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Supported platforms and browsers

Operation flowchart

1. The browser connects to the server via the Websocket protocol and sends the publish command.
2. The browser captures the microphone and the camera and sends the WebRTC stream to the server.
3. The second browser establishes a connection also via Websocket and sends the play command.
4. The second browser receives the WebRTC stream and plays this stream on the page.

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7. Diagrams in chrome://webrtc-internals for the second stream:

Known issues

1) . Possible bug in the Safari browser on iOS leads to freezes while playing via WebRTC

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а) enable the transcoder on the server by setting the following parameter in flashphoner.properties

disable_streaming_proxy=true

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session.createStream({constraints:{audio:true,video:{width:320,height:240}}}).play();

2. Audiocodec PMCU ia used instead of Opus when sttream is published via RTMP and is played via WebRTC

Symptoms: PMCU codec is shown in chrome://webrtc-internals

Solution: switch Avoid Transcoding Alhorithm off using the following parameter in flashphoner.properties

disable_rtc_avoid_transcoding_alg=true