[Asterisk-Users] SIP + NAT = horrible mess

[Kim Lux]

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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-- 
Kim Lux,  Diesel Research Inc.