Sip session » History » Revision 23
Revision 22 (Adrian Georgescu, 04/13/2009 10:54 AM) → Revision 23/44 (Adrian Georgescu, 04/14/2009 09:56 AM)
== sip_session == [[TOC(SipTesting*, sip_*, depth=2)]] === 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 using usine re-INVITE. The default behaviour is to establish outgoing session with both audio and chat media. [[Image(http://www.tech-invite.com/img/cf3665/cf3665-37.gif)]] {{{ adigeo@ag-imac3:~$sip_session -h Usage: sip_session [options] [target-user@target-domain.com] [audio|chat] This script will either sit idle waiting for an incoming session, or start a new session with the specified target SIP address. The program will close the session and quit when CTRL-D is pressed. This scripts supports RTP audio and MSRP chat sessions. Options: -h, --help show this help message and exit -a ACCOUNT_NAME, --account-name=ACCOUNT_NAME The name of the account to use. --no-register Bypass registration. -c FILE, --config_file=FILE The path to a configuration file to use. This overrides the default location of the configuration file. -S, --disable-sound Disables initializing the sound card. -s, --trace-sip Dump the raw contents of incoming and outgoing SIP messages. -j, --trace-pjsip Print PJSIP logging output. --trace-engine Print core's events. -m, --trace-msrp Log the raw contents of incoming and outgoing MSRP messages. --no-relay Don't use the MSRP relay. --msrp-tcp Use TCP for MSRP connections. }}} === Available commands === {{{ adigeo@ag-imac3:~/work/python-sipsimple$sip_session Using account 31208005169@ag-projects.com Type :help to get information about commands and shortcuts Waiting for incoming SIP session requests... Registering "Adrian G." <sip:31208005169@ag-projects.com> at 81.23.228.129:5060 Registered SIP contact address: sip:mdbwqhek@192.168.1.6:61453 (expires in 600 seconds) 31208005169@ag-projects.com> :help :add audio|chat Add a new stream to the current session. :call URI [audio|chat] Make an outgoing call. By default, use audio+chat :dtmf DIGITS Send DTMF digits. Also try CTRL-SPACE for virtual numpad :help Print this help message :hold Put the current session on hold :remove audio|chat Remove the stream from the current session :switch (or CTRL-N) Switch between active sessions :unhold Un-hold the current session }}} === Example of audio only session === {{{ adigeo@ag-imac3:~$sip_session Using account 31208005169@ag-projects.com Press Ctrl-d to quit or Control-n to switch between active sessions Waiting for incoming SIP session requests... Registering "Adrian G." <sip:31208005169@ag-projects.com> at 81.23.228.150:5060 Registered SIP contact address: sip:hctoyfvx@192.168.1.6:61277 (expires in 600 seconds) Incoming Audio request from "Adrian G." <sip:31208005169@ag-projects.com>, do you accept? (y/n) y Connecting SIP session to "Adrian G." <sip:31208005169@ag-projects.com> Session established, using "speex" codec at 32000Hz Audio RTP endpoints 192.168.1.6:50018 <-> 81.23.228.150:58260 Remote SIP User Agent is "sip2sip-0.9.0-pjsip-1.0.2-trunk-r2553" Detected NAT type: Port Restricted Audio to Adrian G. (31208005169@ag-projects.com): }}} === Example of chat only session === {{{ adigeo@ag-imac3:~$sip_session room1@chatserver.ag-projects.com Using account 31208005169@ag-projects.com Press Ctrl-d to quit or Control-n to switch between active sessions Registering "Adrian G." <sip:31208005169@ag-projects.com> at 85.17.186.7:5060 Initiating SIP session from "Adrian G." <sip:31208005169@ag-projects.com> to sip:room1@chatserver.ag-projects.com via udp:81.23.228.146:6060 ... Registered SIP contact address: sip:lpgdqwes@192.168.1.6:61392 (expires in 600 seconds) Connecting SIP session to sip:room1@chatserver.ag-projects.com Remote SIP User Agent is "sip-chatserver-0.9.1" 10:38:55 room1@chatserver.ag-projects.com: Welcome to the room, Adrian G.. You are the only participant in the room Chat to room1@chatserver.ag-projects.com: }}} === DTMF interaction === {{{ adigeo@ag-imac3:~$sip_session 61@ag-projects.com Using account 31208005169@ag-projects.com Type :help to get information about commands and shortcuts Registering "Adrian G." <sip:31208005169@ag-projects.com> at 81.23.228.150:5060 Registered SIP contact address: sip:ijpmwqbg@192.168.1.6:56320 (expires in 600 seconds) Initiating SIP session from "Adrian G." <sip:31208005169@ag-projects.com> to sip:61@ag-projects.com via udp:81.23.228.150:5060 ... Connecting SIP session to sip:61@ag-projects.com (Ok) Session established, using "GSM" codec at 8000Hz Audio RTP endpoints 192.168.1.6:50250 <-> 81.23.228.150:51318 Remote SIP User Agent is "pbxnsip-PBX/3.0.0.2896" +------+-----+------+ | 1 | 2 | 3 | | | ABC | DEF | +------+-----+------+ | 4 | 5 | 6 | | GHI | JKL | MNO | +------+-----+------+ | 7 | 8 | 9 | | PQRS | TUV | WXYZ | +------+-----+------+ | * | 0 | # | +-------------------+ Detected NAT type: Port Restricted > 0000# Disconnecting SIP session to sip:61@ag-projects.com Disconnected SIP session to sip:61@ag-projects.com (Ok) Session ended by local party. Session duration was 13 seconds. 31208005169@ag-projects.com> }}} See [wiki:sip_trace_msrp_rtp] for a sample SIP trace captured with this command line tool.