History, I have a SIP hosting provider that I use for my FreePBX setup. Upon original setup I was using an Asus with Merilin software and doing port forwarding and all was well. I decided to go Pfsense and get rid of Asus.
Fast forward I can’t get the audio (RTP) to work with my hosting provider on PFsense. The only way I can get it to work is to open up the source ip for RTP to the entire internet on my WAN Rule and Nat Rule. But when I switch the source back to my SIP provider audio breaks again.
I have done a pftop and provided to my SIP hosting provider to show that its connecting to other IPs tor it to work and was told no one else is having this problem.
So leaves me with 2 questions:
- Anyone know a way around this or experienced something similar?
- Is my SIP secure having the SIP (5060) only talk to my provider which is working and let RTP just be open?
Rules are as followed:
Working: (192.168.x.x = FreePBX)
Port Forward: Interface (WAN) -> protocol (udp) -> source address () -> source ports () _> Dest Address (WAN address) -> Destination ports (10000:20000) -> Nat IP (192.168.x.x) -> NAT Ports (10000:20000) descritpion Full RTP Test
WAN Rule: Protocol (IPv4 UDP) -> Source () -> port () destination (192.168.x.x) port (10000:20000) -> Que (none) -> description (NAT Full RtP Test)
Freepbx cross post : https://community.freepbx.org/t/not-sure-what-happened/61989/19
I am anal about having something broken and this is making me lose 5 minutes of sleep a day lol, any help again greatly appreciated.