Sip audio session

Version 10 (Adrian Georgescu, 03/30/2009 12:28 pm)

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== sip_audio_session ==
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[[TOC(SipTesting*, sip_*, xcap*,depth=2)]]
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To use this script you must to have a valid [wiki:SipSettingsAPI configuration].
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=== Description ===
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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.
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[[Image(http://www.tech-invite.com/img/cf3665/cf3665-32.gif)]]
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Source code: [source:scripts/sip_audio_session.py scripts/sip_audio_session.py]
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{{{
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adigeo@ag-oxygen:~$sip_audio_session --help
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Usage: sip_audio_session [options] [target-user@target-domain.com]
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This script can sit idle waiting for an incoming audio call, or perform an
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outgoing audio call to the target SIP account. The program will close the
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session and quit when Ctrl+D is pressed.
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Options:
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  -h, --help            show this help message and exit
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  -a NAME, --account=NAME
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                        The account name to use for any outgoing traffic. If
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                        not supplied, the default account will be used.
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  -c [FILE], --config_file=[FILE]
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                        The path to a configuration file to use. This
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                        overrides the default location of the configuration
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                        file.
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  -s [stdout|file|all|none], --trace-sip=[stdout|file|all|none]
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                        Dump the raw contents of incoming and outgoing SIP
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                        messages. The argument specifies where the messages
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                        are to be dumped.
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  -j [stdout|file|all|none], --trace-pjsip=[stdout|file|all|none]
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                        Print PJSIP logging output. The argument specifies
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                        where the messages are to be dumped.
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  -S, --disable-sound   Disables initializing the sound card.
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  --auto-answer         Interval after which to answer an incoming call
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                        (disabled by default). If the option is specified but
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                        the interval is not, it defaults to 0 (answer the call
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                        as soon as it starts ringing).
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  --auto-hangup         Interval after which to hangup an on-going call
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                        (applies only to outgoing calls, disabled by default).
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                        If the option is specified but the interval is not, it
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                        defaults to 0 (hangup the call as soon as it
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                        connects).
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}}}
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=== Example for incoming session ===
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{{{
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adigeo@ag-imac3:~$sip_audio_session
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Using account 31208005169@ag-projects.com
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Available control keys:
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  h: hang-up the active session
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  r: toggle audio recording
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  t: toggle SIP trace on the console
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  j: toggle PJSIP trace on the console
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  <> : adjust echo cancellation
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  SPACE: hold/on-hold
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  Ctrl-d: quit the program
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  ?: display this help message
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Succesfully registered using contact "sip:cwntuzyl@192.168.1.6:61163"
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Detected NAT type: Port Restricted
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Incoming audio session from ""Adrian G." <sip:31208005169@ag-projects.com>", do you want to accept? (y/n)
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Session established, using "PCMU" codec at 8000Hz
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Audio RTP endpoints 192.168.1.6:50132 <-> 85.17.186.7:53358
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Remote SIP User Agent is "CSCO/7"
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Session ended by remote party.
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Session duration was 3 seconds
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}}}
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=== Example for outgoing session ===
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{{{
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adigeo@ag-imac3:~$sip_audio_session ag@ag-projects.com
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Using account 31208005169@ag-projects.com
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Initiating SIP session from "Adrian G." <sip:31208005169@ag-projects.com> to 
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sip:ag@ag-projects.com via udp:81.23.228.150:5060 ...
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Available control keys:
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  h: hang-up the active session
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  r: toggle audio recording
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  t: toggle SIP trace on the console
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  j: toggle PJSIP trace on the console
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  <> : adjust echo cancellation
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  SPACE: hold/on-hold
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  Ctrl-d: quit the program
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  ?: display this help message
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Succesfully registered using contact "sip:ztomvpis@192.168.1.6:61215"
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Ringing...
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Session established, using "speex" codec at 32000Hz
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Audio RTP endpoints 192.168.1.6:50374 <-> 81.23.228.129:52156
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Remote SIP User Agent is "sip2sip-0.9.0-pjsip-1.0.2-trunk-r2553"
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Detected NAT type: Port Restricted
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Ending session...
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Session ended by local party.
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Session duration was 12 seconds
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}}}