SIP SIMPLE Client SDK

== 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 re-INVITE.

[[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 chat

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.