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- Oct 1, 2017
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After completing all three 3CX certifications and doing some research in the forum, it seems like a lot of people get confused (including me) when it's time to configure a router in a STUN + Cloud Hosted PBX environnement
My Setup:
A couple of phones on the LAN + a 3CX server hosted at OVH
My Firewall: PFSense
Incoming ports that I opened: none
I provisioned a few phones with Direct SIP (STUN remote), left all port default (SIP 5065 and RTP 14000-14019) and all seems to be fine.
The only option I had to enable on extenions is "PBX Deliver Audio", because phones have no clue how to re-invite each other without a SBC.
Why would I need to forward different incoming SIP ports for all my phones when the phone initiate a SIP connection toward 3CX and source ports a rewritten with a random port by the router ?
The PBX knows how to reach the phones via the public IP and the random port:
Example: Extension 103 is registered, contact: sip:[email protected]:20106
The router takes care of the rest in it's NAT table (forwarding external incoming port 20106 traffic to internal IP of the phone on right port)
So again, my question is: why would I need to forward any ports on my firewall ?
Another setting that I usually change in this kind of setup (on another evil PBX kind) is the SIP registration Timeout. It's usually set at 3600 and I lower it to 60 seconds so the registration doesn't get killed by the TCP/UCP timeout in the router. On 3CX, it seems to be set the 120 seconds, which is probably fine for remote extensions.
My Setup:
A couple of phones on the LAN + a 3CX server hosted at OVH
My Firewall: PFSense
Incoming ports that I opened: none
I provisioned a few phones with Direct SIP (STUN remote), left all port default (SIP 5065 and RTP 14000-14019) and all seems to be fine.
The only option I had to enable on extenions is "PBX Deliver Audio", because phones have no clue how to re-invite each other without a SBC.
Why would I need to forward different incoming SIP ports for all my phones when the phone initiate a SIP connection toward 3CX and source ports a rewritten with a random port by the router ?
The PBX knows how to reach the phones via the public IP and the random port:
Example: Extension 103 is registered, contact: sip:[email protected]:20106
The router takes care of the rest in it's NAT table (forwarding external incoming port 20106 traffic to internal IP of the phone on right port)
So again, my question is: why would I need to forward any ports on my firewall ?
Another setting that I usually change in this kind of setup (on another evil PBX kind) is the SIP registration Timeout. It's usually set at 3600 and I lower it to 60 seconds so the registration doesn't get killed by the TCP/UCP timeout in the router. On 3CX, it seems to be set the 120 seconds, which is probably fine for remote extensions.