Audio clipping for phone users when the presenter is using web audio.

[MMDTerry]

We have been having a bunch of vendors doing product training sessions with our sales staff. The issue we are having is the presenter's audio clipping for participants who are calling in using the audio bridge. Doesn't matter if they use the conference extension (internal call) or the external DID (external call) the audio from people presenting who are using a headset/camera mic are over driven. Other phone users in the meeting sound fine. People who are using their PC speakers or headset and only the web meeting do not have this issue, the phone users and web users all sound the same to them. Also, the recorded meetings sound fine. It seems like and level-matching issue between the web meeting audio platform and the audio bridge for phone users. Auto gain control is on in the web meeting, and this has been an issue with multiple presenters from different vendors so I don't believe it's an issue on their end. Has anyone else had an issue like this? Any solutions?

 

[LeonidasG_3CX]

Since the issue happens even with a direct call to the meeting from a locally connected user without a sip trunk & the final recording is fine, this indicates an issue on the caller's side, between the caller and the PBX or on the PBX.

Is your PBX using IVP6?

 

[MMDTerry]

no ipv6.

I don't believe it's the endpoint, approx 15 callers had this issue.

Happens to internal callers and outside callers over the sip trunk into the meeting.

phone-to-phone meeting participants are fine, no distortion, the audio level does seem a bit low

phone-to-web meeting participants are fine, no distortion.

web meeting-to-phone users are very loud and distorted. again multiple web users across multiple meetings so it's not a device or user issue on the web side.

standard audio meetings to the same conference extension no issues.

zero other call issues, if it was the endpoint they should be having issues with all calls, not just conferences.

Waiting on the video link and I can post a comparison of what the phone user hear vs what everyone else is hearing.

 

[LeonidasG_3CX]

We're releasing a WebRTC equivalent of wireshark early next week that will allow both us and technical users to troubleshoot things like this. I'll let you know once it's out so we can take a better look with more data and less guesswork.

 

[MMDTerry]

Oh cool, that will be very useful for troubleshooting.

 

[LeonidasG_3CX]

The release of this featureset has been moved ahead by 1 week. We expect it to be available on Monday Nov 14.

 

[Evolute IT]

The release of this featureset has been moved ahead by 1 week. We expect it to be available on Monday Nov 14.

We're releasing a WebRTC equivalent of wireshark early next week that will allow both us and technical users to troubleshoot things like this. I'll let you know once it's out so we can take a better look with more data and less guesswork.

Is it gonna be only for the WebMeeting app or also for Webclient/DesktopApp troubleshooting? Like the "WebRTC Line not registered" error that appears sometimes.

 

[LeonidasG_3CX]

Officially, you can analyze 3CX video conferences made on Chrome or on the 3CX Desktop app. When troubleshooting the 3CX video conferencing application, you'll be presented with a lot more different types of data than what is normally provided by WebRTC.

There is no official support for other products beyond 3CX's video conferencing.

That being said, unofficially, it's possible to also analyze any WebRTC based web app that you can open and run on Chrome as long as you're able to navigate to the following URL chrome://webrtc-internals where you can press the "Create WebRTC Dump" button and load it in the analyzer. The amount of data will be limited to what vanilla webrtc stats are normally allowed by Chrome.

As this is unsupported, questions on the forum regarding usage of the tool for other purposes beyond 3CX's video conferencing product will not be dignified with an answer past this point.

 

[MMDTerry]

The release of this featureset has been moved ahead by 1 week. We expect it to be available on Monday Nov 14.

Has this been released?

 

[LeonidasG_3CX]

Has this been released?

Yes however the feature will only be visible to users who upgrade to U6 once it's out.

 

[MMDTerry]

Yes however the feature will only be visible to users who upgrade to U6 once it's out.

Anything I can try that might fix this issue before then? It's still affecting all dial-in audio bridge users during web conferences.

 

[LeonidasG_3CX]

Anything I can try that might fix this issue before then? It's still affecting all dial-in audio bridge users during web conferences.

Unfortunately no, this feature requires a U6 PBX due to critical changes made on the PBX side needed to support this feature in the first place.

 

[LeonidasG_3CX]

Anything I can try that might fix this issue before then? It's still affecting all dial-in audio bridge users during web conferences.

Please keep in mind that U6 Alpha is out and that you may be able to use try these features out.