playSIP^TM

A command-line SIP session recorder

"playSIP" is a command-line program that can be used to call a SIP IP telephony ('voice over IP') session - i.e., specified by a URL beginning with "sip:" - and (optionally) record the incoming media stream into a file.

Running the program

"playSIP" is run the same way as "openRTSP", with the same command-line arguments, except for the following differences:

Because SIP commands are sent using UDP rather than TCP, the "-t" option is not applicable.
At present, only a single audio session is retrieved (no video). Also, by default, "playSIP" requests that PCM u-law audio be streamed.
To request a different codec, use one of the following options:
- For static RTP payload types, use "-A <codec-number>", where <codec-number> is the static RTP payload format number for the desired codec. So, for example, to request that GSM audio be sent, use "-A 3".
- For dynamic RTP payload types, use "-D <MIME-subtype>", where <MIME-subtype> is the name of the MIME subtype for the desired codec. So, for example, to request that Speex audio be sent, use "-D speex".

Note also that "playSIP" is a 'receive-only' application; it does not send any RTP data of its own. (It does, however, send RTCP "Reception Report" packets.)

Source code

This program uses the "SIPClient", "MediaSession", "FileSink", "QuickTimeFileSink", and several "*RTPSource" modules from the "liveMedia" library, which is distributed as part of the "LIVE555 Streaming Media" source code package. (Other SIP clients could readily be built from this code.)

The source code for the program itself is also bundled with this package, as the files "playSIP.cpp" and "playCommon.cpp", in the "testProgs" directory. See the "LIVE555 Streaming Media" documentation for instructions on how to build this program from source.

"LIVE555", "playSIP", "openRTSP", and the Live Networks logo are trademarks of Live Networks, Inc.

playSIPTM

A command-line SIP session recorder

Running the program

Source code

playSIP^TM