History

« Previous - Version 4/44 (diff) - Next » - Current version
Adrian Georgescu, 03/30/2009 11:33 am

sip_session

<acronym title="SipTesting*, sip_*, depth=2">TOC</acronym>

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

=== Description === {{{
Usage: sip_session [options] []

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
Press Ctrl-d to quit or Control-n to switch between active sessions
Waiting for incoming SIP session requests...
Registering "Adrian G." <sip:> at 81.23.228.150:5060
Registered SIP contact address: sip::61277 (expires in 600 seconds)
Incoming Audio request from "Adrian G." <sip:>, do you accept? (y/n) y
Connecting SIP session to "Adrian G." <sip:>
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. ():

}}}