Poor audio quality through handset but fine when using USB headset or Smartphone app?

[Neil Shaw]

Our 3CX installation has a frustrating issue where handsets intermittently have really bad audio yet USB headset users or people using the smartphone app are fine.

We have a dedicated FTTC 60:20 line for up to 8 simultaneous calls. We're running the PBX as a virtual server on a powerful host and the phones have a dedicated switch.

We've tried different handsets from two manufacturers, moved ports around on the switch, stopped the users piggybacking their LAN to PC on the phone and changed many settings on 3CX with no luck. We've tried another internet connection and replaced the switch/router.

The SIP is provided by Gamma who provide all our other SIP's.

We know this system works as it's one we have at 12 other sites.

Any ideas or help would be greatly appreciated.

TIA,
Neil

 

[Lee Cramman]

Sorry, I read virtual server and assumed a cloud install.

No, you don't need an SBC.

 

[Neil Shaw]

Sorry, I read virtual server and assumed a cloud install.

No, you don't need an SBC.

Apologies for the confusion Lee.

 

[Lee Cramman]

No probs. Any chance there could be something hoovering up your bandwidth?

How are your pings to gamma from the PBX? Might be worth watching how they are affected when calls with failed audio take place. It's probably worth double checking your firewall / port forwarding setup as well.

Is it affecting all calls or just some giving you issues?

 

[Neil Shaw]

I've been watching the bandwidth all morning and it's barely using any.

The ping is steady at 12ms with a choppy call.

It's tricky to say if it's all call as some short ones are OK but all long one are affected. However, some short ones are affected it's a roll of the dice!

 

[Lee Cramman]

How about internal calls?

 

[Neil Shaw]

Internal calls are OK.

Further testing. I've tweaked the QoS and things are much better but all calls still see 1 to 2ms of jitter. Can I tweak 3CX a little to compensate for this?

I'm not sure why QoS has helped because it's a dedicated FTTC connection.

 

[JohnS_3CX]

Keep testing because with so much available bandwidth it is not really justified to have packet loss. QoS can work on the LAN side, but also on the WAN side so I don't know which of the two was enabled on your end, and lets not forget you have a lot of available bandwidth in the first place, it might be placebo to think that QoS helped.

You can at least ensure that you have PBX Delivers audio on all extensions, and if internal calls are perfect, then you can potentially eliminate the route between extensions and PBX from being the problem.

With that part done, this would only leave the incoming route from the provider to the PBX (the part that has bits of silence), and this is what the wireshark capture will show you. A check from your provider's side can also confirm. So the only thing remaining between provider+PBX is your CPE and firewall.

 

[Neil Shaw]

Looking at the router we're not seeing any packet loss at all but jitter ranges from 0 to 3ms on most calls.

We're looking at the suggestions you've mentioned.

 

[Lee Cramman]

That's not enough jitter to worry about and shouldn't be causing noticeable quality degradation.

 

[Neil Shaw]

OK great. We've monitored loads of calls and it seems there's the odd blip here and there but for the most part, it's clean and clean. We're going to monitor calls for another 24 hours but as it stands we're nearly there!

 

[Neil Shaw]

Another update. We've recorded a few calls and they're all crystal clear no matter how long they last.

However, users are complaining about robotic sounding voice on the handsets from time to time. So now I guess it's eithre the PoE switch of handsets?

 

[JohnS_3CX]

So if you recording is clean on its way to the handset, then I would also suspect the link between the PBX and the Phone or the phone itself.

What model and firmware are your handsets?

 

[Neil Shaw]

Yeah, that's what we're thinking.

Grand GXP2170 (1.0.9.108 suggested by Grandstream and .127) and 2135's (.102) and one Fanvil (C600 14.0.0.4.R2).

 

[JohnS_3CX]

Ok,firstly I suggest using the 3CX recommended firmware V1.0.9.102 for the GrandStream phones.

Once that's done, log on to the phone's web interface and change the jitter buffer to adaptive:
Accounts > Account 1 > Audio Settings > Jitter Buffer Type > Adaptive

Then save and apply. Keep in mind that a reprovisioning overnight will reset this to Fixed, so best do it in the morning.

Test it for a day or two and see if the complaints about bad audio go away and let us know if this worked for you

 

[Neil Shaw]

Thanks John. We'll get that sorted and report back. Probably be on Monday.

Fingers crossed!

 

[Neil Shaw]

Update. Users are still having issues even with jitter correction configured, but more so on longer calls.

Any ideas? If we have to rip the system out it's going to cost us a small fortune

 

[JohnS_3CX]

Is this before the jitter buffer reverted to Fixed via daily auto-provisioning?

 

[Neil Shaw]

The phones haven't reset themselves to fixed since we made the changes last week?

 

[JohnS_3CX]

Just double-checking in case they did. We noticed some of those phones had choppy audio due to the buffer setting which has changed now in V16 to Adaptive buffer and seems to solve the issue with choppy audio which comes in after about 4 mins or so of talking and then comes and goes throughout a long conversation.

Do you have an alternate spare phone (i.e. Yealink) you can use at the same site to see if we can replicate the issue on other brands too?

 

[Neil Shaw]

Just seen this on another forum..

We've identified an issue in some Grandstream phones that can cause intermittent call quality or audio distortion (at the phone level) on the user's end of the call. The issue is caused by the phone's jitter buffer settings not accurately managing the packets sent from our servers.
If you are experiencing this issue, please follow the steps below:
1 - Login to the phone's web interface via it's IP Address in your web browser. (IP can be found by pressing UP arrow.)
2 - Under Accounts, Account 1, Audio Settings find and change the following:

• Jitter Buffer Type: Fixed
• Jitter Buffer Length: 400ms

3 - At the bottom, choose Save and Apply.