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 usine re-INVITE. 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.

}}}

=== 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

Detected NAT type: Port Restricted

}}}

=== 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: 

}}}

=== Example of SDP with RTP and MSRP proposal ===

{{{

INVITE sip:61@ag-projects.com SIP/2.0

Via: SIP/2.0/UDP 192.168.1.6:61335;rport;branch=z9hG4bKPjTgHt1tWJgdV0tXsSCX.h9EJBZneF1134

Max-Forwards: 70

From: "Adrian G." <sip:31208005169@ag-projects.com>;tag=yf.gdZqZwOE5qcCB02qcKL9tdjtHK3-r

To: sip:61@ag-projects.com

Contact: <sip:rlafgmkq@192.168.1.6:61335>

Call-ID: slXPhUalLN3tiJYDKI5UnNOspHkV4PNb

CSeq: 16232 INVITE

Route: <sip:85.17.186.7:5060;lr>

Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE

Supported: 100rel

User-Agent: sip2sip-0.9.0-pjsip-1.0.2-trunk-r2553

Proxy-Authorization: Digest username="31208005169", realm="ag-projects.com", 

nonce="49d092ef3dd41223af53ece9bc1b5ce903898ece", uri="sip:61@ag-projects.com", response="6baaa853d66bc376e8e56acbd512d16b"

Content-Type: application/sdp

Content-Length:   592

v=0

o=- 3447394641 3447394641 IN IP4 192.168.1.6

s= 

c=IN IP4 192.168.1.6

t=0 0

m=audio 50048 RTP/AVP 104 103 102 0 8 117 3 9 101

a=rtcp:50049 IN IP4 192.168.1.6

a=rtpmap:104 speex/32000

a=rtpmap:103 speex/16000

a=rtpmap:102 speex/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:117 iLBC/8000

a=fmtp:117 mode=20

a=rtpmap:3 GSM/8000

a=rtpmap:9 G722/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=sendrecv

m=message 2855 TCP/TLS/MSRP *

a=path:msrps://192.168.1.6:2855/f2a8d0dcf07af4869cdd;tcp

a=accept-types:message/cpim text/*

a=accept-wrapped-types:*

}}}