[Asterisk-Users] A solution for SIP and NAT

[Michael Kane]

Maybe I mis-understood the question or the architecture.  I assumed (I
know), the SIP UA sat behind the NAT and Asterisk sat on the public IP
network.(there are inhererent signaling problems in this scenario and will
not work without either the device having the ability to learn the WAN IP
address or the SIP aware firewall performing the translation for the SIP
UA).  If both the SIP UA and Asterisk are behind the NAT I would agree there
is no reason the UA and Asterisk shouldn't work.

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 6:20 PM
Subject: Re: [Asterisk-Users] A solution for SIP and NAT


> Sorry, I still don't know what you're talking about.
>
> Clients behind NAT can talk to Asterisk without difficulty, and I use
> that functionality all the time.  If that is not the case for you,
> I'm afraid you'll have to be much more specific about your problems
> for anyone to help you.  Despite many claims that SIP can't run
> behind a NAT without special configuration, I have proof that they're
> wrong.
>
> JT
>
>
> >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
> ><http://www.intertex.se>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" <<mailto:jtodd at loligo.com>jtodd at loligo.com>
> >To:
<<mailto:asterisk-users at lists.digium.com>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/>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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