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I'm struggling with a SIP installation with a remote user. THe phone boots up, provisions itself from the (remote) TFTP server, performs SIP registration with the PBX, and I can make calls to the phone, it rings, they can hear me, but silence coming back.

Normally at this point I would suspect a NAT problem, but both ends of the link have Draytek Vigor routers and we've set up a site-to-site VPN between them to eliminate any NAT issues. The PBX 'sees' the device with its correct IP address, and in the PBX's dashboard it shows up as registered with no NAT.

so we have:

PBX (192.168.162.11) <--> Draytek <--> VPN <--> Draytek <--> Phone (192.168.250.9)

I would have thought that was a simple routed network. But no, apparently. I'm tearing my hair out trying to get the audio path to work correctly. As far as I can see, no NAT should mean that the traffic just flows directly from the phone to the PBX (over the VPN). I know SIP is problematic with NAT, but that shouldn't be an issue here, should it? Why isn't this working!!!??? Any thoughts would be appreciated.

The PBX is Trixbox (asterisk). The phone is Cisco 7960.

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Hey I have had EXACTLY the same problem as this, however I never really solved it as such. I got a new connection (cable vs dsl) and a new modem/router and it seemed to work fine, although shortly after we removed the need for VPN and authenticate directly so we didn't spend too much time troubleshooting. I can find out more from the SIP guy however I recall it was due to outbound NAT not working properly as it picks a random port and the client side doesn't 'see' it for some reason. Also now I remember, actually reading your post again we might have opposite problem, in that whenever people –  Joshua D'Alton Mar 16 '11 at 14:38
    
What kind of VPN (IPSec, SSL, PPTP, L2TP, etc.)? Are there firewall rules involved? Does the VPN ruleset equate to "tunnel EVERYTHING between sites"? If you start the call from the other side is the one way audio reversed? –  Jed Daniels Mar 16 '11 at 15:32
2  
I strongly suggest running a wireshark capture at different points along the network. Depending on the phone, some of them have a 2nd network port that can mirror the first, so you remotely capture the exact traffic flow, or some have TCP capture built in and you can download the capture. Wireshark has brilliant SIP/RTP debugging tools to find out where the problem is –  Mark Henderson Jun 8 '11 at 4:03
    
there could be an ALG interfering as well, attempting to hop into the protocol to allow it to work through NAT, I agree with Mark, take a packet cap. –  SpacemanSpiff Jun 8 '11 at 4:25
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