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3CX Webmeeting WebRTC Audio Codec

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gavpop

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We use 3CX Webmeeting quite a lot and I'm getting reports from users about bad audio.
This has meant a number of our users are now resorting to only using the webmeeting for video, but then dialling out on a conference phone to create the audio side of the webmeeting. Audio is crystal clear as this then uses the G711a audio codec via the 3CX Phone System.
This is clearly a crazy way of doing things and is making a farce of webmeetings.

I've been doing quite a lot of research into this and it seems WebRTC has two options to use for the audio side of things...G711 or OPUS. OPUS, by all accounts, being the better one to use.
I'd really like to play around with the audio options for webmeetings but cannot find any way of doing so.
I don't know if this would be controlled by 3CX or whether this would be browser (chrome) based config.

If anyone has any insights, I'd love to hear them.
 
WebMeeting only uses Opus.
-This is most likely not a codec issue, but either a networking issue or maybe a bad route between your users and our meeting server endpoints. I would suggest you go to PBX > Settings > Conferencing > Meeting location and switch to various other servers in the list, create NEW meetings and try again. Let us know if your experience improves.

Also for your own troubleshooting, you can type the following in chrome while you are in a meeting and get some idea of what's happening. While you're in a meeting if you enter the following page, you should see 1-2 tabs. Clicking on one of them should show a lot of information. What you're interested in are the dropdown menus that say something like "Audio_recv" and "Audio_sent". If you expand these, you should see audio track information about everyone in the meeting.
chrome://webrtc-internals
 
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