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Sip audio session » History » Revision 8

Revision 7 (Adrian Georgescu, 03/30/2009 11:29 AM) → Revision 8/28 (Adrian Georgescu, 03/30/2009 11:29 AM)

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

 To use this script you must to have a valid [wiki:SipSettingsAPI configuration]. 

 === 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/cf3665-31.gif)]] 

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

 {{{ 
 adigeo@ag-oxygen:~$sip_audio_session --help 
 Usage: sip_audio_session [options] [target-user@target-domain.com] 

 This script can sit idle waiting for an incoming audio call, or perform an 
 outgoing audio call to the target SIP account. 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 [stdout|file|all|none], --trace-sip=[stdout|file|all|none] 
                         Dump the raw contents of incoming and outgoing SIP 
                         messages. The argument specifies where the messages 
                         are to be dumped. 
   -j [stdout|file|all|none], --trace-pjsip=[stdout|file|all|none] 
                         Print PJSIP logging output. The argument specifies 
                         where the messages are to be dumped. 
   -S, --disable-sound     Disables initializing the sound card. 
   --auto-answer           Interval after which to answer an incoming call 
                         (disabled by default). If the option is specified but 
                         the interval is not, it defaults to 0 (answer the call 
                         as soon as it starts ringing). 
   --auto-hangup           Interval after which to hangup an on-going call 
                         (applies only to outgoing calls, disabled by default). 
                         If the option is specified but the interval is not, it 
                         defaults to 0 (hangup the call as soon as it 
                         connects). 
 }}} 


 === Example for incoming session === 

 {{{ 
 adigeo@ag-imac3:~$sip_audio_session 
 Using account 31208005169@ag-projects.com 
 Available control keys: 
   h: hang-up the active session 
   r: toggle audio recording 
   t: toggle SIP trace on the console 
   j: toggle PJSIP trace on the console 
   <> : adjust echo cancellation 
   SPACE: hold/on-hold 
   Ctrl-d: quit the program 
   ?: display this help message 
 Succesfully registered using contact "sip:cwntuzyl@192.168.1.6:61163" 
 Detected NAT type: Port Restricted 
 Incoming audio session from ""Adrian G." <sip:31208005169@ag-projects.com>", do you want to accept? (y/n) 
 Session established, using "PCMU" codec at 8000Hz 
 Audio RTP endpoints 192.168.1.6:50132 <-> 85.17.186.7:53358 
 Remote SIP User Agent is "CSCO/7" 
 Session ended by remote party. 
 Session duration was 3 seconds 

 }}} 

 === Example for outgoing session === 

 {{{ 
 adigeo@ag-imac3:~$sip_audio_session ag@ag-projects.com 
 Using account 31208005169@ag-projects.com 
 Initiating SIP session from "Adrian G." <sip:31208005169@ag-projects.com> to  
 sip:ag@ag-projects.com via udp:81.23.228.150:5060 ... 
 Available control keys: 
   h: hang-up the active session 
   r: toggle audio recording 
   t: toggle SIP trace on the console 
   j: toggle PJSIP trace on the console 
   <> : adjust echo cancellation 
   SPACE: hold/on-hold 
   Ctrl-d: quit the program 
   ?: display this help message 
 Succesfully registered using contact "sip:ztomvpis@192.168.1.6:61215" 
 Ringing... 
 Session established, using "speex" codec at 32000Hz 
 Audio RTP endpoints 192.168.1.6:50374 <-> 81.23.228.129:52156 
 Remote SIP User Agent is "sip2sip-0.9.0-pjsip-1.0.2-trunk-r2553" 
 Detected NAT type: Port Restricted 
 Ending session... 
 Session ended by local party. 
 Session duration was 12 seconds 
 }}} 


 === Ongoing sessions === 

 During an ongoing session you can record the audio stream in a file by pressing r. You can hangup by pressing h.