Fluctuating voice quality

[srttbe]

Hello,

We are evaluating the 3CX and we note fluctuating voice quality while using 3CXPhone.

The PBX is in our private cloud, behind a firewall. We have two desktops, powerful enough, at two distant locations. Both locations have a decent VDSL internet line, the cloud has enough bandwith. Both softphones are connected with the cloud through a VPN. Both 3CXPhones have the same configuration and both extensions also. We use at both sides the same Logitech headset. In order of preference, we've set the codec list G.722, PCMU, PCMA and GSM. The pbx is configured to send audio.

No matter if user A calls user B or if user B calls user A, the problem is the same.

The caller ( A or B ) has a constant good sound quality.
The receiver ( B or A ) has a fluctuing sound quality during the same call, sometimes good, sometimes less, without any visible reason.

Any idea ?

Best regards.

Michel.

 

[leejor]

Poor call quality meaning what? garbled, dropped conversation, echo?

Have you done a latency test. in many cases, the bandwidth may be there, but because of the routing there may be latency issues. have you done tests without the VPN? Depending on what you are using, that may be introducing issues.

Have you done any tests to narrow down the problem to one particular end, calls from either one, to a third location?

Have you done any tests restricting the codecs to only certain ones?

Have you tried changing the "PBX delivers audio" settings?

 

[srttbe]

Thank for your reply Leejor,

The best way to explain what we noticed is that the voice quality at the receiver side decreases and after some seconds backs to a better one. As it jumps ( juste an example for understanding ) from G.722 to GSM, goes back to G.722 and so on.

Latency is pretty good and stable, between 40ms and 45ms at both sides, no more. And of course, we did the tests with any other signifiant load on the computers and on the internet connexions as well.

As I said, the problem is independent of the location. The caller, no matter where he is, has a stable quality, the receiver of the call has not. Since we do not have a third location, I can not try it but what I noticed does not drive me to a network problem. Do not mind , I've a 30 years background in computer and network including the internet. At my previous job, I was one of the team who implemented the very first internet access in this town ... my God, 16 years yet :O So I know that latency and jitter may cause a lot of problem in real time protocols. Believe me, the reason is elswhere.

As I said also, yes, both extensions are set with "PBX delivers audio". Without that, we have a stable quality but the lesser one at both sides.

We tried with and without VPN, same issue.

You're right, we did not yet tried with only one codec activated in both 3CXPones. We will try it Monday.

Have a nice week-end, your help is appréciated.

Michel.

 

[leejor]

Once the Codec is negotiated, when the call is first established, my understanding, is that it should not be changing mid-way through the conversation. Is that what you are suggesting is happening?

Unless there is a bandwidth issue, I would avoid using the lower bit-rate Codecs. As a test I would try setting each end to use the same (G711u/a) Codec and see if it makes a difference.

As I said also, yes, both extensions are set with "PBX delivers audio". Without that, we have a stable quality but the lesser one at both sides.

I'm not sure what you mean by "the lesser one at both sides", can you clarify?

Could there possibly be a common application, running on each PC, that might be having an affect?

Have you tried calls from a VoIP set, to either location?

 

[srttbe]

Okay, by selecting only G711/A, calls are good and the quality is stable !

A question : When our land lines will be connected to the pbx, may we keep only one codec on softphones ?

Once the Codec is negotiated, when the call is first established, my understanding, is that it should not be changing mid-way through the conversation. Is that what you are suggesting is happening?

That the best way I've found to try to tell what I hear

I'm not sure what you mean by "the lesser one at both sides", can you clarify?

What I mean by "the lesser quality at both side" is that the caller and the receiver have both a quality like a GSM.

Thank for your help !

 

[leejor]

As I said, unless bandwidth is an issue, for the best quality, the G711 Codecs should be at the top of the list in all devices, and preferably in the same order. Other Codecs can be in the list (meaning they are available), but unless the phone/device connects to "something" that is unable to use G711, then they will not be chosen.

Glad you were able to resolve the issue.