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. Since build 5.2.1584, H265 codec is also supported for publishing.

Codecs supported

  • H264
  • H265 (since build 5.2.1584)
  • 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.

H265 publishing

Since build 5.2.1584, RTP stream in H265 codec may be published via RTSP

ffmpeg -re -i source.mp4 -c:v libx265 -c:a aac -b:a 160k -bsf:v hevc_mp4toannexb -keyint_min 60 -profile:v main -preset veryfast -x265-params crf=23:bframes=0 -f rtsp -rtsp_transport tcp rtsp://test1.flashphoner.com:554/test

To do this, H265 should be added to codecs supported list

codecs=opus,alaw,ulaw,g729,speex16,g722,mpeg4-generic,telephone-event,h264,vp8,flv,mpv,h265

and to exclusion lists

codecs_exclude_sip=mpeg4-generic,flv,mpv,h265
codecs_exclude_sip_rtmp=opus,g729,g722,mpeg4-generic,vp8,mpv,h265
codecs_exclude_sfu=alaw,ulaw,g729,speex16,g722,mpeg4-generic,telephone-event,flv,mpv,h265

H265 stream will be transcoded to H264 or VP8 to play from server!

RTSP port setup

TCP port 554 is used by default to publish RTP stream via RTSP

rtsp.port=554

However, the port is privileged and is available to listen to applications launched by root only. If WCS is starting from flashphoner user (the default launch mode), the port should be changed, for example

rtsp.port=5554