Web Call Server 5.1 - EN
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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 and AAC and broadcast this video stream to browsers and mobile devices.

Overview

Technical specifications

  • Receiving incoming audio- and video streams via the RTMP protocol

  • Support for the H.264 video codec and AAC audio codec

  • 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

Operation flowchart


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.


Quick manual on testing

Capturing a video stream from an external source and preparing to publishing

1. For the test we use the demo server at demo.flashphoner.com, and as a source of broadcasting - OBS Studio, to play the received stream -  the web application Player in the Chrome browser

https://demo.flashphoner.com/client2/examples/demo/streaming/player/player.html

Configure broadcasting of the RTMP stream to this address rtmp://demo.flashphoner.com:1935/live/, use obsStream as the stream key:

Start broadcasting:

2. To make sure the server receives broadcasting, open the Player web application:


In the "Stream" field specify the stream key and click "Start". You should see the "PLAYING" label:


Now, broadcasting of the captured stream starts.

Call flow

Below is the call flow when an RTMP stream is broadcast from an external source (Live Encoder)to the WCS server

Known issues

1. A stream containing B-frames does not play or plays with artifacts (latencies, lags)

Symptoms:

  • a stream sent by the RTMP encoder does not play or plays with latencies or lags
  • warnings in the client log:
09:32:31,238 WARN 4BitstreamNormalizer - RTMP-pool-10-thread-5 It is B-frame!

Solution: change the encoder settings so, that B-frames were not used (lower encoding profile, specify in the command line etc).

2. AAC frames of type 0 are not supported by decoder and will be ignored while stream pulled playback

In this case, warnings will be displayed in the client log:

10:13:06,815 WARN AAC - AudioProcessor-c6c22de8-a129-43b2-bf67-1f433a814ba9 Dropping AAC frame that starts with 0, 119056e500

3. Some RTMP functions does not supported and will be ignored:

  • FCSubscribe
  • FCPublish
  • FCUnpublish
  • onStatus
  • onUpstreamBase
  • releaseStream


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