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.

[[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-imac3:~$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-imac3:~$sip_audio_session 

Using account 31208005169@ag-projects.com

Logging SIP trace to file "/Users/adigeo/.sipclient/logs/sip_trace.txt"

Logging PJSIP trace to file "/Users/adigeo/.sipclient/logs/pjsip_trace.txt"

Available audio input devices: Built-in Input, Built-in Microphone, Logitech Wireless Headset

Available audio output devices: Built-in Output, Logitech Wireless Headset

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

  <>: adjust echo cancellation

  SPACE: hold/unhold

  Ctrl-d: quit the program

  ?: display this help message

2009-07-23 13:40:02 Registered contact "sip:oedtbzgw@192.168.1.6:50361" for sip:31208005169@ag-projects.com at 81.23.228.129:5060;transport=udp (expires in 600 seconds).

Other registered contacts:

  sip:froghdmq@192.168.1.6:50334 (expires in 547 seconds)

  sip:31208005169@192.168.1.123:5060 (expires in 234 seconds)

  sip:zegoxqlw@192.168.1.6:50298 (expires in 468 seconds)

  sip:31208005169@192.168.1.1;uniq=5B2860C44383A3D6705629A7E1FB8 (expires in 813 seconds)

Detected NAT type: Port Restricted

Incoming audio session from 'sip:31208005169@ag-projects.com', do you want to accept? (y/n)

Audio session established using "speex" codec at 32000Hz

Audio RTP endpoints 80.101.96.20:50406 <-> 81.23.228.150:52916

RTP audio stream is encrypted

Remote SIP User Agent is "sipsimple 0.9.1"

Audio session ended by remote party

Call duration was 4 seconds

}}}

=== Example for outgoing session ===

{{{

adigeo@ag-imac3:~$sip_audio_session ag@ag-projects.com

Using account 31208005169@ag-projects.com

Logging SIP trace to file "/Users/adigeo/.sipclient/logs/sip_trace.txt"

Logging PJSIP trace to file "/Users/adigeo/.sipclient/logs/pjsip_trace.txt"

Available audio input devices: Built-in Input, Built-in Microphone, Logitech Wireless Headset

Available audio output devices: Built-in Output, Logitech Wireless Headset

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

  <>: adjust echo cancellation

  SPACE: hold/unhold

  Ctrl-d: quit the program

  ?: display this help message

Initiating SIP audio session from 'sip:31208005169@ag-projects.com' to 'sip:ag@ag-projects.com' via sip:81.23.228.150:5060;transport=udp...

Audio session established using "speex" codec at 32000Hz

Audio RTP endpoints 80.101.96.20:50400 <-> 81.23.228.150:53734

RTP audio stream is encrypted

Ending audio session...

Audio session ended by local party

Call duration was 5 seconds

}}}

=== Example for bonjour mode ===

In bonjour mode no server is used. This mode is useful for serverless ad-hoc LAN operation. 

> The actual bonjour protocol that uses multicast DNS to broadcast the contact SIP URIs is not implemented.

[[Image(http://www.tech-invite.com/img/cf3665/cf3665-31.gif)]]

'''Called party'''

{{{

adigeo@ag-imac3:~$sip_audio_session -a bonjour@local

Using account bonjour@local

Listening on "sip:wjnrczhi@192.168.1.6:57624;transport=tls"

Listening on "sip:wjnrczhi@192.168.1.6:57623;transport=tcp"

Listening on "sip:wjnrczhi@192.168.1.6:61994"

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

Incoming audio session from "sip:imdyzosg@192.168.1.6", do you want to accept? (y/n)

Session established, using "speex" codec at 32000Hz

Audio RTP endpoints 192.168.1.6:50276 <-> 192.168.1.6:50100

RTP audio stream is encrypted

Remote SIP User Agent is "sip2sip-0.9.0-pjsip-1.0.2-trunk-r2553"

Session ended by remote party.

Session duration was 5 seconds

}}}

'''Calling party'''

{{{

adigeo@ag-imac3:~$sip_audio_session -a bonjour@local "sip:wjnrczhi@192.168.1.6:57624;transport=tls"

Using account bonjour@local

Listening on "sip:imdyzosg@192.168.1.6:57626;transport=tls"

Listening on "sip:imdyzosg@192.168.1.6:57625;transport=tcp"

Listening on "sip:imdyzosg@192.168.1.6:62008"

Initiating SIP session from sip:imdyzosg@192.168.1.6 to sip:wjnrczhi@192.168.1.6:57624;transport=tls via tls:192.168.1.6:57624 ...

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

Ringing...

Session established, using "speex" codec at 32000Hz

Audio RTP endpoints 192.168.1.6:50100 <-> 192.168.1.6:50276

RTP audio stream is encrypted

Remote SIP User Agent is "sip2sip-0.9.0-pjsip-1.0.2-trunk-r2553"

Ending session...

Session ended by local party.

Session duration was 5 seconds

}}}

=== Alarm system ===

sip_audio_session script can be used for end-to-end testing of a SIP service. 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. The failure can be caused by timeout, a negative response code or lack of RTP media after the SIP session has been established.

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.