[64766] in North American Network Operators' Group
OT: RE: IPv6 NAT
daemon@ATHENA.MIT.EDU (Paul Timmins)
Sat Nov 1 15:38:08 2003
From: Paul Timmins <paul@timmins.net>
To: hackerwacker@cybermesa.com
Cc: NANOG list <nanog@merit.edu>
In-Reply-To: <1067715053.1401.25.camel@jameshome>
Date: Sat, 01 Nov 2003 15:37:19 -0500
Errors-To: owner-nanog-outgoing@merit.edu
On Sat, 2003-11-01 at 14:30, james wrote:
> We use the Grandstream via sipphone.com for office to office calls.
> It is using the RTSP. Just doing some cheap testing before we integrate
> this into our Soft Switch, PBX and the PSTN.
>
> The Sipphone has a "STUN" server function that makes doing SIP behind
> NAT/PAT workable. I am a little hazy on its function as I am testing and
> wanted the phones on public IP's. Seems to keep the NAT/PAT translations
> constant by communicating with a remote server. Some users are doing SIP
> with this phone without problems behind NAT/PAT.
On mine I'm using for testing, if you turn NAT traversal on, and erase
the contents of the STUN field (don't leave them at 0.0.0.0, it'll
break), it does NAT traversal if the server supports it, without need
for a STUN server, which I still can't find a copy of. Fortunately it's
unnecessary. It works, as long as I don't try to contact another phone
behind another NAT.
-Paul
--
Paul Timmins <paul@timmins.net>
