I read elsewhere that WebRTC will drop audio, have you had any issues with that?
And generally, you don't need any buffering mechanisms for the clients?
Nice project!
That is 100% controllable! By setting Playout Delay Header[0] you can pick between 'drop everything to stay live' or buffering up to ~40 seconds!
In this project I don't set anything though.
[0] https://webrtc.googlesource.com/src/+/refs/heads/main/docs/n...
That is 100% controllable! By setting Playout Delay Header[0] you can pick between 'drop everything to stay live' or buffering up to ~40 seconds!
In this project I don't set anything though.
[0] https://webrtc.googlesource.com/src/+/refs/heads/main/docs/n...