Lossless audio streaming over webrtc

[Robin Reumers]

I'm looking at a way to set up a server to client connection and stream lossless audio from the server to the client.

The server will do audio processing and then it should send the audio via 24-bit 44.1 kHz (or 96kHz if the bandwidth permits) to the client, lossless (flac, wav, ..).

I currently only see the option of using the Opus audio codec, but is it possible to customize it for lossless audio streaming?

[Benjamin Schwartz]

You can't do lossless audio using WebRTC.  Try a WebSocket instead.

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