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" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> 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" <[EMAIL PROTECTED]> > >To: <[EMAIL PROTECTED]> > >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:[EMAIL PROTECTED]>[EMAIL PROTECTED]> > >> >To: > ><<mailto:[EMAIL PROTECTED]>[EMAIL PROTECTED]> > >> >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 > >> > > >[EMAIL PROTECTED] > >> > > >http://lists.digium.com/mailman/listinfo/asterisk-users > >> >> > >> >> _______________________________________________ > >> >> Asterisk-Users mailing list > >> >> > ><mailto:[EMAIL PROTECTED]>[EMAIL PROTECTED] > >> >> > >> > >>><http://lists.digium.com/mailman/listinfo/asterisk-users>http://lists.dig i > >um.com/mailman/listinfo/asterisk-users > >> >> > >> > >> _______________________________________________ > >> Asterisk-Users mailing list > >> [EMAIL PROTECTED] > >> http://lists.digium.com/mailman/listinfo/asterisk-users > >> > > > >_______________________________________________ > >Asterisk-Users mailing list > >[EMAIL PROTECTED] > >http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users