Re: [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

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