QUESTION VPN, NAT, re-invite - Oh my!

[GomezAddams]

I'm trying to get a SIP softphone working on my iPhone using OpenVPN to connect to my PiaF box. I have the OpenVPN stuff working, as near as I can tell - I can RDP into my windows boxes, ssh into my pbx, do web pages, etc.

However, I can't get a sip softphone working (I've tried several). I have the classic symptom that the call goes through, but no audio in either direction.

From what I understand, by default asterisk will tell the end devices (Vitelity and my sip phone in this case) to bypass asterisk and do RTP directly to each other. To stop this behavior, and keep asterisk in the middle as a proxy, my understanding is that in the sip extension you set "nat" to yes, and "can reinvite" to no.

Unfortunately, this doesn't seem to work. Even with nat = yes, it looks like asterisk is sending the IP address of the softphone (as assigned by OpenVPN) in the packets to Vitelity. That is assuming I'm reading the log file correctly after turning on sip debugging.

What am I missing here?

 

[jroper]

Hi

I believe that canrenvite is not the right thing to use anymore, I've attached Asterisk's V11 sip.conf, which should shed a little light on the matter. I believe "directmedia" is the setting to play with now.

The relevant information is at line 792, regarding NAT and Media.

Joe

 

[GomezAddams]

Thanks jroper! Looking over the configuration section for sip & nat was a big clue stick. I've configured OpenVPN to give clients an IP address outside my normal internal network. I need to add the OpenVPN client range to the internal networks list.

 

[GomezAddams]

Boom baby! That was indeed the issue.