3CX Webmeeting audio codecs

[FridayIT]

Good evening,

We have been noticing quality issues when we have client's dial in to a web meeting. Joining by web seems to be fine but when we get more than a couple of people dial in, the quality of the call between dialed in and web joined client's turns into a garbled mess.

through troubleshooting, we can see that the webmeeting seems to change the dialed in callers from our globally set preferred codec (g711u/alaw) to g722 (confirmed with google chrome and wireshark) which we believe to be causing the issue, this seems to happen even if we remove all other codecs from the bridge.

I was curious to see if anyone has run into this issue.

 

[ls3]

[...]the quality of the call between dialed in and web joined client's turns into a garbled mess. [...]

We have the same problem. Phone dialed in users sounds muffled and like you say, its a garbled mess : - (

Actually we have no solution for that. Maybe this is "normal" ?!

 

[FridayIT]

What codec does your trunk use? we are using g711u/a.

 

[ls3]

We have the problem internally and externally, so once over the trunk, and once without participation of the trunk. But only in phone conferences through the central conferencing number.

Codecs g711/a/u -> g722 -> 729 -> GSM

This order is also in the trunk and in all phones internally as externally . . .

 

[FridayIT]

Do you happen to notice if the issue is intermittent or triggered if a dial in user is muted and unmuted a few times? we were able to more easily replicate when we did this.

Or is the quality always sounding poor for you?

 

[ls3]

It sounds poor in the moment when the central conferencing number is used.
All connections are sounding good but audio conferences using the conferencing extension sound like garbage in all situations (externally internally, mixed, 2 participants 5 participants, 10 participants) all is sounding the same.