Sip session

Version 39 (Adrian Georgescu, 06/23/2011 04:39 pm)

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== sip-session ==
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[[TOC(SipTesting*, sip_*, xcap_*, depth=2)]]
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> This script is available in ''sipclients'' package that must be installed separately from SIP SIMPLe client SDK package.
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=== Description ===
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'''sip-session''' command line script is a show-case for the powerful features of SIP SIMPLE development kit related to establishing, modifying and terminating SIP sessions with multiple media types like VoIP, Instant Messaging and File Transfer.
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The script has the following features:
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 1. Registers with a SIP registrar and is available for incoming sessions
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 1. Switches between multiple sessions and provides in-call controls like Hold and Mute
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 1. Handles outgoing SIP sessions with combinations of media types based on RTP and MSRP protocols
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 1. Performs NAT traversal using ICE and MSRP relay extension
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 1. Provides control for the input, output and alert audio devices
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 1. Records the RTP audio streams (input, output or combined) 
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 1. Enable text input and output for Instant Messaging sessions
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 1. Provides File Transfer capability with progress indicator
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 1. Gives access to real-time traces of involved protocols (DNS, SIP and MSRP)
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=== Example ===
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{{{
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adigeo@ag-imac3:~$sip-session 
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Using account adi@umts.ro
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adi@umts.ro> /help
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General commands:
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  /call {user[@domain]}: call the specified user using audio and chat
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  /audio {user[@domain]} [+chat]: call the specified user using audio and possibly chat
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  /chat {user[@domain]} [+audio]: call the specified user using chat and possibly audio
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  /send {user[@domain]} {file}: initiate a file transfer with the specified user
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  /next: select the next connected session
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  /prev: select the previous connected session
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  /sessions: show the list of connected sessions
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  /trace [[+|-]sip] [[+|-]msrp] [[+|-]pjsip] [[+|-]notifications]: toggle/set tracing on the console
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  /rtp [on|off]: toggle/set printing RTP statistics on the console (ctrl-x p)
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  /mute [on|off]: mute the microphone (ctrl-x u)
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  /input [device]: change audio input device (ctrl-x i)
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  /output [device]: change audio output device (ctrl-x o)
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  /alert [device]: change audio alert device (ctrl-x a)
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  /echo [+|-][value]: adjust echo cancellation (ctrl-x < | ctrl-x >)
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  /quit: quit the program (ctrl-x q)
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  /help: display this help message (ctrl-x ?)
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In call commands:
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  /hangup: hang-up the active session (ctrl-x h)
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  /record [on|off]: toggle/set audio recording (ctrl-x r)
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  /srecord [on|off]: toggle/set audio recording to separate files for input and output
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  /hold [on|off]: hold/unhold (ctrl-x SPACE)
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  /add {chat|audio}: add a stream to the current session
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  /remove {chat|audio}: remove a stream from the current session
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Available audio input devices: None, system_default, Built-in Input, Built-in Microphone
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Available audio output devices: None, system_default, Built-in Output
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Using audio input device: Built-in Microphone (system default device)
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Using audio output device: Built-in Output (system default device)
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Using audio alert device: Built-in Output
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Type /help to see a list of available commands.
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2009-10-29 22:42:14 Registered contact "sip:puioxbqy@192.168.1.124:50150" (expires in 600 seconds).
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Other registered contacts:
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  sip:jiozqyud@192.168.1.124:49569 (expires in 423 seconds)
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Detected NAT type: Port Restricted
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adi@umts.ro> 
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}}}
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ICE connectivity checks results:
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{{{
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adi@umts.ro> /rtp
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Output of RTP statistics and ICE negotiation results on console is now activated
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adi@umts.ro> /audio ag@sip2sip.info
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Initiating SIP session from 'sip:adi@umts.ro' to 'sip:ag@sip2sip.info' via sip:81.23.228.150:5060;transport=udp...
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ICE negotiation succeeded in 0s:644
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Local ICE candidates:
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(RTP)	 95.97.50.27:55656        	 type srflx
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(RTP)	 192.168.1.122:55656      	 type host
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(RTP)	 10.211.55.2:55656        	 type host
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(RTP)	 10.37.129.2:55656        	 type host
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(RTCP)	 95.97.50.27:55890        	 type srflx
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(RTCP)	 192.168.1.122:55890      	 type host
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(RTCP)	 10.211.55.2:55890        	 type host
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(RTCP)	 10.37.129.2:55890        	 type host
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(RTP)	 81.23.228.150:51782      	 type prflx
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(RTCP)	 81.23.228.150:51783      	 type prflx
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Remote ICE candidates:
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(RTP)	 81.23.228.150:51780      	 type relay
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(RTCP)	 81.23.228.150:51781      	 type relay
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(RTP)	 95.97.50.27:55876        	 type srflx
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(RTP)	 192.168.1.122:55876      	 type host
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(RTP)	 10.211.55.2:55876        	 type host
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(RTP)	 10.37.129.2:55876        	 type host
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(RTCP)	 95.97.50.27:54037        	 type srflx
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(RTCP)	 192.168.1.122:54037      	 type host
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(RTCP)	 10.211.55.2:54037        	 type host
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(RTCP)	 10.37.129.2:54037        	 type host
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ICE connectivity check results:
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(RTP)	 192.168.1.122:55656 <--> 192.168.1.122:55876 	Succeeded
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(RTP)	 10.211.55.2:55656 <--> 10.211.55.2:55876 	Succeeded
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(RTP)	 10.37.129.2:55656 <--> 10.37.129.2:55876 	Succeeded
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(RTCP)	 192.168.1.122:55890 <--> 192.168.1.122:54037 	Succeeded
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(RTCP)	 10.211.55.2:55890 <--> 10.211.55.2:54037 	Succeeded
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(RTCP)	 10.37.129.2:55890 <--> 10.37.129.2:54037 	Succeeded
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(RTP)	 95.97.50.27:55656 <--> 95.97.50.27:55876 	Succeeded
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(RTCP)	 95.97.50.27:55890 <--> 95.97.50.27:54037 	Succeeded
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(RTP)	 81.23.228.150:51782 <--> 81.23.228.150:51780 	Succeeded
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(RTCP)	 81.23.228.150:51783 <--> 81.23.228.150:51781 	Succeeded
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Audio session established using "G722" codec at 16000Hz
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Audio RTP endpoints 192.168.1.122:55656 (ICE type host) <-> 192.168.1.122:55876 (ICE type host)
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}}}