[Asterisk-Users] SIP + NAT = horrible mess

[Voip Business]

NAT=yes Rules
STUN=SUCKS
rtp streams =Rules


I have lots of devices connected behind NAT without trouble but in
fact with STUN was a real MESS


regards

Humberto



On Fri, 28 Jan 2005 10:40:25 -0500, Nabeel Jafferali
<nabeel at jafferali.net> wrote:
> > I don't think you can use NAT = yes unless there is a STUN
> > server involved.  See my post yesterday for my Grandstream settings.
> 
> No, I had nat=yes working with my Cisco 7960 which did not provide it's
> public IP. However, you need to tell the IP Phone to start using the IP
> and port that * received the SIP messages from for RTP traffic (use via
> IP address and via port).
> 
> --
> Nabeel Jafferali
> Tel: +1 (416) 628-9342  Toronto
>     +1 (646) 225-7426  New York
> FWD: 46990
> Email/MSN: nabeel<at>jafferali.net
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