[Asterisk-Users] A solution for SIP and NAT

[Michael Kane]

Hello, NAT/Firewall is truely a problem in the ITSP arena.  There is one solution I know of that works well as an  integrated DHCP/NAT/Firewall into a SIP aware firewall.  Check out www.intertex.se  and look at the IXX66 products.  They even have a device that integrates DSL/NAT/Firewall.  Or, one can purchase a SIP device that supports STUN(Grandstream and SNOM are the only vendors I know of that do) and install a STUN server.  If anyone is interested I have a STUN server running to test with.  Hope this helped....

Mike




Michael Kane
To-Talk Communications LLC.
37 Sandusky Dr.
Wareham, Ma. 02571
508-295-2826
----- Original Message ----- 
From: "John Todd" <jtodd at loligo.com>
To: <asterisk-users at lists.digium.com>
Sent: Tuesday, July 01, 2003 3:47 PM
Subject: Re: [Asterisk-Users] A solution for SIP and NAT


> I'm uncertain why you're not able to get SIP working for your user 
> agents (SIP clients.)  With Cisco equipment, as an example, it works 
> quite well and almost every 79xx or ATA-186 I have is behind a NAT, 
> and this configuration is duplicated across a dozen or more systems 
> now running behind almost every conceivable NAT/PAT situation*
> 
> Known working config:
> 
> UA -> (NAT) -> Internet -> Asterisk
> 
> Can you be more specific about your problems with SIP?  Perhaps you 
> have done so in the past, but re-state and maybe someone can see what 
> the problem is.
> 
> JT
> 
> 
> *Note: the Cisco PIX, while supposedly SIP-friendly, has been the one 
> box that has not worked with NAT/PAT SIP sessions.  I have not been 
> the admin on that system, but a fairly clueful Cisco wrangler has 
> been unable to make it work for originating calls in both directions 
> - only one-way origination works.)
> 
> 
> >Hi all.
> >
> >I have come to the conclusion that there just isn't anything out there
> >for allowing SIP and NAT to work together nicely. This is rather amazing
> >considering that as far back as March 2000 there are documents
> >describing how to do it.
> >
> >So I've started a really simple SIP and RTP proxy project, SaRP, on
> >sourceforge.net. Yesterday we uploaded 0.2 of the perl based release.
> >This is the first general release and should work for most people. We
> >are using it quite successfully for standard calls between all sorts of
> >NATed clients. All you need to do is forward UDP/5060 from your
> >firewall/router to the box running SaRP if you want incoming calls to
> >work and also allow UDP traffic from the ports listed in the config file
> >out.
> >
> >The project can be found at http://sarp.sourceforge.net/
> >
> >I would be very interested in any feedback you may have.
> >
> >Regards
> >
> >Andrew Radke.
> >_______________________________________________
> >Asterisk-Users mailing list
> >Asterisk-Users at lists.digium.com
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> _______________________________________________
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