To run online broadcasts you can use special hardware or software video capturing devices (Live Encoder). Such devices or programs capture a video stream and send it to the server via the RTMP protocol.
Web Call Server 5.1 can receive an RTMP video stream from such a device or software (Wirecast, ffmpeg, OBS Studio, FMLE etc.) encoded to H.264 + AAC or Sorenson Spark + Speex and broadcast this video stream to browsers and mobile devices.
Receiving incoming audio- and video streams via the RTMP protocol
Broadcasting of the received video stream to browsers and platforms: any among ones supported by WCS
Uses video stream playback technologies: any among ones supported by WCS
1. Live Encoder establishes a connection to the server via the RTMP protocol and sends the publish command.
2. Live Encoder sends the RTMP stream to the server.
3. The browser establishes a connection via Websocket and sends the play command.
4. The browser receives the WebRTC stream and plays that stream on the page.
Below is the call flow when an RTMP stream is broadcast from an external source (Live Encoder)to the WCS server
Parsing stream URL parametersWhen RTMP stream is published or played on WCS, RTMP connection and stream parameters may be set in stream URL like this:
Where
WCS server passes the parameters to backend server in REST hook in
This feature can be used for example to authenticate client on backend server while publishing or playing RTMP-stream on WCS server. |
4. Some RTMP functions does not supported and will be ignored:
5. Some RTMP-encoders does not support KeepAlive.
Symptoms: disconnection occurs often while stream publishing with RTMP-encoder.
Solution: switch KeepAlive off for RTMP on the server using the following parameter in flashphoner.properties file
keep_alive.enabled=websocket,rtmfp |