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-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

}}}

=== 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.