asterisk users - Mar 2005 - Asterisk, SER, NAT, STUN and the whole debate

Guys.

Im reading a lot about ser, nat, stun, etc. And I noticed there are a lot of
ways to get around nat but I would like to hear some success stories about
handling nat users with multiple voip phones behind nat.

I have my asterisk box behind but ports are forwarded (5060 5004 10000-20000
for rtp and 4569 for iax2) but still.. I can quite figure out what ser and
stund have to do on this scenarios.

I know ser is a sip proxy and stun helps you get your outside ip known but I
would like to hear some actual setups and how people are solving the nat
issue within scenarios like:

Asterisk - nat (ports forwarded) - internet - nat - multiple voup phones

Any good stories? Ive read the wiki, googled, etc. But I guess its time to
hear what people have actually done and works. Why reinvent the wheel.

----- Original Message ----- 
From: "Anton Krall"
<akrall-lists@intruder.com.mx>
> would like to hear some actual setups and how people are solving the nat
> issue within scenarios like:
>
> Asterisk - nat (ports forwarded) - internet - nat - multiple voup phones

I've been playing with this with my friends for awhile now.  We've got
four
different Asterisk servers set up in four different cities:

1. 2 nics - one on internal network, other on external network.  TDM400 card 
with 2 FXO and 1 FXS, 2 different analog lines, and a LiveVoip IAX2 dialout. 
Various SIP phones connected, both from within the internal network and out 
on the internet from behind other NATs.

2. 1 nic - behind NAT (ports forwarded).  X100p with 1 analog line.  Various 
SIP phones, internal network and from behind other NATs.

3 & 4.  Like #2 but no X100p.

All four servers are connected via IAX2 - in all cases we can dial 
extensions for each other's systems and the call gets dumped to the correct 
server.  Also between server 1 & 2 we have local inter-city dialing working 
(if you dial an outside number that is local to the other city and don't put
a 1 in front of the number it dumps to the other server and dials out).

NAT hasn't proven to be a problem for us - the only thing we can't do as
a
result of all the SIP clients being natted is Reinvites - this just means 
that all conversation *must* go through the server as opposed to direct 
client-client transfer.

Servers that are behind nats have the correct IP settings set in SIP.CONF. 
As long as I set the STUN server on the sip clients to a good working STUN 
server everything works like a hot damn.   Nothing special....

regards,

Paul