[Asterisk-Users] A solution for SIP and NAT

[Matteo Brancaleoni]

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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