[Asterisk-Users] SIP + NAT = horrible mess

[Rich Adamson]

Nat=yes with the phone behind a nat box and asterisk on a registered
IP works just fine with Cisco, Snom, Xlite and others (I haven't tried
many of the others, however).

------------------------
> I don't think you can use NAT = yes unless there is a STUN server
> involved.  See my post yesterday for my Grandstream settings. 
> 
> 
> On Fri, 2005-01-28 at 10:28 +0100, Radovan.Mihalik wrote:
> > Hello, 
> > 
> > I try to connect VoIP phones to Asterisk on private network,
> > And use Asterisk as outbound proxy via his public IP.
> > But the localnet and externip with nat=yes, just is not working,
> > I believe it might only rewrite SIP headers but does not touch
> > The rtp stream. Am I right ?
> > 
> > R.
> >  
> > 
> > -----Original Message-----
> > From: asterisk-users-bounces at lists.digium.com
> > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Kim Lux
> > Sent: Friday, January 28, 2005 1:29 AM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] SIP + NAT = horrible mess
> > 
> > Comments below. 
> > 
> > On Fri, 2005-01-28 at 08:18 +0800, Leo Ann Boon wrote:
> > > 
> > > Kim Lux wrote:
> > > 
> > > >I was expecting to have to port forward too and yet our setup doesn't
> > > >require it, not on the laptop nor on the wireless router. 
> > > >
> > > >I think as long as the SIP clients open a port on the NATing device
> > and
> > > >keep them open so the SIP provider can connect to it, all is well,
> > even
> > > >if STUN isn't used.  
> > > >
> > > >I was surprised by how easy it was to NAT the Grandstreams.  I had
> > > >visions of having every device being assigned a static IP and having
> > a
> > > >fistful of port forwards assigned to them on the router.   
> > > >  
> > > >
> > > You're connecting to a SIP provider or just Asterisk? 
> > 
> > Just a provider right now.  I'll tackle asterisk in a few days. 
> > 
> > > Most SIP provider 
> > > use a far-end NAT traversal device like Jasomi, Acmepacket or Kagoo.
> > The 
> > > NAT traversal device has the intelligence to figure out the UDP port 
> > > mapping used by the NAT. SER + nathelper has the effect.
> > 
> > I guess ignorance is bliss in this case. 
> > 
> > >  For my SER 
> > > setup, most of the time we can just plug the SIP phone into a router
> > and 
> > > it will work without any special config. Unfortunately, there're
> > certain 
> > > firewalls like PIX and MS ISA that will fail. In those cases, your
> > best 
> > > bet is to do port forwarding or use an outbound proxy. IIRC, Vonage
> > also 
> > > has the same problem.
> > 
> > Thanks for sharing this.  It may help some poor soul trying to get his
> > SIP device working in these situations. 
> > 
> > 
> > > _______________________________________________
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> > > Asterisk-Users at lists.digium.com
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> > > 
> -- 
> Kim Lux,  Diesel Research Inc.
> 
> 
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