboration Network (POCN)

The deployment architecture includes:

Company A
For a full description of Company A’s deployment, see Single Company Model - Polycom and Cisco
Infrastructure.
Differing or additional configuration requirements for each element of this deployment model are listed
below:
Company A Solution Architecture Components
Component

Version

Description

8.5

Cisco Unified Communication Manager:
CUCM must be configured with a SIP trunk to the Service Provider’s
SBC.

7.6.x

MCU:
Collaboration Server must be configured to send and receive RTP
streams to and from the Service Provider's SBC.

CISCO Equipment
CUCM

Polycom Equipment
Collaboration Server

Company B
Company B Solution Architecture Components

CISCO Equipment
Endpoints

Polycom®, Inc.

Endpoints should register with the Service Provider’s CUCM (or the
local CUCM, if present).

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Appendix I - Polycom Open Collaboration Network (POCN)

Call Flow
Multipoint call via Service Provider - Model 2
In this example:
● Collaboration Server prefix in the gatekeeper: 72
● Virtual Meeting Room in DMA: 725123
● CUCM: According to its Dial Plan forwards calls with prefix 72 to the Collaboration Server

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Appendix I - Polycom Open Collaboration Network (POCN)

Administration
The various deployment combinations and settings within the various Deployment Architectures affects the
administration of the system.

Gatekeepers
Standalone Polycom CMA/DMA System as a Gatekeeper
The Polycom CMA/DMA system can be used as the only gatekeeper for the network. Bandwidth and call
admission control of endpoints registered with the CMA system is split between the CMA system and the
CUCM.
For more information see the Polycom Unified Communications Deployment Guide for Cisco Environments,
Using a Polycom CMA System as a Gatekeeper.
Standalone Cisco IOS Gatekeeper
The Cisco IOS Gatekeeper can be used as the only gatekeeper for the network if the management
capabilities of the Polycom CMA system are not required.
For more information see the Polycom Unified Communications Deployment Guide for Cisco Environments,
Using a Standalone Cisco IOS Gatekeeper.

Neighbored Cisco IOS and Polycom CMA/DMA Gatekeepers
Neighbored gatekeepers make it easier to create a common dial plan and should be considered when
integrating an existing Cisco telephony environment with an existing Polycom network. Neighbored
Gatekeepers allow number translation while maintaining the existing environments.
For more information see the Polycom Unified Communications Deployment Guide for Cisco Environments,
Neighbored Cisco IOS and Polycom CMA Gatekeepers.

DMA
The Polycom DMA system can be configured as a SIP proxy and registrar for the environment. When used
as a SIP peer, the DMA system can host video calls between Cisco endpoints that are registered with the
CUCM and Polycom SIP endpoints that are registered with the DMA system.
For more information see the Polycom Unified Communications Deployment Guide for Cisco Environments,
Using a Polycom DMA System as SIP Peer.

CUCM
When Polycom SIP endpoints (voice and video) are registered directly with CUCM you can take advantage
of supported telephone functions. CUCM may not support the full range of codecs and features available
on the Polycom equipment. CUCM supported codecs and features will be used in such cases.
For more information see the Polycom Unified Communications Deployment Guide for Cisco Environments,
Direct Registration of Polycom Endpoints with the Cisco Unified Communications Manager Participants.

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Appendix I - Polycom Open Collaboration Network (POCN)

Configuring the Cisco and Polycom Equipment
MLA (Multipoint Layout Application) is required for managing CTS 3XXX layouts whether Polycom TPX,
RPX or OTX systems are deployed or not. MLA is a Windows® application that allows conference
administrators to configure and control video layouts for multipoint calls involving Polycom Immersive
Telepresence (ITP) systems.
Call Detail Records (CDR) are generated on both the CMA Gatekeeper and the CUCM for reporting and
billing purposes.

Cisco Equipment
To configure the various Cisco entities the following procedures are required.

CUCM
1 Configure the CUCM to send and receive calls from the H.323 network.
a With Neighbored IOS and CMA Gatekeepers
For more information see the Polycom Unified Communications Deployment Guide for Cisco
Environments, Configuring Cisco Unified Communications Manager for H.323.
b With CMA Gatekeeper
For more information see the Polycom Unified Communications Deployment Guide for Cisco
Environments, Configuring Cisco Unified Communications Manager for H.323.
c With IOS Gatekeeper
For more information see the Polycom Unified Communications Deployment Guide for Cisco
Environments, Configuring Cisco Unified Communications Manager for H.323.
IOS Gatekeeper
● Set up zones and gateway type prefixes to enable dialing to DMA and Collaboration Server systems.
For more information see the Polycom Unified Communications Deployment Guide for Cisco
Environments, Configuring the Cisco IOS Gatekeeper .
IOS and CMA Gatekeepers (Neighbored)
● Configure the Cisco IOS Gatekeeper for two separate zones.
For more information see the Polycom Unified Communications Deployment Guide for Cisco
Environments, Configure the Cisco IOS Gatekeeper for use with a CMA System.

Polycom Equipment
The following table lists the Polycom products supported within the various Deployment Architecture.
Only Collaboration Server configurations are described in detail in this document.
Configuration procedures for all other solution components are described in the Polycom Unified
Communications Deployment Guide for Cisco Environments.

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Appendix I - Polycom Open Collaboration Network (POCN)

Supported Polycom products
Polycom TIP and SIP

Version(s)

Polycom DMA 7000 system

V4.0

Polycom RealPresence Collaboration Server (RMX) 2000 and
RealPresence Collaboration Server (RMX) 4000 systems

V7.6 and higher

Immersive Telepresence Systems:
• RPX 200 and 400 systems
• OTX 300 system
• TPX HD 306 system
• ATX HD 300 system

V3.0.3
Requires TIP option key.
Requires Polycom Touch Control.

HDX Systems:
• 7000 HD Rev C
• 8000 HD Rev B
• 9006
• 4500

V3.0.3
Requires TIP option key.

The following Polycom peripheral:
• Polycom Touch Control

1.3.0

SIP ONLY (no TIP support)

Version(s)

Spectralink wireless phones 8020/8030
Polycom VVX 1500

V4.0

Polycom VVX 1500 C

V3.3.1

KIRK Wireless Server 300/600v3/6000

The following procedures 1 -16 are a summary of the configuration procedures.
The detailed procedures 1 - 16 begin with Procedure 1: Set the MIN_TIP_COMPATIBILITY_LINE_RATE
System Flag.

Configuring the Collaboration Server
1 Set the MIN_TIP_COMPATIBILITY_LINE_RATE System Flag
2 Configuring the Collaboration Server to statically route outbound SIP calls to DMA or CUCM
3 Configuring the Collaboration Server’s H.323 Network Service to register with CMA gatekeeper
4 Configuring a TIP enabled Profile on the Collaboration Server
5 Configuring an Ad Hoc Entry Queue on the Collaboration Server if DMA is not used
6 Configuring a Meeting Room on the Collaboration Server
7 Configuring Participant Properties for dial out calls

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Appendix I - Polycom Open Collaboration Network (POCN)

Configuring DMA
If DMA is present in the configuration perform procedures Configuring DMA to route SIP calls to CUCM
and Configuring a Virtual Meeting Room (VMR), otherwise skip to procedure Configuring CMA to route
H.323 calls to Collaboration Server.
8 Configuring DMA to route SIP calls to CUCM
9 Configuring a Virtual Meeting Room (VMR)
The procedures for configuring DMA are described in detail in the Polycom Unified Communications
Deployment Guide for Cisco Environments.

Configuring CMA
10 Configuring CMA to route H.323 calls to Collaboration Server
11 Configuring CMA for use with Cisco IOS Gatekeeper (Neighbored)
12 Configuring CMA to route H.323 calls to CUCM
13 Configuring CMA to route non-H.323 calls to CUCM
The procedures for configuring CMA are described in detail in the Polycom Unified Communications
Deployment Guide for Cisco Environments.

Configuring Endpoints
14 Configuring H.323 endpoints to register to the CMA or IOS gatekeeper
The procedures for configuring H.323 endpoints are described in detail in the Polycom Unified
Communications Deployment Guide for Cisco Environments.
15 Configuring SIP endpoints to register to:
a DMA as SIP Proxy
b CUCM as SIP Proxy
The procedures for configuring SIP endpoints are described in detail in the Polycom Unified
Communications Deployment Guide for Cisco Environments.
16 Configuring TIP endpoints to register to:
a DMA
b CUCM
The procedures for configuring TIP- enabled endpoints are described in detail in the Polycom Unified
Communications Deployment Guide for Cisco Environments.

Configuring Entry Queues and IVR Services
Conference IVR and Entry Queue/Virtual Entry Queues are supported with AVC TIP protocol in conferences
that include both TIP-enabled and non-TIP-enabled endpoints.
A Virtual Entry Queue can be configured to either IVR Only Service Provider or External IVR Control mode.
TIP-enabled endpoints can be moved from the Entry Queue to the destination conference if the TIP
Compatibility Modes settings in the Profile are identical for both conferencing entities (it is recommended to
use the same Profile for both entities).
TIP IVR users can access the conference directly or enter the Entry Queue/Virtual Entry Queue and provide
a password to access the conference.

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Appendix I - Polycom Open Collaboration Network (POCN)

The IVR services can be enabled for all TIP Compatibility Modes:
● Video only
● Video and Content
● Prefer TIP
IVR media files, WAV for voice messages and JPG for video slides, are all stored on the Collaboration
Server.

Guidelines
● IVR default audio files are enabled for all TIP Compatibility Modes.
● Only Polycom default Welcome slides are available. Custom Welcome slides are not supported.
● TIP-enabled endpoints can send DTMF digits to MCU.
● In an mixed TIP environment there is no support for content in cascaded conferences.

Entry Queue and Virtual Entry Queue Access
TIP endpoints can dial-in to conferences directly using the IVR, Entry Queue/Virtual Entry Queue and IVR
Only Service Provider.
For more information on Multipoint see the Multipoint Call Flows.
Configuring the Conference and Entry Queue IVR Services
The IVR module includes two types of services:
● Conference IVR Service that is used with conferences
● Entry Queue IVR Service that is used with Entry Queues
The configuration process is the same for TIP and non-TIP enabled Conferences and Entry Queues.

Content
Polycom and Cisco endpoints can share Content within a Cisco TelePresence environment. The content
sharing experience depends on whether the endpoints are registered with the DMA or CUCM.
Endpoint Registration Options - Content Sharing Experience
Multipoint Calls on
Collaboration Server

Content
Sharing

People + Content

HDX/ITP to HDX/ITP

Yes

Yes

HDX/ITP to Cisco CTS

Yes

Yes

Cisco CTS to HDX/ITP

Yes

No

HDX/ITP to HDX/ITP

Yes

No

HDX/ITP to Cisco CTS

Yes

No

Cisco CTS to HDX/ITP

No

No

Endpoints Registered to DMA

Endpoints Registered to CUCM

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Appendix I - Polycom Open Collaboration Network (POCN)

● H.239
 A variety of resolutions and frame rates are supported.
For more information see H.239 / People+Content.
 Can be used with SIP and H.323 endpoints, desktop (CMAD), room systems (HDX) and ITP
(OTX, RPX).
 Not supported by Lync clients, IBM clients and Cisco CTS endpoints.
 Cannot be used when HDX endpoints are registered to CUCM.
● TIP
 The resolution is fixed at XGA at 5fps.
 Supported on HDX, Polycom ITP and Cisco CTS systems.
● The following content compatibility options are available:
 Tip not enabled – CTS cannot join the conference, all other endpoints can share H.239 content.
 TIP video compatibility – CTS receives people video, all other endpoints can share H.239 content.
 TIP video and content compatibility – CTS and HDX can share TIP content, all other endpoints
receive only the people video.
For more information see Procedure 4: Configuring a TIP Enabled Profile on the Collaboration
Server.

Procedure 1: Set the MIN_TIP_COMPATIBILITY_LINE_RATE System Flag
The MIN_TIP_COMPATIBILITY_LINE_RATE System Flag determines the minimum line rate at which an
Entry Queue or Meeting Room can be TIP enabled.
CTS version 7 requires a minimum line rate of 1024 kbps and will reject calls at lower line rates, therefore
the SysCollaboration Servertem Flag value must be 1024 or higher.
HD Video Resolutions for TIP calls are determined according to the following table:
TIP HD Video Resolution by Line Rate
Line Rate

Video Resolution

Line Rate >=3Mbps

HD1080p30

3Mbps > Line Rate >= 936kbps

HD720p30

Line Rate < 936kbps

Call is dropped.

For more information see Modifying System Flags.

Procedure 2: Configuring Collaboration Server to statically route outbound SIP
calls to DMA or CUCM
1 In the IP Network Services Properties dialog box, click the SIP Servers tab.
2 In the SIP Server field, select Specify.
3 In the SIP Server Type field, select Generic.
4 Set Refresh Registration every 3600 seconds.

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Appendix I - Polycom Open Collaboration Network (POCN)

5 If not selected by default, change the Transport Type to TCP.
6 In the SIP Servers table:
a Enter the IP address of the DMA or CUCM in both the Server IP Address or Name and Server
Domain Name fields.
b The Port field must be set to it’s default value: 5060. DMA and CUCM use this port number by
default.
7 In the Outbound Proxy Servers table:
a Enter the IP address in the Server IP Address or Name field. (The same value as entered in Step
6a.)
b The Port field must be set to it’s default value: 5060.
(By default, the Outbound Proxy Server is the same as the SIP Server.)

When configuring Collaboration Server to statically route SIP calls to DMA or CUCM, it is important
to also configure the Collaboration Server’s H.323 Network Service to register with CMA gatekeeper.
For more information see Procedure 3: Configuring the Collaboration Server’s H.323 Network
Service to register with CMA gatekeeper.

Procedure 3: Configuring the Collaboration Server’s H.323 Network Service to
register with CMA gatekeeper
1 In the IP Network Services Properties dialog box, click the Gatekeeper tab.

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Appendix I - Polycom Open Collaboration Network (POCN)

2 In the MCU Prefix in Gatekeeper field, enter the prefix that the Collaboration Server uses to register
with the gatekeeper.

Procedure 4: Configuring a TIP Enabled Profile on the Collaboration Server
TIP enabled profiles must be used for the Entry Queues and Meeting Rooms defined on the Collaboration
Server. (Different Profiles can be assigned to Entry Queues and Meeting Rooms, however they must be TIP
enabled.) When TIP is enabled in the Profile, Gathering Settings and Message Overlay options are
disabled.
1 Create a New Profile for the Meeting Room. For more information see Defining a CP Conference
Profile.

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Appendix I - Polycom Open Collaboration Network (POCN)

2 In the New Profile - General tab, set the Line Rate to a value of at least that specified for the
MIN_TIP_COMPATIBILITY_LINE_RATE System Flag in Procedure 1.

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Appendix I - Polycom Open Collaboration Network (POCN)

3 Click the Advanced tab.

4 Select the TIP Compatibility mode according to the Content Sharing Behaviortables that are listed at
the end of this procedure: Video and Content is recommended.

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Appendix I - Polycom Open Collaboration Network (POCN)

5 Click the Video Quality tab.

Content Settings is disabled if TIP Compatibility is set to Video and Content in the Advanced tab.

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Appendix I - Polycom Open Collaboration Network (POCN)

6 Click the Video Settings tab.

7 Set the Telepresence Mode to Auto.
8 Assign the New Profile to the Meeting Room. For more information see Creating a New Meeting
Room.

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Appendix I - Polycom Open Collaboration Network (POCN)

Content Sharing Behavior
The following tables list the system’s Content sharing behavior for the various combinations of TIP
Compatibility mode settings and the following endpoints:

Polycom Immersive Telepresence Systems (ITP) Version 3.0.3:
 RPX 200
 RPX 400
 OTX 300
 TPX HD 306
 ATX HD 300

Polycom video conferencing endpoints (HDX) Version 3.0.3:
 7000 HD Rev C
 8000 HD Rev B
 9006
 4500

Cisco TelePresence® System (CTS) Versions 1.7 / 1.8:
 CTS 1000
 CTS 3000
TIP Compatibility - None
Content Receiver

None

Content
Sender

HDX / ITP

HDX / ITP

Media:
Flow Control:

CTS

CTS

H.264
H.323 via H.239
SIP via BFCP

Not Connected

Not Connected
Not Connected

TIP Compatibility - Video Only
Video Only

Content
Sender

HDX / ITP
CTS

Polycom®, Inc.

Content Receiver
HDX / ITP
Media:
Flow Control:

H.264
H.323 via H.239
SIP via BFCP

None

CTS
None
None

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Appendix I - Polycom Open Collaboration Network (POCN)

TIP Compatibility - Video & Content
Content Receiver

Video & Content

Content
Sender

HDX* / ITP

HDX / ITP
Media:
Flow Control:

CTS

CTS

H.264
H.323 via H.239
SIP via BFCP
TIP via Auto Collaboration

* If HDX supports TIP Content.
Selecting TIP Compatibility as Video and Content disables Content Settings in the Video Quality tab.
TIP Compatibility - Prefer TIP
Prefer TIP

Content
Sender

HDX / ITP
CTS*

Content Receiver
HDX / ITP
Media:
Flow Control:

CTS

H.264
H.323 via H.239
SIP via BFCP
TIP via Auto Collaboration

* CTS Version 1.9.1 and higher support H.264 Content.
In Prefer TIP mode, it is pre-requisite that the CTS and CUCM versions support H.264 base profile
content without restrictions and that the CTS version be 1.9.1 or higher and that CUCM version be
version 9.0 or higher.

Procedure 5: Configuring an Ad Hoc Entry Queue on the Collaboration Server if
DMA is not used
1 Create or select the Entry Queue as described in Entry Queues.

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Appendix I - Polycom Open Collaboration Network (POCN)

2 In the New Entry Queue or Entry Queue Properties dialog box, ensure that Ad Hoc is selected.

3 Ensure that the Entry Queue is designated as the Transit Entry Queue as described in Setting a
Transit Entry Queue.

Procedure 6: Configuring a Meeting Room on the Collaboration Server
The Profile for the Meeting Room must be TIP enabled as described in Procedure 4.
For more information see Creating a New Meeting Room.

Procedure 7: Configuring Participant Properties for dial out calls
Participant Properties must be configured to ensure that defined participants inherit their TIP settings from
the Profile assigned to the Meeting Room.
a Define the New Participant’s General settings. For more information see Adding a Participant to
the Address Book.

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Appendix I - Polycom Open Collaboration Network (POCN)

b Click the Advanced tab.

c Ensure that:
 Call Bit Rate is set to Automatic or at least equal to or greater than the value specified by the
MIN_TIP_COMPATIBILITY_LINE_RATE System Flag.
 Resolution is set to Auto or at least HD 720.
 Video Protocol is set to Auto or at least H.264.

Collaboration with Microsoft and Cisco
This solution enables Polycom, Microsoft and Cisco users, each within their own environment, to participate
in the same conference running on an Collaboration Server.
The Collaboration Server natively inter-operates with Microsoft Lync and Cisco TelePresence Systems,
ensuring optimum quality multi-screen, multipoint calls between:
● Polycom Immersive Telepresence Systems (ITP) Version 3.1.1:
 RPX 200
 RPX 400
 OTX 300
● Polycom video conferencing endpoints
 Standalone HDX
 Polycom Group Series 300/500
● Microsoft
 MS Lync (using MS-ICE)
 RTV 720p
● Cisco TelePresence® System (CTS) Versions 1.10
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Appendix I - Polycom Open Collaboration Network (POCN)

 CTS 1300
 CTS 3010
The deployment architecture in Single company with Polycom and Cisco Infrastructure - Polycom endpoints
using SIP shows a company that has a mixture of Polycom, Cisco and Microsoft endpoints, room systems
and telephony equipment that needs to enable multipoint calls between all its video and audio endpoints
using the Collaboration Server as the conference bridge.
This solution enables Polycom, Microsoft and Cisco users, each within their own environment, to participate
in the same conference running on an MCU.
In the solution described in Single company with Polycom and Cisco Infrastructure - Polycom endpoints
using SIP:
● DMA is required as all calls are dial-in to Virtual Meeting Rooms (VMR) provisioned on the DMA.
● Microsoft and Cisco clients dial the same VMR number to connect to the conference.
● Dial- out calls directly from the RMX are not supported.
● Lync Clients cannot share content with CTS
● SIP trunks are required to the DMA from:
 MS Lync as a Static Route.
 CUCM
Deployment Architecture
● DMA is required as all calls are dial-in to Virtual Meeting Rooms (VMR) provisioned on the DMA.
● Microsoft and Cisco clients dial the same VMR number to connect to the conference.
● Dial- out calls are not supported
● Lync Clients can not share content with CTS
● SIP trunks are required to the DMA from:
 MS Lync as a Static Route.
 CUCM

For more information, see Cisco TIP Support.

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Appendix I - Polycom Open Collaboration Network (POCN)

POCN Polycom, Microsoft and Cisco Infrastructure. Solution Architecture
components.

.POCN Polycom, Microsoft and Cisco Infrastructure. Solution Architecture components
Component

Version

Polycom
HDX

3.0.5

RSS

8.0

DMA

5.0

CMA

6.0.1

CMAD

5.2.3

ITP (OTX, RPX, ATX, TPX)

3.0.5

Conferencing for Outlook (PCO)

1.0.7

Touch Control

1.3

Microsoft
Microsoft Lync 2010 Server

4.0.7577.223(CU10)

Microsoft Lync 2013 Server

5.0.8308.556 (CU3)

Microsoft Lync 2010 client

4.0.7577.4051 CU4

Exchange 2007 R2 SP3

8.3.213.1

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Appendix I - Polycom Open Collaboration Network (POCN)

Component

Version

Exchange 2010 SP2

14.2.247.5

Outlook 2007

12.0.6557.5001 SP2

Outlook 2010

14.0.6112.5000

Cisco
CUCM

8.5, 8.6.2

Cisco Unified Personal communicator

8.5(2),8.5(5)

Cisco Unified IP Phones 7960, 7961, 7962, 7965, 7975

CUCM 8.5 / CUCM 8.6(2) Compatible

CTS

1.7.4, 1.8.1

C90, C20

TC5.0

The following are not supported:
● In the Lync environment:
 Sending or receiving Content.
 Dial-out to Lync clients.
 Presence of VMRs
● In the Cisco environment:
 TLS and SRTP
 OBTP

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Appendix I - Polycom Open Collaboration Network (POCN)

Call Flow
Multipoint Calls using DMA
In this example:
● Endpoint registration: To either DMA, Lync or CUCM.
● DMA dial in Prefix: 72
● Virtual Meeting Room in DMA: 725123
● DMA Meeting Number: Generated by DMA

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Appendix I - Polycom Open Collaboration Network (POCN)

Administration
The various deployment combinations and settings within the Deployment Architecture affects the
administration of the system.

DMA
The DMA system can be configured as a SIP proxy and registrar for the environment as well as a
Gatekeeper for dial in H.323 calls. When configured as a Gateway for dial in H.323 calls, it enables H.323
endpoints to connect to the same VMR as SIP clients.
When used as a SIP peer, the DMA system can host video calls between Cisco endpoints that are registered
with the CUCM, Lync Clients that are registered with the Lync Server and Polycom endpoints that are
registered with the DMA system.
For more information see the Polycom Unified Communications Deployment Guide for Cisco Environments,
Using a Polycom DMA System as SIP Peer.

Microsoft Lync Server
Microsoft Lync Server manages Presence for each registered Polycom endpoint and enables video calls
between Lync Clients and Polycom endpoints allowing Lync contacts to be called without needing to know
their addresses.
RTV video, MS-ICE and Lync-hosted conferencing are supported when Polycom endpoints are registered
to Lync Server. Polycom endpoints use H.264, while Lync Clients use the RTV protocol.

CUCM
When Polycom SIP endpoints (voice and video) are registered directly with CUCM you can take advantage
of supported telephone functions. CUCM may not support the full range of codecs and features available
on the Polycom equipment. CUCM supported codecs and features will be used in such cases.
For more information see the Polycom Unified Communications Deployment Guide for Cisco Environments,
Direct Registration of Polycom Endpoints with the Cisco Unified Communications Manager Participants.

Solution Interoperability Table
The following table lists components and versions of the Collaboration Server, Microsoft and Cisco
Telepresence Systems (CTS) Integration Solution Architecture.
Solution Architecture Components
Component

Version

Description

9.0.1

Cisco Unified Communication Manager:
• CUCM must be configured to route calls to ASR/SBC. CUCM must be
configured with a SIP trunk to the Service Provider’s SBC.
• All endpoints must register once with the CUCM
• SIP trunks from CUCM to Polycom system components (eg. DMA) should
be configured with Music on Hold disabled.

CISCO Equipment
CUCM

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Component

Version

Description

ASR (Cisco SBC)

100x

The Cisco Aggregation Services Routers (ASR) Series includes Cisco IOS
XE Software Internetwork Operating System - Gatekeeper.
It controls and manages real-time multimedia traffic flows between IP/SIP
network borders, handling signaling, data, voice, and video traffic.

DMA

6.0.0_ATT_B
uild_25

Polycom Distributed Media Application
• DMA is an optional component but is essential if Content sharing is to be
enabled.
• All SIP endpoints register to DMA as a SIP Proxy.
• DMA should be configured to route SIP calls (with CTS destination) to
CUCM.
• DMA can be configured with a VMR (Virtual Meeting Room). Incoming
calls are then routed to the Collaboration Server.

Collaboration
Server

8.1.1

MCU:
• Functions as the network bridge for multipoint calls between H.323, SIP
and TIP endpoints.
• The Collaboration Server can be interfaced to CUCM using a SIP trunk,
enabling CTS to join multipoint calls on Collaboration Server. Signaling
goes through the CUCM while the media in TIP format goes directly
between the CTS and Collaboration Server.
• The Collaboration Server must be configured to route outbound SIP calls
to DMA.
• Collaboration Server must be configured to send and receive RTP
streams to and from the Service Provider's SBC.

MLA Server

3.0.5

Multipoint Layout Application
Required for managing multi-screen endpoint layouts for Cisco CTS 3XXX,
Polycom TPX, RPX or OTX systems.

HDX and ITP
Endpoints

3.1.1.1

Telepresence, desktop and room systems.
• Polycom SIP endpoints must register to DMA as SIP Proxy.

Polycom Equipment

Microsoft
Lync 2010

4.0.7577.183
CU4

Lync 2010 client

4.0.7577.405
1 CU4

Exchange 2007
R2 SP3

8.3.213.1

Exchange 2010
SP2

14.2.247.5

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Component

Version

Outlook 2007

12.0.6557.50
01 SP2

Outlook 2010

14.0.6112.50
00

Description

TIP Layout Support & Resource Usage
Cisco Telepresence endpoints using TIP protocol support only one (CTS 1000) or three (CTS 3000) display
screens. Therefore, Polycom Telepresence endpoints will adjust their display to use one or three screens
as follows:
● OTX system - works with three screens, therefore no adjustment is required and it should be set to
work in room switch Telepresence Layout Mode (while avoiding zooming in/out)
● RPX 2xx - This endpoint works with two screens, therefore it will adjust to use only one screen.
● RPX 4xx - This endpoint works with four screens, therefore it will adjust to use only three screens.
● Standalone HDX - behaves as the CTS 1000 and uses only one screen.
● Group system 300/500 - behaves as the CTS 1000 and uses only one screen.
The Polycom MLA Server manages the conference template layouts for Telepresence systems.
The number of screens used by each TIP-enabled endpoint is determined during the capabilities exchange
phase of the dial-in connection. It affects the usage and allocation of resources used with TIP-enabled
endpoints.

Supported TIP Resolutions and Resource Allocation
Supported Resolutions
In a Telepresence TIP-enabled environment, only two video resolutions are available: 720p30 & 1080p30.
Supported resolution per conference line rate
Conference Line Rate

Selected Resolution

3Mb or higher

1080p 30 fps

963kbps to 3Mb

720p 30 fps

Up to 936kbps

Call is disconnected.

Resource Allocation
The MCU media processor (ART) supports up to three TIP-enabled screens as follows:
● One TIP-enabled endpoint with three screens
● Up to three TIP-enabled endpoint with one screen
TIP-enabled endpoint with three screens must be handled by the same media processor. This endpoint may
fail to connect if there is no one fully free media (ART) processor available.

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The MCU will always try to fill up one media processor with up to three TIP-enabled endpoint with one
screen, to save free media processors for TIP-enabled endpoint with three screens.
When monitoring an ongoing Telepresence conference with TIP-enabled endpoints (Cisco and Polycom),
virtual participants are used to indicate the additional screens in the in the Web Client. For example, if the
endpoint has three screens, the system will display three participants, one for each screen.
An additional virtual Audio Only participant is used for the audio only telephone connected to the TIP
endpoint.

Configuring the Microsoft, Cisco and Polycom
Components
1 Configure a SIP Trunk connection between the Polycom DMA system and the Cisco Unified
Communications Manager (CUCM).
For more information see the Polycom Unified Communications Deployment Guide for Cisco
Environments, Using a Polycom DMA System as SIP Peer.
2 Register the Collaboration Server to the Lync Server
a Install a Security Certificate on the Collaboration Server.
The Certificate is obtained from the System Administrator and saved on the Workstation.
b In the SIP Servers tab of the IP Network Services Properties dialog box:
1

In the Certificate Method drop-down menu, select PEM/PFX.

2

Click the Send Certificate button.
The Install File dialog box is displayed.

iii Browse to the saved Certificate on the Workstation and click the Yes button to install the
certificate.

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For more information see:
 Appendix H - Integration Into Microsoft Environments.
 Polycom Unified Communications Deployment Guide for Microsoft Environments, Configuring
Your Collaboration Server System for use with the Lync Server.
3 Register the Collaboration Server with the Lync Server.
a In the IP Network Services Properties dialog box, click the SIP Servers tab.
b In the SIP Server field, select Specify.
c In the SIP Server Type field, select Microsoft.
d Set Refresh Registration every 3600 seconds.
e If not selected by default, change the Transport Type to TLS.
f

In the SIP Servers table, enter the IP address of the Lync Server in both the Server IP Address or
Name and Server Domain Name fields.

g In the SIP Servers table, the Port field must be set to 5061.
h In the Outbound Proxy Servers table, enter the IP address in the Server IP Address or Name field.
(The same value as entered in Step f.)

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i

In the Outbound Proxy Servers table, the Port field must be set to 5061. (The same value as
entered in Step g.)

For more information see the Polycom Unified Communications Deployment Guide for Microsoft
Environments.
4 Set the ITP_CERTIFICATION System Flag to YES.
When set to NO (default), this flag disables the Telepresence features in the Conference Profile.
5 Set the MIN_TIP_COMPATIBILITY_LINE_RATE System Flag.
The MIN_TIP_COMPATIBILITY_LINE_RATE System Flag determines the minimum line rate at
which a Profile can be TIP enabled.
CTS version 1.7 requires a minimum line rate of 1024 kbps and will reject calls at lower line rates,
therefore the System Flag value must be 1024 or higher.
For more information see Modifying System Flags.
6 If required, manually add and set the FORCE_720P_2048_FOR_PLCM_TIP System Flag using
one of the following values:
FORCE_720P_2048_FOR_PLCM_TIP (Default) - Forces HD 720p video resolution and a line rate
of 2048kbps for all Polycom TIP-enabled endpoints that connect to the TIP-enabled Telepresence
conference. This setting is the recommended setting.
FORCE_2048_FOR_PLCM_TIP - Forces a line rate of 2048kbps for all Polycom TIP-enabled
endpoints connecting to the TIP-enabled Telepresence conference.
NO_FORCE - No forcing is applied and Polycom TIP-enabled endpoints can connect to the
TIP-enabled Telepresence conference at any line rate or resolution.
7 Reset the Collaboration Server.
8 For more information see .
9 Register the DMA to the Lync server
For more information see the Polycom Unified Communications Deployment Guide for Microsoft
Environments, Configure a DMA System SIP Peer for the Lync Server.

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10 Register the ITP endpoints to the Lync server
For more information see the Polycom Unified Communications Deployment Guide for Microsoft
Environments, Deployment Process for Polycom Immersive Telepresence Systems.
11 Register Lync Clients to the Lync server.
For more information see the relevant Lync documentation.
12 Register DMA to the CUCUM server
For more information see the Polycom Unified Communications Deployment Guide for Cisco
Environments, Using a Polycom DMA System in a Cisco Environment.
13 Register CTS1000 and CTS3000 endpoints to the CUCUM server
For more information see the relevant Cisco documentation.
14 Register ITP endpoints to the CUCM server.
For more information see the Polycom Unified Communications Deployment Guide for Cisco
Environments, Direct Registration of Polycom Telepresence Systems with the Cisco Unified
Communications Manager.
15 Register HDX endpoints to the DMA as Gatekeeper
For more information see the Polycom® DMA™ 7000 System Operations Guide.
16 Open MLA to configure ITP Layouts
MLA (Multipoint Layout Application) is required for managing CTS 3XXX layouts whether Polycom
TPX, RPX or OTX systems are deployed or not. MLA is a Windows® application that allows
conference administrators to configure and control video layouts for multipoint calls involving
Polycom Immersive Telepresence (ITP) systems.
For more information see the Polycom Multipoint Layout Application (MLA) User’s Guide for Use with
Polycom Telepresence Solutions.
17 Configure a TIP Enabled Profile on the Collaboration Server.
a Create a New Profile for the Meeting Room.
For more information see Defining New Profiles.

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b In the New Profile - General tab, set the Line Rate to a value of at least that specified for the
MIN_TIP_COMPATIBILITY_LINE_RATE System Flag in Procedure 1: Set the
MIN_TIP_COMPATIBILITY_LINE_RATE System Flag.

c Click the Advanced tab.

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d Select the TIP Compatibility mode according to the Content Sharing Behavior tables that are
listed below.
Prefer TIP is recommended if Polycom endpoints are to connect using TIP, otherwise select
Video and Content.
When Prefer TIP is selected Gathering Settings, Skins, Message Overlay, Site Names and Network
Indication(s) are disabled.

Content Sharing Behavior
The following tables list the system’s Content sharing behavior for the various combinations of TIP
Compatibility mode settings and the following endpoints:

Polycom Immersive Telepresence Systems (ITP) Version 3.0.3:
 RPX 200
 RPX 400
 OTX 300
 TPX HD 306
 ATX HD 300

Polycom video conferencing endpoints (HDX) Version 3.0.3:
 7000 HD Rev C
 8000 HD Rev B
 9006
 4500

Cisco TelePresence® System (CTS) Versions 1.7 / 1.8:
 CTS 1000
 CTS 3000
TIP Compatibility - None
Content Receiver

None

Content
Sender

HDX / ITP

HDX / ITP
CTS

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Media:
Flow Control:

H.264
H.323 via H.239
SIP via BFCP

Not Connected

CTS
Not Connected
Not Connected

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TIP Compatibility - Video Only
Content Receiver

Video Only

Content
Sender

HDX / ITP

HDX / ITP
Media:
Flow Control:

CTS

CTS

H.264
H.323 via H.239
SIP via BFCP

None

None
None

TIP Compatibility - Video & Content
Content Receiver

Video & Content

Content
Sender

HDX* / ITP

HDX / ITP
Media:
Flow Control:

CTS

CTS

H.264
H.323 via H.239
SIP via BFCP
TIP via Auto Collaboration

* If HDX supports TIP Content.
Selecting TIP Compatibility as Video and Content disables Content Settings in the Video Quality tab.
TIP Compatibility - Prefer TIP
Prefer TIP

Content
Sender

HDX / ITP
CTS*

Content Receiver
HDX / ITP
Media:
Flow Control:

CTS

H.264
H.323 via H.239
SIP via BFCP
TIP via Auto Collaboration

* CTS Version 1.9.1 and higher support H.264 Content.
In Prefer TIP mode, it is pre-requisite that the CTS and CUCM versions support H.264 base profile
content without restrictions and that the CTS version be 1.9.1 or higher and that CUCM version be
version 9.0 or higher.

Encryption
Encryption between the Polycom® RealPresence® Collaboration Server 800s and Polycom®
RealPresence® Collaboration Server Virtual Edition and a CISCO environment is supported. Media is
encrypted using SRTP, while control is encrypted using SRTCP. TIP is encrypted using SRTCP. SIP is be
encrypted using TLS. When upgrading, the Collaboration Server automatically creates a self-signed
certificate to support encrypted communications with CISCO endpoints.
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For media encryption. the Collaboration Server will first attempt to exchange keys using DTLS. If the
Collaboration Server fails to exchange keys using DTLS, SIP TLS encrypted with SDES is used to exchange
media encryption keys.

Guidelines
● This feature is not supported in Ultra Secure Mode.
● Voice activity metrics and RTP are not encrypted.
● In the event that DTLS negotiation fails, SIP will be encrypted using TLS if enabled in the IP
Management Network properties, SIP Servers tab. DTLS negotiation does not require SIP TLS.
 In a mixed CISCO and Microsoft Lync environment, in order to assure encrypted communications
with both CISCO endpoints and Microsoft Lync in the event of DTLS negotiation failure, the
certificate defined in the IP Management Network Services properties dialog box, SIP Servers
tab, must have been issued by the same certificate authority that issued the certificates used by
both the Microsoft Lync server and the CUCM server.
● The flag, SIP_ENCRYPTION_KEY_EXCHANGE_MODE, is used to control this feature. The
possible values are:
 AUTO (default): Normal encryption flow
 DTLS: Only use DTLS for encryption
 SDES: Only use SDES (SRTP) for encryption
 NONE: Encryption is disabled
● The feature was tested using the following CISCO components:
 Cisco CUCM Version 9.0
 Cisco TPC Version 2.3
 Cisco endpoints running Version 1.9.1
 C20, C40, C60, and C90 running TC5
 CTS500
 CTS1310
 CTS3010

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To enable DTLS negotiation for content encryption:
1 In a new or existing Profile, click the Advanced tab.

2 Set Encryption to either Encrypt All or Encrypt when possible.
3 Set the FORCE_ENCRYPTION_FOR_UNDEFINED_PARTICIPANT_IN_WHEN_
AVAILABLE_MODE System Flag to NO
These setting will enable encrypted and non-encrypted H.323 participants to connect to encrypted or
non-encrypted conferences.
For more information see Encryption.

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a Click the Video Quality tab.

Content Settings is disabled if TIP Compatibility is set to Video and Content in the Advanced tab.

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b Click the Video Settings tab.

c Set the Telepresence Mode to Auto/On and select the Telepresence Layout Mode.
4 Assign the New Profile to the Meeting Room. For more information see Creating a New Meeting
Room.
5 Configure a Virtual Meeting Room (VMR) on the DMA.
The procedures for configuring DMA are described in detail in the Polycom Unified Communications
Deployment Guide for Cisco Environments.

Resolution Configuration
The resolution configuration dialog box is not applicable to TIP-enabled conferences as it uses fixed
settings:

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HD Video Resolutions for TIP calls are determined according to the following table:
TIP HD Video Resolution by Line Rate
Line Rate

Video Resolution

Line Rate >=3Mbps

HD1080p30

3Mbps > Line Rate >= 936kbps

HD720p30

Line Rate < 936kbps

Call is dropped.

Endpoints
1 Configure HDX endpoints to register to Lync Server.
The procedures for configuring HDX endpoints are described in detail in the Polycom Unified
Communications Deployment Guide for Microsoft Environments.
2 Configure H.323 endpoints to register to DMA as SIP Proxy
The procedures for configuring SIP endpoints are described in detail in the Polycom Unified
Communications Deployment Guide for Cisco Environments.
3 Configure SIP endpoints to register to:
 DMA as SIP Proxy
 Lync Server as SIP Proxy
 CUCM as SIP Proxy
The procedures for configuring SIP endpoints are described in detail in the Polycom Unified
Communications Deployment Guide for Cisco Environments.
4 Configure TIP endpoints to register to:
 DMA
 CUCM
The procedures for configuring TIP- enabled endpoints are described in detail in the Polycom
Unified Communications Deployment Guide for Cisco Environments.

Content
Endpoint Registration and Dialing Method affect the Video and Content Sharing characteristics of the
conference as detailed in Table 5-9.

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Video and Content
Endpoint Registration
Lync

CUCM

DMA

ITP /HDX
RTV Key is required for
HDX and ITP

ITP /HDX
TIP Key is required for
HDX

ITP /HDX
TIP Key is required for
HDX

HDX to
Collaboration
Server

•
•
•
•

HD H.264 Video
SIP P+C
Content: XGA,5fps
ICE

•

HD H.264 Video

•
•

No Content
ICE not supported

•
•
•
•

Lync to
Collaboration
Server

•
•
•

HD Video (RTV)
No Content Sharing
Content sent to Lync using Content for Legacy Endpoints

CTS to
Collaboration
Server

•
•
•

HD1080p30
TIP Content Sharing
Content: XGA,5fps

Dialing Method

HD H.264 Video
SIP P+C
Content: XGA,5fps
ICE not supported

Operations During Ongoing Conferences
Moving participants between TIP enabled meetings and non TIP enabled meetings is not possible.

Monitoring
CTS Participants
1 In the Participant List pane double-click the participant entry. Alternatively, right-click a participant
and then click Participant Properties.
The Participant Properties - General dialog box opens.
2 Click the SDP tab.
The following are indicated in the Remote Capabilities, Remote Communication Mode and Local
Communication Mode panes:
 AAC_LD - Audio Protocol

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● Main Profile - Video protocol

When viewing CTS systems in the Participants list, the individual video screens and the Audio Channel
(AUX) of the CTS system are listed as separate participants. The Participant list below shows a connected
CTS 3000, a 3-screen system.

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Lync Participants (RTV)
1 In the Participant List pane double-click the participant entry. Alternatively, right-click a participant
and then click Participant Properties.
The Participant Properties - General dialog box opens.
2 Click the SDP tab.
RTV is indicated in the Remote Capabilities, Remote Communication Mode and Local
Communication Mode panes:

3 Click the Channel Status - Advanced tab

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4 In the Channel Info drop-down menu select Video Out.
Media Info displays RTV Channel Status parameters:

Known Limitations
The following may occur in the collaborative environment:
● Artifacts and ghosting may appear when Lync Clients and CTS endpoints connect to the VMR.
Frequency: Seldom.
● Lync Client receives fast updates (Intra) from CTS 500 endpoints causing the screen to refresh
repeatedly.
Frequency: Often.
● Audio volume and video quality decreases on CTS endpoints.
Frequency: Seldom.
● CTS endpoint connects and then disconnects after a few seconds.
Frequency: Seldom.
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● Lync Clients always connect encrypted to non-encrypted conferences.
● Auto Layout sometimes ignored for CTS and Lync Clients calling through DMA.
Frequency: Rarely.
● Content sent from HDX endpoint is received by all endpoints for 1 second before stopping.
Conference is Content to Legacy enabled and TIP Compatibility is Video Only.
Frequency: Often.

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Restoring Defaults and System Recovery
The USB port of a RealPresence Collaboration Server 800s in secure mode can be used to:
● Restore the RealPresence Collaboration Server 800s to Factory Security Defaults mode (https 
http).
● Perform a Comprehensive Restore to Factory Defaults
● Perform an Emergency CRL (Certificate Revocation List) Update

Restore to Factory Security Defaults
Restore to Factory Security Defaults can be performed by either:
● Inserting a USB device such as a mouse or a keyboard into any USB port of the RealPresence
Collaboration Server 800s causing it to exit secure mode and return to Factory Security Defaults
mode. After performing this procedure, logins to the RealPresence Collaboration Server 800s use the
http command and not the https command.
or
● Using the Polycom USB key that came with the RealPresence Collaboration Server.

To restore the RealPresence Collaboration Server 800s to Factory Security Defaults using
a USB device:
1 Insert a USB device into any USB port of the RealPresence Collaboration Server 800s.
2 Power the RealPresence Collaboration Server 800s Off and then On.
3 Login using http://.

To restore the RealPresence Collaboration Server 800s to Factory Security Defaults using
the Polycom USB key:
1 Inset the Polycom USB key that came with the RealPresence Collaboration Server into your
computer.
The Polycom Documentation window is displayed.
a Select Open Folder to view files using Windows Explorer.

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b Double-click the index.hta file.
The Language Menu is displayed, offering a choice of several languages.

2 Click the documentation language of your choice.
An End-User Licence Agreement for Polycom Software is displayed.
3 Read the agreement and click the Accept Agreement button.
4 In the Product Type window, select RealPresence Collaboration Server 800s.

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Restoring Defaults and System Recovery

5 In the Initial Setup Utility dialog box, click the Restore to Factory Security Defaults link.

The Restore to Factory Security Defaults dialog box is displayed.

6 Click OK.
7 Remove the USB key from the PC workstation.
8 Insert the USB key in any USB port of the RealPresence Collaboration Server 800s.
9 Turn off the RealPresence Collaboration Server 800s, then turn it on.
10 Start the Collaboration Server Web Client application on the workstation.
a In the browser’s address line, enter the IP address of the Control Unit in the format:
http://.

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b Click Enter.
When the Collaboration Server Web Client Login window is displayed, the system was successfully
restored to the factory security mode.

11 Remove the USB key from the RealPresence Collaboration Server 800s.

Comprehensive Restore to Factory Defaults
The Comprehensive Restore to Factory Defaults deletes the following files:
● CDR
● Address Book
● Log Files
● Faults
● Dump Files
● Notes
In addition, all the conferencing entities are deleted:
● Entry Queues
● Profiles
● Meeting Rooms
● IVR Services
● Default Network IP Service
● Log Files
● CFS license information
● Management Network Service
The RealPresence Collaboration Server 800s is restored to the settings it had when shipped from the
factory. The Product Activation Key is required to re-configure the Management Network Service during the
First Entry Configuration.

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Restoring Defaults and System Recovery

Comprehensive Restore to Factory Defaults Procedure
Restoring the RealPresence Collaboration Server 800s to Factory Defaults consists of the following
procedures:

A: Backup Configuration Files
 These files will be used to restore the system in Procedure C.

B: Restore to Factory Defaults
 Restart the system with the configured Polycom USB key plugged into any USB port.

C: Optional. Restore the System Configuration From the Backup
 Apply the backup file created in procedure A.
 Restart the RealPresence Collaboration Server 800s.
(If the RealPresence Collaboration Server 800s is unresponsive after these procedures, a further
restart may be necessary.)

Procedure A: Backup Configuration Files
The Software Management menu is used to backup and restore the configuration files of the RealPresence
Collaboration Server 800s and to download MCU software.

To backup configuration files:
1 On the Collaboration Server Menu, click Administration > Software Management > Backup
Configuration.
The Backup Configuration dialog box is displayed.

2 Click Browse.
3 Browse to the Backup Directory Path and then click Ok.
4 Click Backup.

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Procedure B: Restore to Factory Defaults
To perform a Comprehensive Restore to Factory Default perform the following steps:
1 Insert the Polycom USB key that came with the RealPresence Collaboration Server into your
computer.
The Polycom Documentation window is displayed.
a Select Open Folder to view files using Windows Explorer.
b Double-click the index.hta file.
The Language Menu is displayed, offering a choice of several languages.

2 Click the documentation language of your choice.
An End-User Licence Agreement for Polycom Software is displayed.
3 Read the agreement and click the Accept Agreement button.
4 In the Product Type window, select RealPresence Collaboration Server 800s.

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Restoring Defaults and System Recovery

5 In the Initial Setup Utility window, click the Restore to Factory Defaults link.

The Restore to Factory Security Defaults dialog box is displayed.

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Restoring Defaults and System Recovery

6 Recommended: Click the Lan Configuration check box and modify the following parameters in
the utility’s dialog box using the information supplied by your network administrator.
 Control Unit IP Address
 Subnet Mask
 Default Router IP Address
The restore to Factory Defaults removes the Management Network Service. If the above fields
addresses are not set, the Control Unit IP Address of the Management Network Service will be
192.168.1.254.

7 Click OK.
8 Remove the USB key from the PC.
9 Insert the USB key in any USB port of the RealPresence Collaboration Server 800s.
10 Turn off the RealPresence Collaboration Server 800s, then turn it On.
11 Start the Collaboration Server Web Client application on the workstation.
a In the browser’s address line, enter the IP address of the Control Unit in the following format:
http://,

as defined in the USB key. If no Control Unit IP Address was defined, the IP address will be
192.168.1.254.
b Click Enter.
When the Collaboration Server Web Client Login window is displayed, the system was successfully
restored to factory defaults.

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12 Optional. Restore the system using Procedure C: Restore the System Configuration From the
Backup on page 873 below.

Procedure C: Restore the System Configuration From the Backup
To restore configuration files:
1 On the Collaboration Server Menu, click Administration > Software Management > Restore
Configuration.
2 Browse to the Restore Directory Path where the backed up configuration files are stored.
3 Click the Restore button.
4 When the Restore is complete, restart the RealPresence Collaboration Server 800s.
RealPresence Collaboration Server 800s system settings, with the exception of User data, are
restored.
5 Restore User data by repeating step 1 to step 3 of this procedure.

System Recovery Using the Recovery DVD
You can use the system recovery disk to re-install the Polycom® RealPresence® Collaboration Server 800s
and Polycom® RealPresence® Collaboration Server Virtual Edition application in case of system crashes
or server hard disk damage.
Two DVDs are shipped with the Polycom® RealPresence® Collaboration Server 800s and Polycom®
RealPresence® Collaboration Server Virtual Edition, the Polycom Recovery DVD, and the Dell
Diagnostics DVD.

Preparation for System Recovery
Before performing the recovery procedure it is necessary to modify the Factory Default Management
Network Settings on the USB Utilities and Documentation memory stick. For details, see the
RealPresence Collaboration Server 800s Getting Started Guide.

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Restoring Defaults and System Recovery

Performing the Recovery Process
To re-install the system using the System Recovery DVD:
1 Turn off the RMX™.
2 Insert the USB memory stick into any USB slot.
3 Connect a USB keyboard and monitor.
4 Turn on the RMX™.
5 Insert the Polycom Recovery DVD that comes with your system.
The RMX™ loads the Polycom Recovery DVD.

6 Press 2 and then Enter.
7 The RMX™ installation starts.
The whole procedure may take 15 minutes. When the installation completes, the system reboots.

When the system reboots the DVD is ejected.

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Restoring Defaults and System Recovery

8 Remove the DVD from the server and close the DVD drive.

Completing the System Configuration
To complete the system configuration:
1 Connect to the MCU using the default Polycom user name and Polycom password. For details, see
the RealPresence Collaboration Server 800s Getting Started Guide, Procedure 3:
Connecting to the MCU..
2 Enter the activation key.
3 Define the IP Network Service for the media and signaling using the Fast Configuration Wizard. For
details, see the RealPresence Collaboration Server 800s Getting Started Guide, Fast
Configuration Wizard.
4 Optional: Restore the system configuration. For details, see Software Management.
5 Remove the USB memory stick from the MCU.
The system recovery is complete.

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Appendix K - SIP RFC Support
SIP RFC Support in RealPresence Collaboration Server (RMX) Systems
SIP RFC

Description

1321

MD5

2032

RTP Payload for H.261

2205

RSVP

2327

Session Description Protocol (SDP)

2429

RTP Payload for H.263+

2833

RTP Payload for DTMF

2617

HTTP Authentication

2976

SIP Info Method

3261

SIP

3264

Offer/Answer Model

3265

SIP Specific Event Notification

3311

SIP Update Method

3515

SIP Refer Method

3550

RTP

3551

RTP Profile for Audio/Video

3711

SRTP

3890

Transport Independent Bandwidth Modifier for SDP

3891

SIP Replaces header

Limited support

3892

SIP Referred-by Mechanism

Limited support

3984

RTP Payload format for H.264

4028

Session Timers in SIP

4145

TCP Media Transport in SDP

4566

Session Description Protocol (SDP)

4568

SDP Security Descriptions

4573

H.224 RTP Payload (FECC)

Polycom®, Inc.

Note

Limited support

Limited support

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Appendix K - SIP RFC Support

SIP RFC

Description

4574

SDP Label Attribute

4582

Binary Floor Control Protocol (BFCP)

4583

SDP for BFCP

4796

SDP Content Attribute

5168

XML Schema for Media Control (Fast Update)

cc-transfer

Call Transfer Capabilities in SIP

draft-turn-07

TURN spec for firewall traversal in SIP

draft-rfc3489bis-15

STUN spec for firewall traversal in SIP

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Note

Limited support

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