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VoIP Programming Guide

Rev. 0.5-draft
Jan. 27 2011

VoIP Programming Guide
COPYRIGHT
©2009 Realtek Semiconductor Corp. All rights Reserved. No part of this document may be reproduced,
transmitted, transcribed, stored in a retrieval system, or translated into any language in any form or by
any means without the written permission of Realtek Semiconductor Corp.

DISCLAIMER
Realtek provides this document “as is”, without warranty of any kind, neither expressed nor implied,
including, but not limited to, the particular purpose. Realtek may make improvements and/or changes in

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this document or in the product described in this document at any time. This document could include
technical inaccuracies or typographical errors.

TRADEMARKS

Realtek is a trademark of Realtek Semiconductor Corporation. Other names mentioned in this document
are trademarks/registered trademarks of their respective owners.

USING THIS DOCUMENT

This document provides is intended for the software engineer’s reference and provides detailed
programming information.

Though every effort has been made to ensure that this document is current and accurate, more information
may have become available subsequent to the production of this guide. In that event, please contact your
Realtek representative for additional information that may help in the development process.

REVISION HISTORY
Revision

Release Date

0.1

2009/04/9

First release (preliminary version)

0.2

2009/08/04

Second release

Summary

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VoIP Programming Guide
0.3

2009/10/22

Add new section 6.5 FXS-Pulse Dial Detection
Add new section 6.8 FXO Pulse Dial Generation
Add new section 6.11 FXS/FXO Line Status
Add new API rtk_Get_Session_Statistics() in section 7.3.
Add new API rtk_SetFaxModemDet(),rtk_GetCEDToneDetect() in
section 7.5.

0.4

2010/03/08

Add new section 6.12 FXS/FXO DTMF Detection

2010/04/16

Rename section 6.1 to FXS-line, and add new API in this section

2010/04/23

Add V.152 parameters description in section 13.1

2010/04/23

Add jitter buffer optimization factor zero

2010/05/19

Add new API rtk_Gen_MDMF_FSK_CID(), rtk_Set_FSK_CLID_Para(),

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rtk_Set_FSK_CLID_Para() in section 6.4.4

Add new section 11.2 Caller ID Generation Example 2

Add new section 12.13 to 12.15 for enumeration of FSK Caller ID
Add new section 13.8 Data Structure for FSK Caller ID

2010/05/21

Remove the API rtk_Set_CID_FSK_GEN_MODE() in section 6.4.4
Add new mode for DTMF Caller ID in section 6.4.2

2010/05/24

Add new example code in section 11.1

Add new API rtk_LimitMaxRfc2833DtmfDuration() in section 7.4.1
Add new example code in section 11.3

To correct the wrong API name from rtk_Set_SLIC_Impedance() to
rtk_Set_Impedance() in section 6.1.1

0.5

2010/6/28

Add RTP payload for Redundant Audio Data description in 7.3.1

2010/10/18

Add vlan API and example

2011/01/04

Add busy state return value for API

rtk_Gen_FSK_CID(),rtk_Gen_MDMF_FSK_CID(), and
rtk_Gen_FSK_VMWI().
2011/01/24

Add new API rtk_SetRFC2833TxConfig() and update section 11.3
example.

2011/01/27

Add new parameters for API rtk_Set_Pulse_Digit_Det()
Add pulse dial sample code in section 11.11

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VoIP Programming Guide
1.

OVERVIEW..................................................................................................................................................................1

2.

SYSTEM ARCHITECTURE........................................................................................................................................3

3.

FLASH SETTING FOR VOIP PARAMETERS ..........................................................................................................5

4.

VOIP MIDDLEWARE..................................................................................................................................................6
4.1.

DSP DEFAULT SETTING .........................................................................................................................................6

4.2.

VOIP FEATURE ......................................................................................................................................................7

4.2.1.
4.3.

Software API....................................................................................................................................................8

VOICE GAIN................................................................................................................................................................9
5.1.1.
5.2.

5.2.1.
5.3.
5.4.

5.4.1.
6.

LEC .....................................................................................................................................................................8

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4.3.1.
5.

Software API....................................................................................................................................................8

Software API....................................................................................................................................................9

AUTO GAIN CONTROL ...........................................................................................................................................9
Software API....................................................................................................................................................9

JITTER BUFFER.................................................................................................................................................... 11

MISCELLANEOUS ................................................................................................................................................ 12
Software API.................................................................................................................................................. 12

FXS/FXO.....................................................................................................................................................................12
6.1.

6.1.1.
6.2.

6.2.1.
6.3.

6.3.1.
6.4.

FXS–LINE .......................................................................................................................................................... 13

Software API.................................................................................................................................................. 13

FXS-RING .......................................................................................................................................................... 15

Software API.................................................................................................................................................. 15

FXS-FLASH HOOK TIME .....................................................................................................................................16
Software API.................................................................................................................................................. 16

FXS-CALLER ID .................................................................................................................................................17

6.4.1.

DTMF ...........................................................................................................................................................17

6.4.2.

Software API.................................................................................................................................................. 17

6.4.3.

FSK ...............................................................................................................................................................18

6.4.4.

Software API.................................................................................................................................................. 18

6.4.5.

VMWI............................................................................................................................................................ 21

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VoIP Programming Guide
6.4.6.
6.5.

FXS-P ULSE DIAL DETECTION.............................................................................................................................. 21

6.5.1.
6.6.

6.7.

6.8.

Software API.................................................................................................................................................. 25
FXO PULSE DIAL GENERATION ...........................................................................................................................26

6.8.1.

Software API.................................................................................................................................................. 26

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6.9.1.
6.10.

6.10.1.
6.11.

6.11.1.
6.12.

6.12.1.

FXS EVENT ........................................................................................................................................................ 27

Software API.................................................................................................................................................. 27

FXO EVENT........................................................................................................................................................ 28

Software API.................................................................................................................................................. 28

FXS/FXO LINE STATUS .......................................................................................................................................29
Software API.................................................................................................................................................. 29

FXS/FXO DTMF DETECTION .............................................................................................................................29
Software API.................................................................................................................................................. 29

VOIP SESSION...........................................................................................................................................................30
7.1.
7.2.

7.2.1.
7.3.

SOFTWARE API ...................................................................................................................................................31

VOIPSESSION-TONE............................................................................................................................................ 32
Software API.................................................................................................................................................. 33

VOIPSESSION-RTP .............................................................................................................................................34

7.3.1.

RTP Payload for Redundant Audio Data ........................................................................................................ 35

7.3.2.

Software API.................................................................................................................................................. 36

7.4.

7.4.1.
7.5.
7.5.1.
8.

Software API.................................................................................................................................................. 24
FXO CALLER ID .................................................................................................................................................25

6.7.1.

7.

Software API.................................................................................................................................................. 21
FXO...................................................................................................................................................................22

6.6.1.

6.9.

Software API.................................................................................................................................................. 21

VOIPSESSION-DTMF .........................................................................................................................................38
Software API.................................................................................................................................................. 38
VOIPSESSION-FAX.............................................................................................................................................40
Software API.................................................................................................................................................. 42

IVR.............................................................................................................................................................................. 44
8.1.

SOFTWARE API ...................................................................................................................................................44

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VoIP Programming Guide
9.

VOIP SECURITY .......................................................................................................................................................45
9.1.

SIP OVER TLS .................................................................................................................................................... 46

9.2.

SECURE RTP.......................................................................................................................................................47

10.

VLAN .................................................................................................................................................................... 49

10.1.

VLAN FUNCTION API .......................................................................................................................................... 49

10.2.

VLAN EXAMPLE .................................................................................................................................................. 52

11.

EXAMPLE FOR CALL FEATURES IMPLEMENTATION .............................................................................. 52

11.1.

CALLER ID GENERATION EXAMPLE 1...................................................................................................................52

11.2.

CALLER ID GENERATION EXAMPLE 2...................................................................................................................55

11.3.

RFC2833 SEND .................................................................................................................................................. 63

11.4.

CALL CONFERENCE .............................................................................................................................................64

11.5.

FXO...................................................................................................................................................................73

11.6.

PHONE ................................................................................................................................................................ 89

11.7.

RING ...................................................................................................................................................................91

11.8.

RTP.................................................................................................................................................................... 92

11.9.

VMWI.................................................................................................................................................................. 96

11.10.

TLS REGISTER.................................................................................................................................................96

11.11.

PULSE DIAL GENERATION/DETECTION ............................................................................................................... 101

12.

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APPENDIX A – ENUMERATION ..................................................................................................................... 105

12.1.

ENUMERATION FOR PHONE STATE ....................................................................................................................... 105

12.2.

ENUMERATION FOR FXO EVENT ........................................................................................................................ 106

12.3.

ENUMERATION FOR SUPPORTED RTP PAYLOAD ................................................................................................... 107

12.4.

ENUMERATION FOR RTP SESSION STATE ............................................................................................................. 108

12.5.

ENUMERATION FOR COUNTRY ............................................................................................................................ 109

12.6.

ENUMERATION FOR DSP TONE ........................................................................................................................... 109

12.7.

ENUMERATION FOR DSP PLAY TONE DIRECTION .................................................................................................. 111

12.8.

ENUMERATION FOR IVR PLAY............................................................................................................................ 112

12.9.

ENUMERATION FOR IVR PLAY DIRECTION (PLAY TEXT ONLY)............................................................................. 112

12.10.

ENUMERATION FOR FAX STATE ......................................................................................................................... 112

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VoIP Programming Guide
12.11.

ENUMERATION FOR DTMF TYPE ........................................................................................................................ 113

12.12.

ENUMERATION FOR THE COUNTRY ..................................................................................................................... 113

12.13.

ENUMERATION FOR FSK CALLER ID AREA ........................................................................................................ 114

12.14.

ENUMERATION FOR FSK CALLER ID PARAMETER TYPE ...................................................................................... 114

12.15.

ENUMERATION FOR FSK CALLER ID GAIN ......................................................................................................... 114

13.

APPENDIX B – DATA STRUCTURE................................................................................................................ 116

13.1.

DATA STRUCTURE FOR RTP PAYLOAD TYPE ........................................................................................................ 116

13.2.

DATA STRUCTURE FOR RTP CONFIGURATION ..................................................................................................... 117

13.3.

DATA STRUCTURE FOR RTP SESSION .................................................................................................................. 118

13.4.

DATA STRUCTURE FOR TONE CONFIGURATION .................................................................................................... 119

13.5.

DATA STRUCTURE FOR 3-WAY CONFERENCE CONFIGURATION ............................................................................. 120

13.6.

DATA STRUCTURE FOR RTP STATISTICS .............................................................................................................. 121

13.7.

DATA STRUCTURE FOR VOIP CONFIGURATION .................................................................................................... 121

13.8.

DATA STRUCTURE FOR FSK CALLER ID ............................................................................................................. 127

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VoIP Programming Guide

1.

Overview

The Realtek VoIP SDK provides comprehensive voice codec support including G.711, G.729,
G.723, G.726, iLBC and GSM. All these codecs are tested and measured with Digital Line
Speech Analyzer. The wideband codec, for example G.722, is being developed and will be
released in no time. The Table 1 shows the voice codecs information of RTK VoIP SDK.
Codec Name

Bitrate

Fidelity

Avg. Quality
(PESQ-LQ)1

G.711 ulaw

64kbps

Narrowband

4.33

G.711 alaw

64kbps

Narrowband

4.33

G.729/AB

8kbps

Narrowband

3.90

G.723 5.3k

5.3kbps

Narrowband

3.67

G.723 6.3K

6.5kbps

Narrowband

3.82

G.726 32K

32kbps

Narrowband

4.09

G.726 16K

16kbps

Narrowband

2.94

G.726 24K

24kbps

Narrowband

3.72

G.726 40K

40kbps

Narrowband

4.19

GSM-FR

13.2bps

Narrowband

3.72

20ms

Narrowband

3.87

64kbps

Wideband

4.36

ILBC

2

G.722

Table 1. VoIP Codec for VoIP SDK

The VoIP call features can be accomplished by using the powerful VoIP Interface APIs. The
VoIP Interfaces APIs provide various functions to control the DSP and network interface. The
Table 2 shows the call features supported in RTK VoIP SDK.
Type of Feature

Call Feature
Multi-Proxy REGISTER

REGISTER

Single-Proxy REGISTER

Non-REGISTER

1
2,3

Direct IP CALL

The value of PESQ-LQ depends on platform. The above figure refers to the test report of RTL8186
This is new codec which is available depending on platform.
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VoIP Programming Guide
Incoming Call
Outing Call

Answer call
Drop call
Cancel Call
Drop Call
Call waiting tone

Call Waiting

Call waiting caller id
Alternate call after call waiting

Hold

Hold/Resume call
Consultation call
Immediate Forward

Call Forwarding

Busy Forward
No Answer Forward

Transfer Call
Conference

Blind Transfer call
Attended Transfer call
3-way Conference
Speed Dial
Auto Dial
Off-hook Alarm
Flash Hook Time

Miscellaneous

Hot Line
DND
Off-hook Password-FXS
Off-hook Password-FXO
PSTN Routing Prefix
Reject Direct IP Call

NAT Traversal
VoIP Security

Outbound Proxy
Stun
SRTP
SIP Over TLS
Table 2 Call features for VoIP SDK

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VoIP Programming Guide

2.

System Architecture

The Realtek VoIP architecture includes several components. These components can be
classified into two categories, the user space components and kernel space components. The
user space component, for example sipua, can control the kernel space components through I/O
control and system call. However, the developer should know about the detail of operation of
kernel space components. This is too complex and time consuming for user to understand the
detail of the operations. Owning this, VoIP interface APIs are developed. Infect, these APIs are
conceptual layer for developers who can just concern about the input parameters and return
values of the APIs. Most of the VoIP features can be achieved by calling the VoIP interface APIs.
The Figure 1 shows the main components of VoIP architecture.

Figure 1 VoIP Architecture

Application: The sipua generates SIP message according to RFC3261. As a caller, it sends the
SIP request and receives SIP response over TCP/IP stacks. On the other hands, it receives SIP
request and sends response to caller as it acts as callee. The other application for VoIP is web.
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VoIP Programming Guide
Infact main purpose of the web is the interface that allows user to input the VoIP parameters of
the device. The parameters then are saved in flash memory.
VoIP interface: The user space application, sipua for example, controls the kernel space
components through VoIP interface APIs. Including VLAN, DSP, RTP and phone functions, the
VoIP interface APIs is the basic components for setting up the VoIP function.
Manager interface: This interface accepting the VoIP interface command controls the VoIP
kernel components.
VoIP network interface: This interface accepts the data of RTP set by VoIP interface. There
may be different network interfaces for different platform. For example, the 8186 platform
includes static IP, DHCP, PPPoE, PPTP and L2TP.
Network driver: The RTP packet different form SIP packet is sent by kernel space module. So
the VoIP packets will be set as higher priority in queue. The network driver sends the VoIP data
through wire.
VoIP middeware: This is the core component for VoIP SDK. This DSP toolbox includes codec,
jitter buffer, AEC, LEC, tone detection, dtmf detection etc.
PCM driver: This module accepts the data of VoIP middleware and converts the data for
generating the VoIP packet.
SPI simulator: This module imitates the SPI to control the SLIC.
Components

Module Name

Corresponding Location in SDK

Application

Solar, solar monitor

VoIP-ATA/AP/rtk_voip/src

Application

Webs

VoIP-ATA/AP/goahead-2.1.1/LINUX

VoIP Interface

VoIP interface

VoIP-ATA/AP/rtk_voip/voip_manager

Manager Interface

VoIP manager

VoIP-ATA/linux-2.4.18/rtk_voip/voip_manager
VoIP-ATA/linux-2.4.18/rtk_voip/voip_rx

VoIP Network Interface

VoIP_RX, VoIP_TX

Network Interface

Network Driver

VoIP-ATA/linux-2.4.18/drivers/net

VoIP Middleware

VoIP DSP

VoIP-ATA/linux-2.4.18/rtk_voip/voip_dsp

PCM Driver

VoIP Driver

VoIP-ATA/linux-2.4.18/rtk_voip/voip_drivers

SPI Simulator

VoIP Driver

VoIP-ATA/linux-2.4.18/rtk_voip/voip_drivers

VoIP-ATA/linux-2.4.18/rtk_voip/voip_tx

Table 3 The VoIP components

4

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VoIP Programming Guide

3.

Flash Setting for VoIP Parameters

The VoIP parameters such as proxy server IP, phone number and login name etc are stored in
flash of the system. These parameters can be set or shown in web pages. However, there is
another interface for setting and getting these parameters. By inputting the command in the
console, the VoIP parameters can be extracted. The Figure 2 command will show the all
parameters of VoIP in the flash. The Figure 3 command will show the single parameter of the
VoIP parameter. The Figure 4 command will set the definite value to the VoIP parameter.
#flash voip
Figure 2. Dump the VoIP parameters in the flash

#flash voip get 
Figure 3. Dump the single VoIP parameter

#flash voip set  
Figure 4. Set the VoIP parameter to the definite value

The flash layout is divided into 3 sections. They are the current setting, the default setting and
the hardware setting. The current setting section contains the user settings from the web pages.
The default setting section contains the settings of factory reset. The hardware setting section
contains the hardware information such as MAC address of Ethernet.
The file with the path “VoIP-ATA/AP/rtk_voip/includes/voip_flash.h” contains the structure of
VoIP configure parameters3. The corresponding file “VoIP-ATA/AP/rtk_voip/flash/voip_flash.c”
contains the default setting of the VoIP parameters and the flash read/write functions. The flash
read sample codes are shown in Figure 5 while the flash write sample codes are shown in Figure
6.
voipCfgParam_t *pVoIPCfg;
if (voip_flash_get(&pVoIPCfg) != 0)
return -1;
Figure 5. Flash read for VoIP parameters

3

The location of the files will be different according to different platforms.
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VoIP Programming Guide
voipCfgParam_t *pVoIPCfg;
/*do setting */
…
voip_flash_set(pVoIPCfg);
Figure 6. Flash set for VoIP parameters

4.

VoIP Middleware

The VoIP Middleware supports many features. These include setting the echo tail length, lighting
the LED, setting the speaker and microphone voice gain and setting the speaker and
microphone AGC.

4.1. DSP Default Setting
l

Line Echo Cancellation Tail Length: 2 msec with non-linear linear process

l

Jitter Buffer Control:
n

Min. delay:40ms

n

Max. delay: 130ms

n

Optimization factor: 7

l

Vocie Active Detection(VAD): disable

l

Packet loss concealment(PLC): disable

l

Auto Gain Control:

l

n

Speaker AGC: disable

n

Speaker AGC Required Level: Level 0 (up to level 8)

n

Speaker AGC Gain Up: 5 dB

n

Speaker AGC Gain Down: 5 dB

n

MIC AGC: default disable

n

MIC AGC Required Level: 0 (up to level 8)

n

MIC AGC Gain Up: 5 dB

n

MIC AGC Gain Down: 5 dB

Voice Gain:
n

Speaker (Tx): 0dB

n

MIC(Rx): 0dB
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VoIP Programming Guide
l

Caller ID Generation:
n

Software Caller ID generation (Hardware mode is not supported now)

n

Caller ID before 1st Ring as default

n

No Polarity Reversal before Caller ID

n

No Short Ring before Caller ID

n

No Alert Tone before Caller ID

n

DTMF Caller ID:

n

u

Start digit: A

u

End digit: C

FSK Caller ID:
u

Bellcore as default

u

No Date, Time, and Name

l

Caller ID detection: default detect DTMF caller ID only

l

DTMF mode: Inband DTMF

l

Tone of Country: USA

l

Flash Hook Time: 100 msec ~ 300 msec

4.2. VoIP Feature
The variable g_VoIP_Feature contain the VoIP features that the user selects from
menuconfigure in kernel directory. The initial value for this variable is 0. The function
rtk_Get_VoIP_Feature must be called first to retrieve the VoIP feature value. The Figure 7 shows
the sample codes for VoIP features extraction. The function of rtk_InitDSP must be called before
any use of manipulative function of voice.
rtk_Get_VoIP_Feature();
if ((g_VoIP_Feature & DAA_TYPE_MASK) == NO_DAA)
fxo_type = FXO_NO_DAA;
Figure 7. The extraction of the VoIP features

// init dsp
rtk_InitDSP(0);

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VoIP Programming Guide
rtk_Set_Voice_Gain(0, 0, 0);
rtk_Set_echoTail(0, 4);

// 0db
// 4ms

Figure 8. Calling rtk_InitDSP before calling voice manipulative function

4.2.1.

Software API

Prototype:

Int32 rtk_Get_VoIP_Feature(void)

Description:

Get VoIP Feature

Parameters:

Void

Return Values: 0 success
Reference:

FXO

Prototype:

Int32 rtk_InitDSP(int ch)

Description:

Interface Initialization

Parameters:

ch:The channel number

Return Values: 0 success
Reference:

Call Conference, FXO, RTP

4.3. LEC
The Line Echo Cancellation (LEC) is used to remove the echo caused by the delay of the
telephone network line. The tail length can be set as 2ms, 4ms, 8ms, 16ms and 32ms. The
higher value of tail length requires more computation time. The Figure 9 shows the sample code
for setting the channel 0 for 2ms tail length.
rtk_Set_echoTail(0, 2);
Figure 9. Set the channel 0 for 2ms tail length for LEC

4.3.1.

Software API

Prototype:

Int32 rtk_Set_echoTail(uint32 chid, uint32 echo_tail)

Description:

Set echo tail for LEC

Parameters:

Chid:The channel number
echo_tail:The echo tail length (ms)

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VoIP Programming Guide
Return Values: 0 success
Note:

1~16ms in G.711, 1~4ms in G.729/G.723

Reference:

FXO

5.

Voice Gain

User can set up the volumes of the speaker and microphone by call the function of
“rtk_Set_Voice_Gain”.

5.1.1.

Software API

Prototype:

int32 rtk_Set_Voice_Gain(uint32 chid, int spk_gain, int mic_gain)

Description:

Set the speaker and mic voice gain

Parameters:

chid:The channel number
spk_gain:-32~31 dB (-32: mute, default is 0 dB)
mic_gain:-32~31 dB (-32: mute, default is 0 dB)

Return Values: 0 success
Reference:

FXO, phone

5.2. Auto Gain Control
The function of “rtk_Set_Voice_Gain” sets the volume of the speaker and volume of microphone.
The range is between –32dB and 31dB. The function of “rtk_Set_SPK_AGC” sets the speaker
auto gain control. The value of “support_gain” is either 0 or 1(not support/support). The function
of “rtk_Set_SPK_AGC_LVL” sets the speaker auto gain control level. The parameter of level is
between 0 to 8 and the default value is 0. The functions of “rtk_Set_MIC_AGC_GUP” and
“rtk_Set_MIC_AGC_GDOWN” set the auto gain control down and up value. The parameter of
gain can be 0 to 8. The corresponding functions of microphone are the same as speaker.

5.2.1.

Software API

Prototype:

int32 rtk_Set_SPK_AGC(uint32 chid, uint32 support_gain)

Description:

rtk_Set_SPK_AGC
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VoIP Programming Guide
Parameters:

chid:The channel number
support_gain:support_gain 0: not support, 1:supported

Return Values: 0 success
Prototype:

int32 rtk_Set_SPK_AGC_LVL(uint32 chid, uint32 level)

Description:

rtk_Set_SPK_AGC_LVL

Parameters:

chid:The channel number
level:level

Return Values: 0 success

Prototype:

int32 rtk_Set_SPK_AGC_GUP(uint32 chid, uint32 gain)

Description:

rtk_Set_SPK_AGC_GUP

Parameters:

chid:The channel number
gain:gain

Return Values: 0 success
Prototype:

int32 rtk_Set_MIC_AGC(uint32 chid, uint32 support_gain)

Description:

rtk_Set_MIC_AGC

Parameters:

chid:The channel number
support_gain:support_gain

Return Values: 0 success

Prototype:

int32 rtk_Set_MIC_AGC_LVL(uint32 chid, uint32 level)

Description:

rtk_Set_MIC_AGC_LVL

Parameters:

chid:The channel number
level:level

Return Values: 0 success
rototype:

int32 rtk_Set_MIC_AGC_GUP(uint32 chid, uint32 gain)

Description:

rtk_Set_MIC_AGC_GUP

Parameters:

chid:The channel number
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gain:gain
Return Values: 0 success
Prototype:

int32 rtk_Set_MIC_AGC_GDOWN(uint32 chid, uint32 gain)

Description:

rtk_Set_MIC_AGC_GDOWN

Parameters:

chid:The channel number
gain:gain

Return Values: 0 success

5.3. Jitter Buffer
A jitter buffer is space for storing the arriving packets temporarily. In order to make the voice to
be sounded more smoothly, a jitter buffer is needed to minimize the delay variations. The
minimum delay can be set at the range of 40ms to 100ms. The recommended value is 40ms.
The maximum delay can be set at the range of 130ms to 300ms and the recommended value is
130ms. The optimization factor is the value for adjusting the quality of the voice. This is shown in
Figure 10. The higher value of optimization means the better quality but more delay. The medium
value of 7 is recommended. The value of 0 and 13 are fix delay, but first one is for low buffering
application, the other is for FAX or modem only. These three values can be set to structure of rtp
configuration, TstVoipMgrRtpCfg as shown in Figure 11.

Fix delay

Fix delay

(low buffering)

(for FAX or Modem)

Recommend
Delay

0

1

2

Quality

3

4

5

6

7

8

9

Figure 10 The optimization factor for jitter buffer

11

10 11 12 13

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typedef struct
{
…
Tuint32 nJitterDelay;
Tuint32 nMaxDelay;
Tuint32 nJitterFactor;
…
} TstVoipMgrRtpCfg;
Figure 11 jetter buffer parameters setting

5.4. Miscellaneous
5.4.1.

Software API

Prototype:

int rtk_sip_register(unsigned int chid, unsigned int isOK)

Description:

Light LED if SIP Register OK

Parameters:

chid:The channel number
isOK:1: SIP register OK, 0: SIP register failed

Return Values: 0 success

6.

FXS/FXO

The analogue terminal adaptor (ATA) usually has FXS port, FXO port and Ethernet port. User
can make call through internet or through PSTN. It is possible for an ATA to have two FXS ports.
In this case, each telephone will have individual phone number. By using the VoIP SDK, user
can accomplish the FXS and FXO functions. The following two functions will illustrate the use of
these FXS APIs and FXO APIs.

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Figure 12 The FXS and FXO diagram

6.1. FXS–Line
6.1.1.

Software API

Prototype:

int32 rtk_Set_Country_Impedance(_COUNTRY_ country)

Description:

Set the SLIC impedance according to the Country

Parameters:

country: the selected country
COUNTRY_USA: 600
COUNTRY_UK: 270 ohm + (750 ohm||150nF)
COUNTRY_AUSTRALIA: 220 ohm + (820 ohm||120nF)
COUNTRY_HK: 600
COUNTRY_JAPAN: 600 ohm + 1000nF
COUNTRY_SWEDEN: 200 ohm + (1000 ohm||100nF)
COUNTRY_GERMANY: 220 ohm + (820 ohm||115nF)
COUNTRY_FRANCE: 215 ohm + (1000 ohm||137nF)
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VoIP Programming Guide
COUNTRY_TW: 600
COUNTRY_BELGIUM: 150 ohm + (830 ohm||72nF)
COUNTRY_FINLAND: 270 ohm + (750 ohm||150nF)
COUNTRY_ITALY: 400 ohm + (700 ohm||200nF)
COUNTRY_CHINA: 200 ohm + (680 ohm||100nF)
COUNTRY_CUSTOMER: 600
Return Values: 0 success
Prototype:

int32 rtk_Set_Impedance(uint32 chid, uint8 preset)

Description:

Set the SLIC Impedance from preset table

Parameters:

chid:The FXS channel number
preset: The SLIC Impedance preset table index.
0: 600
1: 900
2: 250+(750||150nf)
3: 320+(1150||230nf)
4: 350+(1000||210nf)

Return Values: 0 success
Reference:
Prototype:

int32 rtk_Set_SLIC_Line_Voltage(uint32 chid, uint8 flag)

Description:

Set SLIC line voltage

Parameters:

chid:The FXS channel number
flag: The SLIC line voltage flag
0: zero voltage
1: normal voltage
2: reverse voltage

Return Values: 0 success
Reference:
Prototype:

int32 rtk_Gen_SLIC_CPC(uint32 chid, uint32 cpc_ms)

Description:

Generate SLIC CPC signal
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Parameters:

chid:The FXS channel number
cpc_ms: SLIC CPC signal period (unit in ms)

Return Values: 0 success
Reference:

6.2. FXS-Ring
As a caller, an FXS device initials a call by passing ring voltage over the line to the attached FXO
device. As a callee, an FXS device receives a call by detecting the line that has been seized. The
function of “rtk_SetRingFXS” sets the FXS to ring. The parameter of bRinging is either 0 or
1(silent or ring). The function of “rtk_Set_SLIC_Ring_Cadence” sets the period of cadence on
and period of cadence off. This function should use with the function “rtk_SetRingFXS”.
rtk_Set_SLIC_Ring_Cadence(0, 2000, 4000);

// 2 sec ON, 4 sec OFF

rtk_SetRingFXS(0, 1); // enable ring
Figure 13. Set the cadence and ring the phone

6.2.1.

Software API

Prototype:

int32 rtk_SetRingFXS (uint32 chid,uint32 bRinging)

Description:

Enable/Disable FXS Ringing

Parameters:

chid:the FXS channel number
bringing:1 enable, 0 disable ring

Return Values: 0 success
Reference:

Caller id generation, FXO, ring

Prototype:

int32 rtk_Set_SLIC_Ring_Cadence(uint32 chid, uint16 cad_on_msec, uint16
cad_off_msec)

Description:

Set the Ring Cadence of FXS

Parameters:

chid:The FXS channel number
cad_on_msec:The time period of Cadence-On
cad_off_msec:The time period of Cadence-Off

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Return Values: 0 success
Reference:

FXO, ring

Prototype:

int32 rtk_Set_SLIC_Ring_Freq_Amp(uint32 chid, uint8 preset)

Description:

Set the Ring frequency and amputide of FXS

Parameters:

chid:The FXS channel number
preset: The Ring frequency, amputide preset table index.
0: 20Hz, 38V
1: 20Hz, 45V
2: 20Hz, 48V (default)
3: 25Hz, 48V
4: 17Hz, 50V
5: 20Hz, 40V

Return Values: 0 success
Reference:

6.3. FXS-Flash Hook Time
6.3.1.

Software API

Prototype:

int32 rtk_Set_Flash_Hook_Time(uint32 chid, uint32 min_time, uint32 time)

Description:

Set the flash hook time for each FXS channel

Parameters:

chid:the FXS channel number
min_time: the minimal acceptabl flash hook period
time: the maximal acceptabl flash hook period

Return Values: 0 success
Reference:

Call Conference, phone

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6.4. FXS-Caller ID
The caller-id will display in the callee’s phone during the ringing signal. It can be classified into
two categories. One is DTMF mode and the other is FSK mode. For DTMF mode, there are three
kinds of DTMF. They are inband, SIP INFO and RFC2833. For FSK mode, there are four kinds of
FSK. They are bellcore, etsi, bt and ntt.

6.4.1.

DTMF

Dual-tone multi-frequency (DTMF) is a signal used in telecommunication. There are 12 DTMF
signals and each of these is made up of two tones. According to the frequency, the tones forming
the DTMF signal can be classified into tow groups.
1209 Hz

1336 Hz

1477 Hz

697 Hz

1

3

3

770 Hz

4

5

6

852 Hz

7

8

9

941 Hz

*

0

#

Table 4. The DTMF signal

6.4.2.

Software API

Prototype:

int32 rtk_Gen_Dtmf_CID(uint32 chid, char* str_cid)

Description:

Caller ID generation via DTMF

Parameters:

chid:The FXS channel number
str_cid:The Caller ID

Return Values: 0 success
Note:

When application calls this API to send DTMF caller ID, DSP ring the phone
automatically after sending caller ID. So, there is no need to ring the phone by
application. If caller ID is configed to send between 1st and 2nd ring, DSP also can
handle it with ring cadence automatically.

Reference:

Caller ID Generation Example 1

Prototype:

int32 rtk_Set_CID_DTMF_MODE(uint32 chid, char area, char cid_dtmf_mode)

Description:

Set the DTMF Caller ID Generation Mode

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Parameters:

chid:The FXS channel number
area:
bit0-2: Skip (set to zero for DTMF CID)
bit 3: Send cid before 1st ring (1) or between 1st ring and 2nd ring (0).
bit 4: Skip (set to zero for DTMF CID)
bit 5: Skip (set to zero for DTMF CID)
bit 6: Reverse Polarity before Caller ID (Line Reverse) (0: Forward, 1: Reverse)
bit 7: Skip (set to zero for DTMF CID)
cid_dtmf_mode:set the caller id start/end digit.
0-1 bits for starting digit, and 2-3 bits for ending digit.
* 00: A, 01: B, 02: C, 03: D
bit 4: Auto start/end digit send
* 0: auto mode: DSP send start/end digit according to the bit 0-3 setting
automatically
* 1: non-auto mode: DSP send caller ID string only. If caller ID need
start/end digits, developer should add them to caller ID strings.

Return Values: 0 success
Reference:

6.4.3.

Caller ID Generation Example 1, FXO

FSK

Frequency-shift keying (FSK) is mainly used for caller ID applications. There are several types of
FSK They are listed as following:
1.

ETSI FSK (formed by the European Telecommunications -1 and –2)

2.

Bellcore FSK (formed by the Telcordia Technologies)

3.

BT FSK (formed by the British Telecom)

4.

NTT FSK

6.4.4.

Software API

Prototype:

int32 rtk_Set_FSK_Area(uint32 chid, char area)

Description:

Set the FSK Area for FSK Caller ID, VMWI Generation

Parameters:

chid:The FXS channel number.
area:The area of FSK. (see CIDAREA),
bit 0-2: Area -> 0: Bellcore, 1: ETSI, 2: BT, 3: NTT

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VoIP Programming Guide
bit 3: Send Caller ID before 1st ring (1) or between 1st ring and 2nd ring (0).
bit 4: Dual Tone before Caller ID (Fsk Alert Tone) (0: False, 1: True)
bit 5: Short Ring before Caller ID (0: False, 1: True)
bit 6: Reverse Polarity before Caller ID (Line Reverse) (0: Forward, 1: Reverse)
bit 7: FSK Date & Time Sync and Display Name (0: False, 1: True)
Return Values: 0 success
Reference:

Caller ID Generation Example 1

Prototype:

int32 rtk_Gen_FSK_CID(uint32 chid, char* str_cid, char* str_date, char*
str_cid_name, char mode)

Description:

FSK Caller ID generation

Parameters:

chid:The FXS channel number
str_cid:The Caller ID
str_date:The Date and Time
str_cid_name:The Caller ID Name
mode:0: Service Type I, 1: Service Type II

Return Values: 0: success, -2: busy state
Note:

When application calls this API to send FSK caller ID, DSP ring the phone
automatically after sending caller ID. So, there is no need to ring the phone by
application. If caller ID is configed to send between 1st and 2nd ring, DSP also can
handle it with ring cadence automatically.

Reference:

Caller ID Generation Example 1, Caller ID Generation Example 2

Prototype:

int32 rtk_Gen_CID_And_FXS_Ring(uint32 chid, char cid_mode, char *str_cid,
char *str_date, char *str_cid_name, char fsk_type, uint32 bRinging)

Description:

FSK Caller ID generation and Ring Control

Parameters:

chid:The FXS channel number
cid_mode: 0: DTMF Caller ID, 1: FSK Caller ID
str_cid:The Caller ID
str_date:The Date and Time
str_cid_name:The Caller ID Name
fsk_type:0: Service Type I, 1: Service Type II
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bRinging: Ringing control, 0: ring off, 1: ring on
Return Values: 0 success
Note:

When application calls this API to send FSK caller ID, DSP ring the phone
automatically after sending caller ID even bRinging is equal to zero. So, there is
no need to ring the phone by application. If caller ID is configed to send between
1st and 2nd ring, DSP also can handle it with ring cadence automatically.
If str_cid[0] is equal to zero, caller ID won’t be send. In this case, this API will ring
on/off the phone according to the parameter bRinging.

Reference:

Caller ID Generation Example 2

Prototype:

int32 rtk_Gen_MDMF_FSK_CID(uint32 chid, TstFskClid* pClid, uint32
num_clid_element)

Description:

MDMF FSK Caller ID generation

Parameters:

chid:The FXS channel number
pClid str_date:The Caller ID Datas
num_clid_element: number of Caller ID elements

Return Values: 0: success, -2: busy state
Note:

When application calls this API to send FSK caller ID, DSP ring the phone
automatically after sending caller ID. So, there is no need to ring the phone by
application. If caller ID is configed to send between 1st and 2nd ring, DSP also can
handle it with ring cadence automatically.

Reference:

Caller ID Generation Example 2

Prototype:

int32 rtk_Set_FSK_CLID_Para(TstVoipFskPara* para)

Description:

Set the FSK Caller ID Parameters

Parameters:

para: FSK Caller ID Parameters

Return Values: 0 success
Reference:

Caller ID Generation Example 2

Prototype:

int32 rtk_Get_FSK_CLID_Para(TstVoipFskPara* para)

Description:

Get the current FSK Caller ID Parameters

Parameters:

para: FSK Caller ID Parameters
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Return Values: 0 success
Reference:

Caller ID Generation Example 2

Prototype:

int32 rtk_GetFskCIDState(uint32 chid, uint32* cid_state)

Description:

get the fsk caller id send complete or not

Parameters:

chid:The FXS channel number.
cid_state:0:fsk cid send complete 1:sending fsk cid

Return Values: 0 success

6.4.5.

VMWI

Voice message waiting indicator is the feature of the illuminate of LED on select telephone to
notify the voice mail messages.

6.4.6.

Software API

Prototype:

int32 rtk_Gen_FSK_VMWI(uint32 chid, char* state, char mode)

Description:

Generate the VMWI via FSK

Parameters:

chid:The FXS channel number
state:The address of value to set VMWI state. (0: off, 1: on)
mode:0: type I, 1: type II

Return Values: 0: success, -2: busy state
Reference:

vmwi

6.5. FXS-Pulse Dial Detection
6.5.1.

Software API

Prototype:

int rtk_Set_Pulse_Digit_Det(uint32 chid, uint32 enable, uint32 pause_time, uint32
min_break_ths, uint32 max_break_ths)

Description:

Set FXS pulse detection

Parameters:

chid:The FXS channel number.
enable:0: disable, 1:enable Pulse detection for FXS

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pause_time: The threshold of the pause duration of the adjacent pulse digit
(msec)
min_break_ths: The threshold of the min. pulse break time (msec)
max_break_ths: The threshold of the max. pulse break time (msec)
Return Values: 0 success

6.6. FXO
As a caller, an FXO device initiates a call by going off-hook to seize the telephone line or DTMF
which identity the destination to be called. As a callee, an FXO device receives a call by
detecting the ring voltage supplied by the FXS device or going off-hook to answer the call.
The phone connecting to FXS can make a call to PSTN. This can be achieved by connecting to
the FXO. The FXO, infect, acts as a converter to transfer the data between FXS and PSTN. The
Figure 14 shows the conceptional architecture for FXS making call to PSTN. As the FXS makes
a call to PSTN, it sends the SIP message of INVITE to FXO. After the FXO received the INVITE
message, it calls the function of “rtk_FXO_RingOn” and the kernel will response with off_hook.
Calling the function of GetFXOEvent can get the status of the FXO. The FXO then responses
with the 200 OK SIP message and the FXS sends its ACK to FXO. The connection of RTP is built
up between FXO and FXS.

Figure 14. Internal connection for FXO to FXS

22

格式化:
字型:(
英文)
Ar
i
a
l

刪除: Figure 14

VoIP Programming Guide

Figure 15. The SIP negoriation for FXO and FXS

As the phone of the PSTN calls the phone of FXS in the ATA, the FXO will send the signal of
offhook and do the SIP negotiation with FXS. The connection of RTP between FXO and FXS is
built up.
The FXO acts as a sipua that can do negotiation with the other sipua such as softphone.

Figure 16. The FXO calls the sipua

The virtual DAA different form real DAA acts as relay between PSTN and FXS. The function of
getFXSEvent can get the five statuses. They are off-hook, on hook, DTMF, RingOn and RingOff.

Figure 17. Virtual DAA

23

VoIP Programming Guide

6.6.1.

Software API

Prototype:

int rtk_DAA_off_hook(uint32 chid)

Description:

Off-Hook in PSTN line

Parameters:

chid:The FXO channel number

Return Values: 0 success
Reference:

FXO

Prototype:

int rtk_DAA_on_hook(uint32 chid)

Description:

On-Hook in PSTN line

Parameters:

chid:The FXO channel number

Return Values: 0 success
Reference:

FXO

Prototype:

int32 rtk_Dial_PSTN_Call_Forward(uint32 chid, uint32 sid, char* cf_no_str)

Description:

Dial PSTN Number for PSTN Call Forward

Parameters:

chid:The FXO channel number
sid:The seesion number
*cf_no_str:The Pointer of the PSTN Number String

Return Values: 0 success
Reference:

FXO

Prototype:

int rtk_FXO_offhook(uint32 chid)

Description:

FXO:off-hook

Parameters:

chid:The FXO channel number

Return Values: 0 success
Reference:

FXO

Prototype:

int rtk_FXO_onhook(uint32 chid)

Description:

FXO on-hook

Parameters:

chid:The FXO channel number

Return Values: 0 success

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VoIP Programming Guide
Note:

Virtual DAA not support

Reference:

FXO

Prototype:

int rtk_FXO_RingOn(uint32 chid)

Description:

FXO Ring

Parameters:

chid:The FXO channel number

Return Values: 0 success
Note:

Virtual DAA not support

Reference:

FXO

Prototype:

int rtk_FXO_Busy(uint32 chid)

Description:

FXO Busy

Parameters:

chid:The FXO channel number

Return Values: 0 success
Note:

Virtual DAA not support

Reference:

FXO

Prototype:

int32 rtk_GetFxoLineVoltage(uint32 chid, uint32 *pval)

Description:

Get the FXO (DAA) Line Voltage

Parameters:

chid:The FXO channel number
pval: FXO line voltage

Return Values: 0 success
Reference:

6.7. FXO Caller ID
6.7.1.

Software API

Prototype:

int32 rtk_Get_DAA_CallerID(uint32 chid, char* str_cid, char* str_date)

Description:

Get the FXO Detected Caller ID

Parameters:

chid:The FXO channel number
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str_cid:The Caller ID String
str_date:The Date String
Return Values: 0 success
Note:

Virtual DAA not support.
How to get FXO Caller ID:
1. When application get FXO Ring event, then it’s safe to get Caller ID.
2. If Caller ID is set to send between 1st ring and 2nd ring, then DSP report FXO
Ring event till 2nd ring, to let application safe to get Caller ID.
3. If no Caller ID is detected, then str_cid[0] = str_date[0] = 0.

Reference:

FXO, rtk_Set_FSK_Area(), rtk_Set_CID_DTMF_MODE(),rtk_GetFxoEvent()

Prototype:

Int32 rtk_Set_CID_Det_Mode(uint32 chid, int auto_det, int cid_det_mode)

Description:

Set the FXO Caller ID Detection Mode

Parameters:

chid:The FXO channel number
auto_det: 0: Disable Caller ID Auto Detection, 1: Enable Caller ID Auto Detection
(NTT Support), 2: Enable Caller ID Auto Detection (NTT Not Support)
cid_det_mode:The Caller ID Mode for Caller ID Detection

Return Values: 0 success
Note:

Virtual DAA not support

Reference:

FXO

6.8. FXO Pulse Dial Generation
6.8.1.

Software API

Prototype:

int rtk_Set_Dail_Mode(uint32 chid, uint32 mode)

Description:

Set FXO dial mode

Parameters:

chid: The FXO channel number
mode: 0:disable, 1:enable Pulse Dial for FXO

Return Values: 0 success
Note:

Virtual DAA not support
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VoIP Programming Guide

Prototype:

int rtk_Get_Dail_Mode(uint32 chid)

Description:

Get FXO dial mode

Parameters:

chid: The FXO channel number

Return Values: 0: disable 1: enable Pulse dial
Note:

Virtual DAA not support

Prototype:

int rtk_PulseDial_Gen_Cfg(char pps, short make_duration, short
interdigit_duration)

Description:

Pulse dial generation config for FXO

Parameters:

pps: The pulse dial gen speed(Pulse per second)
make_duration l: Make duration of the pulse digit, unit:10 msec
interdigit_duration: The pause time between the pulse digit, unit:10 msec

Return Values: 0 success
Note:

Virtual DAA not support

Prototype:

int rtk_Gen_Pulse_Dial(uint32 chid, char digit)

Description:

Generate pulse dial for FXO

Parameters:

chid: The FXO channel number
digit: The pulse dial digit(0~9)

Return Values: 0 success
Note:

Virtual DAA not support

6.9. FXS Event
6.9.1.

Software API

Prototype:

int32 rtk_GetFxsEvent(uint32 chid, SIGNSTATE *pval)

Description:

Get the FXS Event

Parameters:

chid: The FXS channel number

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*pval: The pointer of variable to save FXS event
Return Values: 0 success, -1 unknown FXS event
Note:
Reference:

SIGNSTATE

6.10. FXO Event
6.10.1. Software API
Prototype:

int32 rtk_GetFxoEvent(uint32 chid, SIGNSTATE *pval)

Description:

Get the FXO Event

Parameters:

chid: The FXO channel number
*pval: The pointer of variable to save FXO event

Return Values: 0 success, -1 unknown FXO event
Note:

This API report the FXO event to application with event transformation.
l

Ring Start à SIGN_OFFHOOK

l

Ring Stop à SIGN_ONHOOK

l

Busy Tone à SIGN_ONHOOK

l

Dis-connect Tone à SIGN_ONHOOK

Reference:

SIGNSTATE

Prototype:

int32 rtk_GetRealFxoEvent(uint32 chid, FXOEVENT *pval)

Description:

Get the FXO Event

Parameters:

chid: The FXO channel number
*pval: The pointer of variable to save FXO event

Return Values: 0 success, -1 unknown FXO event
Note:

This API report the real FXO event to application.

Reference:

FXOEVENT

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6.11. FXS/FXO Line Status
6.11.1. Software API
Prototype:

int rtk_line_check(uint32 chid)

Description:

Check the FXS, FXO line status

Parameters:

chid: The FXS/FXO channel number

Return Values: For FXS:
0: Phone dis-connect,
1: Phone connect,
2: Phone off-hook,
3: Check time out ( may connect too many phone set => view as connect),
4: Can not check, Linefeed should be set to active state first.
For FXO:
0: PSTN Line connect,
1: PSTN Line not connect,
2: PSTN Line busy
Note:

The FXS line status is not always reliable. It may depend on SLIC, PCB layout,
length of line and number of Phones. So it’s just for reference. But FXO line
status is reliable.

Reference:

6.12. FXS/FXO DTMF Detection
6.12.1. Software API
Prototype:

int32 rtk_set_dtmf_det_threshold(uint32 chid, int32 threshold)

Description:

Set the DTMF detection threshold

Parameters:

chid: The FXS/FXO channel number
threshold: The threshold of DTMF detection. Range: 0 ~ 40, it means 0 ~ -40

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VoIP Programming Guide
dBm.
Return Values: 0 success
Note:

7.

If this API is not called, the default detection threshold of DSP is -32 dBm.

VoIP Session

The VoIP session is a part of VoIP interface. Each channel contains two sessions: Session0
and Session1. There are four modules in each session. The four modules are tone, RTP, DTMF
and FAX. The Figure 18 shows an example of the phone A initials a call to phone B with session
0, then phone A hold phone B and makes a call with session 1.

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VoIP Programming Guide

Figure 18-1 Phone A call with different sessions to phone B and Phone C

Figure 19-2 Each channel (port) contains two sessions: Session0 and Session1

7.1. Software API
Prototype:

Int32 rtk_SetTranSessionID(uint32 chid, uint32 sid)

Description:

Assign the active session

Parameters:

chid:The channel number
sid:The session number
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Return Values: 0 success
Reference:

Call Conference, FXO, RTP

7.2. VoIPSession-Tone
The tone table locates in "VoIP-ATA/linux-2.4.18/rtk_voip/voip_dsp/dsp_r0/dspparam.c”.
Developer can access the table and modify the setting of tones. The Table 5 shows the different
tones in tone table.
Number

Tone Name

1

Dial tone

2

Stutter-dial tone

3

Message waiting tone

4

Confirmation tone

5

Ring-back tone

6

Busy tone

7

Congestion tone

8

Roh tone

9

Double ringback tone

10

Sit-no circuit tone

11

Site-intercept tone

12

Sit-vacant tone

13

Sit-reorder tone

14

Calling cord with event tone

15

Calling card tone

16

Call waiting #1 tone

17

Call waiting #2 tone

18

Call waiting #3 tone

19

Call waiting #4 tone

20

Ingress ringback tone
Table 5 Tone table

The tone is various in different countries. The VoIP SDK pre-defined the tones using in these
countries. These countries include USA, UK, Australia, Hong Kong, Japan, Sweden, Germany,
32

VoIP Programming Guide
France, Taiwan, Belgium, Finland, Italy and China. Beside this users can set their custom tone
they like.

7.2.1.

Software API

Prototype:

int32 rtk_SetPlayTone(uint32 chid, uint32 sid, DSPCODEC_TONE nTone,
uint32 bFlag, DSPCODEC_TONEDIRECTION Path)

Description:

Play/Stop the assigned Tone

Parameters:

chid:The channel number
sid:The session number
nTone:The tone type
bFlag:1: play, 0: stop
Path:The tone direction

Return Values: 0 success
Reference:

Call Conference, FXO, Phone and RTP

Prototype:

int32 rtk_Set_Country(voipCfgParam_t* voip_ptr)

Description:

Set the country tone, busy tone detection parametes, and SLIC impedance
according to the Country

Parameters:

Voip_ptr The configuration of VoIP

Return Values: 0 success
Prototype:

int32 rtk_Set_Country_Tone (_COUNTRY_ country)

Description:

Set the Tone of Country

Parameters:

country: the selected country

Return Values: 0 success
Prototype:

int32 rtk_Set_Custom_Tone(uint8 custom, st_ToneCfgParam*
pstToneCfgParam)

Description:

Set the custom tone

Parameters:

custom:The custom index
pstToneCfgParam:The custom tone configuration

Return Values: 0 success

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VoIP Programming Guide

Prototype:

int32 rtk_Use_Custom_Tone(uint8 dial, uint8 ringback, uint8 busy, uint8 waiting)

Description:

Use the custom tone to be dial, ringback, busy, and call waiting tone

Parameters:

dial: Customer dial tone
ringback: customer ringing tone
busy: customer busy tone
waiting: customer call waiting tone

Return Values: 0 success
Prototype:

int32 rtk_SIP_INFO_play_tone(unsigned int chid, unsigned int sid,
DSPCODEC_TONE tone, unsigned int duration)

Description:

Play tone when receive SIP INFO

Parameters:

chid:The channel number
sid:The session number
tone:The tone need to play
duration:The tone duration (ms)

Return Values: 0 success

7.3. VoIPSession-RTP
The RTP configuration includes RTP and RTCP. For RTP, the developer can set the payload
type as g.911, g.723, g.726, g.729, GSM and iLBC etc. There are 5 states for session state.
They are rtp_session_undefined, rtp_session_inactive, rtp_session_sendonly,
rtp_session_recvonly and rtp_session_sendrecv. Both RTP sessions are transmitting when
session state is set to be rtp_session_sendrecv. In hold case, the holding peer will set
rtp_session_sendonly which means to stop receiving the voice data through RTP while the held
peer will set rtp_session_recvonly which means to stop sending the voice data through RTP. In
this case, only the held peer can hear the voice data form holding peer. The music on hold can
be achieved by this way.
The rtp_session_inactive can be used for both peers stop sending and receiving any voice data.
The structure of rtp_config is shown as Table 6.
Member

Variable type
34

Description

VoIP Programming Guide
Chid

uint32

Channel id

Sid

uint32

Session id

IsTcp

uint32

Tcp=1 udp=0

RemIp

uint32

Remote ip

RemPort

uint16

Remote prot

ExtIp

uint32

Local ip

ExtPort

uint16

Local port

Tos

uint8

QoS

rfc2833_payload_type_local

uint16

Local RFC2833 payload type

rfc2833_payload_type_remote

uint16

Remote RFC2833 payload
type

RemoteSrtpKey

unsigned char

Remote srtp key

LocalSrtpKey

unsigned char

Local srtp key

RemoteCryptAlg

Int

Remote crypt algorithm

LocalCryptAlg

Int

Local crypt algorithm

rtp_redundant_payload_type_local

uint16

local payload type of rtp
redundant

rtp_redundant_payload_type_remote uint16

remote payload type of rtp
redundant

Table 6. The variable of rtp configure

The member “rfc2833_payload_type_local” and “rfc2833_payload_type_remote” of this
structure MUST be set from application to make sure that the RFC2833 behavior of DSP works
fine. If local or remote side doesn’t support RFC2833, application should set it to ZERO,
otherwise, set to the supported payload type of local and remote side. Application should
negotiate with another VoIP user agent to decide the remote RFC2833 payload type.
The Realtek VoIP SDK supports both symmetric RTP and non-symmetric RTP. The transmitting
RTP port different form the receiving RTP port is said to be non-symmetric RTP.

7.3.1.

RTP Payload for Redundant Audio Data

This feature is an implementation of RFC 2198, and it uses redundant data (bandwidth) to deal
with packet loss. To enable this function, SIP uses a special media attribute called “red” to
negotiate, and it normally looks like:
35

VoIP Programming Guide
a=rtpmap: 121 red/8000/1
a=fmtp: 121 18/18
It says that we use dynamic payload type 121 as redundant RTP payload in RTP header
(“rtp_redundant_payload_type_local” and " rtp_redundant_payload_type_remote” in Table 7.),
and primary and secondary codecs are 18 and 18. In this example, static payload types (i.e. 18)
are assigned for codecs, but we can also assign dynamic payload types to them (e.g. iLBC=98).
This can reference to rtk_SetRtpPayloadType() to give suitable values. However, its limitation is
all redundant codecs to be the same, so SIP negotiation should guarantee this.

7.3.2.

Software API

Prototype:

int32 rtk_SetRtpConfig(rtp_config_t* cfg)

Description:

Set RTP configuration

Parameters:

cfg The RTP configuration

Return Values: 0 success
Reference:

Call Conference, FXO and RTP

Prototype:

int32 rtk_SetRtcpConfig(rtp_config_t *cfg, unsigned short rtcp_tx_interval)

Description:

Set RTCP configuration

Parameters:

cfg:The RTCP configuration
rtcp_tx_interval:The RTCP TX Interval

Return Values: 0 success
Prototype:

int32 rtk_SetRtpPayloadType(payloadtype_config_t* cfg)

Description:

Set RTP payload type

Parameters:

cfg:The RTP payload configuration

Return Values: 0 success
Reference:

Call Conference, FXO and RTP, payloadtype_config_t

Prototype:

int32 rtk_SetRtpSessionState(uint32 chid, uint32 sid, RtpSessionState state)

Description:

Set RTP Session State

Parameters:

chid:The channel number
sid:The session number

36

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VoIP Programming Guide
state:The RTP Session State
Return Values: 0 success
Reference:

Call Conference, FXO and RTP

Prototype:

int32 rtk_SetConference(TstVoipMgr3WayCfg* stVoipMgr3WayCfg)

Description:

Enable/Disable conference

Parameters:

stVoipMgr3WayCfg:The conference setting

Return Values: 0 success
Reference:

Call Conference

Prototype:

int32 rtk_Hold_Rtp(uint32 chid, uint32 sid, uint32 enable)

Description:

Hold/Resume RTP

Parameters:

chid:The channel number
sid:The session number
enable:1: Hold, 0: Resume

Return Values: 0 success
Reference:

Call Conference and RTP

Prototype:

int32 rtk_Get_Rtp_Statistics(uint32 chid, uint32 bReset, TstVoipRtpStatistics*
pstVoipRtpStatistics)

Description:

Get RTP statistics by channel

Parameters:

chid:The channel number
bReset:Reset statistics of the channel
pstVoipRtpStatistics:RTP statistics. (If bReset is set, statistics are all zeros.)

Return Values: 0 success

Prototype:

int32 rtk_Get_Session_Statistics( uint32 chid, uint32 sid, uint32 bReset,
TstVoipSessionStatistics *pstVoipSessionStatistics )

Description:

Get statistics by session

Parameters:

chid:The channel number
sid: The session number

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VoIP Programming Guide
bReset:Reset statistics of the session
pstVoipSessionStatistics: session statistics. (If bReset is set, statistics are all
zeros.)
Return Values: 0 success

7.4. VoIPSession-DTMF
There are three modes for DTMF. They are RFC2833, SIP INFO and inband. For RFC2833
mode, DTMF message is sent through RTP, while DTMF message is send through SIP info for
SIP info mode. The Figure 20 shows the SIP info message for DTMF 5 with an indicated duration
of 160 milliseconds.
INFO sip:7007471000@example.com SIP/2.0
Via: SIP/2.0/UDP alice.uk.example.com:5060
From: ;tag=d3f423d
To: ;tag=8942
Call-ID: 312352@myphone
CSeq: 5 INFO
Content-Length: 24
Content-Type: application/dtmf-relay
Signal=5
Duration=160
Figure 20 SIP info message for DTMF

7.4.1.

Software API

Prototype:

int32 rtk_SetDTMFMODE( uint32 chid, uint32 mode)

Description:

Set the DTMF mode

Parameters:

chid:The channel number
mode: The DTMF mode. (see __DTMF_TYPE__)

Return Values: 0 success
Reference:

Call Conference, FXO and RTP

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VoIP Programming Guide
Prototype:

int32 rtk_SetRFC2833SendByAP(uint32 chid, uint32 config)

Description:

Send RFC2833 by AP or DSP

Parameters:

chid:The channel number
config: 0: by DSP 1: by AP

Return Values: 0 success
Reference:

RFC2833 Send

Prototype:

int32 rtk_SetRTPRFC2833(uint32 chid, uint32 sid, uint32 digit)

Description:

Send DTMF via RFC2833

Parameters:

chid:The channel number
sid:The session number
digit:The digit of user input. (0..11, means 0..9,*,#)

Return Values: 0 success
Reference:

RFC2833 Send, Call Conference and RTP
格式化:
字型:1
2點

Prototype:

int32 rtk_LimitMaxRfc2833DtmfDuration(uint32 chid, uint32 duration_in_ms,
uint8 bEnable)

Description:

Limit the Max. RFC2833 DTMF Duration

Parameters:

chid:The channel number
duration: The limited Max. DTMF duration (ms)
bEnable 0: disable , 1: enable Max. limitation for DTMF duration

Return Values: 0 success
Reference:

RFC2833 Send, Call Conference and RTP

Prototype:

int32 rtk_SetRFC2833TxConfig (uint32 chid, uint32 volume, uint32 tx_mode)

Description:

Set the RFC2833 Tx packet volume field

Parameters:

chid:The channel number
volume: range from 0 to 31 dBm0
tx_mode: 0: DSP mode (RFC2833 packet will send automatically by DSP when
DTMF key is pressed), 1: application mode(RFC2833 packet send by
application)

Return Values: 0 success
39

VoIP Programming Guide
Reference:

RFC2833 Send

7.5. VoIPSession-FAX
There are two kind of FAX transmission. One is T.38 and the other is T.30. For T.38, after SIP
negotiation and the modem tone detection caller and callee will transmit through T.38 protocol.
For T.30, after SIP negotiation the caller and callee will transmit through G.711. The Figure 21
shows the SIP negotiation for T.38 protocol.
Having detected the CED tone, the callee sends the reinvite SIP message to the caller to
propose the T.38 transmission. The Figure 22 shows the SIP re-INVITE message for T.38 and
the Figure 23 shows the sample code for fax tone detection.

格式化:
字型:(
英文)
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格式化:
字型:(
英文)
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,
使
用拼字與文法檢查

刪除: Figure 22
格式化:
字型:(
英文)
Ar
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刪除: Figure 23

40

VoIP Programming Guide

Figure 21. T.38 SIP data flow

INVITE sip:+1-650-555-2222@ss1.wcom.com;user=phone SIP/2.0
Via: SIP/2.0/UDP iftgw.there.com:5060
From: sip:+1-303-555-1111@ift.here.com;user=phone
To: sip:+1-650-555-2222@ss1.wcom.com;user=phone
Call-ID: 1717@ift.here.com
CSeq: 56 INVITE
Content-Type: application/sdp
Content-Length: 320
41

VoIP Programming Guide

v=0
o=faxgw1 2890844527 2890844527 IN IP4 iftgw.there.com
s=Session SDP
c=IN IP4 iftmg.there.com
t=0 0
m=image 49172 udptl t38
a=T38FaxVersion:0
a=T38maxBitRate:14400
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:260
a=T38FaxUdpEC:t38UDPRedundancy
Figure 22. SIP re-INVITE message for T.38

uint32 faxFlag;
rtk_GetCEDToneDetect(0, faxFlag)
if (faxFlag == FAX_DETECT)
{
/*do re-INVITE*/
…
}
Figure 23. Do re-INVITE after fax detection

7.5.1.

Software API

Prototype:

int32 rtk_SetFaxModemDet(uint32 chid, uint32 mode)

Description:

Fax/Modem detection config

Parameters:

chid:The channel number
mode: 0:auto. 1: fax. 2:modem

Return Values: 0 success
Reference:

rtk_GetCEDToneDetect()
42

VoIP Programming Guide

Prototype:

int32 rtk_GetCEDToneDetect(uint32 chid, uint32 * pval)

Description:

FAX/Modem detection

Parameters:

chid:The channel number
pval:0: No Detected, 1: FAX detected, 2: Modem detected

Return Values: 0 success
Note:

When Fax/Modem detect mode is auto, detect result may be 0, 1, or 2.
When Fax/Modem detect mode is fax, detect result may be 0, or 1.
When Fax/Modem detect mode is modem, detect result may be 0, or 2.

Prototype:

int32 rtk_GetFaxDisTxDetect (uint32 chid, uint32 *pval)

Description:

FAX DIS(Digital Identification Signal) TX detection

Parameters:

chid:The channel number
pval: 1: DIS TX detected, 0: No DIS

Return Values: 0 success
Note:

DIS TX event path: ethernet à device à FAX machine

Prototype:

int32 rtk_GetFaxDisRxDetect (uint32 chid, uint32 *pval)

Description:

FAX DIS(Digital Identification Signal) RX detection

Parameters:

chid:The channel number
pval: 1: DIS RX detected, 0: No DIS

Return Values: 0 success
Note:

DIS RX event path: FAX machine à device à ethernet

Prototype:

int32 rtk_GetFaxEndDetect(uint32 chid, uint32 *pval)

Description:

FAX end detection

Parameters:

chid:The channel number
pval:1: FAX end detected, 0: No FAX end.

Return Values: 0 success

43

VoIP Programming Guide

8.

IVR

The interactive voice response (IVR) is the pre-recorded or the dynamically generated audio for
system information such as IP, default gateway and phone number. On the other hand user can
configure the system by passing the special combination of keys. This function is very useful
when web interface malfunctions.

8.1. Software API
Prototype:

int rtk_IvrStartPlaying(unsigned int chid, unsigned char* pszText2Speech)

Description:

Play a text speech

Parameters:

chid:The channel number
pszText2Speech:The text to speech

Return Values: Playing interval in unit of 10ms
Reference:

FXO

Prototype:

int rtk_IvrStartPlayG72363(unsigned int chid, unsigned int nFrameCount, const
unsigned char* pData)

Description:

Play a G.723 6.3k voice

Parameters:

chid:The channel number
nFrameCount:The number of frame to be decoded
pData:Point to data

Return Values: Copied frames, so less or equal to nFrameCount
Prototype:

int rtk_IvrPollPlaying(unsigned int chid)

Description:

Poll whether it is playing or not

Parameters:

chid:The channel number

Return Values: 1:Playing, 0:Stopped
Prototype:

int rtk_IvrStopPlaying(unsigned int chid)

Description:

Stop playing immediately

Parameters:

chid:The channel number
44

VoIP Programming Guide
Return Values: 0 Success
Reference:

9.

FXO

VoIP Security

There are two aspects for VoIP security. One is SIP over TLS transport and the other is security
RTP (SRTP) that is defined in RFC3711. The SDP embedded in SIP INVITE message contains
an attribute of crypto that indicates the algorithm of encryption and the master key for security
RTP. The key exchange protocol is defined in RFC4568. The crypto-suite is a negotiated while
the key parameter is a declarative parameter. The Figure 24 shows the crypto attribute of SDP.
The length of master key is 32, however, it is encoded by base64.
v=0
o=jdoe 2890844526 2890842807 IN IP4 10.47.16.5
s=SDP Seminar
i=A Seminar on the session description protocol
u=http://www.example.com/seminars/sdp.pdf
e=j.doe@example.com (Jane Doe)
c=IN IP4 161.44.17.12/127
t=2873397496 2873404696
m=video 51372 RTP/SAVP 31
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:d0RmdmcmVCspeEc3QGZiNWpVLFJhQX1cfHAwJSoj|2^20|1:32
m=audio 49170 RTP/SAVP 0
a=crypto:1 AES_CM_128_HMAC_SHA1_32
inline:NzB4d1BINUAvLEw6UzF3WSJ+PSdFcGdUJShpX1Zj|2^20|1:32
m=application 32416 udp wb
a=orient:portrait
Figure 24 SDP Secruity Descriptions

The following code will set the local master key, remote master key and negotiated algorithm.
TstVoipMgrSession stVoipMgrSession;
memcpy(stVoipMgrSession.remoteSrtpKey, cfg->remoteSrtpKey, SRTP_KEY_LEN);
memcpy(stVoipMgrSession.localSrtpKey, cfg->localSrtpKey, SRTP_KEY_LEN);

45

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英文)
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刪除: Figure 24

VoIP Programming Guide
stVoipMgrSession.remoteCryptAlg = cfg->negotiated_algorithm;
SETSOCKOPT(VOIP_MGR_SET_SESSION, &stVoipMgrSession, TstVoipMgrSession, 1);
Figure 25 SIP programming for setting local master key , remote master key and negotiated algorithm

9.1. SIP over TLS
SIP message will be encrypted over SSL protocol. As shown in Figure 27, the sip protocol
information is encrypted as application data and transmitted over SSL layer. According to RFC
3261, the TLS_RSA_WITH_AES_128_CBC_SHA ciphersuite must be supported at a minimum
by implementers when TLS is used in a SIP application. The Figure 26 shows the SIP REGISTER
message in TLS transport. The sample codes will show how to create SSL connection with SIP
server supported TLS. In fact, there are several steps for doing this:
1.

Create TCP connection with SIP Server.

2.

The sipua acts as client and the SIP server acts as server in SSL handshake.

3.

The sipua login the SIP server with its id and password.

REGISTER sip:192.168.1.218:5061 SIP/2.0
Via: SIP/2.0/TLS
192.168.1.215:5061;branch=z9hG4bK-d87543-f2542d2c010c3c2a-1--d87543-;rport
Max-Forwards: 70
Contact: 
To: "5119"
From: "5119";tag=740f277d
Call-ID: MTNlNzgwYzM4ZmYzNGMxNDllYTUxMGE5NDVhY2VhOWY.
CSeq: 1 REGISTER
Expires: 3600
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO
User-Agent: sip user agent
Content-Length: 0
Figure 26. SIP message for TLS transport

46

VoIP Programming Guide

Figure 27. Data flow of SSL

9.2. Secure RTP
The SRTP is defined in RFC3711. The algorithms are AES-128 counter mode and AES-128 f8
mode. For AES counter mode, the key stream is generated by a 128-bin integer which is
calculated as follows: (216 x the packet index) XOR (the salting key x 216) XOR (the SSRC x 264).
Figure 28 shows the key-stream generation process.

47

VoIP Programming Guide

Figure 28 AES counter mode

For AES F8 mode, the XOR of the session key and a salting key is generated, and it is used to
encrypt the initialization vector. If the salting key is less than 128 bits in length, it is padded with
alternating zeros and one (0x555…) to 128 bits. The result is known as the internal initialization
vector. The first block of the key stream is generated as the XOR of the internal initialization
vector and a 128-bit variable (j = 0), and the result is encrypted with the session key. The
variable j is incremented, and the second block of the key stream is generated as the XOR of the
internal initialization vector, the variable j, and the previous block of the key stream. The Figure 29
shows the AES F8 mode.

48

VoIP Programming Guide

Figure 29. AES F8 Mode

10.

VLAN

The RTL89xxc supports port-based, protocol-based, and tagged VLANs. VLAN processing is
controlled by a VLAN table and some related register fields. Both VLAN tagged and untagged
packets can be handled. A 802.1Q VLAN tag aware or unaware mode con be configured. VLAN
tag auto insertion and removal is set via the VLAN table. The VLAN table support up to 4K VLAN
entries. User can use following VLAN interfaces to set VLAN

10.1. Vlan Function API
Prototype:

int32 rtl865x_initVlanTable(void);
49

VoIP Programming Guide
Description:

Initialize the VLAN environment. The function enables VLAN function, and it
should be called before using VLAN function. After VLAN initialization, all VLAN
set are empty. The function should be called once.

Parameters:

None

Return Values: 0 success
Prototype:

int32 rtl865x_addVlan(uint16 vid);

Description:

Add a VLAN:The VID should 1 ~ 4095. It only add VLAN without giving its port
member. So user should call rtl865x_addVlanPortMember to set port member.
User should make sure the VID has not been added, or it will return VID-exist
error number.

Parameters:

Vid: vlan id

Return Values: 0 success
Prototype:

int32 rtl865x_delVlan(uint16 vid);

Description:

Delete a VLAN:Delete an existed VLAN and its port member set. If the VLAN
does not exit, it will return VID-not-exist error number.

Parameters:

Vid: vlan id

Return Values: 0 success
Prototype:

int32 rtl865x_reinitVlantable(void);

Description:

Reinitialize the VLAN environment:The function will delete all VLAN.

Parameters:

None

Return Values: 0 success
Prototype:

int32 rtl865x_addVlanPortMember(uint16 vid, uint32 portMask);

Description:

Add Port to a VLAN:The VLAN must exit before add it’s member port. Port should
be 0 ~ 4b
i
t
ma
s
k
.

Parameters:

Vid: vlan id
Portmask: port member for special vlan

Return Values: 0 success

50

VoIP Programming Guide
Prototype:

int32 rtl865x_delVlanPortMember(uint16 vid,uint32 portMask);

Description:

Delete Port to a VLAN: It only removes some ports from the existed VID’s member
set, and doesn’t effect other ports of the member port. If the removed ports are not
in the member port, the function has no effect and returns Success.

Parameters:

Vid: vlan id
Portmask: port member for special vlan

Return Values: 0 success
Prototype:

int32 rtl865x_setVlanPortTag(uint16 vid,uint32 portMask,uint8 tag);

Description:

Set tag/untag port:It sets member port as tag or un-tag port depending on tag
parameter. The tag port will add a VLAN tag on outgoing packets by it’s port VLAN
ID.

Parameters:

Vid: vlan id
Portmask: port member for special vlan
Tag: tag or untag

Return Values: 0 success
Prototype:

int32 rtl865x_setAsicPvid (uint32 portMask , uint16 vid);

Description:

It sets the vlan id for port

Parameters:

Portmask: port member for special vlan
Vid: vlan id

Return Values: 0 success
Prototype:

int 32 swNic_setVlanPortTag(int portmask);

Description:

Set tag id for packet from CPU: Add vlan tag for packet without vlan tag from
dev->hard_start_xmit.

Parameters:

Portmask: port member

Return Values: 0 success

51

VoIP Programming Guide

10.2. Vlan Example
User can echo parameter into /proc/rtk_hw_vlan_support to set swcore to add vlan tag for
packets to WAN port. Refenence Code is in “linux-2.6.30/drivers/net/rtl819x/rtl_nic.c.”
Pr
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/proc/rtk_hw_vlan_support

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11.

Example for Call Features Implementation

11.1. Caller ID Generation Example 1
#include "voip_manager.h"
void ShowUsage(char *cmd)
{
printf("usage: %s   [FSK area] [DTMF mode]\n" \
" - mode => 0 is DTMF, 1 is FSK\n" \
" - caller_id => caller id string\n" \
" - FSK area[2:0] => 0 BELLCORE, 1: ETSI, 2: BT, 3: NTT\n" \
" - FSK area[bit7]=> FSK date & time sync\n" \
" - FSK area[bit6]=> reverse polarity before caller id (For FSK)\n" \
52

VoIP Programming Guide
" - FSK area[bit5]=> short ring before caller id (For FSK)\n" \
" - FSK area[bit4]=> dual alert tone before caller id (For FSK)\n" \
" - FSK area[bit3]=> caller id Prior Ring (FSK & DTMF)\n" \
" - DTMF mode[1:0]=> Start digit, 0:A, 1:B, 2:C, 3:D\n" \
" - DTMF mode[3:2]=> End digit, 0:A, 1:B, 2:C, 3:D\n" \
" - DTMF mode[4]=> Auto start/end digit send, 0:suto mode 1:non-auto\n" \
"

(non-auto mode: DSP send caller ID string only. If caller ID need start/end digits,
developer should add them to caller ID strings.)\n", cmd);

exit(0);
}
int main(int argc, char *argv[])
{
unsigned int mode, fsk_area, dtmf_mode;
if (argc < 3)
{
ShowUsage(argv[0]);
}
mode = atoi(argv[1]);
switch (mode)
{
case 0:
if (argc == 5)
{
fsk_area = atoi(argv[3]);
dtmf_mode = atoi(argv[4]);
//for DTMF caller id prior ring or not. & set the caller id start/end digit
rtk_Set_CID_DTMF_MODE(0, fsk_area, dtmf_mode);
rtk_Gen_Dtmf_CID(0, argv[2]);
}
else if(argc == 4)

53

VoIP Programming Guide
{
fsk_area = atoi(argv[3]);
dtmf_mode = 0; // start/end digit : A.
//for DTMF caller id prior ring or not.
rtk_Set_CID_DTMF_MODE(0, fsk_area, dtmf_mode);
rtk_Gen_Dtmf_CID(0, argv[2]);
}
else if(argc == 3)
{
if (argv[2] == 1)
{
// for DTMF caller id priori 1st ring, auto send start/end digit
rtk_Set_CID_DTMF_MODE(0, 0x8, 0x8);
rtk_Gen_Dtmf_CID(0, "654321");
}
else if (argv[2] == 2)
{
// for DTMF caller id after 1st ring, non-auto
rtk_Set_CID_DTMF_MODE(0, 0, 0x10);
// DSP send caller ID string only. If caller ID need start/end digits,
developer should add them to caller ID strings
rtk_Gen_Dtmf_CID(0, "A654321C");
}
}
break;
case 1:
if (argc == 4)
{
fsk_area = atoi(argv[3]);
if ((fsk_area&7) < CID_DTMF)
{

54

VoIP Programming Guide
/*if ((fsk_area&7) == CID_FSK_NTT)
**{
** printf("not support NTT area currently\n");
** return 0;
**}
*/
rtk_Set_FSK_Area(0, fsk_area);
}
else
ShowUsage(argv[0]);
}
rtk_Gen_FSK_CID(0, argv[2], (void *) 0, (void *) 0, 0); // FSK TypeI
break;
default:
ShowUsage(argv[0]);
}
rtk_SetRingFXS(0, 1);
sleep(1);
rtk_SetRingFXS(0, 0);
return 0;
}

11.2. Caller ID Generation Example 2
#include "voip_manager.h"
int fskgen_main(int argc, char *argv[])
{
if (argc == 7)
{
if (atoi(argv[2]) == 1)//type I
{
55

VoIP Programming Guide
/* area -> 0:Bellcore 1:ETSI 2:BT 3:NTT */
rtk_Set_FSK_Area(atoi(argv[1])/*chid*/, atoi(argv[3])/*area*/);
#if 0

// This case is workable. Call rtk_Gen_FSK_CID() to send FSK Caller ID.
// API to gen FSK caller ID and auto ring by DSP
rtk_Gen_FSK_CID(atoi(argv[1]), argv[4]/*cid*/, argv[6]/*date_time*/,
argv[5]/*name*/, 0);

#elif 0

// This case is workable. Call rtk_Gen_CID_And_FXS_Ring () to send FSK Caller ID.
// API to gen caller ID and auto ring by DSP
rtk_Gen_CID_And_FXS_Ring(atoi(argv[1]), 1/*fsk*/, argv[4], argv[6] /*date_time*/,
argv[5]/*name*/, 0/*type1*/, 0);

#else

// This case is workable. Call rtk_Gen_MDMF_FSK_CID () to send FSK Caller ID.
TstFskClid clid;
clid.ch_id = atoi(argv[1]);
clid.service_type = 0; //service type 1
clid.cid_data[0].type = 0x01;
strcpy(clid.cid_data[0].data, argv[6]); //DATE
if ( (argv[4][0] == 'P') || (argv[4][0] == 'O') )

//Private or Out of area

clid.cid_data[1].type = 0x04;
else
clid.cid_data[1].type = 0x02;
strcpy(clid.cid_data[1].data, argv[4]); //CLI
if ( (argv[5][0] == 'P') || (argv[5][0] == 'O') )

//Private or Out of area

clid.cid_data[2].type = 0x08;
else
clid.cid_data[2].type = 0x07;
strcpy(clid.cid_data[2].data, argv[5]); //CLI_NAME
//Only 3 elements for Caller ID data. Set other element to 0 (MUST)

56

VoIP Programming Guide
clid.cid_data[3].type = 0;
clid.cid_data[4].type = 0;
rtk_Gen_MDMF_FSK_CID(atoi(argv[1])/*chid*/, &clid, 3);
#endif
}
else if (atoi(argv[2]) == 2)//type II
{
//DSP default is soft gen, no need to set soft gen mode
rtk_Set_FSK_Area(atoi(argv[1])/*chid*/, atoi(argv[3])/*area*/);

/* area -> 0:Bellcore

1:ETSI 2:BT 3:NTT */
#if 0
rtk_Gen_FSK_CID(atoi(argv[1])/*chid*/, argv[4]/*cid*/, argv[6], argv[5]/*name*/, 1);
#elif 0
rtk_Gen_CID_And_FXS_Ring(atoi(argv[1]), 1/*fsk*/, argv[4], argv[6],
argv[5]/*name*/, 1/*type1*/, 0);
#else
TstFskClid clid;
clid.ch_id = atoi(argv[1]);
clid.service_type = 1; //service type 2
clid.cid_data[0].type = 0x01;
strcpy(clid.cid_data[0].data, argv[6]); //DATE
if ( (argv[4][0] == 'P') || (argv[4][0] == 'O') )
clid.cid_data[1].type = 0x04;
else
clid.cid_data[1].type = 0x02;
strcpy(clid.cid_data[1].data, argv[4]); //CLI
if ( (argv[5][0] == 'P') || (argv[5][0] == 'O') )

57

VoIP Programming Guide
clid.cid_data[2].type = 0x08;
else
clid.cid_data[2].type = 0x07;
strcpy(clid.cid_data[2].data, argv[5]); //CLI_NAME
clid.cid_data[3].type = 0;
clid.cid_data[4].type = 0;
rtk_Gen_MDMF_FSK_CID(atoi(argv[1])/*chid*/, &clid, 3);
#endif
}
else
printf("wrong fsk type: should be type-I(1) or type-II(2)\n");
}
else if (argc == 5)// no name, date, time
{
if (atoi(argv[2]) == 1)//type I
{
/* area -> 0:Bellcore 1:ETSI 2:BT 3:NTT */
rtk_Set_FSK_Area(atoi(argv[1])/*chid*/, atoi(argv[3])/*area*/);
#if 0
rtk_Gen_FSK_CID(atoi(argv[1]), argv[4]/*cid*/, 0/*date*/, 0/*name*/, 0);
#elif 0
rtk_Gen_CID_And_FXS_Ring(atoi(argv[1]), 1/*fsk*/, argv[4], 0/*date*/, 0/*name*/,
0/*type1*/, 0);
#else
TstFskClid clid;
clid.ch_id = atoi(argv[1]);
clid.service_type = 0;

//service type 1

if ( (argv[4][0] == 'P') || (argv[4][0] == 'O') )

58

VoIP Programming Guide
clid.cid_data[0].type = 0x04;
else
clid.cid_data[0].type = 0x02;
strcpy(clid.cid_data[0].data, argv[4]); //CLI
clid.cid_data[1].type = 0;
clid.cid_data[2].type = 0;
clid.cid_data[3].type = 0;
clid.cid_data[4].type = 0;
rtk_Gen_MDMF_FSK_CID(atoi(argv[1])/*chid*/, &clid, 1);
#endif
}
else if (atoi(argv[2]) == 2)//type II
{
/* area -> 0:Bellcore 1:ETSI 2:BT 3:NTT */
rtk_Set_FSK_Area(atoi(argv[1])/*chid*/, atoi(argv[3])/*area*/);
#if 0
rtk_Gen_FSK_CID(atoi(argv[1])/*chid*/, argv[4]/*cid*/, 0/*date*/, 0/*name*/, 1);
#elif 0
rtk_Gen_CID_And_FXS_Ring(atoi(argv[1]), 1/*fsk*/, argv[4], 0/*date*/, 0/*name*/,
1/*type1*/, 0);
#else
TstFskClid clid;
clid.ch_id = atoi(argv[1]);
clid.service_type = 1;

//service type 2

if ( (argv[4][0] == 'P') || (argv[4][0] == 'O') )
clid.cid_data[0].type = 0x04;
else
clid.cid_data[0].type = 0x02;

59

VoIP Programming Guide
strcpy(clid.cid_data[0].data, argv[4]);
clid.cid_data[1].type = 0;
clid.cid_data[2].type = 0;
clid.cid_data[3].type = 0;
clid.cid_data[4].type = 0;
rtk_Gen_MDMF_FSK_CID(atoi(argv[1])/*chid*/, &clid, 1);
#endif
}
else
printf("wrong fsk type: should be type-I(1) or type-II(2)\n");
}
else if (argc == 11)
{
TstVoipFskPara para;
if (argv[1][0] == 's') //set para
{
para.ch_id = atoi(argv[2]);
格式化:
義大利文 (
義大利)

para.area = atoi(argv[3]);
para.CS_cnt = atoi(argv[4]);
para.mark_cnt = atoi(argv[5]);
para.mark_gain = atoi(argv[6]);
para.space_gain = atoi(argv[7]);
para.type2_expected_ack_tone = argv[8][0];
para.delay_after_1st_ring = atoi(argv[9]);
para.delay_before_2nd_ring = atoi(argv[10]);
rtk_Set_FSK_CLID_Para(¶);
}
}

60

VoIP Programming Guide
else if (argc == 4)
{
TstVoipFskPara para;
if (argv[1][0] == 'g') //get para
{
para.ch_id = atoi(argv[2]);
para.area = atoi(argv[3]);
rtk_Get_FSK_CLID_Para(¶);
printf("Get parameters:\n");
printf("=================================\n");
printf(" - ch seizure cnt = %d\n", para.CS_cnt);
printf(" - mark cnt = %d\n", para.mark_cnt);
printf(" - mark gain = %d\n", para.mark_gain);
printf(" - space gain = %d\n", para.space_gain);
printf(" - type2_expected_ack_tone = %c\n", para.type2_expected_ack_tone);
printf(" - delay_after_1st_ring = %d ms\n", para.delay_after_1st_ring);
printf(" - delay_before_2nd_ring = %d ms\n", para.delay_before_2nd_ring);
printf("=================================\n");
}
}
else
{
printf("Error! Usage:\n");
printf("To send Caller ID ==> fskgen 'chid' 'fsk_type' 'fsk_area' 'caller_id' 'name'
'date_time'\n");
printf("To set parameters ==> fskgen set 'chid' 'fsk_area' 'ch seizure cnt' 'mark cnt' 'mark gain'
'space gain' 'type-2 expexted ack tone' 'delay_after_1st_ring' 'delay_before_2nd_ring'\n");
printf("To set parameters ==> fskgen get 'chid' 'fsk_area'\n");

61

VoIP Programming Guide
printf("Example for Type 1:\n");
printf("'fskgen 0 1 128 035780211 tester_B 01020304' will display number 035780211 and
name, date, and time.(Bellcore)\n");
printf("'fskgen 0 1 129 035780211 tester_E 04030201' will display number 035780211 and
name, date, and time.(ETSI)\n");
printf("'fskgen 0 1 131 035780211 tester_N 06020602' will display number 035780211 and
name, date, and time.(NTT)\n");
printf("'fskgen 0 1 0 035780211' will just display number 035780211 without name, date, and
time.(Bellcore)\n");
printf("'fskgen 0 1 1 035780211' will just display number 035780211 without name, date, and
time.(ETSI)\n");
printf("'fskgen 0 1 3 035780211' will just display number 035780211 without name, date, and
time.(NTT)\n");
printf("Example for set Bellcore parameters:\n");
printf(" fskgen set 0 0 300 180 8 8 D 150 1000\n");
printf("Example for get ETSI parameters:\n");
printf(" fskgen get 0 1\n\n\n");
printf("Note:\n");
printf(" - fsk_type: 1 -> type1, 2 -> type2\n");
printf(" - fsk_area: 0 -> Bellcore, 1 -> ETSI, 2 -> BT, 3 -> NTT\n");
printf(" - gain 0 ~ gain 8: 8dB ~ 0dB\n");
printf(" - gain 9 ~ gain 24: -1dB ~ -16dB\n");
printf(" - type-2 expexted ack tone: character A or B or C or D\n");
}

return 0;
}

62

VoIP Programming Guide

11.3. RFC2833 Send
#include 
#include "voip_manager.h"
int send_2833_main(int argc, char *argv[])
{
if (argc == 5)
{
if (argv[1][0] = 'l')
rtk_LimitMaxRfc2833DtmfDuration(atoi(argv[2]), atoi(argv[3]), atoi(argv[4]));
else
rtk_SetRTPRFC2833(atoi(argv[1]), atoi(argv[2]), atoi(argv[3]), atoi(argv[4]));
}
else if (argc == 3)
{
rtk_SetRFC2833SendByAP(atoi(argv[1]), atoi(argv[2]));
}
else if (argc == 4)
{
rtk_SetRFC2833TxConfig(atoi(argv[1]), atoi(argv[2]), atoi(argv[3]));
}
else
{
printf("****** Usage ******\n");
printf("'send_2833 chid 0' to set RFC2833 sent by DSP.\n");
printf("'send_2833 chid 1' to set RFC2833 sent by application.\n");
printf("'send_2833 chid 10 0' set TX volume to -10 dBm and set RFC2833 sent by DSP.\n");
printf("'send_2833 chid 5 1' set TX volume to -5 dBm and set RFC2833 sent by
application.\n");
63

VoIP Programming Guide
printf("'send_2833 chid sid dtmf_event duration' to send RFC2833 event\n");
printf("Note: above tests are only workable when you set to RFC2833 mode, and during
VoIP call\n");
printf ("'send_2833 limit chid duration(ms) bEnable(0 or 1)' to config RFC2833 DTMF
duation max. limitation\n");
}
return 0;
}
Note: Application MUST set the proper RTP config via API rtk_SetRtpConfig() first to let RFC2833
mode work fine. See here for detail description.

11.4. Call Conference
#include "voip_manager.h"
#define MAX_SESSION (SESS_NUM / VOIP_CH_NUM)
enum {
STATE_IDLE = 0,
STATE_RTP
};
int main(int argc, char *argv[])
{
int chid, ssid;
SIGNSTATE val;
rtp_config_t rtp_config[SLIC_CH_NUM][MAX_SESSION];
payloadtype_config_t codec_config[SLIC_CH_NUM][MAX_SESSION];
TstVoipMgr3WayCfg stVoipMgr3WayCfg;
int ActiveSession[SLIC_CH_NUM];
int session_state[SLIC_CH_NUM][MAX_SESSION];
64

VoIP Programming Guide
int dtmf_val[SIGN_HASH + 1] = {0,1,2,3,4,5,6,7,8,9,0,10,11};
char dial_val[SIGN_HASH + 1] = {'\0', '1', '2', '3', '4', '5', '6', '7', '8', '9', '0', '.', '#'};
char dial_code[SLIC_CH_NUM][256];
int dial_index[SLIC_CH_NUM];
int i, j;
char *src_ip;
int src_port;
if (argc == 2)
{
src_ip = argv[1];
src_port = 9000;
}
else if (argc == 3)
{
src_ip = argv[1];
src_port = atoi(argv[2]);
}
else
{
printf("usage: %s src_ip [src_port]\n", argv[0]);
return 0;
}
// init
for (i=0; i ip: 192.168.1.1, rtp port:9002
pPort = strstr(dial_code[chid], "..");
if (pPort)
{
*pPort = 0;
pPort += 2;
dest_ip = dial_code[chid];
dest_port = atoi(pPort);
}
else
{
dest_ip = dial_code[chid];
dest_port = 9000;
}
// set rtp session
rtp_config[chid][ssid].chid = chid;

// use channel 0

rtp_config[chid][ssid].sid = ssid;
rtp_config[chid][ssid].isTcp = 0;

// use udp

rtp_config[chid][ssid].extIp = inet_addr(src_ip);
rtp_config[chid][ssid].remIp = inet_addr(dest_ip);
rtp_config[chid][ssid].extPort = htons(src_port + chid * 2);
rtp_config[chid][ssid].remPort = htons(dest_port);
rtp_config[chid][ssid].tos = 0;
rtp_config[chid][ssid].rfc2833_payload_type_local = 96;

67

VoIP Programming Guide
rtp_config[chid][ssid].rfc2833_payload_type_remote = 96;
rtk_SetRtpConfig(&rtp_config[chid][ssid]);
// set rtp payload, and other session parameters.
codec_config[chid][ssid].chid = chid;

// use channel 0

codec_config[chid][ssid].sid = ssid;
codec_config[chid][ssid].local_pt = 0;

// PCMU use pt 0

codec_config[chid][ssid].remote_pt = 0;

// PCMU use pt 0

codec_config[chid][ssid].uPktFormat = rtpPayloadPCMU;
codec_config[chid][ssid].nG723Type = 0;

// 6.3K

codec_config[chid][ssid].nFramePerPacket = 1;
codec_config[chid][ssid].bVAD = 0;
codec_config[chid][ssid].nJitterDelay = 4;
codec_config[chid][ssid].nMaxDelay = 13;
codec_config[chid][ssid].nJitterFactor = 7;
codec_config[chid][ssid].nG726Packing = 2; //pack right
rtk_SetRtpPayloadType(&codec_config[chid][ssid]);
// start rtp (channel number = 0, session number = 0)
rtk_SetRtpSessionState(chid, ssid, rtp_session_sendrecv);
session_state[chid][ssid] = STATE_RTP;
printf("%s:%d -> %s:%d\n",
src_ip, src_port + chid * 2, dest_ip, dest_port);
}
else
{
// stop dial tone in current session
rtk_SetPlayTone(chid, ssid, DSPCODEC_TONE_HOLD, 0,
DSPCODEC_TONEDIRECTION_LOCAL);
dial_code[chid][dial_index[chid]++] = dial_val[val];

68

VoIP Programming Guide
}
break;
}
case SIGN_OFFHOOK:
{
// do rtk_SLIC_Offhook_Action at first
rtk_Offhook_Action(chid);
rtk_SetTranSessionID(chid, 0);
ActiveSession[chid] = 0;
dial_index[chid] = 0;
for (i=0; i= SLIC_CH_NUM)

#define NULL

((void *) 0)

enum {
FXO_NO_DAA = 0,
FXO_VIRTUAL_DAA,
FXO_DAA
};

73

VoIP Programming Guide
enum {
STATE_IDLE = 0,
STATE_RING,
STATE_CONNECT,
STATE_DAA_RING,

// virtual daa state

STATE_DAA_CONNECT

// virtual daa state

};
enum {
DIAL_NONE = 0,
DIAL_FXO_0,
DIAL_FXS_0,
DIAL_FXS_1,
DIAL_OTHER,
};
typedef struct channel_s channel_t;
struct channel_s {
// phone info
int chid;
int state;
char dial_code[101];
int dial_index;
// rtp info
unsigned long ip;
unsigned short port;
// remote channel (just peer simulation)
channel_t *remote;
};
void channel_init(channel_t *channel, int chid);

74

VoIP Programming Guide
void channel_reset(channel_t *channel, int chid);
int channel_input(channel_t *channel, SIGNSTATE val);
void channel_dial_local(channel_t *channel, channel_t *remote_channel);
void channel_dial_out(channel_t *channel);
void channel_switch_to_fxo(channel_t *channel); // virtual daa
void channel_rtp_start(channel_t *channel);
void channel_rtp_stop(channel_t *channel);
// global variable
int fxo_type;
channel_t channels[VOIP_CH_NUM];
void channel_init(channel_t *channel, int chid)
{
memset(channel, 0, sizeof(*channel));
// phone
channel->chid = chid;
channel->state = STATE_IDLE;
channel->dial_code[0] = 0;
channel->dial_index = 0;
// rtp
channel->ip = ntohl(inet_addr("127.0.0.1"));
channel->port = 9000 + chid;
// reomte
channel->remote = NULL;
}
int channel_input(channel_t *channel, SIGNSTATE val)
{
if (channel->dial_index >= sizeof(channel->dial_code) - 1)
{
printf("%s: out of buffer\n", __FUNCTION__);

75

VoIP Programming Guide
return DIAL_NONE;
}
if (val >= SIGN_KEY1 && val <= SIGN_KEY9)
channel->dial_code[channel->dial_index++] = (char) '0' + val;

格式化:
義大利文 (
義大利)

else if (val == SIGN_KEY0)
channel->dial_code[channel->dial_index++] = '0';

格式化:
義大利文 (
義大利)

else if (val == SIGN_STAR)
channel->dial_code[channel->dial_index++] = '.';
else if (val == SIGN_HASH)
return DIAL_OTHER;
channel->dial_code[channel->dial_index] = 0;
// check local dial
if (strcmp(channel->dial_code, FUNCKEY_FXO_0) == 0)
return DIAL_FXO_0;
else if (strcmp(channel->dial_code, FUNCKEY_FXS_0) == 0)
return DIAL_FXS_0;
else if (strcmp(channel->dial_code, FUNCKEY_FXS_1) == 0)
return DIAL_FXS_1;
return DIAL_NONE;
}
void channel_dial_local(channel_t *channel, channel_t *remote_channel)
{
if (remote_channel->chid == channel->chid)
{
printf("%s: couldn't dial itself\n", __FUNCTION__);
rtk_SetPlayTone(channel->chid, 0, DSPCODEC_TONE_BUSY, 1,
DSPCODEC_TONEDIRECTION_LOCAL);

76

格式化:
義大利文 (
義大利)

VoIP Programming Guide
return;
}
// 1. create local rtp
channel->remote = remote_channel;
channel_rtp_start(channel);
channel->state = STATE_CONNECT;
// 2. signal remote channel
remote_channel->remote = channel;
if (CHANNEL_IS_FXS(remote_channel->chid))
{
rtk_SetPlayTone(channel->chid, 0, DSPCODEC_TONE_RINGING, 1,
DSPCODEC_TONEDIRECTION_LOCAL);
// signal fxs
rtk_SetRingFXS(remote_channel->chid, 1);
}
else
格式化:
英文 (
美國)

{
// signal fxo
rtk_FXO_RingOn(remote_channel->chid);
}
remote_channel->state = STATE_RING;
}
void channel_dial_out(channel_t *channel)
{
printf("%s: dial %s done.\n", __FUNCTION__, channel->dial_code);
rtk_SetPlayTone(channel->chid, 0, DSPCODEC_TONE_BUSY, 1,
DSPCODEC_TONEDIRECTION_LOCAL);

77

VoIP Programming Guide
}
void channel_switch_to_fxo(channel_t *channel)
{
if (rtk_DAA_off_hook(channel->chid) == 0)
{
printf("%s: ok.\n", __FUNCTION__);
}
else // pstn out failed
{
printf("%s: failed.\n", __FUNCTION__);
rtk_SetPlayTone(channel->chid, 0, DSPCODEC_TONE_BUSY, 1,
DSPCODEC_TONEDIRECTION_LOCAL);
}
}
void channel_rtp_start(channel_t *channel)
{
rtp_config_t rtp_config;
payloadtype_config_t codec_config;
// set rtp session
rtp_config.chid = channel->chid;
rtp_config.sid = 0;
rtp_config.isTcp = 0;

// use udp

rtp_config.extIp = htonl(channel->ip);
rtp_config.remIp = htonl(channel->remote->ip);
rtp_config.extPort = htons(channel->port);
rtp_config.remPort = htons(channel->remote->port);
rtp_config.tos = 0;
rtp_config.rfc2833_payload_type_local = 0;
rtp_config.rfc2833_payload_type_remote = 0;

78

// use in-band
// use in-band

VoIP Programming Guide
rtk_SetRtpConfig(&rtp_config);
// set rtp payload, and other session parameters.
codec_config.chid = channel->chid;
codec_config.sid = 0;
codec_config.local_pt = 0;
codec_config.remote_pt = 0;
codec_config.uPktFormat = rtpPayloadPCMU;
codec_config.nG723Type = 0;
codec_config.nFramePerPacket = 1;
codec_config.bVAD = 0;
codec_config.nJitterDelay = 4;
codec_config.nMaxDelay = 13;
codec_config.nJitterFactor = 7;
codec_config.nG726Packing = 2;//pack right
rtk_SetRtpPayloadType(&codec_config);
// start rtp
rtk_SetRtpSessionState(channel->chid, 0, rtp_session_sendrecv);
}
void channel_rtp_stop(channel_t *channel)
{
rtk_SetRtpSessionState(channel->chid, 0, rtp_session_inactive);
}
void channel_handle_input(channel_t *channel, SIGNSTATE val)
{
int ret;
rtk_SetPlayTone(channel->chid, 0, DSPCODEC_TONE_HOLD, 0,
DSPCODEC_TONEDIRECTION_LOCAL);

79

VoIP Programming Guide
ret = channel_input(channel, val);
switch (ret)
{
case DIAL_FXO_0:
if (fxo_type == FXO_NO_DAA)
{
printf("%s: no fxo\n", __FUNCTION__);
rtk_SetPlayTone(channel->chid, 0, DSPCODEC_TONE_BUSY, 1,
DSPCODEC_TONEDIRECTION_LOCAL);
return;
}
if (fxo_type == FXO_VIRTUAL_DAA)
{
channel_switch_to_fxo(channel);

// switch to fxo

}
else
{
channel_dial_local(channel, &channels[SLIC_CH_NUM]); // dial fxo 0
}
break;
case DIAL_FXS_0:
channel_dial_local(channel, &channels[0]);

// dial fxs 0

break;
case DIAL_FXS_1:
if (SLIC_CH_NUM < 2)
{
printf("%s: no fxs 1\n", __FUNCTION__);
rtk_SetPlayTone(channel->chid, 0, DSPCODEC_TONE_BUSY, 1,
DSPCODEC_TONEDIRECTION_LOCAL);
return;
}
channel_dial_local(channel, &channels[1]);

80

// dial fxs 1

VoIP Programming Guide
break;
case DIAL_OTHER:
channel_dial_out(channel);
break;
default:
// nothing
break;
}
}
void channel_handle_onhook(channel_t *channel)
{
switch (channel->state)
{
case STATE_CONNECT:
channel_rtp_stop(channel);
if (CHANNEL_IS_FXS(channel->chid))
{
// fxs
}
else
{
// daa
rtk_FXO_onhook(channel->chid);
}
// check remote channel is terminated
if (channel->remote->remote == NULL)
break;
// signal remote channel
if (CHANNEL_IS_FXS(channel->remote->chid))

81

VoIP Programming Guide
{
rtk_SetPlayTone(channel->remote->chid, 0, DSPCODEC_TONE_BUSY, 1,
DSPCODEC_TONEDIRECTION_LOCAL);
}
else
{
rtk_FXO_Busy(channel->remote->chid);
}
break;
case STATE_DAA_CONNECT:
// virtual daa
rtk_DAA_on_hook(channel->chid);
break;
default:
break;
}
#ifdef CONFIG_RTK_VOIP_IVR
rtk_IvrStopPlaying(channel->chid); // clear ivr
#endif
rtk_SetTranSessionID(channel->chid, 255); // clear resource count
rtk_SetPlayTone(channel->chid, 0, DSPCODEC_TONE_HOLD, 0,
DSPCODEC_TONEDIRECTION_LOCAL);

// clear tone

channel_init(channel, channel->chid);
}
void channel_handle_offhook(channel_t *channel)
{
// active session 0, and record resource + 1
rtk_SetTranSessionID(channel->chid, 0);
switch (channel->state)

82

VoIP Programming Guide
{
case STATE_IDLE:
if (CHANNEL_IS_FXS(channel->chid))
{
// fxs
rtk_SetPlayTone(channel->chid, 0, DSPCODEC_TONE_DIAL, 1,
DSPCODEC_TONEDIRECTION_LOCAL);
}
else
{
char caller_id[21];
char caller_date[9];
char caller_name[51];
#ifdef FXO_DIAL_FXS0_AUTO
rtk_Get_DAA_CallerID(channel->chid, caller_id, caller_date, caller_name);
if (caller_id[0])
printf("caller id is %s\n", caller_id);
else
printf("no caller id detect\n");
if (caller_name[0])
printf("caller name is %s\n", caller_name);
channel_dial_local(channel, &channels[0]); // dial fxs 0
#else
char text[] = {IVR_TEXT_ID_PLZ_ENTER_NUMBER, '\0'};
usleep(500000); // after phone off-hook, wait for a second, and then play IVR
rtk_FXO_offhook(channel->chid);
#ifdef CONFIG_RTK_VOIP_IVR
rtk_IvrStartPlaying(channel->chid, text);

83

VoIP Programming Guide
#else
// use dial tone to replace if no IVR
rtk_SetPlayTone(channel->chid, 0, DSPCODEC_TONE_DIAL, 1,
DSPCODEC_TONEDIRECTION_LOCAL);
#endif
#endif
}
break;
case STATE_RING:
if (CHANNEL_IS_FXS(channel->chid))
{
// fxs
rtk_SetRingFXS(channel->chid, 0);
rtk_SetPlayTone(channel->remote->chid, 0, DSPCODEC_TONE_RINGING, 0,
DSPCODEC_TONEDIRECTION_LOCAL);
channel_rtp_start(channel);
channel->state = STATE_CONNECT;
}
else
{
// daa
rtk_FXO_offhook(channel->chid);
channel_rtp_start(channel);
channel->state = STATE_CONNECT;
}
// ack remote channel
if (CHANNEL_IS_FXS(channel->remote->chid))
{
// fxs
}
else

84

VoIP Programming Guide
{
// daa
rtk_FXO_offhook(channel->remote->chid);
}
break;
case STATE_DAA_RING:
// virtual daa
rtk_DAA_off_hook(channel->chid);
channel->state = STATE_DAA_CONNECT;
break;
default:
break;
}
}
int main()
{
int i;
SIGNSTATE val;
channel_t *channel;
rtk_Get_VoIP_Feature();
if ((g_VoIP_Feature & DAA_TYPE_MASK) == NO_DAA)
fxo_type = FXO_NO_DAA;
else if ((g_VoIP_Feature & DAA_TYPE_MASK) == VIRTUAL_DAA)
fxo_type = FXO_VIRTUAL_DAA;
else
fxo_type = FXO_DAA;
// init
for (i=0; istate == STATE_IDLE)
{
channel_handle_input(channel, val);
}
break;
case SIGN_ONHOOK:
channel_handle_onhook(channel);
rtk_Onhook_Action(i); // rtk_Onhook_Action have to be the last step in
SIGN_ONHOOK
break;
case SIGN_OFFHOOK:
rtk_Offhook_Action(i); // rtk_Offhook_Action have to be the first step in
SIGN_OFFHOOK
channel_handle_offhook(channel);
break;
case SIGN_RING_ON:

// virtual daa ring on

if (channel->state == STATE_IDLE)
{
channel->state = STATE_DAA_RING;
rtk_SetRingFXS(channel->chid, 1);
}
break;
case SIGN_RING_OFF:

// virtual daa ring off

if (channel->state == STATE_DAA_RING)
{
rtk_SetRingFXS(channel->chid, 0);
channel->state = STATE_IDLE;
}
break;
case SIGN_FLASHHOOK:
break;
case SIGN_OFFHOOK_2: // off-hook but no event

88

VoIP Programming Guide
break;
case SIGN_NONE:
break;
default:
printf("unknown(%d)\n", val);
break;
}
usleep(100000 / VOIP_CH_NUM); // 100ms
}
goto main_loop;
return 0;
}

11.6. phone
#include "voip_manager.h"
int main()
{
int i;
SIGNSTATE val;
for (i=0; i %s:%d\n",

94

VoIP Programming Guide
argv[2], 9000 + chid, argv[3], 9000 + (chid ? 0 : 1));
break;
case SIGN_ONHOOK:
// close rtp
rtk_SetRtpSessionState(chid, 0, rtp_session_inactive);
rtk_SetRtpSessionState(chid, 1, rtp_session_inactive);
bStartRTP = 0;
// call rtk_Offhook_Action at last
rtk_Onhook_Action(chid);
break;
case SIGN_FLASHHOOK:
ActiveSession = !ActiveSession;
rtk_SetTranSessionID(chid, ActiveSession);
if (ActiveSession == 1)
{
// hold session
rtk_Hold_Rtp(chid, 0, 1);
// play dial tone
rtk_SetPlayTone(chid, 1, DSPCODEC_TONE_DIAL, 1,
DSPCODEC_TONEDIRECTION_LOCAL);
}
else
{
// resume session
rtk_Hold_Rtp(chid, 0, 0);
// stop dial tone
rtk_SetPlayTone(chid, 1, DSPCODEC_TONE_HOLD, 0,
DSPCODEC_TONEDIRECTION_LOCAL);
}
break;

95

VoIP Programming Guide

default:
break;
}
usleep(100000); // 100ms
}
return 0;
}

11.9. vmwi
#include "voip_manager.h"
int main()
{
char vmwi_on = 1, vmwi_off = 0;
// VMWI test via FSK Type I
rtk_Gen_FSK_VMWI(0, &vmwi_on, 0);
sleep(2);
rtk_Gen_FSK_VMWI(0, &vmwi_off, 0);
return 0;
}

11.10. TLS REGISTER
#include 
#include 
#include 
#include 

96

VoIP Programming Guide
#include 
#include 
#include 
#include 
#include 
#include 
#include 
#include 
格式化:
義大利文 (
義大利)

#define CA_LIST "cacert.pem"
BIO *bio_err=0;
static void sigpipe_handle(int x){
printf("+++++%s+++++\n",__FUNCTION__);
}
/* Print SSL errors and exit*/
int berr_exit(char *string)
{
BIO_printf(bio_err,"%s\n",string);
ERR_print_errors(bio_err);
exit(0);
}
SSL_CTX *initialize_ctx(){
SSL_METHOD *meth;
SSL_CTX *ctx;
if(!bio_err){
/* Global system initialization*/

97

VoIP Programming Guide
SSL_library_init();
SSL_load_error_strings();
/* An error write context */
bio_err=BIO_new_fp(stderr,BIO_NOCLOSE);
}
/* Set up a SIGPIPE handler */
signal(SIGPIPE,sigpipe_handle);
/* Create our context*/
meth=TLSv1_method();
ctx=SSL_CTX_new(meth);
/* Load the CAs we trust*/
if(!(SSL_CTX_load_verify_locations(ctx,CA_LIST,0)))
berr_exit("Can't read CA list");
#if (OPENSSL_VERSION_NUMBER < 0x00905100L)
SSL_CTX_set_verify_depth(ctx,1);
#endif
return ctx;
}
char* genRegMsg(){
char *sipMsg;
char *sipurl="REGISTER sip:192.168.1.218:5061 SIP/2.0\r\n\0";
char *via="Via: SIP/2.0/TLS
192.168.1.200:5061;rport;branch=z9hG4bK-d87543-d641201b415f0331-1--d87543\r\n\0";
char *forward="Max-Forwards: 70\r\n\0";
char *contact="Contact:
\r\n\0";
char *to="To: \"4487\"\r\n\0";

98

VoIP Programming Guide
char *from="From: \"4487\";tag=7a05a24e\r\n\0";
char *callid="Call-ID: 633697324@192.168.1.200\r\n\0";
char *cseq="CSeq: 13 REGISTER\r\n\0";
char *expire="Expires: 3600\r\n\0";
char *allow="Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO\r\n\0";
char *userAgent="User-Agent: eyeBeam\r\n\0";
char *content="Content-Length: 0\r\n\0";
sipMsg = (char*)malloc(sizeof(char)*4000);
sprintf(sipMsg,"%s%s%s%s%s%s%s%s%s%s%s%s\r\n\0",
sipurl,via,forward,contact,to,from,callid,cseq,expire,allow,userAgent,content);
return sipMsg;
}
int tlsSipSendMsg(char *sipMsg, SSL *ssl, int len){
int ret;
ret=SSL_write(ssl, sipMsg, len);
if(ret < 0)
printf("+++++SSL_write error!+++++\n");
return ret;
}
int main(int argc, char *argv[])
{
int tcpSocket;
char *sipRegMsgStr;
SSL_CTX *ctx;
SSL *ssl;
BIO *sbio;
struct sockaddr_in addr;

99

VoIP Programming Guide

if(argc != 2){
printf("usage: reg \n");
exit(1);
}
/* Build our SSL context*/
ctx=initialize_ctx();
/*build socket*/
tcpSocket = socket(PF_INET, SOCK_STREAM, IPPROTO_TCP);
if(tcpSocket == -1)
perror("open socket error");
addr.sin_family = PF_INET;
addr.sin_addr.s_addr=inet_addr(argv[1]);
addr.sin_port = htons(5061);
connect(tcpSocket, (struct sockaddr*)&addr, sizeof(addr));
/* Connect the SSL socket */
ssl=SSL_new(ctx);
sbio=BIO_new_socket(tcpSocket,BIO_NOCLOSE);
SSL_set_bio(ssl,sbio,sbio);
if(SSL_connect(ssl)<=0)
berr_exit("SSL connect error");
sipRegMsgStr = genRegMsg();
#if 1
printf("---send sip message---\n%s",sipRegMsgStr);
tlsSipSendMsg(sipRegMsgStr, ssl, strlen(sipRegMsgStr));
while(1){
int i;

100

VoIP Programming Guide
char *buf=NULL;
int bufSize=0;
buf = (char*)malloc(4000*sizeof(char));
if((bufSize=SSL_read(ssl, buf, 4000)) == 0){
free(buf);
continue;
}else{
printf("---recv sip message---\n%s",buf);
free(buf);
}
break;
}
#endif
free(sipRegMsgStr);
SSL_CTX_free(ctx);
close(tcpSocket);
return 0;
}

11.11. Pulse Dial Generation/Detection
#include "voip_manager.h"
int pulse_dial_main(int argc, char *argv[])
{
int chid, i, enable;
unsigned int pause_time, max_break_ths, min_break_ths;
if (argc < 2)
goto PULSE_DIAL_USAGE;

101

VoIP Programming Guide
if (argv[1][0] == 'g') // Pulse Dial Generation
{
if (argc == 4)
{
if (argv[1][1] == 'm')
{
rtk_Set_Dail_Mode(atoi(argv[2]), atoi(argv[3]));
if (atoi(argv[2]) == 1)
printf("Enable pulse dial gen for ch=%d\n", chid);
else if (atoi(argv[2]) == 0)
printf("Disable pulse dial gen for ch=%d\n", chid);
}
else if (argv[1][1] == 'g')
{
int sum = 0;
int len = 0;
rtk_debug(0); // disable kernel debug message to avoid effect
pulse dial
chid = atoi(argv[2]);
len = strlen(argv[3]);
rtk_FXO_offhook(chid);
sleep(2);
for (i=0; i < len; i++)
{
//printf("gen %d\n", argv[2][i] - '0');
rtk_Gen_Pulse_Dial(chid, argv[3][i] - '0');
sum = sum + (argv[3][i] - '0');
}

102

VoIP Programming Guide
usleep(100000*sum + 1000*800*(len-1)); // sleep enough time
to make sure pulse dial gen finish, assume interdigit_duration = 800 ms
sleep(2);
rtk_FXO_onhook(chid);
rtk_debug(3); //enable kernel debug message
}
}
else if (argc == 5)
{
rtk_PulseDial_Gen_Cfg(atoi(argv[2]), atoi(argv[3]), atoi(argv[4]));
printf("Config pulse dial gen to %d pps, make: %d ms, interdigit
duration:%d ms\n", atoi(argv[2]), atoi(argv[3]), atoi(argv[4]));
}
else
{
printf("usage error!\n\n");
goto PULSE_DIAL_USAGE;
}
}
else if (argv[1][0] == 'd') // Pulse Dial Detection
{
if (argc == 7)
{
chid = atoi(argv[2]);
enable = atoi(argv[3]);
pause_time = atoi(argv[4]);
min_break_ths = atoi(argv[5]);
max_break_ths = atoi(argv[6]);
rtk_Set_Pulse_Digit_Det(chid, enable, pause_time, min_break_ths,

103

VoIP Programming Guide
max_break_ths);
if (enable == 1)
{
printf("Enable pulse dial det for ch= %d, ", chid);
printf("pause_time= %d ms, min_break_ths = %d ms,
max_break_ths = %d ms\n", pause_time, min_break_ths, max_break_ths);
}
else if (enable == 0)
printf("Disable pulse dial det for ch= %d\n", chid);
}
else
{
printf("usage error!\n\n");
goto PULSE_DIAL_USAGE;
}
}
else
{
PULSE_DIAL_USAGE:
printf("pulse_dial usage:\n");
printf(" - Enable/Disable pulse dial generation: pulse_dial gm chid flag\n");
printf("

+ Example to enable pulse dial gen: pulse_dial g 2 1\n");

printf(" - Enable/Disable pulse dial detection: pulse_dial d chid flag pause_time
min_break_ths max_break_ths\n");
printf("

+ Example to enable pulse dial det: pulse_dial d 0 1 450 20 70\n");

printf(" - Config pulse dial generation: pulse_dial g pps make_time
interdigit_duration\n");
printf("

+ Example1: pulse_dial g 10 40 700\n");

printf("

+ Example2: pulse_dial g 20 20 750\n");

printf(" - Generate pulse dial : pulse_dial gg chid digit(0~9)\n");
printf("

+ Example1 to gen digit 3939889: pulse_dial gg 2 3939889\n");

104

VoIP Programming Guide
return -1;
}
return 0;
格式化:
字型:(
中文)新細明
體

}

12.

Appendix A – Enumeration

12.1. Enumeration for phone state
typedef enum
{
SIGN_NONE = 0 ,
SIGN_KEY1 = 1 ,
SIGN_KEY2 = 2 ,
SIGN_KEY3 = 3 ,
SIGN_KEY4 = 4,
SIGN_KEY5 = 5 ,
SIGN_KEY6 = 6 ,
SIGN_KEY7 = 7 ,
SIGN_KEY8 = 8 ,
SIGN_KEY9 = 9 ,
SIGN_KEY0 = 10 ,
SIGN_STAR = 11,
SIGN_HASH = 12 ,
SIGN_ONHOOK = 13 ,
SIGN_OFFHOOK = 18,
SIGN_FLASHHOOK = 19,
SIGN_AUTODIAL = 20,
SIGN_OFFHOOK_2 = 21,
105

VoIP Programming Guide
SIGN_RING_ON = 22,
SIGN_RING_OFF = 23
} SIGNSTATE ;

12.2. Enumeration for FXO Event
typedef enum
{
FXOEVENT_NONE = 0 ,
FXOEVENT_KEY1 = 1 ,
FXOEVENT_KEY2 = 2 ,
FXOEVENT_KEY3 = 3 ,
FXOEVENT_KEY4 = 4,
FXOEVENT_KEY5 = 5 ,
FXOEVENT_KEY6 = 6 ,
FXOEVENT_KEY7 = 7 ,
FXOEVENT_KEY8 = 8 ,
FXOEVENT_KEY9 = 9 ,
FXOEVENT_KEY0 = 10 ,
FXOEVENT_STAR = 11,
FXOEVENT_HASH = 12 ,
FXOEVENT_RING_START = 13 ,
FXOEVENT_RING_STOP = 14,
//FXOEVENT_PSTN_OFFHOOK = 15,
//FXOEVENT_PSTN_ONHOOK = 16,

// not support yet
// not support yet

FXOEVENT_BUSY_TONE = 17,
FXOEVENT_DIST_TONE = 18,
FXOEVENT_POLARITY_REVERSAL = 19,
FXOEVENT_BAT_DROP_OUT = 20,
FXOEVENT_MAX = 21
} FXOEVENT;

106

VoIP Programming Guide

12.3. Enumeration for supported RTP payload
typedef enum
{
rtpPayloadUndefined = -1,
rtpPayloadPCMU = 0,
rtpPayload1016 = 1,
rtpPayloadG726_32 = 2,
rtpPayloadGSM = 3,
rtpPayloadG723 = 4,
rtpPayloadDVI4_8KHz = 5,
rtpPayloadDVI4_16KHz = 6,
rtpPayloadLPC = 7,
rtpPayloadPCMA = 8,
rtpPayloadG722 = 9,
rtpPayloadL16_stereo = 10,
rtpPayloadL16_mono = 11,
rtpPayloadQCELP = 12,
rtpPayloadMPA = 14,
rtpPayloadG728 = 15,
rtpPayloadDVI4_11KHz = 16,
rtpPayloadDVI4_22KHz = 17,
rtpPayloadG729 = 18,
rtpPayloadCN = 19,
rtpPayloadG726_40

= 21,

rtpPayloadG726_24

= 22,

rtpPayloadG726_16

= 23,

rtpPayloadCelB = 25,
rtpPayloadJPEG = 26,
rtpPayloadNV = 28,
rtpPayloadH261 = 31,
rtpPayloadMPV = 32,

107

VoIP Programming Guide
rtpPayloadMP2T = 33,
rtpPayloadH263 = 34,
// fake codec
rtpPayloadSilence = 35,
// dynamic payload type in rtk_voip
rtpPayload_iLBC = 97,
rtpPayload_iLBC_20ms = 98,
rtpPayloadDTMF_RFC2833 = 101,
rtpPayloadT38_Virtual = 110,
rtpPayloadCiscoRtp = 121,
rtpPayloadL16_8k_mono = 122,
rtpPayload_AMR_NB = 123,
rtpPayload_SPEEX_NB_RATE8 = 111,
rtpPayload_SPEEX_NB_RATE2P15 = 112,
rtpPayload_SPEEX_NB_RATE5P95 = 113,
rtpPayload_SPEEX_NB_RATE11 = 114,
rtpPayload_SPEEX_NB_RATE15 = 115,
rtpPayload_SPEEX_NB_RATE18P2 = 116,
rtpPayload_SPEEX_NB_RATE24P6 = 117,
rtpPayload_SPEEX_NB_RATE3P95 = 118,
} RtpPayloadType;

12.4. Enumeration for RTP session state
typedef enum
{
rtp_session_undefined = -1,
rtp_session_inactive = 0,
rtp_session_sendonly = 1,
rtp_session_recvonly = 2,
rtp_session_sendrecv = 3
108

VoIP Programming Guide
} RtpSessionState;

12.5. Enumeration for country
typedef enum
{
DSPCODEC_COUNTRY_USA,
DSPCODEC_COUNTRY_UK,
DSPCODEC_COUNTRY_AUSTRALIA,
DSPCODEC_COUNTRY_HK,
DSPCODEC_COUNTRY_JP,
DSPCODEC_COUNTRY_SE,
DSPCODEC_COUNTRY_GR,
DSPCODEC_COUNTRY_FR,
DSPCODEC_COUNTRY_TR,
DSPCODEC_COUNTRY_BE,
DSPCODEC_COUNTRY_FL,
DSPCODEC_COUNTRY_IT,
DSPCODEC_COUNTRY_CN,
DSPCODEC_COUNTRY_CUSTOME
} DSPCODEC_COUNTRY;

12.6. Enumeration for DSP tone
typedef enum
{
DSPCODEC_TONE_NONE = -1,
DSPCODEC_TONE_0 = 0,
DSPCODEC_TONE_1,
DSPCODEC_TONE_2,
DSPCODEC_TONE_3,
DSPCODEC_TONE_4,
DSPCODEC_TONE_5,

109

VoIP Programming Guide
DSPCODEC_TONE_6,
DSPCODEC_TONE_7,
DSPCODEC_TONE_8,
DSPCODEC_TONE_9,
DSPCODEC_TONE_STARSIGN,

// *

DSPCODEC_TONE_HASHSIGN,

// #

DSPCODEC_TONE_DIAL,

// code = 66 in RFC2833

DSPCODEC_TONE_STUTTERDIAL,

格式化:
英文 (
美國)

//

DSPCODEC_TONE_MESSAGE_WAITING,
DSPCODEC_TONE_CONFIRMATION,

//
//

DSPCODEC_TONE_RINGING, /* ring back tone */ // code = 70 in RFC2833
DSPCODEC_TONE_BUSY,

// code = 72 in RFC2833

DSPCODEC_TONE_CONGESTION,
DSPCODEC_TONE_ROH,

//
//

DSPCODEC_TONE_DOUBLE_RING,

//

DSPCODEC_TONE_SIT_NOCIRCUIT,

//

DSPCODEC_TONE_SIT_INTERCEPT,

//

DSPCODEC_TONE_SIT_VACANT,

//

DSPCODEC_TONE_SIT_REORDER,

//

DSPCODEC_TONE_CALLING_CARD_WITHEVENT,

//

DSPCODEC_TONE_CALLING_CARD,

//

DSPCODEC_TONE_CALL_WAITING,

// code = 79 in RFC2833

DSPCODEC_TONE_CALL_WAITING_2,

//

DSPCODEC_TONE_CALL_WAITING_3,

//

DSPCODEC_TONE_CALL_WAITING_4,

//

DSPCODEC_TONE_INGRESS_RINGBACK,
DSPCODEC_TONE_HOLD,

// code = 76 in RFC2833

DSPCODEC_TONE_OFFHOOKWARNING,
DSPCODEC_TONE_RING,

//

// code = 88 in RFC2833

// code = 89 in RFC2833

#ifdef SW_DTMF_CID
DSPCODEC_TONE_A,

// DTMF digit A (19)

110

VoIP Programming Guide
DSPCODEC_TONE_B,

// DTMF digit B (20)

DSPCODEC_TONE_C,

// DTMF digit C (21)

DSPCODEC_TONE_D,

// DTMF digit D (22)

#endif
DSPCODEC_TONE_FSK_SAS,

// alert signal (23)

DSPCODEC_TONE_FSK_ALERT,

// alert signal (23)

DSPCODEC_TONE_NTT_IIT_TONE,

// for ntt type 2 caller id

// continous DTMF tone play for RFC2833
DSPCODEC_TONE_0_CONT,
DSPCODEC_TONE_1_CONT,
DSPCODEC_TONE_2_CONT,
DSPCODEC_TONE_3_CONT,
DSPCODEC_TONE_4_CONT,
DSPCODEC_TONE_5_CONT,
DSPCODEC_TONE_6_CONT,
DSPCODEC_TONE_7_CONT,
DSPCODEC_TONE_8_CONT,
DSPCODEC_TONE_9_CONT,
DSPCODEC_TONE_STARSIGN_CONT,
DSPCODEC_TONE_HASHSIGN_CONT,
DSPCODEC_TONE_A_CONT,
DSPCODEC_TONE_B_CONT,
DSPCODEC_TONE_C_CONT,
DSPCODEC_TONE_D_CONT,
DSPCODEC_TONE_KEY

// the others key tone

} DSPCODEC_TONE;

12.7. Enumeration for DSP play tone direction
typedef enum
{
111

VoIP Programming Guide
DSPCODEC_TONEDIRECTION_LOCAL,

// local

DSPCODEC_TONEDIRECTION_REMOTE,

// remote

DSPCODEC_TONEDIRECTION_BOTH

// local and remote

} DSPCODEC_TONEDIRECTION;

12.8. Enumeration for IVR play
typedef enum {
IVR_PLAY_TYPE_TEXT,

/* play a string, such as '192.168.0.0' */

IVR_PLAY_TYPE_G723_63,

/* play a G723 6.3k data (24 bytes per 30 ms) */

IVR_PLAY_TYPE_G729,

/* play a G729 data (10 bytes per 10ms) */

IVR_PLAY_TYPE_G711A,

/* play a G711 a-law (80 bytes per 10ms) */

} IvrPlayType_t;

12.9. Enumeration for IVR play direction (play TEXT only)
typedef enum {
IVR_DIR_LOCAL,
IVR_DIR_REMOTE,
} IvrPlayDir_t;

12.10. Enumeration for FAX State
typedef enum
{
FAX_IDLE = 0,
FAX_DETECT ,
MODEM_DETECT,
FAX_RUN,
MODEM_RUN
} FAXSTATE;

112

VoIP Programming Guide

12.11. Enumeration for DTMF type
enum __DTMF_TYPE__
{
DTMF_RFC2833 = 0,
DTMF_SIPINFO,
DTMF_INBAND,
DTMF_MAX
};

12.12. Enumeration for the Country
typedef enum
{
COUNTRY_USA = 0,
COUNTRY_UK,
COUNTRY_AUSTRALIA,
COUNTRY_HK,
COUNTRY_JAPAN,
COUNTRY_SWEDEN,
COUNTRY_GERMANY,
COUNTRY_FRANCE,
#if 0
COUNTRY_TR57,
#else
COUNTRY_TW,
#endif
COUNTRY_BELGIUM,
COUNTRY_FINLAND,
COUNTRY_ITALY,
COUNTRY_CHINA,
COUNTRY_CUSTOMER,
COUNTRY_MAX
113

VoIP Programming Guide
}_COUNTRY_;

12.13. Enumeration for FSK Caller ID Area
typedef enum
{
FSK_Bellcore = 0,
FSK_ETSI,
FSK_BT,
FSK_NTT
}TfskArea;

12.14. Enumeration for FSK Caller ID Parameter Type
typedef enum
{
FSK_PARAM_NULL = 0,

// No Data

FSK_PARAM_DATEnTIME = 0x01,

// Date and Time

FSK_PARAM_CLI = 0x02,

// Calling Line Identify (CLI)

FSK_PARAM_CLI_ABS = 0x04,

// Reason for absence of CLI

FSK_PARAM_CLI_NAME = 0x07,

// Calling Line Identify (CLI) Name

FSK_PARAM_CLI_NAME_ABS = 0x08, // Reason for absence of (CLI) Name
FSK_PARAM_MW = 0x0b,

// Message Waiting

}TfskParaType;

12.15. Enumeration for FSK Caller ID Gain
typedef enum
{
FSK_CLID_GAIN_8DB_UP = 0,
FSK_CLID_GAIN_7DB_UP,
114

VoIP Programming Guide
FSK_CLID_GAIN_6DB_UP,
FSK_CLID_GAIN_5DB_UP,
FSK_CLID_GAIN_4DB_UP,
FSK_CLID_GAIN_3DB_UP,
FSK_CLID_GAIN_2DB_UP,
FSK_CLID_GAIN_1DB_UP,
FSK_CLID_GAIN_0DB,
FSK_CLID_GAIN_1DB_DOWN,
FSK_CLID_GAIN_2DB_DOWN,
FSK_CLID_GAIN_3DB_DOWN,
FSK_CLID_GAIN_4DB_DOWN,
FSK_CLID_GAIN_5DB_DOWN,
FSK_CLID_GAIN_6DB_DOWN,
FSK_CLID_GAIN_7DB_DOWN,
FSK_CLID_GAIN_8DB_DOWN,
FSK_CLID_GAIN_9DB_DOWN,
FSK_CLID_GAIN_10DB_DOWN,
FSK_CLID_GAIN_11DB_DOWN,
FSK_CLID_GAIN_12DB_DOWN,
FSK_CLID_GAIN_13DB_DOWN,
FSK_CLID_GAIN_14DB_DOWN,
FSK_CLID_GAIN_15DB_DOWN,
FSK_CLID_GAIN_16DB_DOWN,
FSK_CLID_GAIN_MAX
}TfskClidGain;

115

VoIP Programming Guide

13.

Appendix B – Data Structure

13.1. Data Structure for RTP payload type
l

Our jitter buffer is an adaptive one which adjusts buffering delay automatically. One can use
API to set minimum and maximum delay to make delay within specified range. Though
buffering delay always falls into this range, it makes delay as smaller as possible. Because
it is a trade off problem between quality and delay, optimization factor is introduced. This
factor is designed for preference to quality or delay. If someone prefers better quality, delay
will be larger. In contrast, smaller delay causes worse quality. We recommend default
values of minimum and maximum delay are 40 and 130 ms respectively, and set
optimization factor to be 7.

l

nFramePerPacket means the voiced frame number per RTP packet. For G.711u/A,
G.729a/b, G.726, the unit is 10ms. If nFramePerPacket is set to 3, it means RTP packet
includes 30ms voice. For G.723, iLBC-30ms, the unit is 30ms. If nFramePerPacket is set
to 2, it means RTP packet includes 60ms voice.

l

nG726Packing means the G726 RTP packet packing order. 0- non-pack, 1- pack left, 2pack right. See RFC3551 for detail.

l

uPktFormat_vbd, local_pt_vbd and remote_pt_vbd with suffix ‘_vbd’ are used to V.152
application. If SIP negotiation doesn’t support V.152, they have to be filled with
rtpPayloadUndefined. On the other hand, if it supports V.152, uPktFormat_vbd can specify
rtpPayloadPCMU or rtpPayloadPCMA. local_pt_vbd and remote_pt_vbd specify payload
type between 96 to 127. One should ignore rtk_GetCEDToneDetect() event, because CED
event is handled by DSP to start V.152 mechanism.

typedef struct
{
uint32 chid;

///< channel id

uint32 sid;

///< session id

RtpPayloadType uPktFormat;

///< payload type in rtk_voip

RtpPayloadType local_pt;

///< payload type in local

RtpPayloadType remote_pt;

///< payload type in remote answer
116

VoIP Programming Guide
RtpPayloadType uPktFormat_vbd; ///< payload type in rtk_voip
RtpPayloadType local_pt_vbd;

///< payload type in local

RtpPayloadType remote_pt_vbd; ///< payload type in remote answer
int32 nG723Type;

///< G.723 bit rate, 0: 6.3K, 1: 5.3K

int32 nFramePerPacket;

///< voiced frame number per RTP packet

int32 bVAD;

///< VAD on: 1, off: 0

uint32 nJitterDelay;

///< jitter buffer min. delay

uint32 nMaxDelay;

///< jitter buffer max. delay

uint32 nJitterFactor;

///< optimization factor of jitter buffer

uint32 nG726Packing;

///< G.726 packing order

} payloadtype_config_t;

13.2. Data Structure for RTP Configuration
typedef struct
{
uint32 chid;

///< channel id

uint32 sid;

///< session id

uint32 isTcp;

///< tcp = 1 or udp = 0

uint32 remIp;

///< remote ip

uint16 remPort;

///< remote port

uint32 extIp;

///< local ip

uint16 extPort;

///< local port

uint8 tos;

///< QoS

uint16 rfc2833_payload_type_local;

///< rfc2833 payload type

uint16 rfc2833_payload_type_remote;

///< remote rfc2833 payload type

#ifdef CONFIG_RTK_VOIP_SRTP
unsigned char remoteSrtpKey[SRTP_KEY_LEN];
unsigned char localSrtpKey[SRTP_KEY_LEN];
int remoteCryptAlg;
int localCryptAlg;
#endif /*CONFIG_RTK_VOIP_SRTP*/
117

VoIP Programming Guide
#ifdef SUPPORT_RTP_REDUNDANT
uint16 rtp_redundant_payload_type_local; ///< local payload type of rtp redundant
uint16 rtp_redundant_payload_type_remote; ///< remote payload type of rtp redundant
#endif
} rtp_config_t;

13.3. Data Structure for RTP Session
typedef struct
{
uint32 ip_src_addr;
uint32 ip_dst_addr;
uint16 udp_src_port;
uint16 udp_dst_port;
uint8 protocol;
uint8 ch_id;
int32 m_id;
int32 result;
uint8 tos;
uint16 rfc2833_payload_type_local;
uint16 rfc2833_payload_type_remote;
#ifdef CONFIG_RTK_VOIP_SRTP
unsigned char remoteSrtpKey[SRTP_KEY_LEN];
unsigned char localSrtpKey[SRTP_KEY_LEN];
int remoteCryptAlg;
int localCryptAlg;
#endif /*CONFIG_RTK_VOIP_SRTP*/
#ifdef SUPPORT_RTP_REDUNDANT
uint16 rtp_redundant_payload_type_local;
uint16 rtp_redundant_payload_type_remote;
#endif
int32

ret_val;
118

VoIP Programming Guide
}
TstVoipMgrSession;

13.4. Data Structure for Tone Configuration
typedef struct ToneCfgParam st_ToneCfgParam;
struct ToneCfgParam
{
unsigned long toneType;
unsigned short cycle;
unsigned short cadNUM;
unsigned long CadOn0;
unsigned long CadOn1;
unsigned long CadOn2;
unsigned long CadOn3;
unsigned long CadOff0;
unsigned long CadOff1;
unsigned long CadOff2;
unsigned long CadOff3;
unsigned long PatternOff;
unsigned long ToneNUM;
unsigned long Freq1;
unsigned long Freq2;
unsigned long Freq3;
unsigned long Freq4;
long Gain1;
119

VoIP Programming Guide
long Gain2;
long Gain3;
long Gain4;
};

13.5. Data Structure for 3-way Conference Configuration
typedef struct
{
uint8 ch_id;
uint8 enable;
TstVoipMgrRtpCfg rtp_cfg[2];
}
TstVoipMgr3WayCfg;
typedef struct
{
// RTP session info
uint8 ch_id;
int32 m_id;
uint32 ip_src_addr;
uint32 ip_dst_addr;
uint16 udp_src_port;
uint16 udp_dst_port;
uint16 rtcp_src_port;
uint16 rtcp_dst_port;
uint8 protocol;
uint8 tos;
uint16 rfc2833_payload_type_local;
uint16 rfc2833_payload_type_remote;
//Payload info
120

VoIP Programming Guide
RtpPayloadType local_pt; // payload type in local
RtpPayloadType remote_pt; // payload type in remote answer
RtpPayloadType uPktFormat; // payload type in rtk_voip
int32 nG723Type;
int32 nFramePerPacket;
int32 bVAD;
uint32 nJitterDelay;
uint32 nMaxDelay;
uint32 nJitterFactor;
// RTP session state
uint8

state;

int32 result;
}
TstVoipMgrRtpCfg;

13.6. Data Structure for RTP Statistics
typedef struct {
unsigned char chid;

/* input */

unsigned char bResetStatistics;

/* input */

/* follows are output */
unsigned long nRxRtpStatsCountByte;
unsigned long nRxRtpStatsCountPacket;
unsigned long nRxRtpStatsLostPacket;
unsigned long nTxRtpStatsCountByte;
unsigned long nTxRtpStatsCountPacket;
} TstVoipRtpStatistics;

13.7. Data Structure for VoIP Configuration
typedef struct voipCfgParam_s voipCfgParam_t;

121

VoIP Programming Guide
struct voipCfgParam_s {
// voip flash check
unsigned long signature;

// voip flash check

unsigned short version;

// update if big change (size/offset not match..)

unsigned long feature;

// 32 bit feature

unsigned long extend_feature;

// 32 extend feature

unsigned short mib_version;

// update if some mib name change

(ex:

VOIP.PORT[0].NUMBER => VOIP.PORT[0].USER.NUMBER)
// used in import/export flash setting
// if mib version not match, then import will fail
// RFC flags
unsigned int

rfc_flags;

///< RFC flags

//tone

///< unit: 10ms

unsigned char

tone_of_country;

unsigned char

tone_of_custdial;

unsigned char

tone_of_custring;

unsigned char

tone_of_custbusy;

unsigned char

tone_of_custwaiting;

unsigned char

tone_of_customize;

st_ToneCfgParam cust_tone_para[TONE_CUSTOMER_MAX];
8 tones)
//disconnect tone det
unsigned char distone_num;
unsigned char d1freqnum;
unsigned short d1Freq1;
unsigned short d1Freq2;
unsigned char d1Accur;
unsigned short d1Level;

122

// for customized tone (up to

VoIP Programming Guide
unsigned short d1ONup;
unsigned short d1ONlow;
unsigned short d1OFFup;
unsigned short d1OFFlow;
unsigned char d2freqnum;
unsigned short d2Freq1;
unsigned short d2Freq2;
unsigned char d2Accur;
unsigned short d2Level;
unsigned short d2ONup;
unsigned short d2ONlow;
unsigned short d2OFFup;
unsigned short d2OFFlow;
//ring
unsigned char ring_cad;
unsigned char ring_group;
unsigned int

ring_phone_num[RING_GROUP_MAX];

unsigned char ring_cadence_use[RING_GROUP_MAX];
unsigned char ring_cadence_sel;
unsigned short ring_cadon[RING_CADENCE_MAX];
unsigned short ring_cadoff[RING_CADENCE_MAX];
// function key
char

funckey_pstn[FUNC_KEY_LENGTH];

// default is *0

char

funckey_transfer[FUNC_KEY_LENGTH];

// default is *1

// other
unsigned short auto_dial;

// 0, 3~9 sec, default is 5 sec, 0 is disable

unsigned short off_hook_alarm;

// 0, 10~60 sec, default is 30 sec, 0 is disable

unsigned short cid_auto_det_select;

// caller id auto detection selection

unsigned short caller_id_det_mode;

// caller id detection mode

123

VoIP Programming Guide
unsigned char one_stage_dial;

// 0: disable
// 1: enable one stage dial

unsigned char two_stage_dial;

//

- VoIP -> FXO -> PSTN

//

- PSTN number is assigned by "To" filed

// 0: disable
// 1: enable two stage dial
//

unsigned char auto_bypass_relay;

- VoIP -> FXO -> IVR prompt -> PSTN

// bit0: enable/disable bypass relay
//

- SIP Register failed

//

- TODO: Network failed

// bit1: enable/disable warning tone
// pulse dial
unsigned char pulse_dial_gen;

// 0: disable, 1: enable pulse dial generation for FXO port

unsigned char pulse_gen_pps;
unsigned char pulse_gen_make_time;
unsigned short pulse_gen_interdigit_pause;
unsigned char pulse_dial_det;

// 0: disable, 1: enable pulse dial detection for FXS port

unsigned short pulse_det_pause;
// auto config
unsigned short auto_cfg_ver;

// load config if version is newer

unsigned char auto_cfg_mode;

// 0: disable, 1: http

char

auto_cfg_http_addr[DNS_LEN];

unsigned short auto_cfg_http_port;
/***

For auto provision for tftp and ftp ***/

char
char
char
char

auto_cfg_tftp_addr[DNS_LEN];
auto_cfg_ftp_addr[DNS_LEN];
auto_cfg_ftp_user[20];
auto_cfg_ftp_passwd[20];

/*** end ***/
char

auto_cfg_file_path[MAX_AUTO_CONFIG_PATH];

124

VoIP Programming Guide
unsigned short auto_cfg_expire;

// 0..365 days

// fw update setting

// load config if version is newer

unsigned char fw_update_mode;

// 0: disable, 1: TFTP 2: FTP 3: HTTP

char

fw_update_tftp_addr[DNS_LEN];

char

fw_update_http_addr[DNS_LEN];

unsigned short fw_update_http_port;
char

fw_update_ftp_addr[DNS_LEN];

char

fw_update_ftp_user[MAX_FW_UPDATE_FTP];

char

fw_update_ftp_passwd[MAX_FW_UPDATE_FTP];

char

fw_update_file_path[MAX_FW_UPDATE_PATH];

unsigned char fw_update_power_on;
unsigned short fw_update_scheduling_day;
unsigned char fw_update_scheduling_time;
unsigned char fw_update_auto;
char

fw_update_file_prefix[MAX_FW_UPDATE_FILE_PREFIX];

unsigned long fw_update_next_time;
char

fw_update_fw_version[MAX_FW_VERSION];

// VLAN setting of WAN Port
unsigned char wanVlanEnable;
// VLAN for Voice and protocl packets (SIP, STUN, DHCP, ARP, etc.)
unsigned short wanVlanIdVoice;
unsigned char wanVlanPriorityVoice;
unsigned char wanVlanCfiVoice;
// VLAN for Data
unsigned short wanVlanIdData;
unsigned char wanVlanPriorityData;

125

VoIP Programming Guide
unsigned char wanVlanCfiData;
// VLAN for Video
unsigned short wanVlanIdVideo;
unsigned char wanVlanPriorityVideo;
unsigned char wanVlanCfiVideo;
// enable HW-NAT
unsigned char

hwnat_enable;

// bandwidth Mgr
unsigned short bandwidth_LANPort0_Egress;
unsigned short bandwidth_LANPort1_Egress;
unsigned short bandwidth_LANPort2_Egress;
unsigned short bandwidth_LANPort3_Egress;
unsigned short bandwidth_WANPort_Egress;
unsigned short bandwidth_LANPort0_Ingress;
unsigned short bandwidth_LANPort1_Ingress;
unsigned short bandwidth_LANPort2_Ingress;
unsigned short bandwidth_LANPort3_Ingress;
unsigned short bandwidth_WANPort_Ingress;
// FXO valume
unsigned char daa_txVolumne;

///< 1..10

unsigned char daa_rxVolumne;

///< 1..10

// DSCP
unsigned char rtpDscp;
unsigned char sipDscp;
voipCfgPortParam_t ports[VOIP_PORTS];

126

VoIP Programming Guide
// UI
#ifdef CONFIG_RTK_VOIP_IP_PHONE
ui_falsh_layout_t ui;
#endif
};

13.8. Data Structure for FSK Caller ID
#define MAX_CLID_DATA_SIZE
#define FSK_MDMF_SIZE

80

5

typedef struct
{
TfskParaType type;
char data[MAX_CLID_DATA_SIZE];
}
TstFskClid_Data;
typedef struct
{
unsigned char ch_id;
unsigned char service_type;
TstFskClid_Data cid_data[FSK_MDMF_SIZE];
int32

ret_val;

}
TstFskClid;
typedef struct
{
unsigned char ch_id;
TfskArea area;

/* area -> 0:Bellcore 1:ETSI 2:BT 3:NTT */
127

VoIP Programming Guide
unsigned int CS_cnt;

/* channel seizure signal bit count */

unsigned int mark_cnt;

/* mark signal bit count */

TfskClidGain mark_gain;

/* unit in dB */

TfskClidGain space_gain;

/* unit in dB */

unsigned int type2_expected_ack_tone;

/* DTMF A, or B, or C, or D*/

unsigned int delay_after_1st_ring;

/* unit in ms */

unsigned int delay_before_2nd_ring;

/* unit in ms */

int32

ret_val;

}
TstVoipFskPara;

128



---

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