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

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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:

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Start broadcasting:

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

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

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Now, broadcasting of the captured stream starts.

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Call flow

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

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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).