Pion-WebRTC-SFU

All-in-one WebRTC SFU server, an overpowered version of this example.

Overview

Low-latency streaming of a singular audio/video pipeline.

1. Fast

   The underlying WebRTC protocol provides realtime media streaming capabilities.

2. Flexible

   Support for popular protocols like VPX, AV1 and H264/H265. Handles concurrent streaming from multiple servers to multiple users.

3. Expandable

   Can be improved to your specific needs because of its modular structure.

Usage

1. Configure .env and start server

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.

2. Start RTP stream

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'

3. Go to localhost:8080

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.

TODO

• Better documentation
• More and better examples
• PLI handling
• Cleaner code
• More customization
• Better logging
• ...