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

[Martijn van Oosterhout]

On Thu, Apr 07, 2005 at 03:57:11PM -0400, William M. Sandiford wrote:
> 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

Possible, yes. Whether it works depends on the firewall. Your problem
is that UA2 is sending directly to the firewall and the firewall will
block it because it knows nothing about UA2. Or not, if it supports
partial matching on UDP ports. In theory a packet of UA1 to UA2 should
open the back channel, except you run the risk of the firewall
assigning a new port number, thus breaking everything.

This is a problem uPNP was supposed to solve, the client can request an
externally visible port on the router. Never seen any client that does
this though.

If you only have one UA you can get around it with port forwarding on
the firewall... But you need to know in advance what ports SIP is going
to use...

> 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)

Good luck!

-- 
Martijn van Oosterhout
Ecomtel Pty Ltd