WebRTC: audio quality issue on variable low bandwidth connections
Reported by
silvia.p...@coviu.com,
May 1 2018
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Issue descriptionUserAgent: Mozilla/5.0 (Macintosh; Intel Mac OS X 10_13_4) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/66.0.3359.117 Safari/537.36 Steps to reproduce the problem: 1. Set up a 2 endpoint WebRTC call 2. have one endpoint vary their bandwidth between 50 and 250 Kbps 3. audio breaks up and gets unusable What is the expected behavior? audio is prioritized over video, so the audio continues to work, even when the video's quality fluctuates; we don't care if the video stalls completely, but the audio should always get sufficient bandwidth to get through What went wrong? video seems to be prioritized over audio, which makes audio drop out sometimes Did this work before? N/A Does this work in other browsers? Yes Chrome version: 66.0.3359.117 Channel: stable OS Version: OS X 10.13.4 Flash Version:
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May 1 2018
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May 2 2018
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May 2 2018
Stefan, you might have an opinion on this. I guess your signal talk is a bit outdated by now :-)
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May 3 2018
There are some things we can do to improve this situation, such as introducing a congestion window and better manage the send-side queues. Our plan is to look at that in coming quarters.
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May 4 2018
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Comment 1 by silvia.p...@coviu.com
, May 1 2018