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Puzzling issues with call quality on 3CX STUN

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Kobo Williams

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We've deployed twenty two 3CX phone systems for our clients, all using hardware phones deployed at the client location using STUN with the server installed in a data center. Two of these are now having an issue with call quality that we're having trouble diagnosing.

Client A has one phone system with 30 phones deployed over three different physical locations using extension ranges as differentiators. Phones at extension ranges 100-199 and 300-399 have no issue but the phones in the range of 200-299 have the issue. The issue is, at some point during the call, people at that location can only hear every other or every few other words the caller is saying. PBX delivers Audio is set in the SIP Trunk settings, Supports Invite and Supports Replaces are not checked. In the Extension Settings, the working Extensions have all three of those options checked, while the extensions with the calls cutting out have only PBX Delivers Audio checked.

Client B has the same cutting out issue, but to all of their phones. PBX delivers Audio is the only thing checked on the SIP Trunk Options. PBX Delivers Audio is the only thing checked in all of the extension settings. We do have support contracts on all of our 3CX systems. We went through creating call dumps from 1. one of the Yealink T29G phones of the client, 2. the phone system itself, and the SupportInfo documents. After sending them in to 3CX their diagnosis is there was no audio drop in the capture on the PBX from the provider to the PBX or the phone to the PBX. The audio drops occurred on the phone coming from the PBX.

This would normally lead me to believe maybe the problem is with their internet provider. I have contacted them (Comcast), and they provide the internet to both of these clients all of their office locations. IN the case of Client A, they are providing the same packaged service at all locations. In the case of Client B, they have not noticed any issue at the location. A major issue of note with Client B, one of the phones is located at the home of one of the users, not the office location with the rest of the phones and it is having the same breakup issues. Another of the extensions is using the mobile 3CX client and not using a physical phone at all and is having the same issue.

So that's where I'm at now, wondering if the issue may be either a setting in the phone system that is causing the issue. Anyone have an insight that may help me? Much appreciated.

P.S. All 22 of our phone systems are in the same data center, behind a single firewall with all the same ports forwarded for all of the external IPs being used by all the phone systems.
 
Audio issues are almost always network issues. There is no PBX setting or phone setting that can make calls cut out. If it was a firewall or port problem it would either work or not work.
 
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Also, you really should be using either a SBC or VPN for these setups. STUN is a single phone solution and really shouldn't be used beyond that. SBC would help slightly in this cause as it uses a little less bandwidth than all the phones configured as STUN but you still need to identify and resolve the network issue at play here.
 
Thanks, cobaltit! If I'm understanding this correctly, you mean to say, if it were a port or setting issue, it would either work or not, but it wouldn't cut in/out. My concern, especially with Client B, is that the issue was happening with the mobile client, at the office location and at one user's home location. It's possible it's a bandwidth issue at those locations, I just thought that improbable.
 
Correct. The RTP streams don't change ports mid call so if it was something like port forwarding it would either work or not work. Audio degradation is most commonly packet delay or packet loss which can be caused by either insufficient bandwidth or poor network conditions. It can also be caused by not enough CPU/memory on the PBX if you are not using the same codec all the way through but that's less common.
 
I agree with @cobaltit deploying an SBC would be the best solution - I also only ever deploy STUN for single endpoint solutions purely down the the network reliance required for mutli-STUN setup.

The only instance I personally have ever known in regards to call cut-off mid call was related to SIP ALG being enabled on a Firewall (some firewalls have the option in the web GUI but others require this to be turned off via CLI) and calls were cutting off after 30 seconds - but this is a common issue with this setting.
 
It's got nothing to do with using STUN because as you have already identified, it happens on the mobile client and desk phones in multiple locations with different networks and different providers.

My guess as this point is codec transcoding or network activity:
1) what codecs do you have enabled and in what order? (we have found that disabling all other codecs other than g711a/PCMA and g77u/PCMU dramatically improved audio issues because it stopped the PBX from transcoding which wasn't working for us. Disable codecs in the SIP Trunk and the extensions.)
2) Do you have the phones on VLANS and have you prioritised the VLAN? (we have found that if the voice traffic was not prioritised and was just lumped in with all the other traffic, if another user on the network was downloading a program or doing some updates, it would cause terrible jitter).
3) do you have any SIP ALG on your routers? (3CX recommends that this be disabled as it causes issues)
4) do you have QoS enabled in your router? (We have found that this works sometimes but doesn't other times, depending on the router and switches. Generally, QoS is a really expensive layer 3 feature that will only work well on very expensive switches and routers, thus we use VLAN prioritisation instead as it's supported by many more network devices and requires less processing as it's only layer 2)
 
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