WebRTC video stream broadcast
with conversion to RTMP

Web Call Server converts WebRTC audio-video stream to RTMP and sends it to the specified RTMP server. Thus, a broadcast from a web page to Facebook, YouTube Live, and other services and servers broadcasting live video can be created

Specifications

Diagram of WebRTC broadcast with republishing as RTMP

A WebRTC-enabled browser captures video from the camera and audio from the microphone and sends it to the WCS server using the WebRTC technology protocol stack (ICE, DTLS, SRTP), for which the H.264 video codec and Opus audio codec are used. If browser does not support H.264 (mobile browsers, for example), VP8 video codec can be used instead.

The WebRTC stream is converted to RTMP with H.264 and AAC codecs and sent to the specified server or service address that supports RTMP broadcasts.

Step-by-step WebRTC translation scheme with republishing as RTMP

1. The WebRTC browser establishes a connection to the WCS server and starts publishing the WebRTC stream
2. The WCS converts the received H.264 video stream to RTMP and transcodes the received audio to AAC for compatibility with most RTMP services
3. Next, the RTMP stream is sent to the RTMP server or service

Learn more about republishing WebRTC as RTMP on the Testing page.

Download Web Call Server 5

System requirements: Linux x86_64, 1 core CPU, 2 Gb RAM, Java

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Installation:

1. wget https://flashphoner.com/download-wcs5.2-server.tar.gz
2. Unpack and install using 'install.sh'
3. Launch server using command 'service webcallserver start'
4. Open the web interface https://host:8444 and activate your license

If you are using Amazon EC2, you don't need to download anything.

Launch WCS5 on Amazon EC2