WebRTC Call Audio - Quiet, Muffled and Drops
Reported by
mar...@missionlabs.co.uk,
Jul 16
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Issue descriptionUserAgent: Mozilla/5.0 (Macintosh; Intel Mac OS X 10_12_6) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/67.0.3396.99 Safari/537.36 Steps to reproduce the problem: 1. WebRTC call - normally on MacOS but have seen the issue on windows as well. What is the expected behavior? What went wrong? I'm currently experiencing an intermittent issue with some VOIP WebRTC voice calls. The symptom is that the outbound audio can sometimes fade in and out and sounds extremely muffled or even disappears momentarily. The 2 audio files reference here show examples or a snippet from a good call and then a bad call, both very close together. The audio is captured server side. Good quality call - https://s3-eu-west-1.amazonaws.com/audio-samples-mlcl/Good.mp3 Poor quality call - https://s3-eu-west-1.amazonaws.com/audio-samples-mlcl/Poor.mp3 I found a link to previous chrome bug which seems similar but never got resolved. https://bugs.chromium.org/p/webrtc/issues/detail?id=5137 Any ideas/help would be greatly be appreciated. Did this work before? No Does this work in other browsers? N/A Chrome version: 67.0.3396.99 Channel: stable OS Version: Flash Version:
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Jul 18
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Jul 23
Thanks for filing the issue... Unable to reproduce the issue on reported chrome version 67.0.3396.99 using Mac 10.13.5. Steps: --------- 1. Launched reported chrome 2. Navigated to "https://appr.tc/" 3. Joined inbound Call (as we are unable to make Outbound call) As we are able hear the audio clearly with good quality @Reporter: Request you to retry this issue with fresh profile without any extensions/apps or reset all the flags and let us know if issue still persists. Thanks...!
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Jul 27
The 'bad' file sounds like audio packets not coming through in an even flow. Please check your network conditions.
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Jul 27
I get the exact same recording from the local stream so it can't be network connections...
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Jul 27
Thank you for providing more feedback. Adding the requester to the cc list. For more details visit https://www.chromium.org/issue-tracking/autotriage - Your friendly Sheriffbot
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Jul 31
Tested this issue and unable to reproduce the issue on latest chrome stable 68.0.3440.75 using Mac 10.13.5. @Reporter:Could you please upgrade to latest chrome stable 68.0.3440.75, you can download latest chrome builds here:" https://www.chromium.org/getting-involved/dev-channel " and let us know if issue still persists. Thanks..!
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Sep 3
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Sep 3
+hlundin@ since he worked on https://bugs.chromium.org/p/webrtc/issues/detail?id=5137 which is now closed
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Sep 4
marcus@: re comment 5, how is the local stream set up and how do you make the recording?
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Sep 4
The local stream is set up user navigator.mediaDevices.getUserMedia, and the recording is done using recordRTC - I doubt the recording causes the audio drop as the recording on the server sounds the same. My guess is it's something to do with echoCancellation or noiseSuppression, we seem to see the issue more with quality headsets so it's as if the echoCancellation is being applied by both the headset and chrome which then causes the audio drop. What do you think?
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Sep 4
I am trying to understand the full scenario here. Is it correct that clients running WebRTC in Chrome transmits audio streams to a server and that bad audio is detected and recorded at the receiving end? If so, what type of server solution is involved?
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Sep 4
Yes the audio stream is transmitted to a server, the server solution is using Janus - but the bad audio is being recorded on the client locally and the server. I'm pretty certain the audio drop is coming from the local stream that comes from getUserMedia.
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Sep 4
Does the issue reproduce locally using https://webrtc.github.io/samples/src/content/peerconnection/audio/ |
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Comment 1 by vamshi.kommuri@chromium.org
, Jul 17