[Asterisk-Users] NAT/SIP solution?

[Uriel Carrasquilla]

Leif:
I take it that the * on the GW machine remains in the middle of the call
between SIP and the Nat'ed *, correct?
I also suspect that you will be using IAX over Ethernet (to avoid
compression/decompression delays), correct?
Great work!
Regards,
Uriel

-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Leif Madsen
Sent: Sunday, September 28, 2003 3:42 PM
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] NAT/SIP solution?


Stig Hess wrote:
> I meant where Asterisk is behing a NAT... sorry for the confusion.

Hi Stig,

If you are able to run * on your NAT'd box, then I have come up with a
work around (thanks wasim!!!) that will allow you to run an * box behind
your NAT, and still recieve and make SIP calls.

I haven't got the whole thing figured out yet in terms of
extensions.conf (but I am working on that today, will post on my website
later) but this is basically my configuration:

<remote> <--TDM400P--> <*> <--IAX--> <*> <--SIP--> <remote>
                       |-----NAT-----||FW|

So basically the * on the GW machine which is also the NAT / FW box
recieves the connection from the SIP remote end, then forwards all the
traffic over IAX to the NAT'd * box.  I just tested it, and it works
fine!  Once I get some more complex extensions.conf files setup, I will
post them.

Thanks,
Leif Madsen.

BTW:  As for just passing SIP through SIP, I believe it's a limitation
of the SIP protocol as the RTP ports are different than the connection
port, whereas IAX is all the same port for everything (from what I gather)

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