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
- Asterisk cmd Ices: Stream to an icecast server
- Asterisk cmd Monitor
- Asterisk record calls
April 7, 2023
Phone System 