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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.

shh don't tell them about playout delay header haha