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• parameters of your SIP account the call is made from;
• URL of the RTMP server the call is republished to, in this case - specify the URL of the WCS server;
• the name of the stream to rebroadcast the call to (the rtmpStream parameter), for instance, rtmp_stream1;
• the name of your second SIP account where the call is made to.

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3. Receive and answer the call on the softphone:

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4. Open the Player web application. Specify the URL of the RTMP server and the name of the RTMP stream the call is redirected to, then click the "Play" button. The call starts playing:

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5. Terminate the call in the softphone.

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9. Receiving confirmation from the SIP server

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

rtp_session_init_always=true

2. SIP callee does not recognize DTMF signals if audio data generation is no enabled

Symptoms: SIP callee does not recognize PIN code sent as DTMF

Solution: enable audio and video data generation for SIP call with the following parameter

generate_av_for_ua=all