The MediaStream Recording API lets us record WebRTC streams in the browser, but what about recording a live WebRTC stream in a native app or on the server side? LibSourcey has a new WebRTC module available that lets you do exactly that.
Before going into details the full open source demo code is here.
How it Works
Basically, the server operates as a standard WebRTC peer, so you connect to it from the browser (or another native app) as you would normally by initiating a
RTCPeerConnection, and the server reads any media sent from the peer and encodes it in realtime using FFmpeg.
WebRTC signaling happens courtesy of Symple, our propose built messaging protocol for scalable high speed native to browser communications.
For those of you familiar with the native WebRTC C++ codebase, what we are doing is overriding the
public webrtc::AudioTrackSinkInterface in order to capture audio and video packets from the incoming
scy::av::MultiplexEncoder then works with FFmpeg under the hood to encode and multiplex the live streams into the output file/stream.
The main WebRTC StreamRecorder class is here.
Using the Code
To get started compile LibSourcey with FFmpeg and WebRTC, and samples enabled. The
webrtcrecorder binary will be compiled and you can test it with the provided client code (you will need Nodejs installed).
Please refer to the README in the
webrtcrecorder sample directory for more information.
If you find this code useful or end up using it in a real world scenario please share your thoughts and experience with others in the comments below.
All contributions to the codebase are welcome, and we hope to continue to improve our WebRTC integrations over time. Good luck and happy coding!