This repo demonstrates a RTMP server that on every RTMP publish makes the audio/video available via WebRTC playback.
go run *.go- Open http://localhost:8080/
- Publish an RTMP feed to
rtmp://localhost:1935/publish/foobar. It must be H264 and alaw
gst-launch-1.0 videotestsrc ! video/x-raw,format=I420 ! x264enc speed-preset=ultrafast tune=zerolatency key-int-max=20 ! flvmux name=flvmux ! rtmpsink location=rtmp://localhost:1935/publish/foobar audiotestsrc ! alawenc ! flvmux.
Modify from source https://github.com/Glimesh/rtmp-ingest.git thanks Glimesh
brew install opusfile fdk-aac
apt install -y pkg-config build-essential libopusfile-dev libfdk-aac-dev libavutil-dev libavcodec-dev libswscale-dev