versatica / mediasoup

Cutting Edge WebRTC Video Conferencing
https://mediasoup.org
ISC License
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[Rust] Transport with only audio producer results in server-side transport cc is not created. #1400

Closed mstyura closed 4 months ago

mstyura commented 4 months ago

Bug Report

WebRtcTransport with only audio producer with enabled transport congestion control results in transport CC is not actually enabled on server side, i.e. Transport::tccServer is nullptr

Your environment

Issue description

Steps to reproduce:

  1. Run rust/examples/echo
  2. Edit rust/examples-frontend/echo/src/index.ts to disable video and hence video producer
  3. Run rust/examples-frontend/echo
  4. Open echo app in browser http://localhost:3001/

Actual result:

Bitrate of sent audio is started at high values and drops to minimum allowed by webrtc (16Kb/s).

image

Expected result:

Bitrate of sent audio is depend on actual network bandwidth available.

ibc commented 4 months ago

Please paste the browser side SDP after creating the audio Producer and also the RtpParameters given to transport.produce() in server side.

mstyura commented 4 months ago

I've found an issue and will provide MR with fix, this was just a justification issue to link with MR.