open-webrtc-toolkit / owt-server

General server (streaming/conference/transcoding/anayltics) for OWT. (A.k.a. MediaServer)
https://01.org/open-webrtc-toolkit
Apache License 2.0
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Audio routing problems #864

Open Alex-YddeR opened 3 years ago

Alex-YddeR commented 3 years ago

Environment and platform: 4.3.x,CentOS 7.6, Software, Clustered-Deployment, Chrome-Version 87 , SFU

I had reports from 4 independent (not in the same call) users that they had audio problems with calls with about 5 to 9 participants. Some participants could not hear another participant, while other could hear him. They have used mostly Chrome. Any similar experiences? This issue happening from past 4 Months randomly in some or other call

One more fact, it happens while in call, so suddenly the audio from one participant stops to be sent to some of the others. If the participant leaves and re-enters everyone can hear him again.

The video was ok all the time.

There were 3 participants and suddenly one of the participants could only be heard by one of the others: User1: can hear User2 and User3 User2: can hear User1 and User3 User3: can hear User1, but CANNOT hear User2

Thanks

starwarfan commented 3 years ago

Forward audio stream or mixed audio stream?

Alex-YddeR commented 3 years ago

It's with forward stream.

Alex-YddeR commented 3 years ago

For me this is a big issue, as having larger number of users in a room causes, that usually somebody stops to hear another user, while others hear him.

starwarfan commented 3 years ago

Hi, Is it easy to reproduce with 3 participants? To address this issue, can you provide your steps to reproduce?

Alex-YddeR commented 3 years ago

Hi, Thats the problem it randomly happens. I am assuming if the browser is busy or heavily loaded then it is happening in SFU mode. same browser in MIX mode works well with out any drop.

yesterday this problem happened every 5 mins for atleast 5 times. later joined in MIX did not dropped. its with only audio only

Thanks

starwarfan commented 3 years ago

Hi, Was the audio loss only periodically or during the whole connection? Can you check the webrtc-internals to see if there were audio packets in browser if that happens?

Alex-YddeR commented 3 years ago

Audio loss is during the whole connection lets say there were 3 participants and suddenly one of the participants could only be heard by one of the others: User1: can hear User2 and User3 User2: can hear User1 and User3 User3: can hear User1, but CANNOT hear User2

Now when user 2 close and open the session it works, and it continue to happen with user 2. but same user 2 login with mix then no problem. All this time video was all ok with all the users. Sure will check the logs next time if we face the issue. so i need to check with user3 or user2 logs??

Thanks

Alex-YddeR commented 3 years ago

Again I would like to express that this only happens when the end user's cpu load is high so that server queues running on the browser start to handle tasks very slowly.