[Asterisk-Users] SIP + NAT = horrible mess

[David Boyd]

On Thu, 2005-01-27 at 16:06, Kim Lux wrote:
> I've got Grandstreams (SIP devices) working behind double NATs, none the
> less. 
> 
> I recommend turning STUN off and make sure that your SIP devices are
> generating random port numbers.  If they generate static port numbers,
> you'll get port collisions.
> 
> The other parameter to watch is the "keep alive" interval. I'm not an
> expert, but I think this has to be long enough so that the device
> doesn't disconnect from the router while the various signalling is
> getting set up.  (I've got it set to 20 seconds.)    
> 
> Maybe I'm missing something, but I thought it works quite well without
> STUN.  They've never ever dropped a call. 
> 
> 
> 
> On Fri, 2005-01-28 at 00:18 +0400, Jean-Michel Hiver wrote:
> > Hi Guys,
> > 
> > After days of fiddling, I can't really get my SIP device to work 
> > communicate with Asterisk behind NAT. Sometimes the STUN server is 
> > flaky, sometimes the device isn't reachable if the connection is dropped 
> > and then put back on, sometimes it registers OK, sometimes it doesn't, etc.
> > 
> > I've come to the same conclusion as the wiki: it's probably better to 
> > avoid this horrible mess by either using IAX or doing VPN. Letting the 
> > IAX option aside, are you aware of any SIP devices that support some 
> > simple, easy to use VPN protocol?
> > 
> > Cheers,
> > Jean-Michel.
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Will you Please share your configuration, I was ready to give  up,
thinking no one had been successful.

TIA
db