Re: [rtcweb] Sebastien Cubaud's non-ICE mechanism (Re: Summary of ICE discussion)

<sebastien.cubaud@orange.com> Tue, 11 October 2011 11:10 UTC

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From: <sebastien.cubaud@orange.com>
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Subject: Re: [rtcweb] Sebastien Cubaud's non-ICE mechanism (Re: Summary of ICE discussion)
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Hi, sorry for my previous message that I probably sent too rapidly: I have just realized that RFC 5576 couldn't be used to describe synchronization source from remote endpoints. So, even for SSRC, extended procedures would be required.. Too bad! (;
Cheers
Sebastien

-----Message d'origine-----
De : CUBAUD Sebastien RD-CORE-LAN 
Envoyé : mardi 11 octobre 2011 09:48
À : 'Jonathan Rosenberg'; rtcweb@ietf.org
Objet : RE: [rtcweb] Sebastien Cubaud's non-ICE mechanism (Re: Summary of ICE discussion)

Hi Jonathan, 

The use of the SSRC - prior to being accessible via JS for mux/demux purposes- could be an alternative to using the sequence number and/or the timestamp (as suggested by Randell). SIP endpoints interfacing with RTC-Web browsers would then only be required to implement RFC 5576 (which may be less costly than STUN implementation, the SHA-1 computation + CRC compute for the fingerprint)..
Cheers
Sebastien   

-----Message d'origine-----
De : rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] De la part de Jonathan Rosenberg
Envoyé : mardi 11 octobre 2011 05:13
À : rtcweb@ietf.org
Objet : Re: [rtcweb] Sebastien Cubaud's non-ICE mechanism (Re: Summary of ICE discussion)

Security issues aside, the proposed solution does not work with existing 
SIP/RTP implementations. The main drawback of the ICE solution is that 
it won't work with deployed equipment. I see little benefit in 
specifying another solution which does not fix the main limitation we 
have with ICE.

-Jonathan R.

On 10/10/2011 8:11 PM, Randell Jesup wrote:
> On 10/10/2011 7:04 PM, Eric Rescorla wrote:
>> On Mon, Oct 10, 2011 at 1:50 PM, Harald
>> Alvestrand<harald@alvestrand.no>; wrote:
>>> Changing the subject to keep threads separate....
>>>
>>> On 10/09/2011 12:00 PM, sebastien.cubaud@orange.com wrote:
>>>> Here are the steps I foresee before allowing the establishment of a
>>>> media
>>>> session:
>>>>
>>>> - Let's consider A (a RTC-Web compliant browser) connected to server
>>>> S and
>>>> wishing to share real-time media with destination B (potentially a SIP
>>>> endpoint or a browser)
>>>> - A& B learn via the signalling channel the triple @IP, transport proto
>>>> and associated listening port of the remote media
>>>> - A sends a few RTP packets to B (3 as in RFC 2833/4733 or more?).-
>>>> This
>>>> would allow the mechanism to resist against packet loss -. The
>>>> format of such
>>>> packets are to be discussed
>>>> - Assuming B receives these packets, it then sends via the signalling
>>>> channel an information from the media path unknown from S (i.e. not
>>>> accessible via
>>>> JS).
>>>> I propose to use the min of the sequence number of the RTP packets
>>>> received (which is random per RFC 3550)
>>>
>>> The sequence number is a 16-bit number, so there are 16 bits of
>>> randomness
>>> to play with here.
>>> An attack based on just returning a random number will succeed 1 out of
>>> 65.536 times; if any of the 3 packets' sequence numbers are
>>> acceptable, it
>>> will succeed 1 out of 21.845 times.
>
> If you use the sequence number and the timestamp, you have 48 bits of
> entropy...
>
>

-- 
Jonathan D. Rosenberg, Ph.D.                   SkypeID: jdrosen
Skype Chief Technology Strategist
jdrosen@skype.net                              http://www.skype.com
jdrosen@jdrosen.net                            http://www.jdrosen.net

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