[Asterisk-Users] A solution for SIP and NAT

[justin at vergeworks.com]

John,

When you say you have SIP clients working behind NAT is this with ports 
mapped from a public ip to the phone? I.e. can many phones sit behind 1 
public ip and recieve incomming calls, and make outgoing calls?

- Justin

On Tue, 1 Jul 2003, John Todd wrote:

> 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.
> >  > >_______________________________________________
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> >  > >Asterisk-Users at lists.digium.com
> >  > >http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
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> >>
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