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Push a test stream
From a machine with ffmpeg and a local mp4 file, run:
ffmpeg -re -stream_loop -1 -i file.mp4 -c:v copy -c:a libopus -f rtsp -rtsp_transport tcp rtsp://YOUR_SERVER:8554/mystream
Note: -c copy for audio only works over WebRTC if the source audio is already
Opus/G711 (WebRTC here does not support AAC). If your mp4 has AAC audio, transcode it to
Opus as shown above, or switch this page to HLS mode, which plays AAC fine.