I have some dudes about a few things, and I really appreciate any clarification. I don’t want support, only clarifications.
mediasoup acts as a webrtc endpoint, like a peer, so for instance, are browser and mediasoup handling between them all the things, like jitter buffer, etc?
When I use a directTransport for consume RTP packets from produces (browser), I get these packets passed to any mediadoup jitter buffer, or are transparent consumed in directTransport (
directTransport.on('rtp', ...) ) app?
The same happens with plainTransport? (of course, through udp)
I will details my issue. I am using mediasoup only for audio, between a browser (the client) and a server which host mediasoup and opus de/coders to pipe to a local audio system:
browser mic --- > internet ---> server ---> mediasoup direct/plainTransport ---> udp/rtp depayload ---> opus decoder ---> local pcm audio out
browser spk <--- internet <--- server <--- mediasoup direct/plainTransport <---udp/rtp payload <--- opus enconder <--- local pcm audio in
All runs like a charm, but… after few minutes, experiment increasing delay from browser mic over the time. I look at webrtc-internal in chrome and see that totalRoundTrip is high. I am not experiment delays in reverse order, from server to browser.
So, the last question is, I need to make a jitter buffer for received rtp packets in the transport to mitigate this issue?
PD: If developers think this post must go in off topic, please move it.