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- parameters of your SIP account the call will be made from;
- the stream name to republish the call to (the toStream parameter), for example, call_stream1$stream1;
- the name of your second SIP account the call will be made to.
3. Receive and answer the incoming call on the softphone:
4. Open the Player web application and in the "Stream" field specify the name of the stream the call is redirected to (in our example: call_stream1):
5. Click "Play". The stream starts playing:
6. To terminate the call, send /call/terminate from the REST client to the WCS server and pass the call id in the parameters:
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Call
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flow
Below is the call flow when using the SIP as RTMP example to create the call and the Player example to play it
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14. Receiving confirmation from the SIP server
SIP as stream recording
All streams captured from SIP calls can be recorded on server. To do this, set the following parameters in flashphoner.properties file:
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sip_single_route_only=true
sip_record_stream=true |
The following codecs are supported:
- Video: H264
- Audio: opus, PCMA (alaw), PCMU (ulaw)
Stream recording is described here in details.
Known issues
1. Stream captured from SIP call, can not be played, if RTP session is not initialized for this stream.
Symptoms: SIP stream is published on server, but can not be played.
Solution: enable RTP session initializing with the following parameter
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rtp_session_init_always=true |