RTX X8663 DECT Base station User Manual

RTX Hong Kong Ltd. DECT Base station Users Manual

Users Manual

SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: About This Document 1 SME VoIP System Guide Installation & Configuration Network Deployment Operation & Management Technical Reference Document Version 2.6 © Jan-2016 RTX A/S, Denmark
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: About This Document 2 Trademarks RTX and the combinations of its logo thereof are trademarks of RTX A/S, Denmark. Other product names used in this publication are for identification purposes and maybe the trademarks of their respective companies. Disclaimer The contents of this document are provided in connection with RTX products. RTX makes no representations with respect to completeness or accuracy of the contents of this publication and reserves the right to make changes to product descriptions, usage, etc., at any time without notice. No license, whether express, implied, to any intellectual property rights are granted by this publication Confidentiality This document should be regarded as confidential, unauthorized copying is not allowed © Jan-2016 RTX A/S, Denmark, All rights reserved http://www.rtx.dk
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: About This Document 3 Contents SME VoIP System Guide .................................................................................................................................... 1 ........................................................................................................................................................................... 1 Contents ............................................................................................................................................................ 3 1 About This Document ................................................................................................................................ 7 1.1 Audience .................................................................................................................................................. 7 1.2 When Should I Read This Guide .............................................................................................................. 7 1.3 Important Assumptions ........................................................................................................................... 7 1.4 What’s Inside This Guide ......................................................................................................................... 7 1.5 What’s Not in This guide.......................................................................................................................... 8 1.6 Abbreviations ........................................................................................................................................... 8 1.7 References/Related Documentation ....................................................................................................... 9 1.8 Document History .................................................................................................................................... 9 1.9 Documentation Feedback ....................................................................................................................... 9 2 Introduction – System Overview ............................................................................................................. 10 2.1 Hardware Setup ..................................................................................................................................... 10 2.2 Components of SME VoIP System ......................................................................................................... 11 2.3 Wireless Bands ...................................................................................................................................... 11 2.4 System Capacity (in Summary) .............................................................................................................. 12 2.5 Advantages of SME VoIP System ........................................................................................................... 12 3 Installation of Base Stations/Repeater .................................................................................................... 13 3.1 Package – Contents/Damage Inspection ............................................................................................... 13 3.2 RTX Base station Mechanics .................................................................................................................. 14 3.3 RTX Base Unit – Reset feature ............................................................................................................... 14 3.4 Installing the Base Station ..................................................................................................................... 15 3.5 Find IP of Base Station ........................................................................................................................... 16 3.6 Login to Base SME Configuration Interface ........................................................................................... 17 4 Making Handset Ready ............................................................................................................................ 18 4.1 Package – Contents/Damage Inspection ............................................................................................... 18 4.2 Before Using the Phone ......................................................................................................................... 19 4.3 Using the Handset ................................................................................................................................. 20 5 SME VoIP Administration Interface ......................................................................................................... 21 5.1 Web navigation ...................................................................................................................................... 21 5.2 Home/Status .......................................................................................................................................... 23 5.3 Extensions .............................................................................................................................................. 24
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: About This Document 4 5.4 Servers ................................................................................................................................................... 35 5.5 Network ................................................................................................................................................. 40 5.6 Management Settings Definitions ......................................................................................................... 44 5.7 Firmware Update Definitions ................................................................................................................ 48 5.8 Time Server ............................................................................................................................................ 49 5.9 Country .................................................................................................................................................. 51 5.10 Security ................................................................................................................................................ 52 5.11 Central Directory and LDAP ................................................................................................................. 54 5.12 Multi-cell Parameter Definitions ......................................................................................................... 57 5.13 Repeaters ............................................................................................................................................. 64 5.14 Alarm ................................................................................................................................................... 68 5.15 Statistics ............................................................................................................................................... 70 5.16 Settings – Configuration File Setup ..................................................................................................... 74 5.17 Sys log .................................................................................................................................................. 74 5.18 SIP Logs ................................................................................................................................................ 75 6 Multi-cell Setup & Management ............................................................................................................. 76 6.1 Adding Base stations ............................................................................................................................. 76 6.2 Synchronizing the Base stations ............................................................................................................ 80 6.3 Summary of Procedure – Creating a Chain............................................................................................ 82 6.4 Practical Configuration of Multi-cell System ......................................................................................... 83 7 Registration Management - Handset ...................................................................................................... 88 7.1 Register handset to base (non multiline) .............................................................................................. 88 7.2 Register handset to base (multiline) ..................................................................................................... 90 7.3 Register handset to base and specific extension (multiline) ................................................................. 93 8 Firmware Upgrade Procedure ................................................................................................................. 95 8.1 Network Dimensioning .......................................................................................................................... 95 8.2 TFTP Configuration ................................................................................................................................ 95 8.3 Create Firmware Directories ................................................................................................................. 96 8.4 Handset Firmware Update Settings ....................................................................................................... 98 8.5 Handset(s) and Repeater Firmware Upgrade ........................................................................................ 98 8.6 Base Station(s) Firmware Upgrade ...................................................................................................... 100 9 Functionality Overview .......................................................................................................................... 102 9.1 Gateway Interface ............................................................................................................................... 102 9.2 Detail Feature List ................................................................................................................................ 102 Appendix ........................................................................................................................................................ 105
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: About This Document 5 10 Appendix A: Basic Network Server(s) Configuration ......................................................................... 105 10.1 Server setup ....................................................................................................................................... 105 10.2 Requirements .................................................................................................................................... 105 10.3 DNS Server Installation/Setup ........................................................................................................... 105 10.4 DHCP Server Setup ............................................................................................................................ 105 10.5 TFTP Server Setup .............................................................................................................................. 107 10.6 SIP Server Setup ................................................................................................................................. 108 11 Appendix B: Using Base with VLAN Network .................................................................................... 111 11.1 Introduction ....................................................................................................................................... 111 11.2 Backbone/ VLAN Aware Switches ..................................................................................................... 112 11.3 How VLAN Switch Work: VLAN Tagging ............................................................................................ 113 11.4 Implementation Cases ....................................................................................................................... 113 11.5 Base station Setup ............................................................................................................................. 114 11.6 Configure Time Server ....................................................................................................................... 114 11.7 VLAN Setup: Base station .................................................................................................................. 115 12 Appendix C: SME VoIP Network Planning/Optimization ................................................................... 116 12.1 Network Requirements ..................................................................................................................... 116 12.2 Deployment Considerations .............................................................................................................. 116 12.3 Site Planning ...................................................................................................................................... 116 12.4 Cell Coverage / Capacity Planning ..................................................................................................... 117 12.5 Network Dimensioning ...................................................................................................................... 118 12.6 Environmental Considerations .......................................................................................................... 119 12.7 Recommended Base station/Repeater Placement ........................................................................... 119 12.8 Network Assessment/Optimisation .................................................................................................. 120 13 Appendix D: Local Central directory file handling ............................................................................. 122 13.1 Central Directory Contact List Structure ........................................................................................... 122 13.2 Central Directory Contact List Filename Format ............................................................................... 122 13.3 Import Contact List to Central Directory ........................................................................................... 123 13.4 Central directory using server ........................................................................................................... 124 13.5 Verification of Contact List Import to Central Directory ................................................................... 124 14 Appendix E: Network Operations ...................................................................................................... 125 14.1 Introduction ....................................................................................................................................... 125 14.2 System Start Up ................................................................................................................................. 125 14.3 Terminal Attachment......................................................................................................................... 125 14.4 Outgoing Calls .................................................................................................................................... 125
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: About This Document 6 14.5 Incoming Calls .................................................................................................................................... 125 14.6 Handover ........................................................................................................................................... 126 14.7 Roaming ............................................................................................................................................. 127
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: About This Document 7 1 About This Document This document describes the configuration, customization, management, operation, maintenance and trouble shooting of the SME VoIP System (RTX8663 base, RTX8630 handset, RTX8430 handset, RTX8830 ruggedized handset and RTX4024 Repeater) in RTX generic mode. For customer specific modes refer to specific customer agreements, which describe the software operational deviations from this document. For handset detailed user guide refer to [1]. 1.1 Audience Who should read this guide? First, this guide is intended for networking professionals responsible for designing and implementing RTX based enterprise networks. Second, network administrators and IT support personnel that need to install, configure, maintain and monitor elements in a “live” SME VoIP network will find this document helpful. Furthermore, anyone who wishes to gain knowledge on fundamental features in the Beatus system can also benefit from this material. 1.2 When Should I Read This Guide Read this guide before you install the core network devices of VoIP SME System and when you are ready to setup or configure SIP server, NAT aware router, advanced VLAN settings, base stations, and multi cell setup. This manual will enable you to set up components in your network to communicate with each other and also deploy a fully functionally VoIP SME System. 1.3 Important Assumptions This document was written with the following assumptions in mind: 1) You have understanding of network deployment in general 2) You have working knowledge of basic TCP/IP/SIP protocols, Network Address Translation, etc... 3) A proper site survey has been performed, and the administrator have access to these plans 1.4 What’s Inside This Guide We summarize the contents of this document in the table below: Where Is It? Content Purpose Chapter 2 Introduction to the SME VoIP Network To gain knowledge about the different elements in a typical SME VoIP Network Chapter 3 Installation of Base station/Repeater Considerations to remember before unwrapping and installing base units and repeaters Chapter 4 Making Handsets Ready To determine precautions to take in preparing handsets for use in the system Chapter 5 SME VoIP Administration Interface To learn about the Configuration Interface and define full meaning of various parameters needed to be setup in the system. Chapter 6 Multi-Cell Setup & Management Learn how to add servers and setup multiple bases into a multi-cell network Chapter 7 Registration Management - Learn how to register handset and extensions to base
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: About This Document 8 Handsets stations Chapter 8 Firmware Upgrade/Downgrade Management Provides the procedure of how to upgrade firmware to base stations and/or handsets and/or repeaters Chapter 9 System Functionality Overview To gain detail knowledge about the system features. 10 Appendix A Basic Network Servers Configuration To learn about operating the handset and base stations including detail description of handset MMI. 11 Appendix B VLAN Setup Management Examines how to setup VLAN in the SME network 12 Appendix C SME VoIP Network Planning/Optimization To learn radio network planning techniques including dimensioning, detailed capacity, coverage planning and network optimisation 13 Appendix D Local central directory file handling Detailed description of central directory file format and upload. 14 Appendix E Network Operations To study the operation of network elements during system start up, location registration, etc. 1.5 What’s Not in This guide This guide provides overview material on network deployment, how-to procedures, and configuration examples that will enable you to begin configuring your VoIP SME System. It is not intended as a comprehensive reference to all detail and specific steps on how to configure other vendor specific components/devices needed to make the SME VoIP System functional. For such a reference to vendor specific devices, please contact the respective vendor documentation. 1.6 Abbreviations For the purpose of this document, the following abbreviations hold: DHCP: Dynamic Host Configuration Protocol DNS: Domain Name Server HTTP(S): Hyper Text Transfer Protocol (Secure) (T)FTP: (Trivial) File Transfer Protocol IOS: Internetworking Operating System PCMA: A-law Pulse Code Modulation PCMU: mu-law Pulse Code Modulation PoE: Power over Ethernet RTP: Real-time Transport Protocol RPORT: Response Port (Refer to RFC3581 for details) SIP: Session Initiation Protocol SME: Small and Medium scale Enterprise VLAN: Virtual Local Access Network TOS: Type of Service (policy based routing) URL: Uniform Resource Locator UA: User Agent
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: About This Document 9 1.7 References/Related Documentation [1]: RTX8430 Handset_Manual_Operations_v1.1 RTX8630 Handset_Manual_Operations_v1.2 RTX8830_Handset_Manual_Operations V1.3 [2]: How to Deploy SME VOIP System v1.4 [3]: Provisioning of SME VoIP System (10) 1.8 Document History Revision Author Issue Date Comments 2.2 KMR 24-July-2014 Document updated to include the new RTX8830 handset and features 2.1 KMR 2-April-2014 Document updated to match V316 software feature level in generic mode Server page: Added TLS, SRTP, server alias Updated other parts affected by server alias Repeater page: Added repeater alias References versions updated 2.0 KMR 1-Oct-2013 Document updated to match V306 software feature level in generic mode Home status: Base status added Extension page: Sort function added, Registration control added. Added unique extension note. Network: VLAN sync added Management: language moved to country Time: Added save button Country: Added language selection Security: Password double confirm added Central dir/LDAP: Reload option added Multicell: In status added Sync data IP Repeaters: Added stop registration Statistics: Added repeater statistics Section 6.3 multicell – modified sequence 1.9 KMR 17-July-2013 Document updated to match V303 software feature level (security, multiline, time settings). Primary Data Sync IP: Added note about data sync source. 1.8 KMR 18-Feb-2013 Restructured and updated to software V273 operation 2.3 KMR 8-Sep-2014 Updated to V322 operation with RTX8830 handset 2.4 KMR 5-Jan-2015 Aligned with V323B14 operation 2.5 KMR 16-Feb-2015 Aligned with V324 operation 2.6 KMR 12-Jan-2016 Added V355 system size capabilities 1.9 Documentation Feedback We always strive to produce the best and we also value your comments and suggestions about our documentation. If you have any comments about this guide, please enter them through the Feedback link on the RTX website. We will use your feedback to improve the documentation.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Introduction – System Overview 10 2 Introduction – System Overview In a typical telephony system, the network setup is the interconnection between Base-stations, “fat” routers, repeaters, portable parts, etc. The back-bone of the network depends on the deployment scenario but a ring or hub topology is used. The network has centralized monitoring, and maintenance system. The system is easy to scale up and supports from 1 to 50 bases in the same network. Further it is able to support up to 1000 registered handsets (RTX8630, RTX8830 and RTX8430). The Small and Medium Scale Enterprise (SME) VoIP system setup is illustrated below. Based on PoE interface each base station is easy to install without additional wires other than the LAN cable. The system supports the IP DECT CAT-IQ repeater RTX4024 with support up to 5 channels simultaneous call sessions. The following figure gives a graphical overview of the architecture of the SME VoIP System: 2.1 Hardware Setup SME network hardware setup can deployed as follows: Base-station(s) are connected via Layer 3 and/or VLAN Aware Router depending on the deployment requirements. The Layer 3 router implements the switching function. The base-stations are mounted on walls or lamp poles so that each base-station is separated from each other by up to 50m indoor1 (300m outdoor). Radio coverage can extended using repeaters that are installed with same distance to base-station(s). Repeaters are range extenders and cannot be used to solve local call capacity issues. In this case additional bases must be used. The base-station antenna mechanism is based on space diversity feature which improves coverage. The base-stations uses complete DECT MAC protocol layer and IP media stream audio encoding feature to provide up to 10 simultaneous calls. 1 Measured with European DECT radio and depends on local building layout and material
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Introduction – System Overview 11 2.2 Components of SME VoIP System RTX SME VoIP system is made up of (but not limited to) the following components: • At least one RTX Base Station is connected over an IP network and using DECT as air-core interface. • RTX IP DECT wireless Handset. • RTX SME VoIP Configuration Interface; is a management interface for SME VoIP Wireless Solution. It runs on all IP DECT Base stations. Each Base station has its own unique settings. 2.2.1 RTX Base Stations The Base Station converts IP protocol to DECT protocol and transmits the traffic to and from the end-nodes (i.e. wireless handsets) over a channel. It has 12 available channels. In a multi-cell setup, each base station has: • 8 channels have associated DSP resources for media streams. • The remaining 4 channels are reserved for control signalling between IP Base Stations and the SIP/DECT end nodes (or phones). Base Stations are grouped into clusters. Within each Cluster, Base Stations are synchronized to enable a seamless handover when a user moves from one base station coverage to another. For synchronization purposes, it is not necessary for Base Stations to communicate directly with each other in the system. E.g. a Base Station may only need to communicate with the next in the chain. It is advisable for a Base Station to identify more than one Base Station to guarantee synchronization in the situation that one of the Base Stations fails. The 4 control signalling channels are used to carry bearer signals that enable a handset to initiate a handover process. 2.2.2 SME VoIP Administration Server/Software This server is referred to as SME VoIP Configuration Interface. The SME VoIP Configuration Interface is a web based administration page used for configuration and programming of the base station and relevant network end-nodes. E.g. handsets can be registered or de-registered from the system using this interface. The configuration interface can be used as a setup tool for software or firmware download to base stations, repeaters and handsets. Further, it is used to check relevant system logs that can be useful to administrator. These logs can be used to troubleshoot the system when the system faces unforeseen operational issues. 2.2.3 RTX Wireless Handset The handset is a lightweight, ergonomically and portable unit compatible with Wideband Audio (G.722), DECT, GAP standard, CAT-iq audio compliant. The handset includes Colour display with graphical user interface. It can also provide the subscriber with most of the features available for a wired phone, in addition to its roaming and handover capabilities. Refer to the relevant handset manuals for full details handset features. 2.3 Wireless Bands The bands supported in the SME VoIP are summarized as follows: Frequency bands: 1880 – 1930 MHz (DECT) 1880 – 1900 MHz (10 carriers) Europe/ETSI 1910 – 1930 MHz (10 carriers) LATAM 1920 – 1930 MHz (5 carriers) US
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Introduction – System Overview 12 2.4 System Capacity (in Summary) SME network capacity of relevant components can be summarised as follows: Description Capacity Min ## of Bases Single Cell Setup 1 Max ## of Bases in Multi-cell Setup (configurable) 50/127/254 Single/Multi Cell Setup: Max ## of Repeaters 50 bases and 3 repeater per Base 127 bases and 1 repeater pr Base 254 bases and 0 repeaters Multi-cell Setup: Total Max ## of Repeaters 100 Max ## of Users (SIP registrations) per Base 30 Max ## of Users per SME VoIP System limited to 1000 Multi-cell Setup: Max ## of Synchronisation levels 24 Single Cell Setup: Max ## Simultaneous Calls 10 per Base station Multi-cell Setup: Max ## of Calls 8 per Base station Total Max ## Simultaneous Calls (Multi-cell Setup) Limited to 1000 Repeater: Max ## of Calls (Narrow band) 5 Repeater: Max ## of Calls (G722) 2 Quick Definitions Single Cell Setup: SME telephony network composed of one base station Multi-cell Setup: Telephony network that consists of more than one base station Synchronisation Level: Is the air core interface between two base stations. 2.5 Advantages of SME VoIP System They include (but not limited to): 1. Simplicity. Integrating functionalities leads to reduced maintenance and troubleshooting, and significant cost reductions. 2. Flexibility. Single network architecture can be employed and managed. Furthermore, the architecture is amenable to different deployment scenarios, including Isolated buildings for in-building coverage, location with co-located partners, and large to medium scale enterprises deployment for wide coverage. 3. Scalability. SME network architecture can easily be scaled to the required size depending on customer requirement. 4. Performance. The integration of different network functionalities leads to the collapse of the protocol stack in a single network element and thereby eliminates transmission delays between network elements and reduces the call setup time and packet fragmentation and aggregation delays.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Installation of Base Stations/Repeater 13 3 Installation of Base Stations/Repeater After planning the network, next is to determine the proper places or location the relevant base stations will be installed. Therefore, we briefly describe the how to install the base station in this chapter. 3.1 Package – Contents/Damage Inspection Before Package Is Opened: Examine the shipping package for evidence of physical damage or mishandling prior to opening. If there is a proof of mishandling prior to opening, you must report it to the relevant support centre of the regional representative or operator. Contents of Package: Make sure all relevant components are available in the package before proceeding to the next step. Every shipped base unit package/box contains the following items:  2 x mounting screws and 2 x Anchors  1 x Plastic stand  Base unit Damage Inspection: The following are the recommended procedure for you to use for inspection: 1. Examine all relevant components for damage. 2. Make a “defective on arrival – DOA” report or RMA to the operator. Do not move the shipping carton until it has been examined by the operator. If possible send pictures of the damage. The operator/regional representative will initiate the necessary procedure to process this RMA. They will guide the network administrator on how to return the damaged package if necessary. 3. If no damage is found then unwrap all the components and dispose of empty package/carton(s) in accordance with country specific environmental regulations.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Installation of Base Stations/Repeater 14 3.2 RTX Base station Mechanics The base station front end shows an LED indicator that signals different functional states of the base unit and occasionally of the overall network. The indicator is off when the base unit is not powered. The table below summarises the various LED states: LED State State Unlit No power in unit Unlit/Solid red Error condition Blinking green Initialisation Solid red Factory reset warning or long press in BS reset button Blinking red Factory setting in progress Solid green Ethernet connection available (Normal operation) Blinking red Ethernet connect not available OR handset de/registration failed Solid red Critical error (can only be identified by RTX Engineers). Symptoms include no system/SIP debug logs are logged, etc. Orange Press reset button of base station. Blinking orange No IP address received 3.3 RTX Base Unit – Reset feature It is possible to restart or reset the base station unit by pressing a knob at the rear side of the unit. Alternatively, it can be reset from the SME Configuration Interface. We do not recommend this; but unplugging and plugging the Ethernet cable back to the PoE port of the base station also resets the base unit.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Installation of Base Stations/Repeater 15 3.4 Installing the Base Station First determine the best location that will provide an optimal coverage taking account the construction of the building, architecture and choice of building materials. Next, mount the Base Station on a wall to cover range between 50 – 300 meters (i.e. 164 to 984 feet), depending whether it’s an indoor or outdoor installation. Please refer to chapter 10 for important information regarding network requirements, deployment considerations, site planning, cell coverage/capacity planning, environmental considerations and recommended Base station placement. 3.4.1 Mounting the Base Stations/Repeaters: We recommend the base station be mounted an angle other than vertical on both concrete/wood/plaster pillars and walls for optimal radio coverage. Avoid mounting the base units upside down as it significantly reduces radio coverage. Mount the base unit as high as possible to clear all nearby objects (e.g. office cubicles and cabinets, etc.). Occasionally extend coverage to remote offices/halls with lower telephony users by installing Repeaters. Make sure that when you fix the base stations with screws, the screws do not touch the PCB on the unit. Secondly, avoid all contacts with any high voltage lines.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Installation of Base Stations/Repeater 16 3.5 Find IP of Base Station To find IP of the installed base station two methods can be used; Using handset Find IP feature or browser IPDECT feature. 3.5.1 Using handset Find IP feature On the handset press “Menu” key followed by the keys: *47* to get the handset into find bases menu. The handset will now scan for 8663 bases. Depending on the amount of powered on bases with active radios and the distance to the base it can take up to minutes to find a base. - Use the cursor down/up to select the base MAC address for the base - The base IP address will be shown in the display The feature is also used for deployment. For further details refer to reference [2]. 3.5.2 Using browser IPDECT Open any standard browser and enter the address: http://ipdect<MAC-Address-Base-Station> for e.g. http://ipdect00087B00AA10. This will retrieve the HTTP Web Server page from the base station with hardware address 00087B00AA10. This feature requires an available DNS server.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Installation of Base Stations/Repeater 17 3.6 Login to Base SME Configuration Interface STEP 1 Connect the Base station to a private network via standard Ethernet cable (CAT-5). STEP 2 Use the IP find menu in the handset (Menu * 4 7 *) to determine the IP-address of the base station by matching the MAC address on the back of the base station with the MAC address list in the handset. STEP 3 On the Login page, enter your authenticating credentials (i.e. username and password). By default the username and password is admin. Click OK button. STEP 4 Once you have authenticated, the browser will display front end of the SME Configuration Interface. The front end will show relevant information of the base station.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Making Handset Ready 18 4 Making Handset Ready In this chapter we briefly describe how to prepare the handset for use, install, insert and charge new batteries. Please refer to an accompanying Handset User Guide for more information of the features available in the Handset. 4.1 Package – Contents/Damage Inspection Before Package Is Opened: Examine the shipping package for evidence of physical damage or mishandling prior to opening. If there is a proof of mishandling prior to opening, you must report it to the relevant support centre of the regional representative or operator. Contents of Package: Make sure all relevant components are available in the package before proceeding to the next step. Every shipped base unit package/box contains the following items:  2 x mounting screws and 2 x Anchors  1 x Handset hook  1 x A/C Adaptor  1 x Battery  1 x charger  1 x Handset Unit, 1 x Battery cover
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Making Handset Ready 19 Damage Inspection: The following are the recommended procedure for you to use for inspection: 1. Examine all relevant components for damage. 2. Make a “defective on arrival – DOA” report or RMA to the operator. Do not move the shipping carton until it has been examined by the operator. The operator/regional representative will initiate the necessary procedure to process this RMA. They will guide the network administrator on how to return the damaged package if necessary. 3. If no damage is found then unwrap all the components and dispose of empty package/carton(s) in accordance with country specific environmental regulations. 4.2 Before Using the Phone Here are the pre-cautions users should read before using the Handset: Installing the Battery 1. Never dispose battery in fires, otherwise it will explode. 2. Never replace the batteries in potentially explosive environments, e.g. close to inflammable liquids/ gases. 3. ONLY use approved batteries and chargers from the vendor or operator. 4. Do not disassemble, customise or short circuit the battery Using the Charger Each handset is charged through the use of a handset charger. The charger is a compact desktop unit designed to charge and automatically maintain the correct battery charge levels and voltage. The charger Handset is powered by AC supply from 110-240VAC that supplies 5.5VDC at 600mA. When charging the battery for the first time, it is necessary to leave the handset in the charger for at least 10 hours before the battery is fully charged and the handset ready for use. Handset in the Charger For correct charging, ensure that the room temperature is between 0°C and 25°C/32°F and 77°F. Do not place the handset in direct sunlight. The battery has a built-in heat sensor which will stop charging if the battery temperature is too high. If the handset is turned off when placed in charger, only the LED indicates the charging. When handset is turned off, the LED flashes at a low frequency while charging and lights constantly when the charging is finished. There will be response for incoming calls. If the handset is turned on when charging, the display shows the charging status. Open Back Cover 1. Press down the back cover and slide it towards the bottom of the handset. 2. Remove Back Cover from Handset
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Making Handset Ready 20 Handset Serial Number The serial number (IPEI/IPUI number) of each handset is found either on a label, which is placed behind the battery, or on the packaging label. First, lift off handset back cover and lift the battery and read the serial number. The serial number is needed to enable service to the handset. It must be programmed into the system database via the SME VoIP Configuration interface. Replace Battery Remove Back Cover from Handset. Remove the old battery and replace with a new one. 4.3 Using the Handset Please refer handset manual for detailed description of how to use the handset features [1].
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 21 5 SME VoIP Administration Interface The SME VoIP Administration Interface is also known as SME VoIP Configuration. It is the main interface through which the system is managed and debugged. The SME VoIP Configuration Interface is an in-built HTTP Web Server service residing in each base station. This interface is a user friendly interface and easy to handle even to a first time user. Note: Enabling secure web is not possible. For secure configuration use secure provisioning. This chapter seeks to define various variables/parameters available for configuration in the network. 5.1 Web navigation We describe the left menu in the front end of the SME VoIP Administration Interface. Feature Description Home/Status This is the front end of the Base station’s HTTP web interface. This page shows the summary of current operating condition and settings of the Base station and Handset(s). Extensions Administration of extensions and handsets in the system
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 22 Servers On this page the user can define which SIP/NAT server the network should connect to. Network Typically the user configures the Network settings from here. NAT provisioning: allows configuration of features for resolving of the NAT – Network Address Translation. These features enable interoperability with most types of routers. DHCP: allows changes in protocol for getting a dynamic IP address. Virtual LAN: specifies the Virtual LAN ID and the User priority. IP Mode: specifies using dynamic (DHCP) or static IP address for your SME network. IP address: if using DHCP leave it empty. Only write in, when you use static IP address. Subnet mask: if using DHCP, leave it empty. Only write in, when you use static IP address. DNS server: specify if using DHCP, leave it empty. Only write in the DNS server address of your Internet service provider, when you use static IP address. (DNS = Dynamic Name Server) Default gateway: if using DHCP, leave it empty. Write in the IP address of your router, when you use static IP address. Management Defines the Configuration server address, Management transfer protocol, sizes of logs/traces that should be catalogued in the system. Firmware Update Remote firmware updates (HTTP(s)/TFTP) settings of Base stations and handsets. Time Here the user can configure the Time server. It should be used as time server in relevant country for exact time. The time servers have to deliver the time to conform to the Network Time Protocol (NTP). Handsets are synchronised to this time. Base units synchronise to the master using the Time server. Country Specifying the country/territory where the SME network is located ensures that your phone connection functions properly. Note: The base language and country setting are independent of each other. Security The users can administrate certificates and create account credentials with which they can log in or log out of the embedded HTTP web server. Central Directory Interface to common directory load of up to 3000 entries using *csv format or configuration of LDAP directory. Note: LDAP and central directory cannot operate at the same time. Multi cell Specify to connect base station or chain of base stations to the network. Make sure the system ID for the relevant base stations are the same otherwise the multi-cell feature will not work. Repeaters Administration and configuration of repeaters of the system Alarm Administration and configuration of the alarm settings on the system. This controls the settings for alarms that can be sent to the handsets. This feature is only available on certain types of handsets. Statistics Overview of system and call statistics for a system. Configuration This shows detail and complete SME network settings for base station(s), HTTP/DNS/DHCP/TFTP server, SIP server, etc. Syslog Overall network related events or logs are displayed here (only live feed is shown). SIP Log SIP related logs can be retrieved from url link. It is also possible to clear logs from this feature.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 23 5.2 Home/Status We describe the parameters found in the Welcome front end home/status of the SME VoIP Administration Interface. Screenshot Parameter Description System information This base current multi-cell state Phone Type Always IPDECT System Type This base customer configuration RF Band This base RF band setting. The parameter is defined in production and relates to the radio approvals shown on the label of the base. Current local time This base local time Operation time Operation is operation time for the base since last reboot RFPI-Address This base RFPI address MAC-Address This base MAC address IP-Address This base IP address Firmware version This base firmware version Firmware URL Firmware update server address and firmware path on server Base Station Status “Idle” : When no calls on base “In use” : When active calls on base SIP identity status List of extensions present at this base station. Format: “extension”@“this base IP address”(“server name”) followed by status to the right. Below is listed possible status: OK: Handset is ok SIP Error: SIP registration error Reboot Reboot after all connections is stopped on base. Connections are active calls, directory access, firmware update active Forced Reboot Reboot immediately.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 24 5.3 Extensions In this section, we describe the different parameters available whenever the administrator is creating extensions for handsets. Note, it is not possible to add extensions if no servers are defined. As well the section describes the administration of extensions and handsets using the extension list and the extension list menu. Software supports customer configurations with and without the multiline feature. Section 5.3.1 describes “add extensions” without multiline and 5.3.2 with “multiline”. The system can handle maximum 1000 extensions matching 1000 handsets which can be divided between servers. When 1000 handsets are registered it is not possible to add more extensions. With active multiline feature the system can handle maximum 1000 extensions. With 4 active lines in multiline maximum 200 handsets can be active in the system. Note: Within servers or even with multi servers, extensions must always be unique. This means same extension number on server 1 cannot be re-used on server 2. 5.3.1 Add extension (no multiline) Screenshot Parameter Default Value(s) Description Extension Empty Handset phone number or SIP username depending on the setup. Possible value(s): 8-bit string length Example: 1024, etc. Note: The Extension must also be configured in SIP server in order for this feature to function. Authentication User Name Empty Username: SIP authentication username Permitted value(s): 8-bit string length
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 25 Authentication Password Empty Password: SIP authentication password. Permitted value(s): 8-bit string length Display Name Empty Human readable name used for the given extension Permitted value(s): 8-bit string length Mailbox Name Empty Name of centralised system used to store phone voice messages that can be retrieved by recipient at a later time. Valid Input(s): 8-bit string Latin characters for the Name Mailbox Number Empty Dialled mail box number by long key press on key 1. Valid Input(s): 0 – 9, *, # Note: Mailbox Number parameter is available only when it’s enabled from SIP server. Server Server 1 IP FQDN or IP address of SIP server. Drop down menu to select between the defined Servers of SME VoIP Service provider. Call waiting feature: Enabled Used to enable/disable Call Waiting feature. When disabled a second incoming call will be rejected. If enabled a second call will be presented as call waiting. Broadsoft Feature Event Package Disabled If enabled the given SIP extension subscribes for the Broadsoft Application Server Feature Event Package, and it becomes ready for reception of SIP NOTIFY with status on the following Broadsoft Server Services: -Do Not Disturb -Call Forwarding (Always, Busy, No answer) The received status will be displayed in the handset idle display. Reference section 5.3.3 Forwarding Unconditional Number Empty Number to which incoming calls must be re-routed to irrespective of the current state of the handset. Forwarding Unconditional must be enabled to function. Note: Feature must be enabled in the SIP server before it can function in the network Disabled Forwarding No Answer Number Empty Number to which incoming calls must be re-routed to when there is no response from the SIP end node. Forwarding No Answer Number must be enabled to function. Note: Feature must be enabled in the SIP server before it can function in the network Specify delay from call to forward in seconds. Disabled 90 Forwarding On Busy Number Empty Number to which incoming calls must be re-routed to when SIP node is busy. Forwarding On Busy Number must be enabled to function. Note: Feature must be enabled in the SIP server before it can function in the network Disabled
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 26 5.3.1.1 Extensions list (no multiline) The added extensions will be shown in the extension lists. The list can be sorted by any of the top headlines, by mouse click on the headline link. Parameter Description Idx Select / deselect for delete, register and deregister handsets Extension Given extension is displayed Display Name Given display name is displayed. If no name given this field will be empty Server Server IP or URL Server Alias Given server alias is displayed. If no alias given this field will be empty. IPEI Handset IPEI. IPEI is unique DECT identification number. State SIP registration state – if empty the handset is not SIP registered. FW info Firmware version of handset FWU Progress Possible FWU progress states: Off: Means sw version is specified to 0 = fwu is off Initializing: Means FWU is starting and progress is 0%. X% : FWU ongoing Verifying X%: FWU writing is done and now verifying before swap ”Waiting for charger” (HS) / ”Conn. term. wait” (Repeater): All FWU is complete and is now waiting for handset/repeater restart. Complete HS/repeater: FWU complete Error: Not able to fwu e.g. file not found, file not valid etc 5.3.1.2 Handset and extension list top/sub-menus The handset extension list menu is used to control paring or deletion of handset to the system (DECT registration/de-registrations) and to control SIP registration/de-registrations to the system. Above and below the list are found commands for making operations on handsets/and extensions. The top menu is general operations, and the sub menu is always operating on selected handsets/extensions. Screenshots In the below table each command is described.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 27 Actions Description Add extension Access to the “Add extension” sub menu Stop Registration Manually stop DECT registration mode of the system. This prevents any handset from registering to the system Delete Handset(s) Deregister selected handset(s), but do not delete the extension(s). Register Handset(s) Enable registration mode for the system making it possible to register at a specific extension (selected by checkbox) Deregister Handset(s) Deregister the selected handset(s) and delete the extension(s). Note: By power off the handset the handset will SIP deregister the PBX. 5.3.1.3 Edit Extension (no multiline) To edit extension use the mouse to click the link of the extension. Screenshot
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 28 5.3.2 Multiline: Add extension With active multiline feature the system distinguish between extensions, physical handsets and maximum 4 lines. To add a physical handset first an extension must be available. The “add extension” is available from the Extension web top. Screenshot By pressing the link the “add extension” menu will appear. In the following the parameters are explained. Screenshot Parameter Default Value(s) Description Line Name Empty Name of line shown to be used to show from which line the incoming call is coming and used when user must select from which line to make outgoing call. Handset New Handset The extension must be associated to a handset. By default a new handset can be configured, alternatively the user can select an already existing handset Idx.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 29 Extension Empty Handset phone number or SIP username depending on the setup. Possible value(s): 8-bit string length Example: 1024, etc. Note: The Extension must also be configured in SIP server in order for this feature to function. Authentication User Name Empty Username: SIP authentication username Permitted value(s): 8-bit string length Authentication Password Empty Password: SIP authentication password. Permitted value(s): 8-bit string length Display Name Empty Human readable name used for the given extension Permitted value(s): 8-bit string length Mailbox Name Empty Name of centralised system used to store phone voice messages that can be retrieved by recipient at a later time. Valid Input(s): 8-bit string Latin characters for the Name Mailbox Number Empty Dialled mail box number by long key press on key 1. Valid Input(s): 0 – 9, *, # Note: Mailbox Number parameter is available only when it’s enabled from SIP server. Server Server 1 IP DNS or IP address of SIP server. Drop down menu to select between the defined Servers of SME VoIP Service provider. Call waiting feature Enabled Used to enable/disable Call Waiting feature. When disabled a second incoming call will be rejected. If enabled a second call will be presented as call waiting. BroadWorks Shared Call Appearance Disabled If enabled the given SIP extension is considered part of a BroadWorks shared call appearance group (SCA). This enables the Shared Call Appearance Settings section on the Handset page for the handset that the extension is connected to. Note: BroadWorks SCAs and their respective extensions must be configured in the BroadWorks application server web interface prior to being added on this page. The extension entered here must match the extension configured in BroadWorks. Broadsoft Feature Event Package Disabled If enabled the given SIP extension subscribes for the Broadsoft Application Server Feature Event Package, and it becomes ready for reception of SIP NOTIFY with status on the following Broadsoft Server Services: -Do Not Disturb -Call Forwarding (Always, Busy, No answer) The received status will be displayed in the handset idle display. Forwarding Unconditional Number Empty Number to which incoming calls must be re-routed to irrespective of the current state of the handset. Forwarding Unconditional must be enabled to function. Note: Feature must be enabled in the SIP server before it can function in the network Forwarding No Answer Number Disabled Number to which incoming calls must be re-routed to when there is no response from the SIP end node. Forwarding No Answer Number must be enabled to function. Note: Feature must be enabled in the SIP server before it can function in the network Specify delay from call to forward in seconds. Empty
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 30 Forwarding On Busy Number Disabled Number to which incoming calls must be re-routed to when SIP node is busy. Forwarding On Busy Number must be enabled to function. Note: Feature must be enabled in the SIP server before it can function in the network 90 Empty The location selection feature, which is available in the add extension screen in non-multiline mode, is moved to edit handset from the handset and extension list. Edit handset screen is found by pressing the handset IPEI link. Screenshot Then maximum extensions supported per handset are 4. There are no restrictions for adding more, but only the first four will attempt to SIP register. 5.3.2.1 Multiline: Handset and extensions list Added handset and extensions will be shown in the extension list. The extension list is the access to the handset location control and the edit extension feature. The list can be sorted by any of the top headlines, by mouse click on the headline link. Screenshot Parameter Description Idx Index of handsets IPEI Handset IPEI. IPEI is unique DECT identification number. Handset State The state of the given handset: Present@RPNxx: The handset is DECT located at the base with RPNxx Detached: The handset is detached from the system (e.g. powered off) Located: The handset is configured to locate on a specific base, but is has not been possible to do so (e.g if the base is powered off) Removed: The handset has been out of sight for a specified amount of time (~one hour).
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 31 Handset Type FW Info Name of the handset type Firmware version of handset FWU Progress Possible FWU progress states: Off: Means sw version is specified to 0 = fwu is off Initializing: Means FWU is starting and progress is 0%. X% : FWU ongoing Verifying X%: FWU writing is done and now verifying before swap ”Waiting for charger” (HS) / ”Conn. term. wait” (Repeater): All FWU is complete and is now waiting for handset/repeater restart. Complete HS/repeater: FWU complete Error: Not able to fwu e.g. file not found, file not valid etc VoIP Idx Index of the configured SIP extensions. Select/deselect to start SIP registration or delete extension. Extension Given extension is displayed Display Name Given display name is displayed. If no name given this field will be empty Server Server IP or URL Server Alias Given server alias is displayed. If no alias given this field will be empty. State SIP registration state – if empty the handset is not SIP registered. 5.3.2.2 Multiline: Edit Extension To edit extension use the mouse to click the link of the extension. Basically the same options are available for edit extension as for add extension. Screenshot
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 32 5.3.2.3 Multiline: Edit handset Use the mouse to click the handset IPEI link to open the handset edit window. In the handset edit view the handset SIP location can be fixed to either any or a specific base. Screenshot
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 33 Parameter Default Value(s) Description Location ANY Specify a handset to be located at a specific base station or ANY base station. A location of a handset controls the DECT registration and the SIP registrations. Binding a handset to a specific base will bind the SIP registrations to this base. IPEI Handset IPEI Shows the handset IPEI. For an already registered handset changing the IPEI will deregister the handset at next handset location update. AC Handset AC code Shows the handset AC code. AC code is used at handset registration. Changing the AC code for an already registered handset will have no effect. Alarm Line No Alarm Line Selected The line of multilines to be used for alarm call feature Alarm Number Empty Number to be dialled in case of handset alarm key is pressed (Long keypress > 3 seconds on navigation center key ) Alarm Profiles Not configured Check the wanted alarm profiles for the particular handset. Shared Call Appearance Settings Not configured Each of the eight rows in the table represents an SCA status LED on the handset Idle screen. For each row it is possible to specify which shared line an LED should display the state of.  Only shared lines can be selected, that is, only extensions defined for the handset for which BroadWorks Shared Call Appearance is enabled are included in the selector.  A shared line can be reused for several LEDs. Each LED with the same shared line then corresponds to different appearance-indexes for that line (1 LED = appearance-index 1, 2 LEDs = appearance-indexes 1 and 2, and so on). It is not necessary to select a shared line for all of the LEDs. If an LED is not assigned a line, its position on the screen is simply empty. 5.3.2.4 Multiline: Handset and extension list top/sub-menus The handset extension list menu is used to control paring or deletion of handset to the system (DECT registration/de-registrations) and to control SIP registration/de-registrations to the system. Above and below the list are found commands for making operations on handsets/and extensions. The top menu is general operations, and the sub menu is always operating on selected handsets/extensions. Screenshots In the below table each command is described.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 34 Actions Description Add extension Access to the “Add extension” sub menu Stop Registration Manually stop DECT registration mode of the system. This prevents any handset from registering to the system Delete Handset(s) Deregister selected handset(s), but do not delete the extension(s). Register Handset(s) Enable registration mode for the system making it possible to register at a specific extension (selected by checkbox) Deregister Handset(s) Deregister the selected handset(s) and delete the extension(s). Start SIP Registration(s) Manually start SIP registration for selected handset(s). SIP Delete Extension(s) Deregister the selected handset(s) and delete the extension(s). After creation of extensions check the handset Idx and click “Register Handset(s)” to DECT register the handset to the base. First SIP registration is made by the system automatically by the handset DECT registration procedure. For new extensions click “Start SIP Registration(s)” to SIP register the extensions to the defined server. Screenshot Use the same procedure for other handsets, where the reference is the idx. no. when adding new extensions to existing handsets. 5.3.3 Broadsoft Feature Event Package If enabled the given SIP extension subscribes for the Broadsoft Application Server Feature Event Package, and it becomes ready for reception of SIP NOTIFY with status on the following Broadsoft Server Services: -Do Not Disturb -Call Forwarding (Always, Busy, No answer) The received status will be displayed in the handset idle display. After pressing save the extension screen will appear with removed configuration option for the forward feature as shown in the below picture. Note: Call forwarding can as well be configured from the handset by the user (for operation refer to the handset guide).
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 35 Screenshot 5.4 Servers In this section, we describe the different parameters available in the Servers configurations menu. Maximum 10 servers can be configured. Screenshot Parameter Default value Description Server Alias Empty Parameter for server alias NAT Adaption Disabled To ensure all SIP messages goes directly to the NAT gateway in the SIP aware router.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 36 If the system receives a SIP response to a REGISTER request with a “Via” header that includes the “received” parameter (ex: “Via: SIP/2.0/UDP 10.1.1.1:4540;received=68.44.20.1”), the base will adapt its contact information to the IP address from the “received” parameter. Thus, the base will issue another REGISTER request with the updated contact information. If NAT Adaption is disabled, the “received” parameter is ignored. Registrar Empty SIP Server proxy DNS or IP address Permitted value(s): AAA.BBB.CCC.DDD:<Port-Number> or <URL>:<Port-Number> Note: Specifying the Port Number is optional. Outbound Proxy Empty This is a Session Border Controller DNS or IP address (OR SIP server outbound proxy address) Set the Outbound proxy to the address and port of private NAT gateway so that SIP messages sent via the NAT gateway. Permitted value(s): AAA.BBB.CCC.DDD or <URL> or <URL>:<Port-Number> Examples: “192.168.0.1”, “192.168.0.1:5062”, “nat.company.com” and “sip:nat@company.com:5065”. If empty call is made via Registar. Conference Server Empty Broadsoft conference feature. Set the IP address of the conference server. In case an IP is specified pressing handset conference will establish a connection to the conference server. If the field is empty the original 3-party local conference on 8630 is used. Call Log Server Empty Broadsoft call log feature. Set the IP address of the XSI call log server. In case an IP is specified pressing handset will use the call log server. If the field is empty the local call log is used Re-registration time 600 The “expires” value 36nalyse36n in SIP REGISTER requests. This value indicates how long the current SIP registration is valid, and hence is specifies the maximum time between SIP registrations for the given SIP account. Permitted value(s): A value below 60 sec is not recommended, Maximum value 65636 SIP Session Timers: Disabled RFC 4028. A “keep-alive” mechanism for calls. The session timer value specifies the maximum time between “keep-alive” or more correctly session refresh signals. If no session refresh is received when the timer expires the call will be terminated. Default value is 1800 s according to the RFC. Min: 90 s. Max: 65636. If disabled session timers will not be used. Session Timer Values (s): 1800 Default value is 1800s according to the RFC. If disabled session timers will not be used. Permitted value(s): Minimum value 90, Maximum 65636
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 37 SIP Transport UDP Select UDP, TCP, TLS 1.0 Signal TCP Source Port Disabled When SIP Transport is set to TCP or TLS, a TCP (or TLS) connection will be established for each SIP extension. The source port of the connection will be chosen by the TCP stack, and hence the local SIP port parameter, specified within the SIP/RTP Settings (see 5.5.5) will not be used. The “Signal TCP Source Port” parameter specifies if the used source port shall be signaled explicitly in the SIP messages. Use One TCP/TLS Connection per SIP Extension: Disabled When using TCP or TLS as SIP transport, choose if a TCL/TLS connection shall be established for each SIP extension or if the base station shall establish one connection which all SIP extensions use. Please note that if TLS is used and SIP server requires client authentication (and requests a client certificate), this setting must be set to disabled. 0: Disabled. (Use one TCP/TLS connection for all SIP extensions) 1: Enabled. (Use one TCP/TLS connection per SIP extensions). RTP from own base station: Disabled If disabled RTP stream will be send from the base, where the handset is located. By enable the RTP stream will always be send from the base, where the SIP registration is made. This setting is typically enabled for operation with Cisco. Keep Alive Enabled This directive defines the window period (30 sec.) to keep opening the port of relevant NAT-aware router(s), etc. Show Extension on Handset Idle Screen Enabled If enabled extension will be shown on handset idle screen. Hold Behaviour RFC 3264 Specify the hold behaviour by handset hold feature. RFC 3264: Hold is signalled according to RFC 3264, i.e. the connection information part of the SDP contains the IP Address of the endpoint, and the direction attribute is sendonly, recvonly or inactive dependant of the context RFC 2543: The ”old” way of signalling HOLD. The connection information part of the SDP is set to 0.0.0.0, and the direction attribute is sendonly, recvonly or inactive dependant of the context Attended Transfer Behaviour Hold 2nd Call When we have two calls, and one call is on hold, it is possible to perform attended transfer. When the transfer soft key is pressed in this situation, we have traditionally also put the active call on hold before the SIP REFER request is sent. However, we have experienced that some PBXes do not expect that the 2nd call is put on hold, and therefore attended transfer fails on these PBXes. The "Attended Transfer Behavior" feature defines whether or not the 2nd call shall be put on hold before the REFER is sent.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 38 If "Hold 2nd Call" is selected, the 2nd call will be held before REFER is sent. If "Do Not Hold 2nd Call" is selected, the 2nd call will not be held before the REFER is sent Use Own Codec Priority Disabled Default disabled. By enable the system codec priority during incoming call is used instead of the calling party priority. E.g. If base has G722 as top codec and the calling party has Alaw on top and G722 further down the list, the G722 will be chosen as codec for the call. DTMF Signalling RFC 2833 Conversion of decimal digits (and ‘*’ and ‘#’) into sounds that share similar characteristics with voice to easily traverse networks designed for voice SIP INFO: Carries application level data along SIP signalling path (e.g.: Carries DTMF digits generated during SIP session OR sending of DTMF tones via data packets in the same internet layer as the Voice Stream, etc.). RFC 2833: DTMF handling for gateways, end systems and RTP trunks (e.g.: Sending DTMF tones via data packets in different internet layer as the voice stream) Both: Enables SIP INFO and RFC 2833 modes. DTMF Payload Type 101 This feature enables the user to specify a value for the DTMF payload type / telephone event (RFC2833). Remote Caller ID Source Priority FROM SIP information field used for Caller ID source: PAI - FROM FROM ALERT_INFO - PAI - FROM Codec Priority G.711U G.711A G.726 Defines the codec priority that base stations uses for audio compression and transmission. Possible Option(s): G.711U,G.711A, G.726, G.729, G.722. Note: Modifications of the codec list must be followed by a “reset codes” and “Reboot chain” on the multipage in order to change and update handsets. Note: With G.722 as first priority the number of simultaneous calls per base station will be reduced from 10 (8) to 4 calls. With G.722 in the list the codec negotiation algorithm is active causing the handset (phone) setup time to be slightly slower than if G.722 is removed from the list. With G.729 add on DSP module for the base is required. Contact RTX sales for purchase number 96101203. RTP Packet size 20ms The packet size offered as preferred RTP packet size by 8630 when RTP packet size negotiation. Selections available: 20ms, 40ms, 60ms, 80ms Secure RTP Disabled With enable RTP will be encrypted (AES-128) using the key negotiated via the SDP protocol at call setup. Secure RTP Auth Disabled With enable secure RTP is using authentication of the RTP packages. Note: with enabled SRTP authentication maximum 4
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 39 concurrent calls is possible per base in a single or multicell system. SRTP Crypto Suites AES_CM_128_HMAX_SHA1_32 AES_CM_128_HMAX_SHA1_80 Field list of supported SRTP Crypto Suites. The device is born with two suites. Note: Within servers or even with multi servers, extensions must always be unique. This means same extension number on server 1 cannot be re-used on server 2.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 40 5.5 Network In this section, we describe the different parameters available in the network configurations menu. 5.5.1 IP Settings Screenshot Parameter Default Values Description DHCP/Static IP DHCP If DHCP is enabled, the device automatically obtains TCP/IP parameters. Possible value(s): Static, DHCP DHCP: IP addresses are allocated automatically from a pool of leased address. Static IP: IP addresses are manually assigned by the network administrator. If the user chooses DHCP option, the other IP settings or options are not available. IP Address NA 32-bit IP address of device (e.g. base station). 64-bit IP address will be supported in the future. Permitted value(s): AAA.BBB.CCC.DDD Subnet Mask NA Is device subnet mask. Permitted value(s): AAA.BBB.CCC.DDD This is a 32-bit combination used to describe which portion an IP address refers to the subnet and which part refers to the host. A network mask helps users know which portion of the address identifies the network and which portion of the address identifies the node. Default Gateway NA Device’s default network router/gateway (32-bit). Permitted value(s): AAA.BBB.CCC.DDD e.g. 192.168.50.0 IP address of network router that acts as entrance to other network. This device provides a default route for TCP/IP hosts to use when communicating with other hosts on hosts networks. DNS (Primary) NA Main server to which a device directs Domain Name System (DNS) queries. Permitted value(s): AAA.BBB.CCC.DDD or <URL> This is the IP address of server that contains mappings of DNS domain names to various data, e.g. IP address, etc. The user needs to specify this option when static IP address option is chosen. DNS (Secondary) NA This is an alternate DNS server.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 41 5.5.2 VLAN Settings Enable users to define devices (e.g. Base station, etc.) with different physical connection to communicate as if they are connected on a single network segment. The VLAN settings can be used on a managed network with separate Virtual LANs (VLANs) for sending voice and data traffic. To work on these networks, the base stations can tag voice traffic it generates on a specific “voice VLAN” using the IEEE 802.1q specification. Screenshot Parameter Default Values Description VLAN id 0 Is a 12 bit identification of the 802.1Q VLAN. Permitted value(s): 0 to 4094 (only decimal values are accepted) A VLAN ID of 0 is used to identify priority frames and ID of 4095 (i.e. FFF) is reserved. Null means no VLAN tagging or No VLAN discovery through DHCP. VLAN User Priority 0 This is a 3 bit value that defines the user priority. Values are from 0 (best effort) to 7 (highest); 1 represents the lowest priority. These values can be used to prioritize different classes of traffic (voice, video, data, etc). Permitted value(s): 8 priority levels (i.e. 0 to 7) VLAN Synchronization Disabled Default disabled. By enabled the VLAN ID is automatic synchronised between the bases in the chain. Bases will be automatic rebooted during the synchronization. For further help on VLAN configuration refer to Appendix. 5.5.3 DHCP Options Screenshot Parameter Default Values Description Plug-n-Play Disabled Enabled: DHCP option 66 to automatically provide PBX IP address to base.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 42 5.5.4 NAT Settings We define some options available when NAT aware routers are enabled in the network. Screenshot Parameter Default Values Description Enable STUN Disabled Enable to use STUN STUN Server NA Permitted value(s): AAA.BBB.CCC.DDD (Currently only Ipv4 are supported) or url (e.g.: firmware.rtx.net). STUN Bindtime Determine Enabled STUN Bindtime Guard 80 Permitted values: Positive integer default is 90, unit is in seconds Enable RPORT Disabled Enable to use RPORT in SIP messages. Keep alive time 90 This defines the frequency of how keep-alive are sent to maintain NAT bindings. Permitted values: Positive integer default is 90, unit is in seconds 5.5.5 SIP/RTP Settings These are some definitions of SIP/RTP settings: Screenshot Parameter Default Values Description Use Different SIP Ports Disabled If disabled, the Local SIP port parameter specifies the source port used for SIP signalling in the system. If enabled, the Local SIP Port parameter specifies the source port used for first user agent (UA) instance. Succeeding UA’s will get succeeding ports.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 43 RTP Collision Detection Enabled Enable: If two sources with same SSRC, the following RTX is discarded. Disabled: No check – device will accept all sources. Local SIP port 5060 The source port used for SIP signalling Permitted values: Port number default 5060. SIP ToS/QoS 0x68 Priority of call control signalling traffic based on both IP Layers of Type of Service (ToS) byte. ToS is referred to as Quality of Service (QoS) in packet based networks. Permitted values: Positive integer, default is 0x68 RTP port 50004 The first RTP port to use for RTP audio streaming. Permitted values: Port number default 50004 (depending on the setup). RTP port range 40 The number of ports that can be used for RTP audio streaming. Permitted values: Positive integers, default is 40 RTP TOS/QoS 0xB8 Priority of RTP traffic based on the IP layer ToS (Type of Service) byte. ToS is referred to as Quality of Service (QoS) in packet based networks. See RFC 1349 for details. “cost bit” is not supported. o Bit 7..5 defines precedence. o Bit 4..2 defines Type of Service. o Bit 1..0 are ignored. Setting all three of bit 4..2 will be ignored. Permitted values: Positive integer, default is 0xB8
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 44 5.6 Management Settings Definitions The administrator can configure base stations to perform some specific functions such as configuration of file transfers, firmware up/downgrades, password management, and SIP/debug logs. Screenshot Parameter Default value Description Base Station Name: SME VoIP It indicates the title that appears at the top window of the browser and is used in the multicell page. Maximum characters: 35 Management Transfer Protocol TFTP The protocol assigned for configuration file and central directory Valid Input(s): TFTP, HTTP, HTTPs HTTP Empty The folder location or directory path that contains the configuration
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 45 Management upload script files of the Configuration server. The configuration upload script is a file located in e.g. TFTP server or Apache Server which is also the configuration server. Permitted value(s): /<configuration-file-directory> Example: /CfgUpload Note: Must begin with (/) slash character. Either / or \ can be used. HTTP Management password Empty Password that should be entered in order to have access to the configuration server. HTTP client credentials are formed by “User name” and the “HTTP Management password”. Permitted value(s): 8-bit string length Configuration server address Empty Server/device that provides configuration file to base station. Type: DNS or IP address Permitted value(s): AAA.BBB.CCC.DDD or <URL> Base Specific File Empty Base configuration file Multi Cell Specific File Empty The file name must be the chain id of the system. E.g 00087b0a00b3.cfg Permitted value(s): Format of file is chain ID.cfg Configuration File Download Disabled Base Specific file: Used when configuring a single cell base Multicell Specific File: Used when configuring a multicell based system Base and Multicell Specific File: Used on out of factory bases to specify VLAN and Multicell ID and settings. DHCP Controlled Config Server Disabled Provisioning server options. DHCP Option 66: Look for provision file by TFTP boot up server. DHCP Custom Option: Look for provision file by custom option DHCP Custom Option & Option 66: Look for provision file by first custom option and then option 66. DHCP Custom Option Empty By default option 160, but custom option can be defined. An option 160 URL defines the protocol and path information by using a fully qualified domain name for clients that can use DNS. DHCP Custom Option Type Empty URL: URL of server with path. Example of URL: http://myconfigs.com:5060/configs Default configuration file on server must follow the name: MAC.cfg IP Address: IP of server with path. Text Messaging Disabled Disable/enable messaging with Mobicall server The third option is to “Enable Without Server”. With this setting handset can send messages to other handsets, which support messaging. Note: Contact Mobicall to get the proper version and setup for Mobicall server Text Messaging & Alarm server Empty Permitted value(s): AAA.BBB.CCC.DDD or <URL> Text Messaging Port 1300 Port number of message server. Text Messaging Keep Alive 30 This defines the frequency of how keep-alive are sent Permitted values: Positive integer, unit is in minutes
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 46 (m) Text Messaging Response (s) 30 This defines the frequency of how response timeout Permitted values: Positive integer, unit is in seconds Text Messaging TTL 0 This defines the text messaging time to live Permitted values: Positive integer, unit is in seconds SIP Log Server Address Empty Permitted value(s): AAA.BBB.CCC.DDD or <URL> Requires a predefined folder named: \SIP Upload of SIP Log Disabled Enable this option to save low level SIP debug messages to the server. The SIP logs are saved in the file format: <MAC_Address><Time_Stamp>SIP.log Syslog Server IP-Address Empty Permitted value(s): AAA.BBB.CCC.DDD or <URL> Syslog Server Port Empty Port number of syslog server. Syslog Level Off Off: No data is saved on syslog server Normal Operation: Normal operation events are logged, incoming call, outgoing calls, handset registration, DECT location, and call lost due to busy, critical system errors, general system information. System Analyze: Handset roaming, handset firmware updates status. The system 46nalyse level also contains the messages from normal operation. Debug: Used by RTX for debug. Should not be enabled during normal operation. Enable Automatic Prefix Disabled Disabled: Feature off. Enabled: The base will add the leading digit defined in “Set Prefix for Outgoing Calls”. Enabled + fall through on * and #: Will enable detection of * or # at the first digit of a dialled number. In case of detection the base will not complete the dialled number with a leading 0. Examples: 1: dialed number on handset * 1234 - > dialed number to the pabx *1234 2: dialed number on handset #1234 - > dialed number to the pabx #1234 3: dialed number on handset 1234 - > dialed number to the pabx 01234 Set Maximum Digits of Internal Numbers 0 Used to detect internal numbers. In case of internal numbers no prefix number will be added to the dialled number. Set Prefix for Outgoing Calls Empty Prefix number for the enabled automatic prefix feature. Permitted value(s): 1 to 9999 There are three ways of configuring the system. 1. Manual configuration by use of the Web server in the base station(s) 2. By use of configuration files that are uploaded from a disk via the “Configuration” page on the Web server. 3. By use of configuration files which the base station(s) download(s) from a configuration server. For further details refer to doc reference [3].
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SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 48 5.7 Firmware Update Definitions In this page, the system administrator can configure how base stations and SIP nodes upgrade/downgrade to the relevant firmware. Handset firmware update status can be found in the extensions page and repeater firmware update status in the repeater page. Base firmware update status is found in the multicell page. Screenshot Parameter Default Value(s) Description Firmware update server address Empty IP address or DNS of firmware update files source Valid Inputs: AAA.BBB.CCC.DDD or <URL> Example: firmware.rtx.net or 10.10.104.41 Firmware path Empty Location of firmware on server (or firmware update server path where firmware update files are located). Example: /East_Fwu Note: Must begin with (/) slash character Required Version Type Empty Version of firmware to be upgraded (or downgraded) on handset type or repeater. Valid Input(s): 8-bit string length. E.g. 280 Note: Value version 0 will disable firmware upgrade for handsets and/or repeater Note: Two handset types will be serial firmware upgraded. First type 8630 then type 8430. Required Version base Empty Version of firmware to be upgraded (or downgraded) on Base station. Base units are referred to as gateways over here. Valid Input(s): 8-bit string length. E.g. 280
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 49 5.8 Time Server In this section, we describe the different parameters available in the Time Server menu. The Time server supplies the time used for data synchronisation in a multi-cell configuration. As such it is mandatory for a multi-cell configuration. The system will not work without a time server configured. As well the time server is used in the debug logs and for SIP traces information pages, and used to determine when to check for new configuration and firmware files. NOTE: It is not necessary to set the time server for standalone base stations (optional). Press the “Time PC” button to grab the current PC time and use in the time server fields. NOTE: When time server parameters are modified/changed synchronisation between base stations can take up to 15 minutes before all base stations are synchronised, depending on the number of base stations in the system. Changing time settings will require a reboot of system. Screenshot
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 50 Parameter Default Values Description Time Server Empty DNS name or IP address of NTP server. Enter the IP/DNS address of the server that distributes reference clock information to its clients including Base stations, Handsets, etc. Valid Input(s): AAA.BBB.CCC.DDD or URL (e.g. time.server.com) Currently only Ipv4 address (32-bit) nomenclature is supported. Allow broadcast NTP Checked By checked time server is used. Refresh time (h) Empty The window time in hours within which time server refreshes. Valid Inputs: positive integer Set timezone by country/region Checked By checked country setting is used (refer to country web page). Time Zone 0 Refers to local time in GMT or UTC format. Min: -12:00 Max: +13:00 Set DST by country/region Checked By checked country setting is used (refer to country web page). Daylight Saving Time (DST) Disabled The system administrator can Enable or Disable DST manually. Automatic: Enter the start and stop dates if you select Automatic. DST Fixed By Day Use Month and Date You determine when DST actually changes. Choose the relevant date or day of the week, etc. from the drop down menu. DST Start Month March Month that DST begins Valid Input(s): Gregorian months (e.g. January, February, etc.) DST Start Date 25 Numerical day of month DST comes to effect when DST is fixed to a specific date Valid Inputs: positive integer DST Start Time 3 DST start time in the day Valid Inputs: positive integer DST Start Day of Week Monday Day within the week DST begins DST Start Day of Week, Last in Month Last in Month Specify the week that DST will actually start. DST Stop Month October The month that DST actually stops. DST Stop Date 1 The numerical day of month that DST turns off. Valid Inputs: positive integer (1 to 12) DST Stop Time 2 The time of day DST stops Valid Inputs: positive integer (1 to 12) DST Stop Day of Week Sunday The day of week DST stops DST Stop Day of Week Last in Month First in Month The week within the month that DST will turn off.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 51 5.9 Country The country setting controls the in-band tones used by the system. To select web interface language go to the management page. Screenshot Parameter Default Values Description Select Country Germany Supported countries: Australia, Belgium, Brasil, Denmark, Germany, Spain, France, Ireland, Italia, Luxembourg, Nederland, New Zealand, Norway, Portugal, Swiss, Finland, Sweden, Tyrkey, United Kingdom, US/Canade, Austria State / Region NA Only shown by country selection US/Canada, Autralia, Brasil Select Language English Web interface language. Number of available languages: English, Dansk, Italiano, Tyrkie, Deutsch, Portuguese, Hrvatski, Srpski, Slovenian, Nederlands, Francaise, Espanol, Russian, Polski. Set timezone by country/region checked When checked timezone will follow country/region Set DST by country/region checked When checked DST will follow country/region Notes Empty Only showing notes to time setting for countries: US/Canada, Brasil NOTE: By checked timezone and DST the parameters in web page Time will be discarded. The following types of in-band tones are supported: - Dial tone - Busy tone - Ring Back tone - Call Waiting tone - Re-order tone
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 52 5.10 Security The security section is used for loading of certificates and for selecting if only trusted certificates are used. Furthermore, web password can be configured. The Security web is divided into three sections: Certificates (trusted), SIP Client Certificates (and keys) and Password administration. To setup secure fwu and configuration file download select HTTPs for the Management Transfer Protocol (refer to management web). SIP and RTP security is server dependent and in order to configure user must use the web option Servers (refer to servers web). 5.10.1 Certificates The certificates list contains the list of loaded certificates for the system. Using the left column check mark it is possible to check and delete certificates. To import a new certificate use the mouse “select file” and browse to the selected file. When file is selected, use the “Load” bottom to load the certificate. The certificate format supported is DER encoded binary X.509 (.cer). Screenshot Certificates list Parameter Default Values Description Idx Fixed indexes Index number Issued To Empty IP address – which is part of the certificate file Issued To Empty Organisation, Company – which is part of the certificate file Valid Until Empty Date Time Year – which is part of the certificate file Screenshot By enabling Use Only Trusted Certificates, the certificates the base will receive from the server must be valid and loaded into the system. If no valid matching certificate is found during the TLS connection establishment, the connection will fail. When Use Only Trusted Certificates is disabled, all certificates received from the server will be accepted. sNote: It is important to use correct date and time of the system when using trusted certificates. In case of time/date not defined the certificate validation can fail.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 53 5.10.2 SIP Client Certificates To be able to establish a TLS connection in scenarios, where the server requests a client certificate, a certificate/key pair must be loaded into the base. This is currently supported only for SIP. To load a client certificate/key pair, both files must be selected at the same time, and it is done by pressing “select files” under “Import SIP Client Certificate and Key Pair” and then select the certificate file as well as the key file at the same time. Afterwards, press load. The certificate must be provided as a DER encoded binary X.509 (.cer) file, and the key must be provided as a binary PKCS#8 file. Note: Use Chrome for loading SIP Client Certificates Screenshot 5.10.3 Password In the below the password parameters are defined. Screenshot Parameter Default Values Description Username Admin Can be modified to any supported character and number Maximum characters: 15 Current Password Admin Can be modified to any supported character and number New Password Empty Change to new password Maximum characters: 15 Confirm Password Empty Confirm password to reduce accidently wrong changes of passwords Password valid special signs: @/|<>-_:.!?*+# Password valid numbers: 0-9 Password valid letters: a-z and A-Z
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 54 5.11 Central Directory and LDAP The SME VOIP system support two types of central directories, a local central directory or LDAP directory. For both directories caller id look up is made with match for 6 digits of the phone number. 5.11.1 Local Central Directory Select local and save for local central directory. Screenshot Parameter Default Values Description Local Local Drop down menu to select between local central directory and LDAP based central directory Server Empty The parameter is used if directory file is located on server. Valid Inputs: AAA.BBB.CCC.DDD or <URL> Refer to appendix for further details. Filename Empty The parameter is used if directory file is located on server. Refer to appendix for further details Phonebook reload interval (s) 0 The parameter is controlling the reload interface of phonebook in seconds. The feature is for automatic reload the base phonebook file from the server with intervals. It is recommended to specify a conservative value to avoid overload of the base station. With default value setting 0 the reload feature is disabled. 5.11.1.1 Import Central Directory The import central directory feature is using a browse file approach. After file selection press the load button to load the file. The system support only the original *.csv format. Please note that some excel csv
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 55 formats are not the original csv format. The central directory feature can handle up to 3000 contacts. For further details of the central directory feature refer to appendix. 5.11.2 LDAP Select LDAP Server and save for LDAP server configuration. Screenshot Screenshot Parameter Default Values Description LDAP Server LDAP Server Drop down menu to select between local central directory and LDAP based central directory. LDAP Server is displayed when LDAP server is selected. Server Empty IP address of the LDAP server. Valid Inputs: AAA.BBB.CCC.DDD or <URL>
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 56 Port Empty The server port number that is open for LDAP connections. Sbase Empty Search Base. The criteria depends on the configuration of the LDAP server. Example of the setting is CN=Users, DC=umber, DC=loc LDAP filter Empty LDAP Filter is used to as a search filter, e.g. setting LDAP filter to (|(givenName=%*)(sn=%*)) the IP-DECT will use this filter when requesting entries from the LDAP server. % will be replaced with the entered prefix e.g searching on J will give the filter (|(givenName=J*)(sn=J*)) resulting in a search for given name starting with a J or surname starting with J. Bind Empty Bind is the username that will be used when the IP-DECT phone connects to the server Password Empty Password is the password for the LDAP Server Virtuel Lists Disabled By enable, virtual list searching is possible Name Empty The name can be used to specify if sn+givenName or cn (common name) is return in the LDAP search results Work Number Empty Work number is used to specify that LDAP attribute that will be mapped to the handset work number Home Number Empty Home number is used to specify that LDAP attribute that will be mapped to the handset home number Mobile Number Empty Mobile number is used to specify that LDAP attribute that will be mapped to the handset mobile number 5.11.3 Characters supported The below table shows which characters are supported in the communication between RTX8663 and handset.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 57 5.12 Multi-cell Parameter Definitions In this section, we describe the different parameters available in the Multi-cell configurations menu. 5.12.1 Settings for Base Unit Description of Settings for Specific Base units is as follows: Screenshot Multicell status covers status of data synchronization. The status “Keep-alive” means normal operation. Parameter Default values Description Multi cell system Disabled Enable this option to allow the Base unit to be set in multi-cell mode (can be set either as master or slave in the multi-cell chain system – refer to MAC-units in Chain section for details). Valid Inputs: Enable, Disable Must “save and reboot” after change from disabled to enable. System chain ID Empty This is an identifier (in string format e.g. 2275) that is unique for a specific multi-cell system. The Chain ID value MUST not be equal to a used SIP account. The Chain ID use up a SIP account with this value. Note: Chain ID is used as SIP account for check Sync. Default value is 512, which means extension 512 must not be used – unless the chain ID is modified. Chain ID can be modified by provisioning only. Note: There can be several multi-cell systems in SME network. Up to 24 levels of base stations chains are permitted in a setup. Valid Input: The Web site allow max 5 digits in this field. Synchronization time (s) 60 sec This specifies the period in seconds when elements/nodes (e.g. Base units) in a specific Multi-cell will synchronise to each other.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 58 If no keep-alive packets are received within a period of 2*NETWORK_SYNC_TIME, the base will be indicated as lost in the multi cell configuration. The parameter is also used with “Auto create multi primary” feature. Data Sync: Multicast To select between multicast or Peer to Peer data synchronisation mode. The multicast port range and IP addresses used is calculated from the chain id. The multicast feature uses the port range: 49200 – 49999 The multicast feature IP range: 224.1.0.0 – 225.1.0.0 Multicast uses UDP. Primary Data Sync IP Empty IP of base station data sync source – the base handling the data synchronisation. Using multicast this base IP is selected automatically. The data sync feature uses the port range: 49200 – 49999 NOTE: Using Peer to Peer mode the IP of the base used for data sync. source MUST be defined. NOTE: Using Peer to Peer mode with version below V306 limits the system automatic recovery feature – as there is no automatic recovery of the data sync. source in Peer to Peer mode. Multi cell debug None Enable this feature, if you want the system to catalogue low level multi-cell debug information or traces. Options: Data Sync: Writes header information for all packets received and sent to be used to debug any special issues. Generates LOTS of SysLog signaling and is only recommended to enable shortly when debugging. Auto Tree: Writes states and data related to the Auto Tree Configuration feature. Both: Both Data Sync and Auto Tree are enabled. NOTE: Must only be used for debug purpose and not enabled on a normal running system 5.12.2 DECT System Settings Description of DECT Settings for Specific Base units is as follows: Screenshot Parameter Default values Description DECT system RFPI Not able This is a radio network identity accessed by all Base units in a specific multi-cell system. It composed of 5 octets. It is actually 5 different variables combined together. RFPI Format: XX XX XX XX XX (where XX are HEX values)
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 59 Allow multi primary: Disabled This feature is used for multi-location setups. Allows two or more primary in the same system. The two cells will be unsynchronized and handover will not be possible. “Auto Configure DECT sync source tree” must be enabled for this feature to also be enabled Auto create multi primary: Disabled By enabled the system can generate cells in case a base goes into faulty mode. Two cells will only be generated in case no radio connection between the two cells is present. In order to recover the full system after establish of the faulty base, the system must be rebooted. Allow multi primary must be enabled for this feature to also be enabled. Auto configure DECT sync source tree Enabled Enable this to allow the system to automatically synchronise the multi-cell chain/tree. NOTE: Must be enabled in order to allow a new primary recover in case the original primary goes into faulty mode. Note: To run with a system with two separate primary in two locations “Allow multi primary” and “Auto configure DECT sync source tree” must be enabled. To add the second primary the slave must manually be configured as primary. Alternatively the “Auto create multi primary” must be enabled. 5.12.3 Base System Settings Description of SIP Settings for Specific Base units is as follows: Screenshot Parameter Default Values Description Number of SIP accounts before distributed load 8 The maximum number of handsets or SIP end nodes that are permitted to perform location registration on a specific Base unit before load is distributed to other base units. The parameter can be used to optimize the handset distribution among visible basestations. Note: A maximum of 8 simultaneous calls can be routed through each Base unit in a multi-cell setup. Permitted Input: Positive Integers (e.g. 6) SIP Server support for multiple registrations per account Disabled Disable this option so it is possible to use same extension (i.e. SIP Account) on multiple phones (SIP end nodes). These phones will ring simultaneously for all incoming calls. When a phone (from a SIP account group) initiates a handover from Base X to Base Y, this phone will de-register from Base X, and register to Base Y after a call. Permitted Input: Disabled: No SIP de-registration will be made when a handset
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 60 roams to another base station Enabled: The old SIP registration will be deleted with a SIP Deregistration, when a handset roams to another base station System combination (Number of base stations/Repeaters per base station): 50/3 Select between basic base configurations. 50/3 : 50 bases and 3 repeaters 127/1 : 127 bases and 1 repeater 254/0 : 254 bases and 0 repeater The configuration cannot be modified after a system is established. The configuration must be set during first multicell configuration.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 61 5.12.4 Base Station Group The Base station group list various parameter settings for base stations including chain level information. Screenshot: Parameters Description ID Base unit identity in the chained network. Permitted Output: Positive Integers RPN The Radio Fixed Part Number is an 8-bit DECT cell identity allocated by the installer. The allocated RPN within the SME must be geographically unique. Permitted Output: 0 to 255 (DEC) OR 0x00 to 0xFF (HEX) Version Base station current firmware version. Permitted Output: positive Integers with dot (e.g. 273.1) MAC Address Contains the hardware Ethernet MAC address of the base station. It varies from Base station to Base stations. IP Status Current Base station behaviour in the SME network. Possible Outputs Connected: The relevant Base station(s) is online in the network Connection Loss: Base station unexpectedly lost connection to network This Unit: Current Base station whose http Web Interface is currently being accessed DECT Sync source With setting “Auto configure DECT sync source tree” set to Enable, this three will automatically be generated. If manual configured the administrator should choose the relevant “multi cell chain” level its wants a specific Base unit be placed. Maximum number of “multi-cell chain” levels is 12. Format of the selection: “AAAAAxx: RPNyy (-zz dBm)” AAAAA: indication of sync. source for the base. Can be “Primary” or “Level xx” xx: Sync. source base sync. level yy: Sync. source base RPN zz: RSSI level of sync. source base seen from the actual base “(Any) RPN”: When a base is not synchronized to other base. State after reboot of chain. Dect Property Base station characteristics in connection to the current multi cell network.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 62 Possible Output(s) Primary: Main Base station unto which all other nodes in the chain synchronises to. Locked: The Base unit is currently synchronized and locked to the master Base unit. Searching: Base unit in the process of locating to a Master/slave as specified in Dect sync source Free Running: A locked Base unit that suddenly lost synchronisation to the Master. Unknown: No current connection information from specific Base unit Assisted lock: Base has lost DECT sync. source and Ethernet is used for synchronization Sync. Lost: Handset has an active DECT connection with the base. But the base has lost DECT sync. source connection. The base will stay working as long as the call is active and will go into searching mode when call is stopped. Base Station Name Name from management settings. 5.12.5 DECT Chain Below the Base Group Table is the DECT Chain tree. The DECT Chain tree is a graphical presentation of the Base Group table levels and connections. Repeaters are shown with green highlight. Screenshot: DECT Chain tree of above configuration Screenshot: Example of part of DECT Chain tree with repeaters
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 63 Screenshot: Example of part of DECT Chain tree with units in Base Group but not in tree by various reasons. When a base or repeater has not joined the tree it will be shown with read background below the tree.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 64 5.13 Repeaters Within this section we describe the repeater parameter, and how to operate the repeater. 5.13.1 Add repeater From repeaters web select “Add Repeater” Screenshot Then select “DECT Sync mode” Screenshot Parameters Description Name Repeater name. If no name specified the field will be empty DECT sync mode Manually: User controlled by manually assign “Repeater RPN” and “DECT sync source RPN” Local Automatical: Repeater controlled by auto detects best base signal and auto assign RPN.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 65 5.13.1.1 Manually User controlled by manually assign “Repeater RPN” and “DECT sync source RPN”. The parameters are selected from the drop down menu. Screenshot Parameters Description Idx System counter RPN SINGLE CELL SYSTEM: The base has always RPN00, first repeater will then be RPN01, second repeater RPN02 and third RPN03 (3 repeaters maximum per base) MULTI CELL SYSTEM: Bases are increment by 2^2 in hex, means first base RPN00 second base RPN04 etc., in between RPN01, 02, 03 addressed for repeaters at Primary base and 05, 06, 07 addressed for Secondary base (3 repeaters maximum per base) DECT sync source Select the base or repeater the repeater has to be synchronized to.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 66 5.13.1.2 Local Automatical Repeater controlled by auto detects best base signal and auto assign RPN. The RPN and DECT sync source are greyed out. The repeater RPN is dynamic assigned in base RPN range. With local automatical mode repeater on repeater (chain) is not supported. 5.13.2 Register Repeater Adding a repeater makes it possible to register the repeater. Registration is made by select the repeater and pressing register repeater. The base window for repeater registration will be open until the registration is stopped. By stopping the registration all registration on the system will be stopped inclusive handset registration. 5.13.3 Repeaters list
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 67 Parameters Description IDx Repeater unit identity in the chained network. Permitted Output: Positive Integers RPN The Radio Fixed Part Number is an 8-bit DECT cell identity allocated by the installer. The allocated RPN within the SME must be geographically unique. Permitted Output: 0 to 255 (DEC) OR 0x00 to 0xFF (HEX) Name/IPEI Contains the name and the unique DECT serial number of the repeater. If name is given the field will be empty. DECT sync Source The “multi cell chain” connection to the specific Base/repeater unit. Maximum number of chain levels is 12. Sync. source format: “RPNyy (-zz dBm)” yy: RPN of source zz: RSSI level seen from the actual repeater DECT sync Mode Manually: User controlled by manually assign “Repeater RPN” and “DECT sync source RPN” Local Automatical: Repeater controlled by auto detects best base signal and auto assign RPN. Chaining Automatical: Base controlled by auto detects best base or repeater signal and auto assign RPN. This feature will be supported in a future version State Present@unit means connected to unit with RPN yy FW info Firmware version FWU Progress Possible FWU progress states: Off: Means sw version is specified to 0 = fwu is off Initializing: Means FWU is starting and progress is 0%. X% : FWU ongoing Verifying X%: FWU writing is done and now verifying before swap ”Conn. term. wait” (Repeater): All FWU is complete and is now waiting for connections to stop before repeater restart. Complete HS/repeater: FWU complete Error: Not able to fwu e.g. file not found, file not valid etc
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 68 5.14 Alarm In the Alarm Settings menu, it is controlled how an alarm appears on the handset. For example if the handset detects “Man Down”, then it is defined in this menu what alarm signal this type of alarm will send out and if a pre-alarm shall be signaled etc. All configuration of the handset Alarm Settings is done from the base station. The concept is that on the “Alarm” page on the web server, eight different alarm profiles can be configured. Afterwards for each handset, it can be selected which of the configured alarm profiles, the given handset shall subscribe to. When this is done the selected alarm profiles are sent to the handset. See section 5.3.2.3 Multiline: Edit handset The parameters that can be configured are: Parameters Description Idx Indicates the index number of a specific alarm. Profile Alias An alias or user-friendly name to help identify the different profiles when selecting which profiles to enable for the individual handsets. Alarm Type The type of alarm is dependent of what kind of event that has triggered the alarm on the handset. The handset supports either of the following categories: Man Down No Movement Running Pull Cord Emergency Button Disabled Alarm Signal The way the alarm is signalled as it received on the handset. Message: A text message to an alarm server. Call: An outgoing call to the specified emergency number. Stop Alarm from Handset Enable/disable the possibility to stop/cancel the alarm from the handset. Trigger Delay The period from when the alarm has fired until the handset shows a pre-alarm warning. If set to 0, there will be no pre-alarm warning, and the alarm will be signalled immediately. The alarm algorithm typically needs about 6 sec. to detect e.g. man down etc. Stop Pre-Alarm from Handset Enable/disable the possibility to stop/cancel the pre-alarm from the handset. Pre-Alarm Delay The period from the pre-alarm warning is shown until the actual alarm is
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 69 signalled. The maximum value is 255. Howling Enable/disable if howling shall be started in the handset, when the alarm is signalled. If disabled, only the configured signal is sent (call or message). This alarm feature is only available on some types of handsets (e.g. the RTX8830) After configuration, the handset must be rebooted. 5.14.1 Use of Emergency Alarms As described above, it can be configured if it shall be possible to stop an alarm from the handset. If the possibility to stop an alarm from the handset is disabled, it is ensured that an alarm is not stopped before someone at e.g. an emergency center has received the alarm and reacted upon it. The behavior of a handset when an alarm “is sent” depends on the configured Alarm Signal:  Call: When the Alarm Signal is configured as “Call”, the handset will make a call to the specified emergency number, and the alarm is considered stopped when the call is terminated. If it is not allowed to stop the alarm from the handset, it will not be possible to terminate the call from handset, and the alarm will be considered as stopped only when the remote end (e.g. the emergency center) terminates the call.  Message: When the Alarm Signal is configured as “Message”, the handset will send an alarm message to the specified alarm server, and enable auto answer mode. If Howling is enabled, the handset will also start the Howling tone. The alarm will not stop until a call is made, and since auto answer mode is enabled, the emergency center can make the call, and the person with the handset does not have to do anything to answer the call, it will answer automatically. Again, the alarm is considered stopped, when the call is terminated with the same restrictions as for the Call alarm signal. All type of alarms have the same priority. This means that once an alarm is active, it cannot be overruled by another alarm until the alarm has been stopped. However, if the alarm is not yet active, i.e. if it is in “pre-alarm” state and an alarm configured with no pre-alarm is fired, then the new alarm will become active and stop the pending alarm. Alarms with no pre-alarm are considered important, and there is no possibility to cancel them before they are sent, and therefore alarms with no pre-alarm, are given higher priority than alarms in pre-alarm state. The Emergency Button could be an example of an alarm which would be configured without pre-alarm. Thus, when the Emergency Button is pressed you want to be sure the alarm is sent. However, If another alarm was already in pre-alarm state, it could potentially be cancelled, and if the Emergency Button alarm was ignored in this case, no alarm would be sent. This is the reason why alarms with no pre-alarm, are given higher priority than alarms in pre-alarm state.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 70 5.15 Statistics The statistic feature is divided into four administrative web pages, which can be access from any base. 1. System 2. Calls 3. Repeater 4. DECT data All four views have an embedded export function, which export all data to comma separated file. By pressing the clear button all data in the full system is cleared. 5.15.1 System data The system data web is access by http://ip/SystemStatistics.html and data is organised in a table as shown in below example. Screenshot The table is organised with headline row, data pr. base rows and with last row containing the sum of all base parameters. Parameters Description Base Station Name Base IP address and base station name from management settings Operation/Duration D-H:M:S Operation is operation time for the base since last reboot. Duration is the operation time for the base since last reset of statistics, or firmware upgrade. Busy Busy Count is the number of times the base has been busy. Busy Duration D-H:M:S Busy duration is the total time a base has been busy for speech (8 or more calls active). SIP Failed Failed SIP registrations count the number of times a SIP registration has failed Handset Removed Handset removed count is the number of times a handset has been marked as removed Searching Base searching is the number of times a base has been searching for it’s sync
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 71 source Free Running Base free running is the number of times a base has been free running DECT Source Changed Number of time a base has changed sync source 5.15.2 Call data The call data web is access by http://ip/CallStatistics.html and data are organised in a table as shown in below example. Screenshot The table is organised with headline row, data pr. base rows and with last row containing the sum of all base parameters. Parameters Description Base Station Name Base IP address and base station name from management settings Operation time/Duration Total operation time for the base since last reboot or reset Duration is the time from data was cleared or system has been firmware upgraded. Count Counts number of calls on a base. Dropped Dropped calls are the number of active calls that was dropped. E.g. if a user has an active call and walks out of range, the calls will be counted as a dropped call. An entry is stored in the syslog when a call is dropped. No response No response calls is the number of calls that have no response, e.g. if a external user tries to make a call to a handset that is out of range the call is counted as no response. An entry is stored in the syslog when a call is no response. Duration Call duration is total time that calls are active on the base. Active Active call shows how many active calls that are active on the base (Not active DECT calls, but active calls). On one base there can be up to 30 active calls. Max Active Maximum active calls are the maximum number of calls that has been active at the same time. Codecs Logging and count of used codec types on each call. Handover Success Counts the number of successful handovers. Handover Failed Counts the number of failed handovers.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 72 5.15.3 Repeater data The table is organised with headline row, data pr. base rows and with last row containing the sum of all base parameters. Parameters Description Idx/Name Base IP address and base station name from management settings Operation D-H:M:S Total operation time for the repeater since last reboot or reset Duration is the time from data was cleared or system has been firmware upgraded. Busy Busy Count is the number of times the repeater has been busy. Busy Duration D-H:M:S Busy duration is the total time a repeater has been busy for speech (5 or more calls active). Max Active Maximum active calls are the maximum number of calls that has been active at the same time. Searching Repeater searching is the number of times a repeater has been searching for it’s sync source Recovery In case the sync source is not present anymore the repeater will go into lock on another base or repeater and show recovery mode DECT Source Changed Number of time a repeater has changed sync source Wide Band Number of wideband calls on repeaters Narrow Band Number of narrow band calls on repeaters
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 73 5.15.4 DECT data The DECT data web is access by http://ip/DectStatistics.html and data is organised in a table as shown in below example. Screenshot Please note 3 frequencies are manually removed in the example system.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 74 5.16 Settings – Configuration File Setup This page provides non editable information showing the native format of entire SME VoIP Configuration parameter settings. The settings format is exactly what is used in the configuration file. The configuration file is found in the TFTP server. The filename for the configuration server is <MAC_Address>.cfg. The configuration file is saved in the folder /Config in the TFTP sever. There are three ways to edit the configuration file or make changes to the settings page: 1) Using the SME VoIP Configuration interface to make changes. Each page of the HTTP web interface is a template for which the user can customise settings in the configuration file. 2) Retrieving the relevant configuration file from the TFTP and modify and enter new changes. This should be done with an expert network administrator. 3) Navigate to the settings page of the VoIP SME Configuration interface > copy the contents of settings > save them to any standard text editor e.g. notepad > modify the relevant contents, make sure you keep the formatting intact > Save the file as <Enter_MAC_Address_of_RFP>.cfg > upload it into the relevant TFTP server. For details refer to [3]. An example of contents of settings is as follows: ~RELEASE=UMBER_FP_V0054 %GMT_TIME_ZONE%:16 %COUNTRY_VARIANT_ID%:18 %FWU_POLLING_ENABLE%:0 %FWU_POLLING_MODE%:0 %FWU_POLLING_PERIOD%:86400 %FWU_POLLING_TIME_HH%:3 %FWU_POLLING_TIME_MM%:0 %DST_ENABLE%:2 %DST_FIXED_DAY_ENABLE%:0 %DST_START_MONTH%:3 %DST_START_DATE%:1 .... .... 5.17 Sys log This page shows live feed of system level messages of the current base station. The messages the administrator see here depends on what is configured at the Management settings. The Debug logs can show only Boot Log or Everything that is all system logs including boot logs. The Debug log is saved in the file format <Time_Stamp>b.log in a relevant location in the TFTP server as specified in the upload script. A sample of debug logs is as follows: 0101000013 [N](01):DHCP Enabled 0101000013 [N](01):IP Address: 192.168.10.101 0101000013 [N](01):Gateway Address: 192.168.10.254 0101000013 [N](01):Subnet Mask: 255.255.255.0 0101000013 [N](01):TFTP boot server not set by DHCP. Using Static. 0101000013 [N](01):DHCP Discover completed 0101000013 [N](01):Time Server: 192.168.10.11 0101000013 [N](01):Boot server: 10.10.104.63 path: Config/ Type: TFTP 0101000013 [N](01):RemCfg: Download request of Config/00087b077cd9.cfg from 10.10.104.63 using TFTP
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: SME VoIP Administration Interface 75 0101000014 [N](01):accept called from task 7 0101000014 [N](01):TrelAccept success [4]. Listening on port 10010 0101000019 [N](01):RemCfg: Download request of Config/00087b077cd9.cfg from 10.10.104.63 using TFTP 0101000019 [W](01):Load of Config/00087b077cd9.cfg from 10.10.104.63 failed To dump the log simply copy and page the full contents. 5.18 SIP Logs This page shows SIP server related messages that are logged during the operation of the SME system. The full native format of SIP logs is saved in the TFTP server as <MAC_Address><Time_Stamp>SIP.log These logs are saved in 2 blocks of 17Kbytes. When a specific SIP log is fully dumped to one block, the next SIP logs are dumped to the other blocks. An example of SIP logs is shown below: ..... Sent to udp:192.168.10.10:5080 at 12/11/2010 11:56:42 (791 bytes) REGISTER sip:192.168.10.10:5080 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.101:5063;branch=z9hG4bKrlga4nkuhimpnj4.qx Max-Forwards: 70 From: <sip:Ext003@192.168.10.10:5080>;tag=3o5l314 To: <sip:Ext003@192.168.10.10:5080> Call-ID: p9st.zzrfff66.ah8 CSeq: 6562 REGISTER Contact: <sip:Ext003@192.168.10.101:5063> Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, INFO, PRACK Expires: 120 User-Agent: Generic-DPV-001-A-XX(Generic_SIPEXT2MLUA_v1) Content-Type: application/X-Generic_SIPEXT2MLv1 Content-Length: 251 ..... To dump the log simply copy and page the full contents.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Multi-cell Setup & Management 76 6 Multi-cell Setup & Management This chapter seeks to describe how to install, add and synchronize one or multiple base stations to the network. There are two main procedures involved: 1) Proper placement of the base stations (which is called network dimensioning). The present chapter does not address this issue. Refer to Chapter 12 for details. 2) Creating and adding base station profiles to the network via the SME Configuration Tool (to form a multi-cell system). This chapter describes the second procedure. 6.1 Adding Base stations Here are the recommended steps to add Base stations to network: STEP 1 Connect the Base station to a private network via standard Ethernet cable (CAT-5). STEP 2 Use one of the two methods to determine the base station IP address. a. Use the IP find menu in the handset (Menu * 4 7 *) to determine the IP address of the base station by matching the MAC address on the back of the base station with the MAC address list in the handset. b. Use the IPdect feature STEP 3 Open browser on the computer and type in the IP address of the base. Press “Enter” to access the base Login to base station. STEP 4 Once you have authenticated, the browser will display front end of the SME Configuration Interface. The front end will show relevant information of the base station.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Multi-cell Setup & Management 77 6.1.1 Time Server Setup STEP 5 Navigate to the Time settings and configure it. Scroll on the left column and click on Time url link to Open the Time Settings Page. Use the PC time feature or enter the relevant parameters on this page and press the Save and Reboot button. Make sure there is contact to the “Time server” otherwise the Multi-cell feature will not work. You can verify whether the Time server is reachable af rebooting the base station by verifying the correct Time Server IP address is still in place. 6.1.2 SIP Server (or PBX Server) Setup STEP 6 Create the relevant SIP server (or PBX Server) information in the system. Each service provider/customer should refer SIP server vendor on how to setup SIP servers. Click the link “Server” at the left hand column of home page, you can add your SIP server for base station use. Next, from the Server page, click on the Add Server URL and enter the relevant SIP server information (an example is shown below). Choose “Disabled” on NAT adaption parameter if NAT function of the SIP aware router is not enabled. Enter the relevant parameters based on the description in the table below. Select Save button.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Multi-cell Setup & Management 78 6.1.3 Multi-cell Setup STEP 7 Click on Multi Cell url link in the SME VoIP Configuration to view the current Multi cell settings status of the current base station. Brand new base stations have Multi cell system feature disabled by default.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Multi-cell Setup & Management 79 STEP 8 Next, the system administrator needs to create and Enable Multi Settings profile for the current base station. On the Multi Cell settings Page, choose Enable option from the drop down menu of the Multi cell system parameter. Enable the Multi cell debug option if the system administrator wants some Multi-cell related logs to be catalogued by the system. STEP 9 On the same Multi Cell Settings page > Enter the relevant values for System chain ID and Synchronization time (s) respectively. The System chain ID is a geographically unique DECT cell identity allocated to bridge several base stations together in a chain. An example is 55555. The Synchronization time (s) parameter is defined as window/period of time in seconds a specific base station synchronises to the master base station unit (by default 60). Note: Do NOT use a chain ID similar to an extension. Click on Save button to keep modified changes of multi cell settings into the base station.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Multi-cell Setup & Management 80 The Multi Cell data synchronization ONLY works when the relevant Time Server is set in the system before Server/Subscriber profile is added or created. Refer to STEP 5. IMPORTANT: Base stations must be rebooted after the time server has been set. STEP 10 Repeat STEP 1-9 as explained above for each base stations. IMPORTANT: It takes up to 5 minutes (synchronization time) to add a new base station to a Multi Cell System. 6.2 Synchronizing the Base stations STEP 11 On each SME VoIP Configuration interface for the base station(s) navigate to the Home/Status page and Click the Reboot button. This will trigger Are you sure you want to reboot base station? window. Click OK button on this window. A successful restart of the base stations will lead to a display of the page: Gateway has been reset.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Multi-cell Setup & Management 81 STEP 12 Navigate back to the Multi cell settings page by clicking Multi-cell url link at the left column. The revised Multi cell settings page shows the relevant base stations synchronized together. By default, the system uses the first registered base station as the master base unit. STEP 13 On the Multi-cell settings page, scroll to the DECT system settings and Enable or Disable the “Auto configure DECT sync option source tree” (See description in the table below). The DECT system RFPI parameter is computed by the system (It’s often greyed in a multi-cell system configuration). STEP 14 Next, on the MAC-units in chains section, you can manually configure the synchronisation source tree of the multi-cell system. Multi-cell settings page, scroll to the DECT system settings and Enable or Disable the “Auto configure DECT sync option source tree” (See description in the table below). The DECT system RFPI parameter is computed by the system (Its often greyed in a multi-cell system)
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Multi-cell Setup & Management 82 6.3 Summary of Procedure – Creating a Chain We enumerate the short version of how to add 3 base stations units in a multi-cell setup. This can be applied for up to 40 number of base units. This procedure is divided into four (4) main stages. Apply this procedure if all base unit are straight from production. 6.3.1 Stage 1 Skip this stage if relevant base stations are already in the network. a) Add 3 base stations i.e. RFP1, RFP2, RFP3 > Disable the “Multi cell system” option and “Save” b) RFP1, RFP2, RFP3: Reboot from the HTTP SME Configuration Main Page c) RFP1, RFP2, RFP3: Default by pressing reset button 12-sec. 6.3.2 Stage 2 Choosing 1st base unit i.e. RFP1 as Primary a) RFP1: Define Time server and “Save and reboot” from the Time page b) RFP1: Reboot automatically c) RFP1: Press “Add server” and define SIP server IP and “Save” from the Servers page d) RFP1: On the extension page add one extension (no handset needs to be registered). This step is important for allow secondary base to join e) RFP1: Multi cell system = enable and “Save” from the Multi-cell page f) RFP1: Reboot (Verify from Debug log “SYNCMGR: This base is ready to be Primary in a Chain”) 6.3.3 Stage 3 Choose another base unit,RFP2 as Secondary a) RFP2: Select chain ID same as RFP1. b) RFP2: Multi cell system = enable and “Save” c) RFP2: Reboot (Verify from Debug log “SYNCMGR: This base is ready to join into another Chain”) d) RFP1, RFP2: Wait 2min for stable Primary-Secondary chain (check for the message: SYNCMGR: Socket#10 creation success) 6.3.4 Stage 4 Choose the 3rd base unit, RFP3 as Secondary e) RFP3: Multi cell system = enable and “Save” f) RFP3: Reboot (Verify Debug log “SYNCMGR: This base is ready to join into another Chain”) g) RFP1, RFP3: Wait 2min for stable Master-Slave chain (SYNCMGR: Socket#10 creation success) h) RFP3: Check mark ID2/RPN08 and select dropdown “1 – RPN: 04” and “Save” i) RFP3: Reboot (SYNCMGR: Socket#8 creation success) Multi-cell chain of 3 base stations has been created successfully. Next step involves adding extensions to the system.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Multi-cell Setup & Management 83 6.4 Practical Configuration of Multi-cell System In this chapter we describe what exactly to configure in the SME VoIP Configuration Interface ensure these scenarios really work. 6.4.1 Case ##1: Isolated Buildings The optimal configuration for isolated buildings is standalone base stations setting. In this setting, you must: A. Using the figure below as illustration, log into the Configuration Interface of each base station. B. Configure the Time Server, SIP Server, Extensions as described in the previous chapters. C. On the main page of the configuration interface, click Network URL > disable the Multi-cell parameter of each base station > Save and Reboot each base to complete the Case ##1 setup. Disable Multi Cell option of Base Stations
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Multi-cell Setup & Management 84 6.4.2 Case ##2: Location with co-located partners Example includes Department shops, Retail location with co-located photo kiosk or pharmacy, etc. To illustrate this setup, two slave base stations are synchronised to one master base in the two office blocks. Here is diagram to illustrate Case ##2. The procedure: STEP 1 Follow the steps described in section 0 STEP 2 On the Network page of each base define the DECT sync source settings as illustrated in the table below. STEP 3 Save and reboot each base to complete case ##2 setup Multi Cell Page of Base Stations Recommended settings of MAC-units in Chain section of page (Other different settings exist): RPN Ver MAC Addr IP Addr IP Status DECT sync source DECT Property 00 XX XX:XX:XX… XXX.XXX… Connected 1: RPN:04 04 XX XX:XX:XX… XXX.XXX… Connected 1: RPN:04 Primary 08 XX XX:XX:XX… XXX.XXX… Connected 1: RPN:04
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Multi-cell Setup & Management 85 6.4.3 Case ##3: Large to Medium Sized Enterprises In this scenario, we have five slave bases synchronised to one master base. The master base is located in office block B while the slave bases are spread across the whole enterprise. No base station is deployed in the lift because it has high attenuation properties that will drastically reduce radio signals. The procedure: STEP 1 Follow the steps described in sections 0 STEP 2 On the Network page of each base define the DECT sync source settings as illustrated in the table below. STEP 3 Save and reboot each base to complete case ## 3 setup Multi Cell Page of Base Stations Recommended settings of MAC-units in Chain section of page (Other valid setting exists): NOTE: The number of chains cannot exceed 24 levels. RPN Ver MAC Addr IP Addr IP Status DECT sync source DECT Property 00 XX XX:XX:XX… XXX.XXX… Connected 1: RPN:04 04 XX XX:XX:XX… XXX.XXX… Connected 2: RPN:08 08 XX XX:XX:XX… XXX.XXX… Connected 3: RPN:0C 0C XX XX:XX:XX… XXX.XXX… Connected 4: RPN:10 10 XX XX:XX:XX… XXX.XXX… Connected 4: RPN:10 Primary 14 XX XX:XX:XX… XXX.XXX… Connected 4: RPN:10
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Multi-cell Setup & Management 86 6.4.4 Case ##4: Large Enterprises at Different Locations In this scenario, multi-cell systems are deployed at different locations; geographically separated from each other. Each location has a master base station with more than one slave base synchronise to it. The procedure: STEP 1 Follow the steps described in sections 6 STEP 2 On the Network page of each base define the DECT sync source settings as illustrated in the table below. STEP 3 Save and reboot each base to complete case ## 4 setup STEP 4 Important for this configuration is to enable “Allow multi primary” on the multi cell page. This allows the handset to locate on both systems make it possible for the user move from one location to the other and still use the same handset. Multi Cell Page of Base Stations Recommended settings of MAC-units in Chain section of page (Other valid setting exists): RPN Ver MAC Addr IP Addr IP Status DECT sync source DECT Property 00 XX XX:XX:XX… XXX.XXX… Connected 0: RPN:00 Primary for HQ 04 XX XX:XX:XX… XXX.XXX… Connected 0: RPN:00 08 XX XX:XX:XX… XXX.XXX… Connected 1: RPN:04
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Multi-cell Setup & Management 87 0C XX XX:XX:XX… XXX.XXX… Connected 3: RPN:0C Primary for Offices 10 XX XX:XX:XX… XXX.XXX… Connected 3: RPN:0C 14 XX XX:XX:XX… XXX.XXX… Connected 4: RPN:10 18 XX XX:XX:XX… XXX.XXX… Connected 6: RPN:18 Primary for Enterprises 1C XX XX:XX:XX… XXX.XXX… Connected 6: RPN:18 20 XX XX:XX:XX… XXX.XXX… Connected 7: RPN:1C
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Registration Management - Handset 88 7 Registration Management - Handset In this chapter we briefly describe how to register handsets in the SME VoIP Network. A precondition for handset registration is a proper configured single or multi-base system. For this refer to chapter 6.1. 7.1 Register handset to base (non multiline) This section describes how to register the wireless handset to the base station. NOTE: Minimum one server must be registered to the base (system), otherwise a handset cannot be registered to the system. Please see chapter 6.1.2. STEP 1 Login to a base station. STEP 2 Select “Extensions” URL and click “Add extension” link STEP 3 Fill out the form and click “Save”. In the example below we add the extension “3020” and this SIP account got the same number as “Authentication User Name” and “Password”. The “Server 1” is selected by default as server for this extension. STEP 4 In the extensions list set a Check mark on the extension which shall be assigned to the handset you want to register and click “Register handset (s)”. The base is now open (ready state) for handset registrations for 5 minutes.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Registration Management - Handset 89 STEP 5 Start the registration procedure on the handset by following step “a” to “d” below. a) Select main menu “Connectivity” b) Select menu ”Register” c) Type in the “AC code” and press “OK” to start the registration. The default AC code is “0000”. d) After a while the handset is registered, and the idle display is shown. NOTE: The unique handset IPEI is displayed on sheet “Extensions” when the handset is successfully registered. The web page must be manually updated by pressing “F5” to see that the handset is registered; otherwise the handset IPEI (International Portable Equipment Identity) isn’t displayed on the web page. We illustrate how extensions page will be when you register several handsets.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Registration Management - Handset 90 7.2 Register handset to base (multiline) This section describes how to register the wireless handset to a system with active multiline feature. NOTE: Minimum one server must be registered to the base (system), otherwise a handset cannot be registered to the system. Please see chapter 6.1.2. STEP 1 Login to a base station. STEP 2 Select “Extensions” URL and click “Add extension” link STEP 3 Fill out the form and click “Save”. In the example below we add the extension “3020” and this SIP account got the same number as “Authentication User Name”, “Password” and “Display Name”. STEP 4 In the handset and extensions list set a Check mark on the handset Idx, which you want to register and click “Register handset (s)”. The base is now open (ready state) for handset registrations for 5 minutes. STEP 5 Start the registration procedure on the handset by following step “a” to “d” below.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Registration Management - Handset 91 a) Select main menu “Connectivity” b) Select menu ”Register” c) Type in the “AC code” and press “OK” to start the registration. The default AC code is “0000”. d) After a while the handset is registered, and the idle display is shown STEP 6 Confirm the registration from the unique handset IPEI is displayed in column “IPEI” when the handset is successfully registered. Note: The web page must be manually updated by pressing “F5” to see that the handset is registered; otherwise the handset IPEI (International Portable Equipment Identity) isn’t displayed on the web page. STEP 7 SIP registration for first extension is made automatically. For the second extension start the SIP registration procedure on the base by Check the extension and click “Start SIP Registration(s)”.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Registration Management - Handset 92 STEP 8 Confirm the SIP registration by SIP State in right column. Note: The web page must be manually updated by pressing “F5” to see that the handset is SIP registered; otherwise the handset SIP state isn’t displayed on the web page.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Registration Management - Handset 93 7.3 Register handset to base and specific extension (multiline) This section gives an example of controlling handset registration to a specific extension. STEP 1 Login to a base station. STEP 2 Select “Extensions” URL and click “Add extension” link STEP 3 Fill out the form and click “Save”. In the example below we add the extension “2512” with display name “Reserved” STEP 4 Extension 2512 is now listed with handset IPEI “FFFFFFFFF”. Click the IPEI “FFFFFFFFFF” link and get the below view. STEP 5 Enter the IPEI of handset which must register to this particular extension and press “Save”
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Registration Management - Handset 94 STEP 6 Check mark the handset Idx in left column and press “Register Handset(s)” STEP 7 With the handset run the handset registration procedure using AC code “0000” STEP 8 Confirm the registration success by “Handset State” column. Note: It is possible with similar procedure to register using a different AC code than 0000 simply by changing the AC code in step 5. Note: It is possible to SIP register a handset with extensions on different servers by using the edit extension link and pair the extension to a handset Idx.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Firmware Upgrade Procedure 95 8 Firmware Upgrade Procedure This step-by-step chapter describes how to upgrade or downgrade base station(s) and/or handset(s) / repeater (s) to the relevant firmware provided by RTX. 8.1 Network Dimensioning In principle, a number of hardware and software components should be available or be satisfied before base station/handset update can be possible. The minimum hardware and software components that are required to be able update via TFTP include the following (but not limited to):  Handsets  Base stations  TFTP Server (Several Windows and Linux applications are available)  DHCP Server (Several Windows and Linux applications are available)  Workstation (e.g. Normal terminal or PC)  Any standard browser (e.g. Firefox)  Public/Private Network 8.2 TFTP Configuration This section illustrate TFTP Server configuration using “SolarWinds” vendor TFTP Server. Create the following relevant folders as shown in the snap shots and choose defaults settings for the remaining options and save.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Firmware Upgrade Procedure 96 NOTE: If TFTP server timeout settings are too short firmware upgrade might not complete. Recommended time out setting is more than 3 seconds. 8.3 Create Firmware Directories The admin from the service provider’s side must create the relevant firmware directory in the server where both old and new firmware(s) can be placed in it. (See the STEP above)
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Firmware Upgrade Procedure 97 8.3.1 Base: On the TFTP server root, create directory “Beatus”. Copy Base station firmware to the named directory. IMPORTANT: The BeatUs directory name cannot be changed. 8.3.2 Handsets/Repeaters: On the TFTP server root, create directory “Pegasus” or “Raffle” or “Razor” or “DECT4024” depending on type. Copy handset/repeater firmware to the named directory. IMPORTANT: The Raffle, Pegasus, DECT4024 directory names cannot be changed.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Firmware Upgrade Procedure 98 8.4 Handset Firmware Update Settings Scroll down and Click on Firmware Update url link in the SME VoIP Configuration Interface to view the Firmware Update Settings page. Type IP address and firmware path followed by save. For Http download the firmware update server settings must be entered as follows: 8.5 Handset(s) and Repeater Firmware Upgrade On the Firmware Update Settings page enter the relevant handset/repeater firmware for each type (e.g. 273 for V273) to upgrade or downgrade > press Save button to initialize the process of updating all handsets.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Firmware Upgrade Procedure 99 NOTE: To disable handset/repeater firmware process type version 0 in the required version field, followed by the save button. It is recommended to use version 0 after all units are upgraded. NOTE: For handset TFTP/HTTP download only one handset type can be downloaded at the same time. In case two handset models are defined for fwu at the same time fwu will fail. 8.5.1 Monitor handset firmware upgrade Handset firmware upgrade status is monitored on the handset extensions page, right column. Handset firmware update time from start to complete takes minimum 40 minutes. 8.5.2 Monitor Repeater firmware upgrade Repeater firmware upgrade status is monitored on the Repeater page, right column. Repeater firmware upgrade time from start to complete takes minimum 20 minutes.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Firmware Upgrade Procedure 100 8.5.3 Verification of Firmware Upgrade The firmware upgrade is confirmed by the FWU Progress status in the second and first right column on the handset extension list or repeater list. The “FWU info” column contains the software version and the “FWU Progress” column contains the status. In case status is “Complete”, the unit is firmware upgraded. Alternatively the handset firmware can be verified from the Handset Menu by navigate to Settings > Scroll down to Status this will list information regarding Base station and Handset firmware versions. 8.6 Base Station(s) Firmware Upgrade On the Firmware Update Settings page > scroll down to the Update Gateways section > Enter the relevant firmware version of the base station to upgrade or to downgrade. Enter 202 for base version V0202. For multi cell upgrade remember to check “update all Base Stations”. Efter entering required version choose Start update button > select OK button from the dialog window to start the update/downgrade procedure. The relevant base station(s) will automatically reboot and retrieve the firmware specified from the server and update itself accordingly. The base firmware update behaviour is: Base will fetch the fwu file for approximately 3 minutes, then reboot and start flashing the LED - indicated by LED fast flashing for approximately 3 minutes and reboots in new version. Note: All on-going voice calls are dropped from the base station(s) immediately the firmware update procedure starts.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Firmware Upgrade Procedure 101 8.6.1 Base firmware confirmation Base station firmware version status in a multicell environment can be seen in the multicell base station group overview page, column 4. 8.6.2 Verification of Firmware Upgrade Syslog information when Management Syslog level is set to “Debug” [ FWU Downloading File tftp://10.1.24.101/FwuPath/Beatus/BeatusSw_4181_v0202.fwu] [ Base FWU started] [ Base FWU ended with exit code 2101 (NE_FILE_TRANSFER_EOF): End of file] The log window of the TFTP server:
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Functionality Overview 102 9 Functionality Overview So far we have setup our SME VoIP system. Next, in this chapter we list what features and functionalities are available in the system. The SME VOIP system supports all traditional and advanced features of most telephony networks. In addition, 3rd party components handle features like voice mail, call forward, conference calls, etc. A brief description of SME VOIP network functionalities are:  Outgoing/incoming voice call management: The SME VOIP system can provide multiple priority user classes. Further, up to 3 repeaters can be linked to a Base-station.  Internal handover: User locations are reported to SIP Server in order to provide differentiated services and tariff management. Within a DECT traffic area, established calls can seamlessly be handover between Base-stations using connection handover procedures.  Security: The RTX SME VOIP system also supports robust security functionalities for Base-stations. Most security2 functionality is intrinsically woven into the SME VOIP network structure so that network connections can be encrypted and terminal authentication can be performed. 9.1 Gateway Interface Connector interfaces Power Connector: Ethernet PoE (Ethernet adaptor for normal power) IEEE 802.3: Power class 2 (3.84 – 6.49W) LAN Interface Standard : 10BASE-T(IEEE 802.3 100Mbps) Connector: RJ45 8/8 Internet Protocol:  IPv4  IPv6 Keys 1 x Reset key LED indicator One Status LED (multicolor, red, green, orange) RF Frequency Bands 1880 – 1900 MHz (EMEA) 1910 – 1930 MHz (Latam) 1920 – 1930 MHz (USA) This is software settings and to be set when it is packed in factory. Output Power <250 mW (for USA < 140mW) Antenna Two antennas for diversity Software update Downloadable Remote firmware update HTTPS/TFTP 9.2 Detail Feature List CODECs G.711 PCM A-law & U-law Uncompressed voice Silence suppression ( No) G.722 Allows HD sound for the handset G.726 ADPCM, 32 Kbps G.729 A G.729.1 (ehem. G.729 EV) 2 With active security with authentication 4 channels is supported
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Functionality Overview 103 Note: Only with additional module, this is an extra option that requires a board connector mounted in Gateway. Per default not mounted. SIP RFC2327 SDP: Session Description Protocol RFC2396 Uniform Resource Identifiers (URI): Generic Syntax RFC2833 In-Band DTMF/Out of band DTMF support RFC2976 The SIP INFO method RFC3261 SIP 2.0 RFC3262 Reliability of Provisional Responses in the Session Initiation Protocol (PRACK) RFC3263 Locating SIP Servers (DNS SRV, redundant server support) RFC3264 Offer/Answer Model with SDP RFC3265 Specific Event Notification RFC3311 The Session Initiation Protocol UPDATE Method RFC3325 P-Asserted Identity RFC3326 The Reason Header Field for the Session Initiation Protocol (SIP) RFC3489 STUN RFC3515 REFER: Call Transfer RFC3550 RTP: A Transport Protocol for Real-Time Application RFC3581 Rport RFC3842 Message Waiting Indication RFC3891 Replace header support RFC3892 The Session Initiation Protocol (SIP) Referred-By Mechanism RFC3960 Early Media and Ringing Tone Generation in the Session Initiation Protocol (SIP) RFC4475 Session Initiation Protocol (SIP) Torture Test Messages SIPS SRTP Will limit number of active calls pr. base when enabled. Web server Embedded web server HTTP Other features Quality of service Type of Service (ToS) including DiffServ Tagging, and QoS per IEEE 802.1p/q IP Quality Warning – Network outage, VoIP service outage Adaptive Jitter Buffer support Automatic DST Tone Scheme Country Depend Tone Scheme Ethernet features Speed duplex 10 & 100 duplex VLAN VLAN (802.1p/q) DHCP Support Static IP TLS 1.0 For secure connections (AES 128) TFTP For configuration download. HTTP For configuration download. HTTPS For secure configuration download. TCP/IP/UDP SNTP For internet clock synchronization Quality of service Type of Service (ToS) including DiffServ Tagging, and QoS per IEEE 802.1p/q DHCP option 66 DNS srv DECT DECT CAP Connectionless handover, enhanced location registration CAT-IQ v1.0 Wideband Speech General Telephony Handset Support 10 simultaneous handsets supported (single cell) (10 call / single cell and 8 call/Multi cell) Total 1000 simultaneous call supported / system
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Functionality Overview 104 VoIP Accounts 30 VoIP accounts per base – (maximum 254 bases per installation) Total 1000 VoIP accounts / system Maximum 1000 handsets per installation Simultaneous Calls 4 Wideband calls (g.722) or 10 single cell, 8 multi cell narrowband calls (PCMA, PCMU, G.726) or mixed wideband and narrowband. Call features Codec Negotiation Codec Switching Missed call notification Voice message waiting notification Date and Time synchronization Parallel calls Common parallel call procedures Call transfer unannounced Call transfer announced Conference Call Waiting Calling line identity restriction Outgoing call Call Toggle Incoming call Line identification Multiple Lines Multiple calls Call identification Calling Name Identification Presentation (CNIP) Calling Line Identification Presentation (CLIP) Call Hold List of registered handsets Call log 50 mixed between Incoming, outgoing, missed calls Phone Book Common Phonebook with up to 3000 entries (Import via csv format) Common Phonebook LDAP V2.0 Local Phonebook (100 entries 8630 and 50 entries 8430) DND Do Not Disturb Call Forward All No Answer Busy Individual Speed dial Programmable Function keys
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Appendix A: Basic Network Server(s) Configuration 105 Appendix 10 Appendix A: Basic Network Server(s) Configuration In this chapter we describe how to setup the various server elements in the system. 10.1 Server setup In the SME network, the server environment is installed as a centralized system. The main server types hosted on the network include SIP, DNS/DHCP and HTTP/TFTP Servers. These servers can be hosted both in one or multiple windows and/or Linux Server environment. Management servers are normally installed to monitor and manage the network in detail. Each Base-station status can be checked. Each Repeater and each Subscriber Terminal can be monitored over the air from a centralized location. Further, new software can be uploaded to all system elements from the centralized location (typically a TFTP server) on an individual basis. This includes Subscriber Handsets where the latest software is downloaded over the air. 10.2 Requirements Regardless of whether or not you will be installing a centrally provisioned system, you must perform basic TCP/IP network setup, such as IP address and subnet mask configuration, to get your organization’s phones up and running. 10.3 DNS Server Installation/Setup Name server is a name server service installed in a server for mapping or resolution of humanly memorable domain names and hostnames into the corresponding numeric Internet Protocol (IP) addresses. The customer should refer to the platform vendor either windows or Linux vendor for detail step-by-step guide on how to install and configure Domain Name System for internet access. In this section, we briefly describe hints on how to setup DNS behind NAT or Firewall. 10.3.1.1 Hints on how to Configure DNS behind a Firewall/NAT Proxy and Network Address Translation (NAT) devices can restrict access to ports. Set the DNS to use UDP port 53 and TCP port 53. For windows Servers, set the RCP option on the DNS Service Management console and configure the RCP to use port 135. These settings should be enough to resolve some of potential issues that may occur when you configure DNS and firewalls/NAT. 10.4 DHCP Server Setup A DHCP Server allows diskless clients to connect to a network and automatically obtain an IP address. This server is capable of supplying each network client with an IP address, subnet mask, default gateway, an IP address for a WINS server, and an IP address for a DNS server. This is very often used in enterprise networks to reduce configuration efforts. All IP addresses of all computers/routers/bases are stored in a database that resides on a server machine.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Appendix A: Basic Network Server(s) Configuration 106 The network administrator should contact the relevant vendors for detail information or step-by-step procedure on how to install and setup DHCP process or service on windows/Linux servers. In this section, we will provide some hints of how to resolve potential problems to be encountered you setup DHCP Servers. 10.4.1 Hint: Getting DHCP Server to Work Windows Server: 1) Clients are unable to obtain an IP address If a DHCP client does not have a configured IP address; it generally means that the client has not been able to contact a DHCP server. This is either because of a network problem or because the DHCP server is unavailable. If the DHCP server has started and other clients have been able to obtain a valid address, verify that the client has a valid network connection and that all related client hardware devices (including cables and network adapters) are working properly. 2) The DHCP server is unavailable When a DHCP server does not provide leased addresses to clients, it is often because the DHCP service has failed to start. If this is the case, the server may not have been authorized to operate on the network. If you were previously able to start the DHCP service, but it has since stopped, use Event Viewer to check the system log for any entries that may explain the cause. Next, restart the DHCP service, click Start, click Run, type cmd, and then press ENTER. Type net start dhcpserver, and then press ENTER. Linux Platform: Troubleshooting DHCP, check the following: 1) Incorrect settings in the /etc/dhcpd.conf file such as not defining the networks for which the DHCP server is responsible; 2) NAT/Firewall rules that block the DHCP bootp protocol on UDP ports 67 and 68; 3) Routers failing to forward the bootp packets to the DHCP server when the clients reside on a separate network. Always check your /var/logs/messages file for dhcpd errors. 4) Finally restart the dhcpd service daemon
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Appendix A: Basic Network Server(s) Configuration 107 10.5 TFTP Server Setup There are several TFTP servers in the market place; in this section we describe how to setup a commonly used TFTP Server. 10.5.1 TFTP Server Settings The administrator must configure basic parameters of the TFTP application:  Specify UDP 69 port – for TFTP incoming requests and TCP 12000 – for remote management of the server. For file transmission the server opens UDP ports with random numbers. In case the option Enable NAT or firewall support is activated on the server, the server uses the same port for files transmission and listening to the TFTP incoming requests (UDP 69 port on default).  Specify the interface bindings, TFTP root directory, port which the TFTP Server will listen, timeout and number of retries, and TFTP options supported by the server.  Configure the relevant TFTP virtual folder in the server. The TFTP virtual folder is the file folder, visible for TFTP clients under a certain name. You can set security settings separately for every virtual TFTP folder. Next, set rights to access TFTP folders according to the relevant clients.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Appendix A: Basic Network Server(s) Configuration 108 10.6 SIP Server Setup SIP server is one of the main components of an SME network, dealing with the setup of all SIP calls in the network. A SIP server is also referred to as a SIP Proxy or a Registrar. Although the SIP server is the most important part of the SIP based phone system, some servers only handles call setup and call tear down. It does not actually transmit or receive any audio. This is done by the media server in RTP. The RTX SME family of network phones are fully interoperable with the most of SIP Server applications. There are many off-the-shelf vendor and open source SIP servers. In this section, we will briefly explain settings required to take full advantage of FreePBX SIP Server feature set. The settings are similar for other SIP servers. 10.6.1 FreePBX SIP Server FreePBX is an easy to use GUI (graphical user interface) that controls and manages Asterisk, which the most popular open source telephony engine software. The administrator should refer to the relevant detail step-by-step procedure of how to install FreePBX SIP server. This section briefly describes SIP Server setup parameters. 1) SIP Server Setup Settings Description NAT This option determines the settings for users connecting to an asterisk server. Possible values: Yes, No, Never, Route NAT=route Asterisk will send the audio to the port and IP where it’s receiving the audio from. Instead of relying on the addresses in the SIP and SDP messages. This will only work if the phone behind NAT send and receive audio on the same port and if they send and receive the signalling on the same port. (The signalling port does not have to be the same as the RTP audio port). NAT=No Asterisk will add an RPORT to the via header of the SIP messages NAT=never This will cause asterisk not to add an RPORT in the VIA line of the sip invite
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Appendix A: Basic Network Server(s) Configuration 109 header Other NAT Settings Choose the relevant option or enter the settings in IP configuration, External IP, Local Network. Codecs Some SIP Servers supports dynamic codec support. Codecs are algorithm used to compress or decompress speech or audio signals. The user should select the relevant Codecs and other speech compression techniques whose traffic will be routed to the SME network. Video Codecs The user should enable this option if SME network supports video telephony. Media & RTP Settings This option should be enabled to provide for deliver media streams (e.g., audio and video) or out-of-band events signalling (DTMF in separate payload type). 2) Extensions This feature allows administrators create handset profiles in the SME network. In other words, Extensions describes the Dial plan for the PBX SIP system. Enter the relevant parameters
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Appendix A: Basic Network Server(s) Configuration 110
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Appendix B: Using Base with VLAN Network 111 11 Appendix B: Using Base with VLAN Network In this chapter we describe how to setup a typical VLAN in the network. 11.1 Introduction In this chapter, we describe how to setup VLAN to typical network. There are three main stages involved in this procedure: 1) Configure a VLAN Aware Switch to a specific (un)tagged VLAN ID, so the SME system can process untagged frames forwarded to it. 2) Setup the Time Server (NTP Server) and other relevant network servers. 3) Configure the HTTP server in relevant Base stations to access the features in the PBX or SME system. VLAN allows administrators to separate logical network connectivity from physical connectivity analogous to traditional LAN which is limited by its physical connectivity. Normally, users in a LAN belong to a single broadcast domain and communicate with each other at the Data Link Layer or “Layer 2”. LANs are segmented into smaller units for each IP subnets and here communication between subnets is possible at the Network Layer or “Layer 3”, using IP routers. A VLAN can be described as a single physical network that can be logically divided into discrete LANs that can operate independently of each other. An Illustration of using VLANs to create independent broadcast domains across switches is shown below: The figure above highlights several key differences between traditional LANs and VLANs.  All switches are interconnected to each other. However, there are three different VLANs or broadcast domains on the network. Physical isolation is not required to define broadcast domains. If the figure was a traditional LAN without VLAN-aware switches, all stations would belong to one broadcast domain.  All switch ports can communicate with one another at the Data Link Layer, if they become members of the same VLAN.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Appendix B: Using Base with VLAN Network 112  The physical location of an end station does not define its LAN boundary. 1. An end station can be physically moved from one switch port to another without losing its “view of the network”. That is, the set of stations it can communicate with at the Data Link Layer remains the same, provided that its VLAN membership is also migrated from port to port. 2. By reconfiguring the VLAN membership of the switch port an end station is attached to, you can change the network view of the end station easily, without requiring a physical move from port to port. 11.2 Backbone/ VLAN Aware Switches To implement a VLAN in your network, you must use VLAN-aware switches. Before we continue, let consider two rules to remember regarding the functioning of a regular LAN switch: 1. When the switch receives a broadcast or multicast frame from a port, it floods (or broadcasts) the frame to all other ports on the switch. 2. When the switch receives a unicast frame, it forwards it only to the port to which it is addressed. A VLAN-aware switch changes the above two rules as follows: 1. When the switch receives a broadcast or multicast frame from a port, it floods the frame to only those ports that belong to the same VLAN as the frame. 2. When a switch receives a unicast frame, it forwards it to the port to which it is addressed, only if the port belongs to the same VLAN as the frame. 3. A unique number called the VLAN ID identifies each VLAN. Which VLAN Does a Frame Belong To? The previous section notes that a frame can belong to a VLAN. The next question is—how is this association made?  A VLAN-aware switch can make the association based on various attributes of the type of frame, destination of MAC address, IP address, TCP port, Network Layer protocol, and so on.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Appendix B: Using Base with VLAN Network 113 An illustration of IEEE 802.1Q VLAN tag in Ethernet frame is as follows: 11.3 How VLAN Switch Work: VLAN Tagging VLAN functionality can be implemented via explicit frame tagging by switches and end stations. Network switches and end stations that know about VLANs are said to be VLAN aware. Network switches and end stations that can interpret VLAN tags are said to be VLAN tag aware. VLAN-tag-aware switches and end stations add VLAN tags to standard Ethernet frames–a process called explicit tagging. In explicit tagging, the end station or switch determines the VLAN membership of a frame and inserts a VLAN tag in the frame header (see figure above for VLAN tagging), so that downstream link partners can examine just the tag to determine the VLAN membership. 11.4 Implementation Cases Common types of usage scenarios for VLANs on typical VLAN switches: port-based VLANs, protocol-based VLANs, and IP subnet-based VLANs. Before figuring out which usage scenario suits your needs, you must understand what each type of usage scenario implies.  Port-based VLAN: All frames transmitted by a NIC are tagged using only one VLAN ID. The NIC does not transmit or receive any untagged frames. All protocols and applications use this virtual interface’s virtual PPA to transmit data traffic. Therefore all frames transmitted by that NIC port are tagged with the VLAN ID of that Virtual Interface.  Protocol-based VLAN: The NIC assigns a unique VLAN ID for each Layer 3 protocol (such as IPv4, IPv6, IPX, and so on). Therefore, the VLAN ID of outbound frames is different for each protocol. An inbound frame is dropped if the protocol and VLAN ID do not match.  IP subnet-based VLAN: The NIC assigns a unique VLAN ID for each IP subnet it belongs to. Therefore, the VLAN ID of outbound frames is different for different destination subnets. An inbound frame is dropped if the IP subnet and VLAN ID do not match.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Appendix B: Using Base with VLAN Network 114 11.5 Base station Setup After the admin have setup the Backbone switch, next is to configure the Base stations via HTTP interface. STEP 1 Connect the Base station to a private network via standard Ethernet cable (CAT-5). STEP 2 Use one of the two methods to find the base IP STEP 3 On the Login page, enter your authenticating credentials (the username and password is admin by default unless it is changed). Click OK button. STEP 4 Once you have authenticated, the browser will display front end of the SME Configuration Interface. The front end will show relevant information of the base station. STEP 5 Create the relevant SIP server information in the system. Each service provider/customer should refer SIP server vendor on how to setup SIP servers. 11.6 Configure Time Server STEP 6 Navigate to the Time settings and configure it. Scroll on the left column and click on Time url link to Open the Time Settings Page. Enter the relevant parameters on this page and press the Save button.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Appendix B: Using Base with VLAN Network 115 11.7 VLAN Setup: Base station STEP 7 Navigate to the Network url > On the network page enter the relevant settings in the VLAN section > VLAN Id should be the same as those configured into the backbone.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Appendix C: SME VoIP Network Planning/Optimization 116 12 Appendix C: SME VoIP Network Planning/Optimization In this chapter, we describe SME VoIP radio network planning techniques including dimensioning, detailed capacity and coverage planning, and network optimisation. 12.1 Network Requirements Network requirement is essential to determine elements necessary to achieve the overall expectations of the customer. Typical network requirements includes (but not limited to):  The geographical area to be covered  The type or architecture of building and/or topology, etc.  The estimated traffic on each zone or region or building  The blocking criteria in each traffic area.  The relevant quality targets expected to be achieved 12.2 Deployment Considerations The following radio considerations must be examined before deploying a SME VoIP System. These includes (but not limited to): Building Penetration: When a signal strikes a building it is diffracted or absorbed; therefore to some extend the signal is reduced. The amount of absorption is dependent of the kind of building and its environment, the amount of solid structure. This is an important consideration in coverage planning. Interference Sources: Signals to receiving antenna can be weakened by virtue of interference from other signals. These signals may be from the same network or other man-made objects. Interference sources must be identified and avoided or minimized. 12.3 Site Planning 12.3.1 Location Probability The quality of coverage is determined by location probability. Please see the How to Deploy SME VOIP System manual for more information. 12.3.2 Handover Mechanics/Planning Handsets should seamlessly move between coverage areas. In other words, handset should be able to move in a multi-cell setup of base stations and/or repeaters from one base station to another without terminating or causing hindrance while receiving continuous service and maintaining call-sessions in progress. For efficient handover of conversations between Base stations in multi-cell setup, deploy Base stations with wide overlap between them (i.e., plan for some areas to be covered by more than one Base station).
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Appendix C: SME VoIP Network Planning/Optimization 117 Overlaps are necessary to maintain seamless handover and to establish synchronization chains. A good example may be a cafeteria during lunch hour where temporary concentrations of handsets may occur. The overlap carries the excess call load to adjacent Base stations to provide uninterrupted services to subscribers. 12.4 Cell Coverage / Capacity Planning 12.4.1 Cell Coverage Due to the unexpected nature of RF propagation in an indoor environment, an actual on-site test must be performed before the deploying and/or installing core network elements. An extensive guide to effective RF coverage planning is outside the scope of this document. This should be noted: The Base stations provides typical RF coverage of up to 50 meters/164 feet in a typical indoor office environment and up 300 meters/984 feet in an open area (line-of-sight-LOS), extending in all directions from the Base stations (i.e. Omni directional). The exact coverage range depends on the building architecture, wall material and surroundings. The figure below show the correct installation of base stations spaced at the recommended distances from each other: Typically, installations such as office buildings, hotels and hospitals should be equipped with both base stations and repeaters on several floors to create uniform and complete radio coverage. Open areas can be covered with a sparse network of base stations. In such deployments, the base stations and/or repeaters cover an extended range due to the extended line-of-sight radio propagation capability The figure below shows an example of an installation where base stations are not spaced at the right distances:
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Appendix C: SME VoIP Network Planning/Optimization 118 12.4.2 Capacity Planning This is where the Network provider estimates how many calls will be initiated in a typical window/time frame and how many users will be initiating them. Another aspect of capacity planning should address the user growth pattern of a typical SME VoIP network. How many users will be using this network in for example two years window, etc. Based on these estimations, the network dimensioning can be planned orderly bearing in mind the SME capacity. 12.5 Network Dimensioning After the network requirements are clearly defined, the number of users that are expected to use the network must be estimated. Based on that, you should estimate and identify the number and type of equipment required in order to cater for the capacity, coverage and quality requirements. The more accurate the dimensioning, the more efficient the network rollout. Handsets/SIP End-Nodes: In a typical setup, the system can support up to 1000 handsets depending on the configuration. Base stations/Repeaters: The system can easily scale up to 50 base stations. Depending on the network setup, coverage can be extended by up to 3 repeaters pr. base station. The planner should bear in mind that base stations can support 10 simultaneously call sessions while repeaters can support 5 call sessions.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Appendix C: SME VoIP Network Planning/Optimization 119 Core Network Equipment: This equipment is at the premises of the service operator or data centre. Depending on the network requirements the following devices should be available: VLAN/NAT aware router(s), Session Border Controller, DHCP/TFTP/FTP Servers, STUN Server, Media Server, Access Gateway, SIP Server, etc. 12.6 Environmental Considerations In this section, we enumerate some environmental conditions that need to be considered prior to planning, deployment and optimisation of the SME network. The considerations are as follows: • Ensure that the installation area is clean and dry. • Ensure that the floor of the installation area is finished with linoleum, vinyl, ceramic, wooden flooring, computer floor tiles, or polished sealed concrete. • Ensure that the ceiling of the installation area is finished or treated to prevent particle discharge. • Ensure that the installation area is well lit, and that the light source is uniformly diffused without shadows. Adequate lighting should provide a comfortable reading level and allow the identification of wire insulator colours without undue eye fatigue. Lighting should be comparable to an office work environment, with a minimum level of 21 meter/68.9 feet at each work surface. As a rule of thumb, in a room with a 2.5 meters/8.2 feet ceiling, one 1.2 meters/4 feet fluorescent tube provides sufficient illumination for 1.9-2.4 square meters/20.5-25.9 square feet. • Ensure that ventilation of the installation area is capable of maintaining an ambient temperature of 0-40°C/32-104°F, and a relative humidity of 20-80% non-condensing, while the system is operating. The maximum power rating of a base station under full load should not exceed 315W/1070 BTU/Hr. These figures are for each cabinet only, and do not take into account heat generated by other equipment. In particular, charging fully-discharged batteries may generate a considerable amount of heat, depending on battery capacity and rate of charge. Refer to the equipment manufacturer data for more information. • Ensure that the installation area is free of caustic or corrosive liquids, substances, or materials. If batteries will be installed as part of the system, ensure that adequate precautions are taken (such as special ventilation) to prevent corrosive emissions from the batteries. Check local building codes for additional requirements. 12.7 Recommended Base station/Repeater Placement There is no one strategy for deploying base stations. These are some recommended Base station and/or Repeater placement strategies: Around Corridors: Base stations/repeater should be deployed vertically preferably at corridor intersections where propagation patterns follow the corridor patterns. The base station/repeater should point towards the corridor and preferably in the middle height between the floor and the actual ceiling. In case there are high objects in the area, the base station/repeater should be installed above those objects but still kept distant from the ceiling. Multi-Storey Buildings: Base stations and repeaters can be installed on opposite sides of the floors to take advantage of the floor-to-floor coverage. The coverage design cannot rely entirely on floor-to-floor propagation; each case must be verified due to variations in local attenuation patterns.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Appendix C: SME VoIP Network Planning/Optimization 120 Open Areas/ Large Halls: Base stations and repeaters can be deployed in open areas for buildings that contain a central open space area with windows to the other areas. This provides a good coverage for the rooms in the inner circle on all floors (e.g. hotels). In large halls, Base stations/repeater should be installed vertically in the middle of the space below the drop ceiling. Mounting Positions: When Base stations and repeaters are mounted vertically on a wall, the radio coverage in front of these devices is twice as large as the coverage at the rear. Repeaters should be installed in the middle of corridors and small rooms. Metallic Structures/Objects: Base stations and repeaters should not be deployed near large metallic objects. Reinforced Concrete Structures: These structures have a high attenuation factor inside the building. They reduce the radio coverage range of the Base stations and repeaters and therefore require a higher number of base stations or repeaters in the building. Lighter types of construction materials require fewer base stations since attenuation figures are considerably lower. Others Recommendations: • Maximum distance between two base stations varies depending on material and construction of buildings, but there must always be synchronization chains and radio coverage overlap between the two base stations or handover between radio units. The time it takes a person to cross the common coverage area must be 10 seconds or more, as the handset needs time to scan for an alternative base stations. • Ensure that the installation area is located no closer than 6.1meters/20.0 feet from electric devices that produce large electro-magnetic fields (EMF) or high levels of radio frequency energy. Possible EMF sources are radio transmitters, electric arc welding machines, copying machines, electric motors, refrigeration units, power transformers, electric load centres, and main circuit breaker panels. • Ensure that the electrical service is sufficient and located in close proximity to the Base stations. 12.8 Network Assessment/Optimisation This involves monitoring, verifying and improving the performance of the SME VoIP network. Depending on the network setup and varying deployment conditions and network usage some requirements have to be monitored and corrected. The main focus of network optimisation should be telephony quality, handovers, network traffic and other related measurements. The quality of the network is ultimately determined by the satisfaction of users of the network. Therefore before SME VoIP Networks are handed over to customers, Network providers must perform deployment testing. Please see the How to Deploy SME VOIP System manual for more information.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Appendix C: SME VoIP Network Planning/Optimization 121 Collect statistics of the network an example is illustrated in the table below: Parameters Value Comments ## Call Setup failures ## Dropped calls ## HO successes ## HO failures Traffic Blocking Rate (%) Traffic Blocking (Erl) Receiver level (dBm) Receiver Quality (%) After collecting the necessary information, you should fine tune signalling and radio resource sharing parameters. Network optimisation is a continuous process during and after the launch of the network. For example, if it is found that an area within a building has low signal level. There should be an immediate scrutiny of base station and/or repeater locations, heights and tilts. The problem is sorted out by moving the relevant devices and altering the tilts of these devices. For buildings/halls constructed with high signal attenuation materials, deploying additional base stations will be one of the solutions.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Appendix D: Local Central directory file handling 122 13 Appendix D: Local Central directory file handling In this appendix the Local Central Directory file format, import and configuration is described. 13.1 Central Directory Contact List Structure The structure of Contact List is simple. The figure below shows an example of structure of Contact List in Text format and in Xml format. Contact name must not contain more than 23 characters and contact number must not contain more than 21 digits. .csv or .txt .xml Txt file limitations:  Contact name must NOT be longer than 23 characters (name will be truncated)  Contact name must NOT contain “,”  Contact number must be limited to 21 digits (entry will be discarded, no warning)  Contact number digits must be: +0123456789  Contact number does not support SIP-URI  Spaces between name section “,” and number section is not supported 13.2 Central Directory Contact List Filename Format The Contact list is saved as file format: .txt .csv or .xml
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Appendix D: Local Central directory file handling 123 13.3 Import Contact List to Central Directory On the Central Directory page, the admin should click on Browse button and the Choose File to Load dialog window will be shown. On the Choose File to Upload dialog window, navigate to the directory or folder that contains the right file to be imported to the base station > Click on Open button. Next, click on the Load button. This will import the contents of contacts in the selected file into the relevant Base station. The figure below shows the import procedure is in process.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Appendix D: Local Central directory file handling 124 13.4 Central directory using server Alternative way to import a Contact List is to get it from a server. First click on Management url to get Management Settings page, then select the protocol of your server (TFTP/HTTP) in Management Transfer Protocol, then save the setting by clicking Save. Go back to Central Directory page and enter Server IP address (inclusive the path in the end of the address) and Filename of the contact list, then save the setting by clicking Save. (See example below). Then reboot the Base station to ensure that the changes take effect. 13.5 Verification of Contact List Import to Central Directory On the Handset, navigate to Central Directory where the correct contact list should populate to the contacts uploaded to the Base station.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Appendix E: Network Operations 125 14 Appendix E: Network Operations 14.1 Introduction In this chapter, we will provide an overview of the operation of the network during system start-up, location registration and speech calls including illustration of different call scenarios. 14.2 System Start Up When a Base station Unit is powered up, it achieves IP address from DHCP server and time from the Time-Server. Optionally the base retrieves its configuration from a file on the TFTP server. This configuration file describes used network and cluster configuration parameters (optional and not needed). The SME VoIP network has successfully started up. 14.3 Terminal Attachment When a DECT Terminal (also called handset or SIP node) is turned on or moved into the coverage area of a Base-station it has to get attached to the network. When more Base-stations are available, the Terminal selects the one with best RF signal. This procedure, called Location Registration, always keeps the network informed about where a Terminal is located and enables it receive or originate calls. This procedure also authenticates the Terminal and checks the validity of the associated subscription. 14.4 Outgoing Calls Outgoing calls are initiated by the Terminal. It selects the Base-station with best RF signal and establishes a radio communication link to Base-station. DECT call control messages are exchanged between Terminal, Base station and other servers. This server forwards the outgoing call as SIP messages to the external SIP Server. The RTP stream is established between the involved Base-station (and the Media Gateway for PSTN calls). If the call is between two Terminals the media stream may be routed directly between the two involved Base-stations depending on the SIP Server routing strategy. 14.5 Incoming Calls Incoming calls are initiated by SIP INVITE messages from the SIP Server to the Base unit; inviting it to participate in an incoming session. The system sends paging messages to all the Base-station where the Terminal last performed a Location Registration. When the paging is received the Terminal establishes a radio communication link to the best available Base-station and sends a response back to DECT controller. DECT call control messages are exchanged and the Terminal starts ringing. When the user answers the call, a connect message is sent to the IP DECT controller that completes the incoming call by sending 200 OK back to the SIP Server and establishes an RTP media stream between Base-station (and Media Gateway from PSTN line). For internal calls the media stream may be routed directly between the involved Base-stations.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Appendix E: Network Operations 126 14.6 Handover 14.6.1 RTP Stream Remains at Initial Base Station When the call is set up, the handset is located at base station 1. Thus, the DECT communication takes place between the handset and station 1, and the SIP signalling as well as the RTP stream takes place between base station 1 and the SIP server/media gateway. The figure below illustrates this application: SIP Server/Media GatewayBase station 1DECT ConnectionRTP StreamSIP SignalingBase station 2 Base station n Stage 1: Before handover the handset is located at BS 1. After handover, the handset is located at base station 2, and hence the DECT communication goes on between the handset and base station 2. However, to avoid disruption of the audio, the RTP stream is relayed via the initial base station, since a transfer of the RTP stream to another base station may cause the media gateway (or whatever the remote endpoint is) to re-initialize the RTP stream with a small disruption of the RTP stream as consequence. Thus, from the point of view of the remote endpoint, the RTP stream is not affected by the handover, and since the call control also remains at base station 1 the SIP signalling is also unaffected, as shown below: Stage 2: After handover to BS 2, the HS is located at BS 2, and the RTP stream is relayed via BS 1. Since the call control and hence the SIP User Agent remains at the initial base station, the SIP registration is also unaffected by the handover.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Appendix E: Network Operations 127 If the handset makes yet another handover, the RTP stream will still be relayed via the base station at which, the call was established (here base station 1). This is illustrated as follows: After handover to BS n the handset is located at BS n, and the RTP stream is relayed via BS 1. 14.7 Roaming By roaming means the handset moves its SIP and DECT registration from one base station to another base station. Roaming can only be initiated from idle. Roaming does not immediately result in a new SIP registration, because this may cause a lot of unnecessary signalling. Therefore, the handset will not perform a new DECT Location Registration until it has resided on the same base station for a defined period of time. Since the SIP registration is initiated by the completion of the Location Registration, a new SIP registration will also not be done until this procedure has completed on a new base station. Thus, a handset must stay on the same base station as given in the rules stated below, before a new SIP registration will be made. Timing Criteria for Location Registration; or roaming will be initiated when: 1. Handsets lose contact to first base unit due to reset/power off/heavy DECT traffic. 2. After 5 minutes (configuration is possible) but before 5+2 minutes a. The plus maximum 2 minutes will occur when service connection traffic is signaled at the same time as location should happen. In this case the location registration procedure will be delayed. If an incoming call arrives while the handset has moved to another base station (base station 2) but still not performed a new Location Registration, the SIP call will arrive at the initial base station (base station 1), but the RTP stream will be set up between base station 2 and the remote endpoint (refer to figure below). Alternatively, in the case of an outgoing call, the SIP call will be established from the initial base station, and the RTP stream will be set up between base station 2 and the remote endpoint.
SME VoIP System Guide, Version 2.6 Proprietary and Confidential Chapter: Appendix E: Network Operations 128 SIP Server/Media GatewayBase station 1Base station 2 Base station nRTP StreamDECT ConnectionSIP ControlAudio Control An illustration of Handset moving to another base station, but call control is still handled by the initial base station.
FCC Caution: Any Changes or modifications not expressly approved by the party responsible for compliance could void the user's authority to operate the equipment. This device complies with part 15 of the FCC Rules. Operation is subject to the following two conditions: (1) This device may not cause harmful interference, and (2) this device must accept any interference received, including interference that may cause undesired operation. Note: This equipment has been tested and found to comply with the limits for a Class B digital device, pursuant to part 15 of the FCC Rules. These limits are designed to provide reasonable protection against harmful interference in a residential installation. This equipment generates, uses and can radiate radio frequency energy and, if not installed and used in accordance with the instructions, may cause harmful interference to radio communications. However, there is no guarantee that interference will not occur in a particular installation. If this equipment does cause harmful interference to radio or television reception, which can be determined by turning the equipment off and on, the user is encouraged to try to correct the interference by one or more of the following measures: —Reorient or relocate the receiving antenna. —Increase the separation between the equipment and receiver. —Connect the equipment into an outlet on a circuit different from that to which the receiver is connected. —Consult the dealer or an experienced radio/TV technician for help. Privacy of communications may not be ensured when using this telephone! Chapter: Appendix E: Network Operations129SME VoIP System Guide, Version 2.6Proprietary and Confidential FCC Radiation Exposure Statement: This equipment complies with FCC radiation exposure limits set forth for an uncontrolled environment .This equipment should be installed and operated with minimum distance 20cm between the radiator& your body. This transmitter must not be co-located or operating in conjunction with any other antenna or transmitter.
IC: This device complies with Industry Canada licence-exempt RSS standard(s). Operation is subject to the following two conditions: (1) this device may not cause interference, and (2) this device must accept any interference, including interference that may cause undesired operation of the device. Le présent appareil est conforme aux CNR d'Industrie Canada applicables aux appareils radio exempts de licence. L'exploitation est autorisée aux deux conditions suivantes : (1) l'appareil ne doit pas produire de brouillage, et (2) l'utilisateur de l'appareil doit accepter tout brouillage radioélectrique subi, même si le brouillage est susceptible d'en compromettre le fonctionnement SME VoIP System Guide, Version 2.6Proprietary and Confidential Chapter: Appendix E: Network Operations130The device is compliance with RF exposure guidelines, users can obtain Canadian information on RF exposure and compliance.The minimum distance from body to use the device is 20cm. Le présent appareil est conforme Après examen de ce matériel aux conformité ou aux limites d’intensité de champ RF, les utilisateurs peuvent sur l’exposition aux radiofréquences et la conformité and compliance d’acquérir les informations correspondantes. La distance minimale du corps à utiliser le dispositif est de 20cm.

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