SIP SIMPLE Client SDK

== sip_audio_session ==

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

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

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

=== 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.

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] [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=NAME

                        The account name from which to read account settings.

                        Corresponds to section Account_NAME in the

                        configuration file. If not supplied, the section

                        Account will be read.

  --sip-address=SIP_ADDRESS

                        SIP address of the user in the form user@domain

  -p PASSWORD, --password=PASSWORD

                        Password to use to authenticate the local account.

                        This overrides the setting from the config file.

  -n DISPLAY_NAME, --display-name=DISPLAY_NAME

                        Display name to use for the local account. This

                        overrides the setting from the config file.

  -o IP[:PORT], --outbound-proxy=IP[:PORT]

                        Outbound SIP proxy to use. By default a lookup of the

                        domain is performed based on SRV and A records. This

                        overrides the setting from the config file.

  -s, --trace-sip       Dump the raw contents of incoming and outgoing SIP

                        messages (disabled by default). The argument specifies

                        where the messages are to be dumped.

  -t EC_TAIL_LENGTH, --ec-tail-length=EC_TAIL_LENGTH

                        Echo cancellation tail length in ms, setting this to 0

                        will disable echo cancellation. Default is 50 ms.

  -r SAMPLE_RATE, --sample-rate=SAMPLE_RATE

                        Sample rate in kHz, should be one of 8, 16 or 32kHz.

                        Default is 32kHz.

  -c CODECS, --codecs=CODECS

                        Comma separated list of codecs to be used. Default is

                        "speex,g711,ilbc,gsm,g722".

  -S, --disable-sound   Do not initialize the soundcard (by default the

                        soundcard is enabled).

  -j, --trace-pjsip     Print PJSIP logging output (disabled by default).

  --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).

  --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).

}}}

=== Example for incoming session ===

{{{

adigeo@ag-imac3:~/Business/Personal$sip_audio_session 

Accounts available: 'alice', 'as', 'bob', 'ew', 'ewt', 'mrg', 'pbx', 's', 'tf', 'umts', 'umts_test', 'unet', 'unet_test', default

Using default account: 31208005169@ag-projects.com

Registering ""Adrian G." <sip:31208005169@ag-projects.com>" at 81.23.228.129:5060

REGISTER was successful

Contact: sip:HZ1BYFQtHR@192.168.1.6:49421;transport=udp (expires in 300 seconds)

Available control keys:

  h: hang-up the active session

  r: toggle audio recording

  t: toggle SIP trace on the console

  <> : adjust echo cancellation

  SPACE: hold/on-hold

  Ctrl-d: quit the program

  ?: display this help message

Waiting for incoming session...

Detected NAT type: Port Restricted

Incoming session...

Incoming audio session from "sip:adi@umts.ro", do you want to accept? (y/n)

Session established, using "speex" codec at 32000Hz

Audio RTP endpoints 192.168.1.6:40064 <-> 81.23.228.150:56618

Remote SIP User Agent is "sip2sip-0.4.0-pjsip-1.0.1-r2453"

Call is put on hold

Call is taken out of hold

Ending session...

Session ended by local party.

Session duration was 6 seconds

}}}

=== Example for outgoing session ===

{{{

adigeo@ag-imac3:~$sip -a umts ag@ag-projects.com

Accounts available: 'alice', 'as', 'bob', 'ew', 'ewt', 'mrg', 'pbx', 's', 'tf', 'umts', 'umts_test', 'unet', 'unet_test', default

Using account 'umts': adi@umts.ro

Call from "Adi UMTS" <sip:adi@umts.ro> to sip:ag@ag-projects.com through proxy udp:85.17.186.7:5060

Available control keys:

  h: hang-up the active session

  r: toggle audio recording

  t: toggle SIP trace on the console

  <> : adjust echo cancellation

  SPACE: hold/on-hold

  Ctrl-d: quit the program

  ?: display this help message

Ringing...

Ringing...

Ringing...

Session established, using "speex" codec at 32000Hz

Audio RTP endpoints 192.168.1.6:40048 <-> 81.23.228.150:56616

Remote SIP User Agent is "sip2sip-0.4.0-pjsip-1.0.1-r2453"

Remote party has put the call on hold

Remote party has taken the call out of hold

Session ended by remote party.

Session duration was 6 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.