Bad audio quality everywhere

[Dave Smith]

Hi all,

I am quite new to 3CX and just had it running this week, we are an IT company looking to for a good phone system to resell as part of our offerings and 3CX is looking promising apart from some audio issues we cant seem to get around at the moment.

I have read some similar posts with similar setups but they dont really say if they ever fixed it.

We have the 3CX phone system installed on a Windows 2012R2 Server running on a Hyper-v cluster (4 cores and 4GB RAM for about 5-6 users) hosted in a data center, all our staff work remotely and while we are testing they are all using the 3CX Windows and mobile client.

I have tried various codec options as suggested in other posts and had the 'PBX Delivers audio' option on and off but it seems to make little difference. We are coming from FreePBX and the audio quality was fine on that, every call was crystal clear with no issues and it was also running on the same hyper-v cluster.

At the moment the client side codec's are set to -


GSM-FR
G729
G.711 U-LAW
G.711 A-LAW
G722
PBX delivers audio is on

SIP trunk codecs -

G.711 A-LAW
G.711 U-LAW
GSM-FR
G722
PBX delivers audio is on

This seems to be the best / most stable mix at the moment but the call quality isnt great, just usable. When calling between extensions id expect calls to be excellent but they are quite muffled.

Has anyone come across this before and managed to fix the issue? or have any suggestions?

Thanks,
Dave

 

[eddv123]

PBX delivers audio should only really be used if you are using remote extensions using the STUN method or if you are recording calls on the extension(s) in question. I have sometimes used in if I have 2 LAN's on my 3CX or using across a VPN.

If you are using STUN from your data center I would highly recommend you change to the 3CX SBC and see if it improves things: https://www.3cx.com/docs/3cx-tunnel-session-border-controller/

Another reason STUN is not too good is because of bandwidth, each phone passes its audio from its remote site back to the PBX, if you use the SBC then audio and signalling is kept local to the endpoints. If you consider Bandwidth and latency if you have a lot of phone then it may explain the issue you are having.

You can calculate using this guide: https://www.3cx.com/blog/docs/bandwidth-utilised-for-voip/

 

[Dave Smith]

Thanks Ed, we are all using the 3CX Windows client and the extensions are set to 'use 3CX Tunnel for remote connections' so i assumed this was already happening.

We have setup our firewall with a 1:1 NAT straight to the 3CX server and opened all the listed ports, i dont think bandwidth would be an issue as its on 100Mb line although we are going to set aside some bandwidth just to make sure.

Something i didnt notice before on the Hyper-v install page was - Refrain from using Generation 2 VMs on Hyper-V, only Generation 1 will work. is that still the case? as this is a Gen 2 machine

 

[eddv123]

For Hyper-V perhaps review this:
https://www.3cx.com/docs/installing-microsoft-hyper-v/

The only experience I have of installing on Virtual machines is that I find it much better to bridge the network interface rather than using it NAT'd.

 

[Dave Smith]

I've done some more testing and built a 2012R2 Gen1 virtual server, the call quality is slightly better and id say more consistent than before but still not HD even just between extensions.

Again tried different codec options, 3CX tunnel on and off, PBX delivers audio on and off but the changes in call quality are still very slight and most of the time worse, there never seems to be a drastic jump either way.

I tried following this https://www.3cx.com/docs/troubleshooting-remote-extensions/ but the settings are different in the latest version of the windows client, you cant seem to add accounts manually and even if you change the existing one the settings it refers to for STUN under advanced arent there.

 

[eddv123]

Are you using purely 3CX Client or are you using Desk phones as well ?

It would be useful (as a test) to see whether the issue is still apparent on a hard phone (like a Yealink for example) let me know if you have a phone you can test with.

 

[giwm]

This may not be terribly helpful, but at least a point of reference. We have 3CX running on a 2012 R2 VM set up as Gen2 without issue. This install supports 19 total locations (18 remote) with connections as slow as 7 Mbps DSL. Granted, if that DSL connection is streaming video, the call quality drops noticeably, despite QoS. We do add a bit of overhead as all locations are VPN connected. The install has 4 cores and is utilizing 3.1 GB RAM (out of 8 available).

On the other hand, we're using Yealink phones versus software clients. This does allow QoS throughout as we can leverage voice VLAN segregations.

 

[Dave Smith]

So far all the testing has been done using the 3CX client for Windows and with the iOS App

I have a yealink DECT handset here configured with my ext. calling from the yealink phone to a 3cx Windows client sounds the same though.

Thanks giwm for your input, are the phones setup to go over the VPN between the sites? rather than setup as remote handsets?

We are going to order a couple of yealink phones next week, we have been holding off a bit but today the audio seems better.

What model Yealinks do you use? I was looking at the T42S's

Although we all work remotely, 2 of our biggest customers are waiting for us to sell them a phone system but pretty much all of their users will be local to the phone system which is something we havent tested much of yet. If we can get the same sort of quality we have now for their remote users I think we'll be fine.

 

[Robertos]

hi Yealink T42S that you mentioned we have used and call quality is nice especially G.722

Just a note about your problem, you may be actually negotiating a narrowband codec in your scenario. Please check this.
https://www.3cx.com/community/threads/codec-order.48537/

Also, does this "muffled audio" issue occur on all calls or at certain times of day? Do all callers experience this? Most network related issues are usually one-way (usually the same direction from server-to-client download traffic flow but there are exceptions of course). This would manifest itself as remote site cannot hear HQ but HQ can hear remote site fine).

Still looking like a process of elimination type thing here.

 

[Dave Smith]

Thanks Robertos, ill take a look at that post.

Ill be ordering a couple of T42S's today. Yeah you're right its just process of elimination at the moment trying to find the cause is a little tricky. I was sort of hoping id just missed a setting somewhere

All our staff work remotely and they all have fibre pretty much to themselves, i think the slowest connection is 35Mbps down and 8Mbps up, the phone system is in a data center with 100Mbps connection with 10Mbps just for the phone system just in case things get busy.

The issue doesnt have a pattern, its happening on every call, changing the codecs, pbx audio just doesnt seem to make much difference at all. Going to the Gen 1 hyper-v server seem to clear things up a bit and make it more stable and clear but still not HD.

Things seem slightly different when using the mobile clients, from my android phone its louder and clearer.

We'll see what the handsets are like when they come.

 

[DSXVOICE]

There is a chance that your issue is being caused by the clock source on the VM. Be certain you have the Time Synchronization Service installed and running. This is almost guaranteed to be an issue if the VM was a P2V incarnation.

Kirk

 

[giwm]

Wow... 35/8? You've got better speeds than all but a few of our locations! What kind of latency do you have though? That could play a huge role. 10 Mbps for the phone system is way more than you should ever need with that phone count, so at least you can rule that out.

Yeah, we run everything over the VPN. We have T40P, T42G, T42S, T46G, and T48G. All perform quite well and Yealink support is excellent.

 

[Dave Smith]

Yeah i think 35/8 is the slowest, latency is pretty good on all of them, they range from 8ms - 19ms at the most. I dont think bandwidth is an issue for us, we have run FreePBX with Polycom IP Phones for years in exactly the same way and its always been crystal clear HD audio.

All the VM's for running 3CX have been newly built and updated just for running our 3CX server.

The phones are on order and will be here tomorrow so ill see how that goes, otherwise ill log a support call with 3CX i think.

 

[Dave Smith]

The phones arrived today and the call quality on them is much better than the Windows client.

From the desk phone to the Windows client seems to be clearer as well but the volume from the windows client side listening to the desk phone side is quite quiet, however in the other direction its loud and clear.

Getting more phone sent out to other staff for another bit of testing but otherwise all looks good.