[Asterisk-Users] A solution for SIP and NAT
Michael Kane
mkane at to-talk.com
Tue Jul 1 17:58:16 MST 2003
Your correct, Cisco devices stuff the WAN address in the Via: header which
in turn allows the proxy to correctly register the UA for an incoming call
attempt to that UA. If Mark is mentioning STUN as I said before, the only
devices I'm aware of are the SNOM 100 and Grandstream 101. These devices
rely on an external mechanism to properly construct the Via: header
otherwise the proxy has the incorrect return IP address of the UA.
Michael Kane
To-Talk Communications LLC.
37 Sandusky Dr.
Wareham, Ma. 02571
www.to-talk.com
508-295-2826
----- Original Message -----
From: "John Todd" <jtodd at loligo.com>
To: <asterisk-users at lists.digium.com>
Sent: Tuesday, July 01, 2003 8:16 PM
Subject: Re: [Asterisk-Users] A solution for SIP and NAT
>
> No, it works fine. SIP UA behind the NAT. Asterisk outside the NAT.
> "nat=1" set on the SIP peer. Works fine. Really. It does.
>
> I use Cisco equipment for my UA's. The catch might be that the Cisco
> devices are "more" clever than their counterparts, and will compare
> the "Via:" header against their own known IP address and re-issue
> their REGISTERs and INVITEs after they learn of their external
> addresses. However, I think Mark had this working with non-Cisco
> devices as well by using "actual" port numbers instead of
> SIP-reported port numbers, which breaks the RFC but makes for
> functional SIP calls.
>
> JT
>
>
> >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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> >> >>
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> >> >>
> >>
>
>>><http://lists.digium.com/mailman/listinfo/asterisk-users>http://lists.dig
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> >um.com/mailman/listinfo/asterisk-users
> >> >>
> >>
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> >>
> >
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