I mean, if you're streaming audio and video Real Time, is there really any point to TCP? If a few frames get dropped, then bursted back once the connection stabilizes, does that improve the user experience in any way?
WebRTC seems like a perfect candidate for UDP communications for the actual media streams.
The goal of the topic is to explore simple option for server-client communication using low-latency communication, without reliability and without ordered delivery.
WebRTC can be used for such case, although it is not designed for it. Due to that implementation is very complex and not much adopted. This is something we trying to explore, either new API or simplifications to WebRTC to make it simple choice for UDP in server-client scenarios.
UDP - is something been used by many industries for long time and is well known protocol to make a foundation from.
With UDP you have to create your own feedback mechanism to find the optimal bitrate to stream to the far side at.