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How to get best voice quality?

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Frank86

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Running 3CX Enterprise (latest version) on premises (About 40 users on a flat network, with no dedicated voice VLAN) using Flowroute as VoIP provider on a 100 Mbps fiber circuit.

Call quality is mostly good, with the occasional days when it's choppy and staticky when talking to outside landlines and cell phones. 3CX voice codecs are the default ones.

Ownership wants to raise voice quality to what we had before (on-prem Mitel + T1 PRI), which was never choppy or staticky.

What is the best way to achieve this result?
  1. Segregate 3CX server and phones to their own VLAN, still using Flowroute on the fiber data circuit
  2. Option 1 + Flowroute on a separate T1 line
  3. Option 1 + T1 PRI instead of Flowroute
  4. Other?
Thank you.
 
As it appears to be only affecting outside calls, try putting QOS on the router for voip traffic
 
Sorry, forgot to mention that. We have already implemented QoS for VoIP on the Meraki firewall.
 
Step 1 is identify the source of the issue. Otherwise you are just shooting in the dark. Option 3 would definitely work but would likely cost more.

1. We do VLANs for VoIP but it's rarely a QoS concern. It's typically more of a security (PCI, etc) security or for management purposes. It's pretty rare that you will have something on the LAN generate enough traffic to cause call quality issues. And unless you have a consistent data event that corresponds with your bad call quality days this is probably not going to do anything other than rule out a possibility. If already have the equipment to do so and you are comfortable with networking then by all means do it but I wouldn't expect it to fix the problem.
2. This would only fix the problem if the cause saturation of the fiber link, a configuration issue with the CPE or just poor routing between your fiber and the Flowroute POP. If your Meraki is causing the issue and you run the T-1 through the Meraki then you'd have the same issue. If you got the T-1 from the same ISP and it's a routing issue then you're still going to have a problem.
3. Assuming the same provider/setup as what was feeding the Mitel then you should have the same call quality as you did before. The only recommendation would be assuming you had a vPRI and not a true copper PRI then just get a SIP hand-off from that provider instead. You should still get the same quality but you wouldn't need to get a PRI gateway. Downside of this is you are tied to your premise system. Using Flowroute (or any other BYOB SIP trunking provider) gives you the flexibility of moving your 3CX instance to the cloud on short notice for business or disaster recovery reasons.
4. Get a backup/test trunk from another provider and see if you have call quality issues with them. One of the possible culprits could be the routing from your fiber to the Flowroute POP or it could be their carrier handoff to the PSTN. We could help in this respect.

You should engage Flowroute to see what they say. We have the ability to go back 48 hours on request for call flows and you can see/hear which side the call quality issue is coming from and I assume they can as well. You can also run something like PingPlotter to see if your network is degrading when the call quality issues are being reported. Flowroute may be able to make adjustments based on NPA-NXX as I know years ago when I used Vitelity we would report call quality issues to certain numbers and they would make some changes to alleviate. Flowroute may also be able to home you to another POP to see if that changes your experience.
 
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QoS needs to be implemented end to end and on both layer 2 and 3. You only really have the power to control this on the local site so need to work with the ITSP to get the best out of your system.

On the local side there are different techniques to achieve this also.

You can use VLAN's for segregating Voice and Data traffic as mentioned but also on 3CX you can provision the phones with both a VLAN ID and priority setting:
https://www.3cx.com/sip-phones/vlan-configuration/

You can daisy chain phones (use the switch port on the phones to attach a PC) to save on wall ports (in the above you will see that you can VLAN ID the WAN and PC ports) however if I can avoid it I wont daisy chain phones. If you do then the attached port on the switch will need to be a trunk port.

On the router (layer 3) I would normally use the diffserv setting: https://en.wikipedia.org/wiki/Differentiated_services which will provide priority for voice traffic if set with a DSCP code (normally 46 for expedited forwarding).

Codecs are another thing you can look at. Unfortunately (as per the below string) there maybe other elements within the full end to end call network which can dictate which codec is used: https://www.3cx.com/community/threads/poor-sound-quality.62788/#post-264310

For example commonly G.729 compressed voice is used across WAN links which offers less bandwidth utilization than and G.711 or an HD codec but a lower MOS score (call quality score).

As @cobaltit involve your provider. Hopefully they have/had offered you some trunks to test/qualify the deployment before go-live. I don't like to recommend jumping straight in with a provider without testing their trunks first.
 
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