I’ve been researching on ways to smooth out video and audio in a mediasoup router, potentially those that make mediasoup reselient against network congestions and other disruptions considering the fact that now people are streaming and having meetings more than ever.
My questions would be specific to this header extension: https://webrtc.googlesource.com/src/+/refs/heads/master/docs/native-code/rtp-hdrext/playout-delay. Apparently this extension allows the sender to configure the jitter buffer on the receiver and help it in smoothing out playback on its end.
From what I understand this requires changes to the sdp and appending a line that configures it on the sender’s side. How would one go about adding this to mediasoup and testing it out, because I believe it can have a major impact on perceived performance, especially in very low bandwidth scenarios.
And since this is related to WebRTC and not necessarily related to mediasoup, I’ve chosen to post this in the ‘Off Topic’ group.
Any and all the help / questions / critique will be appreciated.