FusionPBX within VPN and NAT setting for RTP audio

[davemc]

FusionPBX VM on local LAN with subnet, eg 10.1.1.0/24

Phones connected on LAN, and also phones connected over VPN from other sites, or from different subnets via local routers, eg 10.0.0.0/8

There is no NAT within the VPN and local routed networks, it's direct connect.

When SDP invite goes out, because the endpoint is not within the local subnet, it invites RTP on the external IP, which is not what I want.

I've figured this might be SIP Profile::internal::apply-nat-acl , which is default nat.auto. I can see in the log fs making wrong decision, doing an apply-nat on a vpn subnet address.

What is the best/recommended way to resolve this? I've read lots around the issue, but not found a concise and definitive receipe or example.

Another method, display ext-rtp-ip and ext-sip-ip.

 

[DigitalDaz]

I don't have any solution to this but I think its going to lie maybe within the LAN ACL, you somehow need to make freeswitch aware of what are to be considered LAN.

 

[EasyBB]

I am not sure this would be of any help: https://www.pbxforums.com/threads/server-and-multiple-local-subnets-behind-same-nat-firewall.462/

 

[davemc]

We ended up setting Advanced... SIP Profile... Internal

name: ext-rtp-ip   value: $${internal_rtp_ip}  enabled: true     note: leave off or rtp to external ip  default   $${external_rtp_ip}
 
name: ext-sip-ip   value: $${internal_sip_ip}  enabled: true    note: default $${external_sip_ip}