[Asterisk-Users] A solution for SIP and NAT

[Michael Kane]

Sorry to answer your question, you need to down load the source from vovida
and compile it.  Follow the instrustion in the readme on the main page.  Do
not use ports indicated 10000 and 1000x.  Use 3478 and 3479.  Oh for the
alternate stun server (-a option) add 127.0.0.1.  It's really straight
forward.  I compiled STUND on RH 8.0.

Mike

Michael Kane
To-Talk Communications LLC.
37 Sandusky Dr.
Wareham, Ma. 02571
508-295-2826
----- Original Message ----- 
From: "Matteo Brancaleoni" <mbrancaleoni at espia.it>
To: <asterisk-users at lists.digium.com>
Sent: Tuesday, July 01, 2003 6:32 PM
Subject: Re: [Asterisk-Users] A solution for SIP and NAT


> Could you give some details about setting up a stun server?
> I'm doing some tests, and were successful using snom + stund
> from vovida . But I got a no-go with budgetones
> (that needs stund on a standard port that's 3478).
> When my snom contacts the stund server, I get a lot
> of info about the connection type, the ip, blah blah
> When the budgetone contacts it, I get only "Receive something len[20]"
> 3 times. Nothing more.
>
> Matteo.
>
> Scrive Michael Kane <mkane at to-talk.com>:
>
> > 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
> > >
> > > _______________________________________________
> > > Asterisk-Users mailing list
> > > Asterisk-Users at lists.digium.com
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
>
>
> -- 
>
> Matteo Brancaleoni
> Espia System Administrator
> http://www.espia.it
>
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