CorentinJ commented 1 year ago

Please read first!

Please use discuss-webrtc for general technical discussions and questions. If you have found an issue/bug with the native libwebrtc SDK or a browser's behaviour around WebRTC please create an issue in the relevant bug tracker. You can find more information on how to submit a bug and do so in the right place here

[x] I understand that issues created here are only relevant to the samples in this repo - not browser or SDK bugs
[x] I have provided steps to reproduce
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Browser affected

Tested on Chrome 107.0.5304.107 (Official Build) (64-bit) and Firefox 107.0 (64-bit)

Description

All examples under the RTCPeerConnection category output audio that is intermittently cut.

Steps to reproduce

I have been able to reproduce this issue with different browsers (chrome & firefox as above) and on different hardware, including a recent Ubuntu install and win64.

Any of the following samples will reproduce this issue: Basic peer connection demo in a single tab Basic peer connection demo between two tabs Audio-only peer connection demo

The samples in the first category do not present this same issue Record stream Audio-only getUserMedia() output to local audio element

Expected results

A smooth audio output

Actual results

This video where both input and output audio is recorded illustrates the problem:

https://user-images.githubusercontent.com/12038136/203829510-f1a5758c-e9a8-44b8-b76e-d158ca653dee.mp4

fippo commented 1 year ago

I think you're hitting AEC and "double talk". https://groups.google.com/g/discuss-webrtc/c/ddS08EgXfSU/m/Sq9OIsVXCwAJ has some hints