[Asterisk-Users] SIP + NAT = horrible mess

Kim Lux lux at diesel-research.com
Thu Jan 27 17:29:19 MST 2005


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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-- 
Kim Lux,  Diesel Research Inc.





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