[Asterisk-Users] SIP UA behind NAT and REINVITE ???

[William M. Sandiford]

Hello:
 
I've read through the list archives and found tonnes of threads on this topic but there has been no definitive answer, so hopefully someone can give me one.
 
Can a proper 2-way audio call be established when the UA is behind a NAT firewall and REINVITE is enabled?
 
Original Call Made
SIP UA 1<--> NAT FIREWALL <---> Asterisk <--> SIP UA 2
 
Then REINVITE occurs and
SIP UA 1<--> NAT FIREWALL <------------------------> SIP UA 2
 
Is this possible?
 
Will using a STUN server help this at all?
 
I have tried and tried and tried to get this working but with no luck (well, I can get it to work with canreinvite=no, but thats not what I want.  I want * out of the audio path)
 
I have even tried putting the private IP of SIP UA 1 in the DMZ of the NAT Firewall and still no luck.
 
Any Suggestions???
 
Bill

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