SIP SIMPLE Client SDK

== sip_session ==

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

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

=== Description ===

This script can be used to establish SIP sessions with more than one media type. One can add and remove RTP audio and MSRP chat to the same SIP session. The defaul behaviour is to establish outgoing session with both audio and chat media.

{{{

Usage: sip_session [options] [target-user@target-domain.com]

This script will either sit idle waiting for an incoming MSRP session, or

start a MSRP session with the specified target 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 ACCOUNT_ID, --account-id=ACCOUNT_ID

  -c [FILE], --config_file=[FILE]

                        The path to a configuration file to use. This

                        overrides the default location of the configuration

                        file.

  -S, --disable-sound   Disables initializing the sound card.

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

                        messages.

  -j, --trace-pjsip     Print PJSIP logging output.

  --trace-engine        Print core's events.

  -m, --trace-msrp      Log the raw contents of incoming and outgoing MSRP

                        messages.

  --no-relay            Don't use the MSRP relay.

  --msrp-tcp            Use TCP for MSRP connections.

}}}

=== Example ===

{{{

adigeo@ag-imac3:~$sip_session   

Using account 31208005169@ag-projects.com

Press Ctrl-d to quit or Control-n to switch between active sessions

Waiting for incoming SIP session requests...

Registering "Adrian G." <sip:31208005169@ag-projects.com> at 81.23.228.150:5060

Registered SIP contact address: sip:hctoyfvx@192.168.1.6:61277 (expires in 600 seconds)

Incoming Audio request from "Adrian G." <sip:31208005169@ag-projects.com>, do you accept? (y/n) y

Connecting SIP session to "Adrian G." <sip:31208005169@ag-projects.com>

Session established, using "speex" codec at 32000Hz

Audio RTP endpoints 192.168.1.6:50018 <-> 81.23.228.150:58260

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

Detected NAT type: Port Restricted

Audio to Adrian G. (31208005169@ag-projects.com): 

}}}