Web Call Server 5.2 - EN
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Overview

Since build 5.2.902 it is possible to publish RTP stream via RTSP directly to WCS. Both TCP interleaved and UDP transports are supported. This can be useful to publish H264+Opus streams from hardware or software encoder supporting RTSP.

Codecs supported

  • H264
  • VP8
  • AAC
  • Opus

Operation flowchart

  1. Live Encoder connects to the server via RTSP and sends ANNOUNCE command.
  2. Live Encoder sends RTP 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.

Quick manual on testing

1. For test we use:

  • WCS server
  • ffmpeg
  • Player web application in Chrome browser to play the stream

2. Start RTSP H264+Opus stream publishing using ffmpeg

ffmpeg -stream_loop -1 -re -i bunny360p.mp4 -c:a libopus -ac 2 -ar 48000 -c:v copy -b:a 96K -b:v 500K -f rtsp -rtsp_transport tcp rtsp://test1.flashphoner.com:554/test

3. Open Player application. Set stream name to "Stream" field and click "Start". The stream captured playback begins.

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