Forgive me if this is the wrong place to ask.
Is there any way to configure mediasoup so a specific producer/consumer values stability over latency? I’m trying to play audio (music) and video (webcam), but get way too much stuttering on the audio stream.
Don’t get me wrong, it’s not super bad, but there is what sounds like a minor “skip” in the sound from time to time (like average every 5-10 seconds, up to 30 seconds, down to 1 second).
I’ve been trying to search google about webrtc and how to improve audio stability, but I can’t find anything on the subject.
- Is there any way to force a high constant FEC along with a stream to correct any problems?
- Or set a target latency higher than the connection capability?
- Is it possible to disable or change the behaviour of acceleration / deceleration?
- Does webrtc (udp) do retransmissions at all?
Typical use case for this modified behaviour is one-to-many. However the music streamer will also participate in many-to-many conference at the same time (which shouldn’t be affected if possible, but can be run with different server and/or client code, so doesn’t matter).
I’ve already considered other technologies, but really wish for the stream to be realtime.