Solved Increment SIP ports with STUN Provisioning

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JoeB

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Please clarify for me. I've read in several places that the phones provisioned using STUN should be incremented.
Yet the admin manual recommends leaving the default:

1595944071328.png

We're a small operation but support some larger ones. Our system runs fine as it is, but if best practice is to increment the ports, then
I'll change all of them. What is the upper limit? I saw some forum posts where older versions had a parameter for this, but don't see one in V16.

Any assistance is appreciated.

3CX Version - Enterprise Annual / 16.0.619
Server OS - Hosted Linux
Is the 3CX Server Hosted and where? - Yes. Cloudco
IP Phone Make/Model/Firmware - Yealink T46S - 66.85.0.5
Provisioning Method: Local / VPN / STUN / SBC - STUN
Trunk Provider or Gateway Make/Model - CloudCo
Has the Firewall Checker passed: YES / NO - Yes
Are custom Phone Templates being used: YES / NO - No
 
You will only need to increment this IF you have MULTIPLE devices at the SAME location (Remote Office)

We have more information on this in the 3CX Academy (Intermediate - Remote Extensions)

1595944977831.png
 
Yes, we have 23 phones. All are remote as it's a hosted account.
 
For this amount of phones, if at the same location I would install a sbc server

using stun you have to

1. Give each phone a fixed ip address (can be via dhcp reservation)
2. Give each phone a different sip Port, different range of rtp ports
3. Setup port forwarding on the router protecting the phone , so that the correct sip, rtp ports are going to the correct ip address of the phone
 
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The screenshot I posted from the manual is likely from the point of view of one remote phone,
not an entire installation of them as we have.
Thank you for the replies.
This thread can be closed.
 
A router that doesn't account for multiple endpoints using the same local ports will break more than VOIP.

Most correctly configured routers use dynamic NAT/PAT. An internal IP/port will be mapped to an often random external IP/port. This mapping will be retained in the router's state table until it expires. Thus it doesn't matter whether a phone uses the same local ports or not.

The PBX itself needs to be configured to use static 1 to 1 NAT without PAT, inbound and outbound. For RTP, a phone will punch a hole through its own NAT and send media to the PBX on the IP/port the PBX announced in SDP. The PBX will then send media back to the IP/port it received the media from, ignoring the phone's SDP. The same applies for SIP signaling, but in this case the Via/Contact headers from the phone are ignored.
 
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