Call quality issues

[eddv123]

Codec selection is commonly done also for combat of bandwidth related issues, for example you can compress a codec (like G729) so that you use less bandwidth, this is why it is used quite a bit across WAN links. Be aware that compressed codecs however can have an effect if you use Fax or DTMF with the classic inband method.

For latency issues I would check your network for (as you have already identified) bandwidth related issues as well as bottlenecks in the network:

This helpful 3CX bandwidth calculator will help you work out how much you are using:
https://www.3cx.com/blog/docs/bandwidth-utilised-for-voip/

 

[Peter Richardson]

Definitely not a bandwidth issue - our city is all on 100mbit fibre. And we're talking tiny clients - only 4 simultaneous calls...usually only 1!

 

[Peter Richardson]

So would it reduce latency by having both legs using the same codec? I guess that would mean that no transcoding is needed, hence less lag, right? (we've been experiencing horrible latency since the beginning, and I'm hoping that this will solve it. Having disabled all codecs except G722 on our own PBX, this seems to have improved the latency greatly).

So what exactly does "PBX delivers audio" do exactly? From what I can gather from my own testing and by reading another forum called "What does pbx delivers audio exactly do", it seems that this option forces the PBX to deliver the audio, whether you've got an SBC or not, whether the PBX and the extensions are on the same LAN or not, and it also enables encoding, is that right? I just don't fully understand why there's 2 places to enable it (trunk and extension) and why everyone recommends that it's always disabled in the extension and enabled in the trunk. This is not a feature/setting in Asterisk so it's obviously a 3CX implementation, but there is no documentation on it, despite it being a problem that is brought up regularly in the forums.

I need some really clear information about this, perhaps even a couple of examples.

 

[leejor]

So what exactly does "PBX delivers audio" do exactly? From what I can gather from my own testing and by reading another forum called "What does pbx delivers audio exactly do", it seems that this option forces the PBX to deliver the audio, whether you've got an SBC or not, whether the PBX and the extensions are on the same LAN or not, and it also enables encoding, is that right?

In a nutshell, it prevents two extensions from communicating directly if that would otherwise be possible.

I just don't fully understand why there's 2 places to enable it (trunk and extension) and why everyone recommends that it's always disabled in the extension and enabled in the trunk.

It will always be enabled on a trunk call (for reasons that were mentioned previously), and should only be enabled on an extension if there are audio issues that require it be used. It puts an additional strain on the PBX, that in most cases is not necessary.

there is no documentation on it, despite it being a problem that is brought up regularly in the forums.

It is actually there to solve a problem, usually involving a remote set, or sets, using STUN. That why it is disabled by default on the extensions.

Keep in mind that when using low bit rate Codecs, depending on the set, coding decoding requires more processing power than the hight rate Codecs. I'm not sure if this is a reason for your latency, but... maybe.
I would recommend sticking with G711a/u as your default Codecs, especially over trunks, unless there is some pressing reason not to, like bandwidth constraints.