All-in-one WebRTC SFU server, an overpowered version of this example.
Low-latency streaming of a singular audio/video pipeline.
- The underlying WebRTC protocol provides realtime media streaming capabilities.
- Support for popular protocols like VPX, AV1 and H264/H265. Handles concurrent streaming from multiple servers to multiple users.
- Can be improved to your specific needs because of its modular structure.
cp .env_sample .env
now edit the .env file with desired configuration. Note that RTP codec must match RTP stream contents. Start the server with go run . in project root.
Start an RTP stream from local or remote device and send the udp packets with mtu=1200 (check .env) to the server. GStreamer or FFmpeg can be used for this. Two simple examples runnning on the local device are shown below:
GStreamer (VP8 - video only)
gst-launch-1.0 videotestsrc ! video/x-raw,width=640,height=480,format=I420 ! vp8enc error-resilient=default keyframe-max-dist=10 auto-alt-ref=true cpu-used=8 deadline=1 ! rtpvp8pay mtu=1200 ssrc=12345 ! udpsink host=127.0.0.1 port=5004
FFmpeg (x264 - video only)
ffmpeg -re -f lavfi -i testsrc=size=640x480:rate=30 -pix_fmt yuv420p -c:v libx264 -g 10 -preset ultrafast -tune zerolatency -ssrc 12345 -f rtp 'rtp://127.0.0.1:5004?pkt_size=1200'
Enter the ssrc of the RTP stream and start the stream. In the above RTP streams, ssrc is set to 12345; so enter that. ssrc for video and audio tracks must be the same if both are used simultaneously.
- Better documentation
- More and better examples
- PLI handling
- Cleaner code
- More customization
- Better logging
- ...