SIP SIMPLE Client SDK

== sip-audio-session ==

[[TOC(SipTesting*, sip_*, xcap*,depth=2)]]

=== Description ===

This script can be used for interactive audio session or for scripting alarms. The script returns appropriate shell response codes for failed or successful sessions. The script can be setup to auto answer and auto hangup after predefined number of seconds, detects SIP negative response codes, missing ACK and the lack of RTP media after a session has been established.

> This script is available in ''sipclients'' package that must be installed separately from SIP SIMPLe client SDK package.

{{{

adigeo@ag-blink:~$sip-audio-session -h

Usage: sip-audio-session [options] [user@domain]

This script can sit idle waiting for an incoming audio session, or initiate an

outgoing audio session to a SIP address. The program will close the session

and quit when Ctrl+D is pressed.

Options:

  -h, --help            show this help message and exit

  -a NAME, --account=NAME

                        The account name to use for any outgoing traffic. If

                        not supplied, the default account will be used.

  -c FILE, --config-file=FILE

                        The path to a configuration file to use. This

                        overrides the default location of the configuration

                        file.

  -s, --trace-sip       Dump the raw contents of incoming and outgoing SIP

                        messages.

  -j, --trace-pjsip     Print PJSIP logging output.

  -n, --trace-notifications

                        Print all notifications (disabled by default).

  -S, --disable-sound   Disables initializing the sound card.

  --auto-answer         Interval after which to answer an incoming session

                        (disabled by default). If the option is specified but

                        the interval is not, it defaults to 0 (accept the

                        session as soon as it starts ringing).

  --auto-hangup         Interval after which to hang up an established session

                        (disabled by default). If the option is specified but

                        the interval is not, it defaults to 0 (hangup the

                        session as soon as it connects).

  -b, --batch           Run the program in batch mode: reading input from the

                        console is disabled and the option --auto-answer is

                        implied. This is particularly useful when running this

                        script in a non-interactive environment.

  -D, --daemonize       Enable running this program as a deamon. This option

                        implies --disable-sound, --auto-answer and --batch.

}}}

=== Example for incoming session ===

{{{

adigeo@ag-blink:~$sip-audio-session 

Using account 31208005169@ag-projects.com

Logging SIP trace to file "/Users/adigeo/Library/Application Support/Blink/logs/sip_trace.txt"

Logging PJSIP trace to file "/Users/adigeo/Library/Application Support/Blink/logs/pjsip_trace.txt"

Available audio input devices: None, system_default, Built-in Input, Built-in Microphone

Available audio output devices: None, system_default, Built-in Output

Using audio input device: Built-in Microphone

Using audio output device: Built-in Output

Using audio alert device: Built-in Output

Available control keys:

  s: toggle SIP trace on the console

  j: toggle PJSIP trace on the console

  n: toggle notifications trace on the console

  p: toggle printing RTP statistics on the console

  h: hang-up the active session

  r: toggle audio recording

  m: mute the microphone

  i: change audio input device

  o: change audio output device

  a: change audio alert device

  <>: adjust echo cancellation

  SPACE: hold/unhold

  Ctrl-d: quit the program

  ?: display this help message

2009-08-25 16:37:12 Registered contact "sip:hxsyungk@192.168.1.124:59164" for sip:31208005169@ag-projects.com at 81.23.228.150:5060;transport=udp (expires in 600 seconds).

Other registered contacts:

  sip:31208005169@192.168.1.123:5060 (expires in 274 seconds)

  sip:kwbfxyvl@192.168.1.124:59116 (expires in 522 seconds)

  sip:ilmegvkp@192.168.1.124:59003 (expires in 339 seconds)

  sip:31208005169@192.168.1.1;uniq=5B2860C44383A3D6705629A7E1FB8 (expires in 1162 seconds)

Detected NAT type: Port Restricted

Incoming audio session from 'sip:adi@umts.ro', do you want to accept? (y/n)

Audio session established using "speex" codec at 16000Hz

Audio RTP endpoints 192.168.1.124:50378 <-> 85.17.186.6:58868

RTP audio stream is encrypted

Remote SIP User Agent is "Blink-0.9.0"

Remote party has put the audio session on hold

Audio session is put on hold

Audio session ended by remote party

Session duration was 6 seconds

2009-08-25 16:37:44 Registration ended.

}}}

=== Example for outgoing session ===

{{{

adigeo@ag-blink:~$sip-audio-session -a umts ag@ag-projects.com

Using account adi@umts.ro

Logging SIP trace to file "/Users/adigeo/Library/Application Support/Blink/logs/sip_trace.txt"

Logging PJSIP trace to file "/Users/adigeo/Library/Application Support/Blink/logs/pjsip_trace.txt"

Available audio input devices: None, system_default, Built-in Input, Built-in Microphone

Available audio output devices: None, system_default, Built-in Output

Using audio input device: Built-in Microphone

Using audio output device: Built-in Output

Using audio alert device: Built-in Output

Available control keys:

  s: toggle SIP trace on the console

  j: toggle PJSIP trace on the console

  n: toggle notifications trace on the console

  p: toggle printing RTP statistics on the console

  h: hang-up the active session

  r: toggle audio recording

  m: mute the microphone

  i: change audio input device

  o: change audio output device

  a: change audio alert device

  <>: adjust echo cancellation

  SPACE: hold/unhold

  Ctrl-d: quit the program

  ?: display this help message

Initiating SIP audio session from 'sip:adi@umts.ro' to 'sip:ag@ag-projects.com' via sip:85.17.186.7:5060;transport=udp...

Audio session established using "speex" codec at 16000Hz

Audio RTP endpoints 192.168.1.124:50054 <-> 85.17.186.6:58866

RTP audio stream is encrypted

Audio session is put on hold

Remote party has put the audio session on hold

Detected NAT type: Port Restricted

Ending audio session...

Audio session ended by local party

Session duration was 7 seconds

}}}

=== Session with sip trace enabled ===

Use -s parameter you can see on the console detailed trace of all DNS queries/responses and SIP traffic exchanged during the session.

{{{

adigeo@ag-imac3:~$sip-audio-session -s -a umts ag@ag-projects.com

Using account adi@umts.ro

Logging SIP trace to file "/Users/adigeo/Desktop/FileTransfers/sip_trace.txt"

Logging PJSIP trace to file "/Users/adigeo/Desktop/FileTransfers/pjsip_trace.txt"

Logging notifications trace to file "/Users/adigeo/Desktop/FileTransfers/notifications_trace.txt"

Available audio input devices: None, system_default, Built-in Input, Built-in Microphone, Logitech Wireless Headset

Available audio output devices: None, system_default, Built-in Output, Logitech Wireless Headset

Using audio input device: Logitech Wireless Headset

Using audio output device: Logitech Wireless Headset

Using audio alert device: Built-in Output

Available control keys:

  s: toggle SIP trace on the console

  j: toggle PJSIP trace on the console

  n: toggle notifications trace on the console

  p: toggle printing RTP statistics on the console

  h: hang-up the active session

  r: toggle audio recording

  m: mute the microphone

  i: change audio input device

  o: change audio output device

  a: change audio alert device

  <>: adjust echo cancellation

  SPACE: hold/unhold

  Ctrl-d: quit the program

  ?: display this help message

2009-09-24 22:31:24.118467: DNS lookup SRV _stun._udp.umts.ro succeeded, ttl=10758: 0 0 3478 stun1.dns-hosting.info., 0 0 3479 stun2.dns-hosting.info.

2009-09-24 22:31:24.120425: DNS lookup NAPTR ag-projects.com succeeded, ttl=244: 20 0 "s" "SIP+D2U" "" _sip._udp.ag-projects.com.

2009-09-24 22:31:24.126619: DNS lookup A stun1.dns-hosting.info. succeeded, ttl=845: 81.23.228.150

2009-09-24 22:31:24.128383: DNS lookup SRV _sip._udp.ag-projects.com. succeeded, ttl=18: 0 0 5060 proxy.sipthor.net.

2009-09-24 22:31:24.132502: DNS lookup A stun2.dns-hosting.info. succeeded, ttl=845: 85.17.186.6

2009-09-24 22:31:24.136754: DNS lookup A proxy.sipthor.net. succeeded, ttl=5: 85.17.186.7, 81.23.228.129

Initiating SIP audio session from '"Adrian G." <sip:adi@umts.ro>' to 'sip:ag@ag-projects.com' via sip:85.17.186.7:5060;transport=udp...

2009-09-24 22:31:24.145751: DNS lookup SRV _stun._udp.umts.ro succeeded, ttl=10758: 0 0 3478 stun1.dns-hosting.info., 0 0 3479 stun2.dns-hosting.info.

2009-09-24 22:31:24.150530: DNS lookup A stun1.dns-hosting.info. succeeded, ttl=845: 81.23.228.150

2009-09-24 22:31:24.155510: DNS lookup A stun2.dns-hosting.info. succeeded, ttl=845: 85.17.186.6

2009-09-24 22:31:24.572498: SENDING: Packet 1, +0:00:00

192.168.1.6:62054 -(SIP over UDP)-> 85.17.186.7:5060

INVITE sip:ag@ag-projects.com SIP/2.0

Via: SIP/2.0/UDP 192.168.1.6:62054;rport;branch=z9hG4bKPjWy0ZCjWb9Ro6Cy15cBX3FE3H.er7.wzB

Max-Forwards: 70

From: "Adrian G." <sip:adi@umts.ro>;tag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF

To: <sip:ag@ag-projects.com>

Contact: <sip:pfxtjskq@192.168.1.6:62054>

Call-ID: TvSQ8UaRQkYIz53p8itOYiV.MLKdlzC3

CSeq: 16887 INVITE

Route: <sip:85.17.186.7;lr>

Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE

Supported: 100rel

User-Agent: blink-0.9.0

Content-Type: application/sdp

Content-Length:  1087

v=0

o=- 3462813084 3462813084 IN IP4 192.168.1.6

s=blink-0.9.0

c=IN IP4 80.101.96.20

t=0 0

m=audio 62066 RTP/AVP 104 103 102 3 9 0 8 101

a=rtcp:62067 IN IP4 80.101.96.20

a=rtpmap:104 speex/32000

a=rtpmap:103 speex/16000

a=rtpmap:102 speex/8000

a=rtpmap:3 GSM/8000

a=rtpmap:9 G722/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:eQ0XcBiuyy33zR2HEHLiaS5LCxA1T9rvP9J8GLw6

a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:zWQU33HIZ0a7otihkQe2Y4jvqpKpXtotNwoW9Xl8

a=ice-ufrag:0aa3379a

a=ice-pwd:619764ea

a=candidate:S 1 UDP 31 80.101.96.20 62066 typ srflx raddr 192.168.1.6 rport 62066

a=candidate:H 1 UDP 23 192.168.1.6 62066 typ host

a=candidate:H 1 UDP 23 10.211.55.2 62066 typ host

a=candidate:H 1 UDP 23 10.37.129.2 62066 typ host

a=candidate:S 2 UDP 30 80.101.96.20 62067 typ srflx raddr 192.168.1.6 rport 62067

a=candidate:H 2 UDP 22 192.168.1.6 62067 typ host

a=candidate:H 2 UDP 22 10.211.55.2 62067 typ host

a=candidate:H 2 UDP 22 10.37.129.2 62067 typ host

a=sendrecv

--

2009-09-24 22:31:24.601167: RECEIVED: Packet 2, +0:00:00.028669

85.17.186.7:5060 -(SIP over UDP)-> 192.168.1.6:62054

SIP/2.0 100 Giving a try

Via: SIP/2.0/UDP 192.168.1.6:62054;rport=62054;branch=z9hG4bKPjWy0ZCjWb9Ro6Cy15cBX3FE3H.er7.wzB;received=80.101.96.20

From: "Adrian G." <sip:adi@umts.ro>;tag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF

To: <sip:ag@ag-projects.com>

Call-ID: TvSQ8UaRQkYIz53p8itOYiV.MLKdlzC3

CSeq: 16887 INVITE

Server: SIP Thor on OpenSIPS XS 1.4.5

Content-Length: 0

--

2009-09-24 22:31:24.621860: RECEIVED: Packet 3, +0:00:00.049362

85.17.186.7:5060 -(SIP over UDP)-> 192.168.1.6:62054

SIP/2.0 407 Proxy Authentication Required

Via: SIP/2.0/UDP 192.168.1.6:62054;received=80.101.96.20;rport=62054;branch=z9hG4bKPjWy0ZCjWb9Ro6Cy15cBX3FE3H.er7.wzB

From: "Adrian G." <sip:adi@umts.ro>;tag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF

To: <sip:ag@ag-projects.com>;tag=e7d4d6b46afb9bf88242924a8d869ebf.962b

Call-ID: TvSQ8UaRQkYIz53p8itOYiV.MLKdlzC3

CSeq: 16887 INVITE

Proxy-Authenticate: Digest realm="umts.ro", nonce="4abbd73a48ba8c7fc6617208684ad122088d2207"

Server: SIP Thor on OpenSIPS XS 1.4.5

Content-Length: 0

--

2009-09-24 22:31:24.622019: SENDING: Packet 4, +0:00:00.049521

192.168.1.6:62054 -(SIP over UDP)-> 85.17.186.7:5060

ACK sip:ag@ag-projects.com SIP/2.0

Via: SIP/2.0/UDP 192.168.1.6:62054;rport;branch=z9hG4bKPjWy0ZCjWb9Ro6Cy15cBX3FE3H.er7.wzB

Max-Forwards: 70

From: "Adrian G." <sip:adi@umts.ro>;tag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF

To: <sip:ag@ag-projects.com>;tag=e7d4d6b46afb9bf88242924a8d869ebf.962b

Call-ID: TvSQ8UaRQkYIz53p8itOYiV.MLKdlzC3

CSeq: 16887 ACK

Route: <sip:85.17.186.7;lr>

User-Agent: blink-0.9.0

Content-Length:  0

--

2009-09-24 22:31:24.622214: SENDING: Packet 5, +0:00:00.049716

192.168.1.6:62054 -(SIP over UDP)-> 85.17.186.7:5060

INVITE sip:ag@ag-projects.com SIP/2.0

Via: SIP/2.0/UDP 192.168.1.6:62054;rport;branch=z9hG4bKPjwxj-gfiYVdjLvEWkt0l-pLfriN3gjo-T

Max-Forwards: 70

From: "Adrian G." <sip:adi@umts.ro>;tag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF

To: <sip:ag@ag-projects.com>

Contact: <sip:pfxtjskq@192.168.1.6:62054>

Call-ID: TvSQ8UaRQkYIz53p8itOYiV.MLKdlzC3

CSeq: 16888 INVITE

Route: <sip:85.17.186.7;lr>

Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE

Supported: 100rel

User-Agent: blink-0.9.0

Proxy-Authorization: Digest username="adi", realm="umts.ro", nonce="4abbd73a48ba8c7fc6617208684ad122088d2207", uri="sip:ag@ag-projects.com", response="cb85bbe3dbe0dcd71820c6ceaa027566"

Content-Type: application/sdp

Content-Length:  1087

v=0

o=- 3462813084 3462813084 IN IP4 192.168.1.6

s=blink-0.9.0

c=IN IP4 80.101.96.20

t=0 0

m=audio 62066 RTP/AVP 104 103 102 3 9 0 8 101

a=rtcp:62067 IN IP4 80.101.96.20

a=rtpmap:104 speex/32000

a=rtpmap:103 speex/16000

a=rtpmap:102 speex/8000

a=rtpmap:3 GSM/8000

a=rtpmap:9 G722/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:eQ0XcBiuyy33zR2HEHLiaS5LCxA1T9rvP9J8GLw6

a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:zWQU33HIZ0a7otihkQe2Y4jvqpKpXtotNwoW9Xl8

a=ice-ufrag:0aa3379a

a=ice-pwd:619764ea

a=candidate:S 1 UDP 31 80.101.96.20 62066 typ srflx raddr 192.168.1.6 rport 62066

a=candidate:H 1 UDP 23 192.168.1.6 62066 typ host

a=candidate:H 1 UDP 23 10.211.55.2 62066 typ host

a=candidate:H 1 UDP 23 10.37.129.2 62066 typ host

a=candidate:S 2 UDP 30 80.101.96.20 62067 typ srflx raddr 192.168.1.6 rport 62067

a=candidate:H 2 UDP 22 192.168.1.6 62067 typ host

a=candidate:H 2 UDP 22 10.211.55.2 62067 typ host

a=candidate:H 2 UDP 22 10.37.129.2 62067 typ host

a=sendrecv

--

2009-09-24 22:31:24.656088: RECEIVED: Packet 6, +0:00:00.083590

85.17.186.7:5060 -(SIP over UDP)-> 192.168.1.6:62054

SIP/2.0 100 Giving a try

Via: SIP/2.0/UDP 192.168.1.6:62054;rport=62054;branch=z9hG4bKPjwxj-gfiYVdjLvEWkt0l-pLfriN3gjo-T;received=80.101.96.20

From: "Adrian G." <sip:adi@umts.ro>;tag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF

To: <sip:ag@ag-projects.com>

Call-ID: TvSQ8UaRQkYIz53p8itOYiV.MLKdlzC3

CSeq: 16888 INVITE

Server: SIP Thor on OpenSIPS XS 1.4.5

Content-Length: 0

--

2009-09-24 22:31:24.721041: RECEIVED: Packet 7, +0:00:00.148543

85.17.186.7:5060 -(SIP over UDP)-> 192.168.1.6:62054

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 192.168.1.6:62054;rport=62054;received=80.101.96.20;branch=z9hG4bKPjwxj-gfiYVdjLvEWkt0l-pLfriN3gjo-T

Record-Route: <sip:85.17.186.7;lr;ftag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF;did=2f3.9e165924>

Record-Route: <sip:81.23.228.150;lr;ftag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF;did=2f3.e5ffeb2>

Record-Route: <sip:85.17.186.7;lr;ftag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF;did=2f3.8e165924>

Call-ID: TvSQ8UaRQkYIz53p8itOYiV.MLKdlzC3

From: "Adrian G." <sip:adi@umts.ro>;tag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF

To: <sip:ag@ag-projects.com>;tag=FkXkUNDcrT80u8GHaUIUuF4OrIJI6O8f

CSeq: 16888 INVITE

Server: blink-0.9.0

Contact: <sip:iwralmqz@80.101.96.20:61962>

Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE

Content-Length:  0

--

2009-09-24 22:31:24.878489: RECEIVED: Packet 8, +0:00:00.305991

85.17.186.7:5060 -(SIP over UDP)-> 192.168.1.6:62054

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 192.168.1.6:62054;received=80.101.96.20;rport=62054;branch=z9hG4bKPjwxj-gfiYVdjLvEWkt0l-pLfriN3gjo-T

Record-Route: <sip:85.17.186.7;lr=on;ftag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF;did=2f3.9e165924>

Record-Route: <sip:81.23.228.150;lr=on;ftag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF;did=2f3.e5ffeb2>

Record-Route: <sip:85.17.186.7;lr=on;ftag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF;did=2f3.8e165924>

From: "Adrian G." <sip:adi@umts.ro>;tag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF

To: <sip:ag@ag-projects.com>;tag=96A4E0ACA527F9AF

Call-ID: TvSQ8UaRQkYIz53p8itOYiV.MLKdlzC3

CSeq: 16888 INVITE

Contact: <sip:31208005169@80.101.96.20:5060;uniq=5B2860C44383A3D6705629A7E1FB8>

User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.56 (May  1 2008)

Content-Length: 0

--

2009-09-24 22:31:25.154425: RECEIVED: Packet 9, +0:00:00.581927

85.17.186.7:5060 -(SIP over UDP)-> 192.168.1.6:62054

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 192.168.1.6:62054;received=80.101.96.20;rport=62054;branch=z9hG4bKPjwxj-gfiYVdjLvEWkt0l-pLfriN3gjo-T

From: "Adrian G." <sip:adi@umts.ro>;tag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF

To: <sip:ag@ag-projects.com>;tag=000c854663c02cf2799a9168-4ae390b1

Call-ID: TvSQ8UaRQkYIz53p8itOYiV.MLKdlzC3

CSeq: 16888 INVITE

Server: CSCO/7

Contact: <sip:31208005169@80.101.96.20:61000>

Record-Route: <sip:81.23.228.129;lr=on;ftag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF;did=2f3.985cae24>,<sip:85.17.186.7;lr=on;ftag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF;did=2f3.9e165924>,<sip:81.23.228.150;lr=on;ftag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF;did=2f3.e5ffeb2>,<sip:85.17.186.7;lr=on;ftag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF;did=2f3.8e165924>

Content-Length: 0

--

2009-09-24 22:31:25.368613: RECEIVED: Packet 10, +0:00:00.796115

85.17.186.7:5060 -(SIP over UDP)-> 192.168.1.6:62054

SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.1.6:62054;received=80.101.96.20;rport=62054;branch=z9hG4bKPjwxj-gfiYVdjLvEWkt0l-pLfriN3gjo-T

From: "Adrian G." <sip:adi@umts.ro>;tag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF

To: <sip:ag@ag-projects.com>;tag=000c854663c02cf2799a9168-4ae390b1

Call-ID: TvSQ8UaRQkYIz53p8itOYiV.MLKdlzC3

CSeq: 16888 INVITE

Server: CSCO/7

Contact: <sip:31208005169@80.101.96.20:61000>

Record-Route: <sip:81.23.228.129;lr=on;ftag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF;did=2f3.985cae24>,<sip:85.17.186.7;lr=on;ftag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF;did=2f3.9e165924>,<sip:81.23.228.150;lr=on;ftag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF;did=2f3.e5ffeb2>,<sip:85.17.186.7;lr=on;ftag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF;did=2f3.8e165924>

Content-Type: application/sdp

Content-Length: 197

v=0

o=Cisco-SIPUA 8420 8964 IN IP4 192.168.1.123

s=SIP Call

c=IN IP4 81.23.228.150

t=0 0

m=audio 51974 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

--

2009-09-24 22:31:25.369124: SENDING: Packet 11, +0:00:00.796626

192.168.1.6:62054 -(SIP over UDP)-> 85.17.186.7:5060

ACK sip:31208005169@80.101.96.20:61000 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.6:62054;rport;branch=z9hG4bKPjkq3Y5tZfK3d.zASBBAHQHZMavNNRQw0W

Max-Forwards: 70

From: "Adrian G." <sip:adi@umts.ro>;tag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF

To: <sip:ag@ag-projects.com>;tag=000c854663c02cf2799a9168-4ae390b1

Call-ID: TvSQ8UaRQkYIz53p8itOYiV.MLKdlzC3

CSeq: 16888 ACK

Route: <sip:85.17.186.7;lr;ftag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF;did=2f3.8e165924>

Route: <sip:81.23.228.150;lr;ftag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF;did=2f3.e5ffeb2>

Route: <sip:85.17.186.7;lr;ftag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF;did=2f3.9e165924>

Route: <sip:81.23.228.129;lr;ftag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF;did=2f3.985cae24>

User-Agent: blink-0.9.0

Content-Length:  0

--

Audio session established using "PCMU" codec at 8000Hz

Audio RTP endpoints 80.101.96.20:62066 <-> 81.23.228.150:51974

Detected NAT type: Port Restricted

2009-09-24 22:31:40.495793: RECEIVED: Packet 12, +0:00:15.923295

85.17.186.7:5060 -(SIP over UDP)-> 192.168.1.6:62054

NOTIFY sip:80.101.96.20:62054 SIP/2.0

Via: SIP/2.0/UDP 85.17.186.7:5060;branch=0

From: sip:keepalive@85.17.186.7;tag=7c29b7d5

To: sip:80.101.96.20:62054

Call-ID: 296fc4b6-56ba6860-24109f@85.17.186.7

CSeq: 1 NOTIFY

Event: keep-alive

Content-Length: 0

--

2009-09-24 22:31:40.495929: SENDING: Packet 13, +0:00:15.923431

192.168.1.6:62054 -(SIP over UDP)-> 85.17.186.7:5060

SIP/2.0 405 Method Not Allowed

Via: SIP/2.0/UDP 85.17.186.7:5060;received=85.17.186.7;branch=0

Call-ID: 296fc4b6-56ba6860-24109f@85.17.186.7

From: <sip:keepalive@85.17.186.7>;tag=7c29b7d5

To: <sip:80.101.96.20>

CSeq: 1 NOTIFY

Server: blink-0.9.0

Content-Length:  0

--

2009-09-24 22:31:43.425393: RECEIVED: Packet 14, +0:00:18.852895

85.17.186.7:5060 -(SIP over UDP)-> 192.168.1.6:62054

BYE sip:pfxtjskq@80.101.96.20:62054 SIP/2.0

Record-Route: <sip:85.17.186.7;lr=on;ftag=000c854663c02cf2799a9168-4ae390b1>

Record-Route: <sip:81.23.228.150;lr=on;ftag=000c854663c02cf2799a9168-4ae390b1>

Record-Route: <sip:85.17.186.7;lr=on;ftag=000c854663c02cf2799a9168-4ae390b1>

Max-Forwards:  7

Record-Route: <sip:81.23.228.129;lr=on;ftag=000c854663c02cf2799a9168-4ae390b1>

Via: SIP/2.0/UDP 85.17.186.7;branch=z9hG4bK9c22.feada044.0

Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bK9c22.3cf12dd3.0

Via: SIP/2.0/UDP 85.17.186.7;branch=z9hG4bK9c22.eeada044.0

Via: SIP/2.0/UDP 81.23.228.129;branch=z9hG4bK9c22.877deec6.0

Via: SIP/2.0/UDP 192.168.1.123:5060;rport=61000;received=80.101.96.20;branch=z9hG4bK63eb02c1

From: <sip:ag@ag-projects.com>;tag=000c854663c02cf2799a9168-4ae390b1

To: "Adrian G." <sip:adi@umts.ro>;tag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF

Call-ID: TvSQ8UaRQkYIz53p8itOYiV.MLKdlzC3

CSeq: 101 BYE

User-Agent: CSCO/7

Content-Length: 0

RTP-RxStat: Dur=18,Pkt=29,Oct=4640,LatePkt=0,LostPkt=0,AvgJit=0

RTP-TxStat: Dur=18,Pkt=889,Oct=142240

--

2009-09-24 22:31:43.425554: SENDING: Packet 15, +0:00:18.853056

192.168.1.6:62054 -(SIP over UDP)-> 85.17.186.7:5060

SIP/2.0 200 OK

Via: SIP/2.0/UDP 85.17.186.7;received=85.17.186.7;branch=z9hG4bK9c22.feada044.0

Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bK9c22.3cf12dd3.0

Via: SIP/2.0/UDP 85.17.186.7;branch=z9hG4bK9c22.eeada044.0

Via: SIP/2.0/UDP 81.23.228.129;branch=z9hG4bK9c22.877deec6.0

Via: SIP/2.0/UDP 192.168.1.123:5060;rport=61000;received=80.101.96.20;branch=z9hG4bK63eb02c1

Record-Route: <sip:85.17.186.7;lr;ftag=000c854663c02cf2799a9168-4ae390b1>

Record-Route: <sip:81.23.228.150;lr;ftag=000c854663c02cf2799a9168-4ae390b1>

Record-Route: <sip:85.17.186.7;lr;ftag=000c854663c02cf2799a9168-4ae390b1>

Record-Route: <sip:81.23.228.129;lr;ftag=000c854663c02cf2799a9168-4ae390b1>

Call-ID: TvSQ8UaRQkYIz53p8itOYiV.MLKdlzC3

From: <sip:ag@ag-projects.com>;tag=000c854663c02cf2799a9168-4ae390b1

To: "Adrian G." <sip:adi@umts.ro>;tag=tv6vh5PXicua6Zuu0ZCv9smnXR.J-CxF

CSeq: 101 BYE

Server: blink-0.9.0

Content-Length:  0

--

Audio session ended by remote party

Session duration was 18 seconds

}}}

=== Alarm system ===

sip-audio-session script can be used for end-to-end testing of a SIP service including the RTP media path. The following failures can be detected:

 * Timeout

 * Negative response code

 * Lack of RTP media after the SIP session has been established

 * Missing ACK

To setup the alarm system start periodically a caller script from a monitoring software using the following arguments:

  {{{

sip-audio-session --auto-hangup user@domain

  }}}

Where the user@domain has been configured as the SIP account of the listener, can be an answering machine on the PSTN network. The caller script hangs up after each call. The shell return code can be used to determine if the session setup has failed.

To receive calls and answer them automatically you can also use sip_audio_session script as follows:

  {{{

sip-audio-session --daemonize

  }}}

You must run the script as user root. The --daemonize option puts the client in the background and the logging goes to /var/log/syslog. The program saves its pid file to /var/run/sip_audio_session.pid.