Subject: [rtcweb] Fwd: I-D Action: draft-johnston-rtcweb-media-privacy-00.txt

From: Alan Johnston <[email-masked]>
Date: Tue, 31 May 2011 22:21:02 -0500

All,

Here is a new draft with some thoughts on media privacy in RTCWEB.  We
discuss the challenge of providing media privacy for browser users
when there is no standardization of the signaling protocol.  We also
discuss the need to sometimes allow trusted Man in the Middle (MitM)
servers to provide media services. We also discuss how ZRTP ("Media
Path Key Agreement for Unicast Secure RTP" RFC 6189) meets these
requirements.

Comments most welcome.

- Alan -

---------- Forwarded message ----------
From:  <[email-masked]>
Date: Tue, May 31, 2011 at 10:15 PM
Subject: I-D Action: draft-johnston-rtcweb-media-privacy-00.txt
To: [email-masked]

A New Internet-Draft is available from the on-line Internet-Drafts directories.

       Title           : RTCWEB Media Privacy
       Author(s)       : Alan Johnston
                         Philip Zimmermann
       Filename        : draft-johnston-rtcweb-media-privacy-00.txt
       Pages           : 7
       Date            : 2011-05-31

  RTCWEB is the joint effort between the IETF and the W3C to add real-
  time voice, video, and communication capabilities to browsers.  This
  document looks at the requirements for media privacy and existing
  mechanisms.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-johnston-rtcweb-media-privacy-00.txt

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/

This Internet-Draft can be retrieved at:
ftp://ftp.ietf.org/internet-drafts/draft-johnston-rtcweb-media-privacy-00.txt
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---

From: Justin Uberti <[email-masked]>
Date: Tue, 31 May 2011 20:41:31 -0700

Interesting read. One comment - I believe the statement in this draft

"In the RTCWEB architecture, there is no standardization of the signaling
protocol possible, and hence this [SRTP SDES] approach can not be used"
is not completely accurate. While the syntax of the signaling protocol (e.g.
HTTP, XMPP, SIP) is left up to the web application, we must have a clear
definition of the protocol semantics, as well as the format of session
descriptions (as understood by the browser), so that they can be mapped to
SDP / XMPP if desired.

Therefore there is no reason we cannot include SRTP keying information in
these session descriptions.

On Tue, May 31, 2011 at 8:21 PM, Alan Johnston <[email-masked]>wrote:

> rtcweb mailing list
> [email-masked]
> https://www.ietf.org/mailman/listinfo/rtcweb
>