This is a many-to-many websocket based SFU. This has the following features.
- Trickle ICE
- Re-negotiation
- Basic RTCP
- Multiple inbound/outbound tracks per PeerConnection
- No codec restriction per call. You can have H264 and VP8 in the same conference.
- Support for multiple browsers
- Concept of rooms. The conference tracks are forwarded within a given room.
- Supports JWT based authentication to connect to websocket to initiate Webrtc peer connection.
It has a npm module to easily get the ag-webrtc-sfu.js file from here for use on client-sides.
This code can be run in dev mode which requires you to clone the repo since it will be serving static HTML and a JS file. If you do not use the -dev option then the HTML & JS files won't be served.
git clone git@github.com:applegrew/ag-webrtc-sfu.git
cd ag-webrtc-sfu
Execute go build then ./ag-webrtc-sfu -dev -addr=:9900
Open http://localhost:9900. This will automatically connect and send your video. Now join from other tabs and browsers!