asterisk users - Nov 2003 - SIP and NAT: try, try again.

In response to the SIP and NAT discussion, I have updated the ticket 
on the subject that seemed to be getting the most attention: #104. 
There are enough clueful people here that perhaps someone can come up 
with a patch that handles NAT in the elegant way that I describe in 
the bugnotes, as I am but a mere integrator who has limited C skills.

In the absence of such a patch being offered, we await William 
Waites' patch and disclaimer which will at least be more sufficient 
than the current externip= method.

Those with an interest in the discussion of how Asterisk should 
handle being put behind a NAT should direct their attention to:

http://bugs.digium.com/bug_view_page.php?bug_id=0000104

JT

...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