[VOIPSEC] sip nat traversal solution

[dhiraj.2.bhuyan at bt.com]

My colleague and myself proposed a NAT traversal and securing the signalling packets solution to 3GPP. Our work was taken up by Siemens and is now one of the three options in the feasibility study phase. You can find details about the solution from the 3GPP website.

http://www.3gpp.org/ftp/Specs/html-info/33802.htm


Dhiraj
--
Dhiraj Bhuyan, CISSP
Senior Security Researcher,
British Telecom, UK

-----Original Message-----
From: Voipsec-bounces at voipsa.org [mailto:Voipsec-bounces at voipsa.org]On
Behalf Of Randell Jesup
Sent: 08 November 2005 03:23
To: Diana Cionoiu
Cc: Voipsec at voipsa.org
Subject: Re: [VOIPSEC] sip nat traversal solution


Diana Cionoiu <diana-liste at diana.null.ro> writes:
>We've found a nice way to get rid of the stun and ice from the SIP client.
>It's already implemented in Yate0.9.0pre4 which is GPL. A nice diagram 
>can be found at http://yate.null.ro/pmwiki/index.php/Main/SIPNAT

        Unless I misread the documenation on your website, you don't
solve the general NAT problem  that ICE and TURN are designed to solve. 
Mostly YATE seems to rely on symmetric RTP combined with (potentially)
RTP proxying.

        The documenation is so disjointed on this I'm not even sure
of what it's supposed to be doing.  Even reading the mailing list didn't
help.

        Your code may be clean (I didn't look), but your documentation 
is missing very important information on _what_ it's doing.

-- 
Randell Jesup, Worldgate (developers of the Ojo videophone), ex-Amiga OS team
rjesup at wgate.com


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