Sip session » History » Revision 6
Revision 5 (Adrian Georgescu, 03/30/2009 11:35 AM) → Revision 6/44 (Adrian Georgescu, 03/30/2009 11:36 AM)
== 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.
[[Image(http://www.tech-invite.com/img/cf3665/cf3665-37.gif)]]
{{{
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):
}}}