PlaySip

playSIP

http://www.live.com/playSIP/

Description

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

Details

"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.)

Potential usage with Asterisk

  • place caller and callee into a MeetMe conference and add the playSIP client as a third (quiet) partner



See also


playSIP

http://www.live.com/playSIP/

Description

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

Details

"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.)

Potential usage with Asterisk

  • place caller and callee into a MeetMe conference and add the playSIP client as a third (quiet) partner



See also


Created by: JustRumours, Last modification: Sun 12 of Apr, 2009 (05:46 UTC) by admin
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