[Asterisk-Users] SIP and NAT: try, try again.

[Olle E. Johansson]

...and to solve another problem, there's my suggestion on support for outbound SIP proxy.
http://bugs.digium.com/bug_view_page.php?bug_id=0000359

There are corporate networks that use a "SIP proxy proxy" as an ALG, application layer gateway,
for all outbound and inbound SIP traffic in the DMZ. This should work in conjunction with
netmask/STUN -
   if host does not belong to my network
	send SIP transaction to outbound proxy
   else
	send SIP transaction to host
   done

This cleverness may cause problems with inside networks consisting of several networks with
different netmasks and complicated routing...

I believe outbound proxy should be configured on a host by host basis for sip clients/peers
as well as an "default" outbound proxy to use in other situations.

In order to support SIP URL dialling, we have to use a netmask/STUN solution to sort out if
the SIP proxy we're trying to reach is ourself, someone on the inside or someone on the outside
of our NAT.

/O