[64764] in North American Network Operators' Group
RE: IPv6 NAT
daemon@ATHENA.MIT.EDU (james)
Sat Nov 1 14:33:58 2003
From: james <hackerwacker@cybermesa.com>
Reply-To: hackerwacker@cybermesa.com
To: NANOG list <nanog@merit.edu>
In-Reply-To: <DD7FE473A8C3C245ADA2A2FE1709D90B06C708@server2003.arneill-py.sacramento.ca.us>
Date: 01 Nov 2003 12:30:54 -0700
Errors-To: owner-nanog-outgoing@merit.edu
On Sat, 2003-11-01 at 11:57, Michel Py wrote:
> but on other SIP phones such as the
> popular $70 Grandstream if you know the IP address and port of the
> remote SIP phone you want to join you can dial it directly.
>
> Michel.
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.
--
James Edwards
Routing and Security
jamesh@cybermesa.com
At the Santa Fe Office: Internet at Cyber Mesa
505-988-9200 SIP:747-669-1965
