Media RTP FLOW

[Leonardo Santana]

Hi,

If my 3CX PBX is on cloud and my phones are at site A, the media flow between the phones (INTERNAL CALLS) at site will remain at site A?

 

[NickD_3CX]

Hey!

It depends how your phones have been configured.
If you have a 3CX SBC at the remote site and your deskphones are provisioned with the SBC (assuming you have recordings disabled), then yes, the audio will stay on Site A.
If you configure your phones are Remote STUN, then no, audio will travel all the way to the server and the back again.

 

[rc179]

Nick, I'd like a little clarification on this as well.

The solution is a 3CX in the cloud with a public IP, the extensions are on the local LAN with private IPs and a Firewall providing NAT/PAT. There will be nothing else, don't assume another piece of equipment is installed.

Assuming there are extension 101-121.
It's my understanding that the audio of a call between the outside world and an extension WILL go through the 3CX.
What path does the sudio take when the call is between ext101 and ext111, with recording off?
What path does the sudio take when the call is between ext101 and ext111, with recording on?

If this is configuration dependent please explain, but please do not say "If you have X additional equipment" because that is outside the scope of the solution.

Thanks

 

[NickD_3CX]

Nick, I'd like a little clarification on this as well.

The solution is a 3CX in the cloud with a public IP, the extensions are on the local LAN with private IPs and a Firewall providing NAT/PAT. There will be nothing else, don't assume another piece of equipment is installed.

Assuming there are extension 101-121.
It's my understanding that the audio of a call between the outside world and an extension WILL go through the 3CX.
What path does the sudio take when the call is between ext101 and ext111, with recording off?
What path does the sudio take when the call is between ext101 and ext111, with recording on?

If this is configuration dependent please explain, but please do not say "If you have X additional equipment" because that is outside the scope of the solution.

Thanks

In this case, this means that you phone will be configured as Direct SIP / STUN.
This in turn means that audio will always go via 3CX.

This is the expected way, and to ensure this it is also sometimes recommend to enable "PBX Delivers audio" in each extension, although technically speaking it is not required.

 

[simply7]

I can confirm what NickD is saying, I have a site with around 10 phones that are direct SIP/STUN phones (Yealink T46S) and the PBX delivers audio had to be enabled or I would not have audio on internal calls. Use a SBC and you will not have that issue.

 

[rc179]

I ran my own tests with packet captures to confirm the IPs and audio path. I found what simply7 has said, that the ext to ext audio (direct sip/stun, cloud pbx) doesn't work unless the pbx delivers audio is checked. Otherwise the phones are attempting to connect (based on the sip signaling from the pbx) to the public ip on the firewall (alg is disabled). So contrary to what 3CX says, this is required.

I'm OK with the requirement, but I'm glad I'm doing my own testing and creating my own install procedure, before rolling this out to my customers.