RTMP to Janus Videoroom
Warning I am not a Go programmer
This is very much in the work-in-progress/proof-of-concept phase!
I really don't know go, I learned just enough so I could make use of the pion/webrtc library.
This is adapted from the Janus example in Pion's example-webrtc-applications repo.
Usage
rtmp-janus <listen host:port> ws://janus-host:port
# example: rtmp-janus :1935 ws://127.0.0.1:8188What does this do?
When launched, this app:
- Starts listening for incoming RTMP sessions.
- Connects to Janus gateway via websocket, establishes a session.
When an RTMP connection is received, it parses the RTMP "key" for a room ID.
Example: rtmp://127.0.0.1:1935/live/1234 - the 1234 part of that URL becomes the room ID.
The live part of the URL (the application name) can be whatever you'd like, it's ignored.
The app then joins the janus videoroom and establishes a WebRTC session.
I'm assuming all incoming RTMP sessions are using H264 for video and AAC for audio.
H264 data is re-packed from FLV format into Annex-B format, but otherwise passed as-is (no decoding/encoding).
AAC audio is resampled to 48000kHz, stereo audio and encoded to Opus using ffmpeg.
TODO
- Figure out what events from Janus I should handle (I just blast audio/video).
- Check more sources for timing info (like the SPS NAL).
- Support more audio samplerates (maybe?).
- Generalize audio decoding, support more input audio codecs.
LICENSE
MIT (see LICENSE)
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