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how 3cx audio flow

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davidlee

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hi , i need seek for help. i can't find the answer on 3cx admin and user manual .

appreciate you can answer me below question :

how the 3cx audio flow between the 2 extension on below case :

1. how the audio flow between 2 remote extension on same remote location , one using sbc and one is direct stun

is peer to peer or peer to sbc to peer ?

2. how is audio recording was turn on on either one extension . is the audio flow peer to 3cx then peer .. please advise how can i check this on 3cx management or other tools that i can use to check this ? like wireshark ?
 
Hi there,

The call flow between remote extensions using STUN is handled by the PBX, even if the other extension is through SBC.
If you had two remote extensions using SBC then after the connection is established it would be a peer to peer flow (not if you have PBX delivers audio).

If you turn on call recording and PBX delivers audio then audio is always handled by the PBX.

Wireshark is a great tool to use to see this and troubleshoot any issues.
 
Thank you Yiannish , just to confirm you mean as long as one of the extension using stun either locally or remotely. the audio flow is handle by PBX not peer to peer between the 2 extension right ?
 
That is correct, when one of the two extensions in a call is connected through STUN then the call flow is handled by the PBX.

Reason is that when remote extensions register on the PBX through STUN they use the external IP of the remote site and a port number assigned to that extension (also dedicated RTP ports). The internal IP of the remote extension is not known so the PBX is needed to handle the call flow.

SBC on the other hand is different, if two remote extensions are on the same remote site and are connected through the same SBC then the SBC can act as a server and connect the two extensions directly.

Hope this makes sense.
 
yes. make sense. thank for the information . since remote extension i uisng stun is using local RTP port. so i can assume is actually no need to open port from main PBX server for port 9000-9255 too right in order for remote stun working property ?
 
In order to configure remote extensions via STUN you will need the following ports open on the PBX side

1) 5060 TCP/UDP
2) 5090 TCP/UDP
3) http and https provisioning ports (5000/5001 if you use Abyss or 80/443 if you use IIS)
4) External RTP port range 9000-9499 (WebRTC ports included in this range) as the remote phones will speak to the PBX on the external RTP port range

On the remote Site Firewall you will need to port forward the following:

SIP port and RTP port range specified in the Extension Settings>> Provisioning tab so by default for the first remote extension you will set in the provisioning tab SIP port 5065 and RTP port range 14000-14009 for example
The next remote phone with STUN set the SIP port 5066 and RTP port range 14010-14019 ...etc
And on the remote firewall Port forward port 5065 and RTP 14000-14009 to IP address of Phone A and port forward 5066 and RTP range 14010-14019 to IP Address of phone B, and repeat all 3 steps for each remote phone with STUN that you add.
 
hi , sorry i a bit confuse here. i thought if remote extension using stun no need open port on remove location. i try to provisioning using stun to yealink ip phone and it work. and the remote location is dynamic ip address and not a public ip address too. tcp/utp port 5090 only need to 3cx for windows/ios and android client right and SCB controller but endpoint like ip phone no need becouse not support 3cx provide stunk (5090) . please advise ?
 
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