[Asterisk-Users] SIP + NAT = horrible mess

[Kim Lux]

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. 
> 
> 
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >    http://lists.digium.com/mailman/listinfo/asterisk-users
> > 
-- 
Kim Lux,  Diesel Research Inc.