Sip audio session

Version 23 (Adrian Georgescu, 09/24/2009 10:34 pm) → Version 24/28 (Adrian Georgescu, 01/24/2010 03:24 pm)

== sip-audio-session 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.

[[Image(http://www.tech-invite.com/img/cf3665/cf3665-32.gif)]]

Source code: [source:scripts/sip_audio_session.py scripts/sip_audio_session.py]

{{{
adigeo@ag-blink:~$sip-audio-session adigeo@ag-blink:~$sip_audio_session -h
Usage: sip-audio-session 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:~$sip-audio-session adigeo@ag-blink:~$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:~$sip-audio-session adigeo@ag-blink:~$sip -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:~$sip-audio-session adigeo@ag-imac3:~$sip -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 follow 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 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 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.