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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Thu, 1 Mar 2012 10:22:10 +0100
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Subject: Re: [rtcweb] Consensus call on JSEP/ROAP
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2012/3/1 Ravindran, Parthasarathi <pravindran@sonusnet.com>:
> Inaki,
>
> I agree that we have two different architecture in mind.

Partha, 4 months ago you talked all the time about "SIP to ROAP
gateway". Now you talk about "SIP to JSEP" gateway. I'm not sure
whether you properly understand that ROAP/JSEP is just a signaling
protocol/mechanism/API between JavaScript and the browser's WebRTC
stack to handle *WebRTC media* sessions:

http://tools.ietf.org/html/draft-uberti-rtcweb-jsep-02#section-2
------------------------------------------------------------------
   JSEP's handling of session descriptions is simple and
   straightforward. Whenever an offer/answer exchange is needed, the
   initiating side creates an offer by calling a createOffer() API on
   PeerConnection. The application can do massaging of that offer, if it
   wants to, and then uses it to set up its local config via a
   setLocalDescription() API. The offer is then sent off to the remote
   side over its preferred signaling mechanism (e.g. WebSockets); upon
   receipt of that offer, the remote party installs it using a
   setRemoteDescription() API.
--------------------------------------------------------------------

Please read http://tools.ietf.org/html/draft-sipdoc-rtcweb-open-wire-protoc=
ol-00
(just replace ROAP with JSEP everywhere), and then please let me know
if you understand that ROAP/JSEP information can be sent in
***multiple*** ways between web browser and HTTP/WebSocket server (and
vice versa), but "something more" than a media negotiation body is
*required*.

If you still think that JSEP is "mapeable" to SIP, then why does SIP
exist? why don't you directly send an SDP body over UDP or TCP?

Response: you need SIP because you need to tell the server who you
want to call to, and that information is NOT in the SDP, but in the
INVITE's Request URI. So ok, JSEP has NOT "Request URI", so the
desired destination of the media session must be indicated in some
other way (not always needed, see example 5.3. "Poker Game" in the
above draft) and you CANNOT standarize that because each web site has
his own HTTP logic, his own users identificator format (numeric ID,
URI, token...), his own everything. Please see the examples in the
above draft.

I hope this is clear after 4 months repeating it.



> Ranjit/Inaki,
>
> After IETF-83, I can write the informational mapping document to show how=
 to implement simple standard SIP using JSEP. Let implementers choose wheth=
er JSEP-SIP gateway or SIP over WebSocket is the way to go.

See above (JSEP is not mapeable to SIP), so you cannot build a JSEP to
SIP gateway.

It seems that you want to build the "standard-and-unique-Web-to-SIP-Gateway=
".
I'm sorry but this is NOT a competition. I'm not a vendor and I don't
sell gateways. And people can use SIP over WebSocket or a custom
protocol with a SIP gateway or whatever they want.


Best regards.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From fluffy@iii.ca  Thu Mar  1 09:22:22 2012
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Given we are still before the draft deadline, this is a very draft =
agenda. Roughly speaking the plan is to spend the first session on =
security and second session on other things.=20

Cullen, Magnus, & Ted



Agenda=20

First Session   -------------------

Admin Update - 10 min=20
- update from mmusic, avtcore, and other WG=20

W3C Update - 10 min

Identity Proxy  - 50 min=20

What to do about RTP and/or SDES - 80 min=20



Second Session  -----------------------

Admin - 10 min=20
- scheduling of next interim meeting=20

Signaling - 60 min =20

Data Channel - 30  min

Codec Selection - 50 min








From fluffy@iii.ca  Thu Mar  1 09:26:29 2012
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From: Cullen Jennings <fluffy@iii.ca>
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Uh, note the subject should have been RTCWeb not WebRTC. Sorry.

On Mar 1, 2012, at 10:22 AM, Cullen Jennings wrote:

>=20
> Given we are still before the draft deadline, this is a very draft =
agenda. Roughly speaking the plan is to spend the first session on =
security and second session on other things.=20
>=20
> Cullen, Magnus, & Ted
>=20
>=20
>=20
> Agenda=20
>=20
> First Session   -------------------
>=20
> Admin Update - 10 min=20
> - update from mmusic, avtcore, and other WG=20
>=20
> W3C Update - 10 min
>=20
> Identity Proxy  - 50 min=20
>=20
> What to do about RTP and/or SDES - 80 min=20
>=20
>=20
>=20
> Second Session  -----------------------
>=20
> Admin - 10 min=20
> - scheduling of next interim meeting=20
>=20
> Signaling - 60 min =20
>=20
> Data Channel - 30  min
>=20
> Codec Selection - 50 min
>=20
>=20
>=20
>=20
>=20
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From igor.faynberg@alcatel-lucent.com  Thu Mar  1 10:46:42 2012
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Subject: Re: [rtcweb] Draft Agenda for WebRTC at IETF83
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Thanks, Cullen!

I always wanted security to be on the agenda for the first session (to 
stimulate more discussions in between) , and I am happy that this is 
happening now.

Igor

On 3/1/2012 12:22 PM, Cullen Jennings wrote:
> Given we are still before the draft deadline, this is a very draft agenda. Roughly speaking the plan is to spend the first session on security and second session on other things.
>
> Cullen, Magnus,&  Ted
>
>
>
> Agenda
>
> First Session   -------------------
>
> Admin Update - 10 min
> - update from mmusic, avtcore, and other WG
>
> W3C Update - 10 min
>
> Identity Proxy  - 50 min
>
> What to do about RTP and/or SDES - 80 min
>
>
>
> Second Session  -----------------------
>
> Admin - 10 min
> - scheduling of next interim meeting
>
> Signaling - 60 min
>
> Data Channel - 30  min
>
> Codec Selection - 50 min
>
>
>
>
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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Good to get things out early!

Some other things that may want to be on the agenda, either as reporting 
from other groups or as items to discuss whether to take up elsewhere:

- SDP updates: msid and bundle - our use of those functions being 
defined in MMUSIC
- Use of multiplexing functions described by AVTCORE
- Congestion control as discussed in the ICCRG meeting
- RTP feature profiling: What do we use? (mandatory  / permitted / 
recommended-against)
- SDP feature profiling: What do we use? (same alternatives)
- Video resolution negotiation per SSRC: What are the alternatives? Can 
we recommend one?

And of course:

- Implementation feedback that should lead to specification updates, if any


On 03/01/2012 06:22 PM, Cullen Jennings wrote:
> Given we are still before the draft deadline, this is a very draft agenda. Roughly speaking the plan is to spend the first session on security and second session on other things.
>
> Cullen, Magnus,&  Ted
>
>
>
> Agenda
>
> First Session   -------------------
>
> Admin Update - 10 min
> - update from mmusic, avtcore, and other WG
>
> W3C Update - 10 min
>
> Identity Proxy  - 50 min
>
> What to do about RTP and/or SDES - 80 min
>
>
>
> Second Session  -----------------------
>
> Admin - 10 min
> - scheduling of next interim meeting
>
> Signaling - 60 min
>
> Data Channel - 30  min
>
> Codec Selection - 50 min
>
>
>
>
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


From harald@alvestrand.no  Fri Mar  2 04:41:28 2012
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In preparing for the Paris meeting, I stumbled across this freshly 
expired draft:

http://tools.ietf.org/html/draft-ietf-mmusic-sctp-sdp-00

It seems to me that it would be of interest to this WG to make sure this 
draft:

a) doesn't die
b) defines a "DTLS/SCTP" transport for SCTP on top of DTLS

Is there anything we need to do here?

                 Harald


From salvatore.loreto@ericsson.com  Fri Mar  2 04:47:04 2012
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Hi Harald,

I am going to refresh the draft soon ... I have on my todo list
and I will do most likely next week after the 00 deadline.

I am going do define labels for SCTP on top of DTLS and also
for SCTP encapsulated in UDP (but this is not of interest for RTCWeb).

thanks
Salvatore

On 3/2/12 2:41 PM, Harald Alvestrand wrote:
> In preparing for the Paris meeting, I stumbled across this freshly
> expired draft:
>
> http://tools.ietf.org/html/draft-ietf-mmusic-sctp-sdp-00
>
> It seems to me that it would be of interest to this WG to make sure this
> draft:
>
> a) doesn't die
> b) defines a "DTLS/SCTP" transport for SCTP on top of DTLS
>
> Is there anything we need to do here?
>
>                   Harald
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From Michael.Tuexen@lurchi.franken.de  Fri Mar  2 10:38:36 2012
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On Mar 2, 2012, at 1:41 PM, Harald Alvestrand wrote:

> In preparing for the Paris meeting, I stumbled across this freshly =
expired draft:
>=20
> http://tools.ietf.org/html/draft-ietf-mmusic-sctp-sdp-00
>=20
> It seems to me that it would be of interest to this WG to make sure =
this draft:
>=20
> a) doesn't die
> b) defines a "DTLS/SCTP" transport for SCTP on top of DTLS
As far as I understand, DTLS over SCTP is considered (the document
refers to RFC 6083) and not SCTP over DTLS.

Best regards
Michael
>=20
> Is there anything we need to do here?
>=20
>                Harald
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


From dburnett@voxeo.com  Fri Mar  2 13:17:56 2012
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Group,

I have just submitted a new draft [1] to create a registry for HTML =
Media constraints for use by the W3C getusermedia and WebRTC specs.  An =
early email proposal on this subject (not the registry, but the use of =
it) is available [2] for background.  I will be updating that with a =
fuller proposal that incorporates changes discussed on the Media Capture =
Task Force teleconference [3], so don't be surprised if you notice =
slight differences in the syntax in the registry from what was proposed =
in email [2].

I don't know how much discussion will happen on this list for this =
draft, but at the least this group should be aware of the W3C need for =
it.

-- dan

[1] =
https://datatracker.ietf.org/doc/draft-burnett-rtcweb-constraints-registry=
/
[2] http://lists.w3.org/Archives/Public/public-webrtc/2012Feb/0198.html
[3] =
http://lists.w3.org/Archives/Public/public-media-capture/2012Feb/att-0053/=
minutes-2012-02-28.html


From keith.drage@alcatel-lucent.com  Fri Mar  2 17:14:11 2012
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: Dan Burnett <dburnett@voxeo.com>
Date: Sat, 3 Mar 2012 02:14:00 +0100
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Pedantic process issue.

Do we need a proposed standard to create an IANA registry which only needs =
expert review to add a value, or will informational suffice?

Keith

> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
> Of Dan Burnett
> Sent: 02 March 2012 21:18
> To: <rtcweb@ietf.org>
> Subject: [rtcweb] New draft-burnett-rtcweb-constraints-registry-00
>=20
> Group,
>=20
> I have just submitted a new draft [1] to create a registry for HTML Media
> constraints for use by the W3C getusermedia and WebRTC specs.  An early
> email proposal on this subject (not the registry, but the use of it) is
> available [2] for background.  I will be updating that with a fuller
> proposal that incorporates changes discussed on the Media Capture Task
> Force teleconference [3], so don't be surprised if you notice slight
> differences in the syntax in the registry from what was proposed in email
> [2].
>=20
> I don't know how much discussion will happen on this list for this draft,
> but at the least this group should be aware of the W3C need for it.
>=20
> -- dan
>=20
> [1] https://datatracker.ietf.org/doc/draft-burnett-rtcweb-constraints-
> registry/
> [2] http://lists.w3.org/Archives/Public/public-webrtc/2012Feb/0198.html
> [3] http://lists.w3.org/Archives/Public/public-media-capture/2012Feb/att-
> 0053/minutes-2012-02-28.html
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From internet-drafts@ietf.org  Sat Mar  3 21:17:09 2012
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Subject: [rtcweb] I-D Action: draft-ietf-rtcweb-jsep-00.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies. This draft is a work item of the Real-Time Communication in WEB-browse=
rs Working Group of the IETF.

	Title           : Javascript Session Establishment Protocol
	Author(s)       : Justin Uberti
                          Cullen Jennings
	Filename        : draft-ietf-rtcweb-jsep-00.txt
	Pages           : 27
	Date            : 2012-03-03

   This document proposes a mechanism for allowing a Javascript
   application to fully control the signaling plane of a multimedia
   session, and discusses how this would work with existing signaling
   protocols.

   This document is an input document for discussion.  It should be
   discussed in the RTCWEB WG list, rtcweb@ietf.org.


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


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FYI

the following draft has been submitted to tsv wg

cheers
Salvatore

-------- Original Message --------
Subject: 	New Version Notification for 
draft-tuexen-tsvwg-sctp-dtls-encaps-00.txt
Date: 	Sun, 4 Mar 2012 11:53:54 +0100
From: 	internet-drafts@ietf.org <internet-drafts@ietf.org>
To: 	tuexen@fh-muenster.de <tuexen@fh-muenster.de>
CC: 	Salvatore Loreto <salvatore.loreto@ericsson.com>, 
"randall@lakerest.net" <randall@lakerest.net>, "randell_ietf@jesup.org" 
<randell_ietf@jesup.org>



A new version of I-D, draft-tuexen-tsvwg-sctp-dtls-encaps-00.txt has been successfully submitted by Michael Tuexen and posted to the IETF repository.

Filename:	 draft-tuexen-tsvwg-sctp-dtls-encaps
Revision:	 00
Title:		 DTLS Encapsulation of SCTP Packets for RTCWEB
Creation date:	 2012-03-04
WG ID:		 Individual Submission
Number of pages: 8

Abstract:
    The Stream Control Transmission Protocol (SCTP) is a transport
    protocol originally defined to run on top of the network protocols
    IPv4 or IPv6.  This memo document specifies how SCTP can be used on
    top of the Datagram Transport Layer Security (DTLS) protocol.  SCTP
    over DTLS is used by the RTCWeb protocol suite for transporting non-
    media data between browsers.




The IETF Secretariat


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    FYI<br>
    <br>
    the following draft has been submitted to tsv wg<br>
    <br>
    cheers<br>
    Salvatore<br>
    <br>
    -------- Original Message --------
    <table class="moz-email-headers-table" border="0" cellpadding="0"
      cellspacing="0">
      <tbody>
        <tr>
          <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Subject: </th>
          <td>New Version Notification for
            draft-tuexen-tsvwg-sctp-dtls-encaps-00.txt</td>
        </tr>
        <tr>
          <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Date: </th>
          <td>Sun, 4 Mar 2012 11:53:54 +0100</td>
        </tr>
        <tr>
          <th align="RIGHT" nowrap="nowrap" valign="BASELINE">From: </th>
          <td><a class="moz-txt-link-abbreviated" href="mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a> <a class="moz-txt-link-rfc2396E" href="mailto:internet-drafts@ietf.org">&lt;internet-drafts@ietf.org&gt;</a></td>
        </tr>
        <tr>
          <th align="RIGHT" nowrap="nowrap" valign="BASELINE">To: </th>
          <td><a class="moz-txt-link-abbreviated" href="mailto:tuexen@fh-muenster.de">tuexen@fh-muenster.de</a> <a class="moz-txt-link-rfc2396E" href="mailto:tuexen@fh-muenster.de">&lt;tuexen@fh-muenster.de&gt;</a></td>
        </tr>
        <tr>
          <th align="RIGHT" nowrap="nowrap" valign="BASELINE">CC: </th>
          <td>Salvatore Loreto <a class="moz-txt-link-rfc2396E" href="mailto:salvatore.loreto@ericsson.com">&lt;salvatore.loreto@ericsson.com&gt;</a>,
            <a class="moz-txt-link-rfc2396E" href="mailto:randall@lakerest.net">"randall@lakerest.net"</a> <a class="moz-txt-link-rfc2396E" href="mailto:randall@lakerest.net">&lt;randall@lakerest.net&gt;</a>,
            <a class="moz-txt-link-rfc2396E" href="mailto:randell_ietf@jesup.org">"randell_ietf@jesup.org"</a> <a class="moz-txt-link-rfc2396E" href="mailto:randell_ietf@jesup.org">&lt;randell_ietf@jesup.org&gt;</a></td>
        </tr>
      </tbody>
    </table>
    <br>
    <br>
    <pre>A new version of I-D, draft-tuexen-tsvwg-sctp-dtls-encaps-00.txt has been successfully submitted by Michael Tuexen and posted to the IETF repository.

Filename:	 draft-tuexen-tsvwg-sctp-dtls-encaps
Revision:	 00
Title:		 DTLS Encapsulation of SCTP Packets for RTCWEB
Creation date:	 2012-03-04
WG ID:		 Individual Submission
Number of pages: 8

Abstract:
   The Stream Control Transmission Protocol (SCTP) is a transport
   protocol originally defined to run on top of the network protocols
   IPv4 or IPv6.  This memo document specifies how SCTP can be used on
   top of the Datagram Transport Layer Security (DTLS) protocol.  SCTP
   over DTLS is used by the RTCWeb protocol suite for transporting non-
   media data between browsers.

                                                                                  


The IETF Secretariat
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Thread-Topic: I-D Action: draft-muthu-behave-consent-freshness-00.txt
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From: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
To: <rtcweb@ietf.org>
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Subject: [rtcweb] FW: I-D Action: draft-muthu-behave-consent-freshness-00.txt
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This I-D describes a new STUN usage for enabling WebRTC implementations
to perform consent freshness. It is far from complete, but comments
(especially on the design considerations/approach) are welcome..

Muthu

-----Original Message-----
From: i-d-announce-bounces@ietf.org
[mailto:i-d-announce-bounces@ietf.org] On Behalf Of
internet-drafts@ietf.org
Sent: Monday, March 05, 2012 2:14 AM
To: i-d-announce@ietf.org
Subject: I-D Action: draft-muthu-behave-consent-freshness-00.txt


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

	Title           : STUN Usage for Consent Freshness
	Author(s)       : Muthu Arul Mozhi Perumal
                          Dan Wing
                          Hadriel Kaplan
	Filename        : draft-muthu-behave-consent-freshness-00.txt
	Pages           : 10
	Date            : 2012-03-04

   This document describes a STUN usage that enables WebRTC
   implementations to verify the peer consent for continuing to receive
   traffic on a candidate pair ICE is using for a media component after
   session establishment.  Verification of peer consent is necessary to
   ensure that a malicious JavaScript cannot use the browser as a
   platform for launching attacks.  This form of consent verification
   also serves the purpose of refreshing NAT bindings.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-muthu-behave-consent-freshness
-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-muthu-behave-consent-freshness-
00.txt

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From harald@alvestrand.no  Sun Mar  4 13:56:37 2012
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Subject: [rtcweb] draft-jesup-rtp-congestion-reqs-00.txt: Requirements for RTP congestion control
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For the WG's interest.
This (among other inputs) will be discussed in the IRTF ICCRG WG on 
Tuesday in Paris.

               Harald

-------- Original Message --------
Subject: 	New Version Notification for 
draft-jesup-rtp-congestion-reqs-00.txt
Date: 	Sun, 04 Mar 2012 06:19:29 -0800
From: 	internet-drafts@ietf.org
To: 	harald@alvestrand.no
CC: 	randell-ietf@jesup.org



A new version of I-D, draft-jesup-rtp-congestion-reqs-00.txt has been successfully submitted by Harald Alvestrand and posted to the IETF repository.

Filename:	 draft-jesup-rtp-congestion-reqs
Revision:	 00
Title:		 Congestion Control Requirements For Real Time Media
Creation date:	 2012-03-04
WG ID:		 Individual Submission
Number of pages: 7

Abstract:
    Congestion control is needed for all data transported across the
    Internet, in order to promote fair usage and prevent congestion
    collapse.  The requirements for interactive, point-to-point real time
    multimedia, which needs by low-delay, semi-reliable data delivery,
    are different from the requirements for bulk transfer like FTP or
    bursty transfers like Web pages, and the TCP algorithms are not
    suitable for this traffic.

    This document attempts to describe a set of requirements that can be
    used to evaluate other congestion control mechanisms in order to
    figure out their fitness for this purpose.





The IETF Secretariat



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    For the WG's interest.<br>
    This (among other inputs) will be discussed in the IRTF ICCRG WG on
    Tuesday in Paris.<br>
    <br>
    Â Â Â Â Â Â Â Â Â Â Â Â Â  Harald<br>
    <br>
    -------- Original Message --------
    <table class="moz-email-headers-table" border="0" cellpadding="0"
      cellspacing="0">
      <tbody>
        <tr>
          <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Subject: </th>
          <td>New Version Notification for
            draft-jesup-rtp-congestion-reqs-00.txt</td>
        </tr>
        <tr>
          <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Date: </th>
          <td>Sun, 04 Mar 2012 06:19:29 -0800</td>
        </tr>
        <tr>
          <th align="RIGHT" nowrap="nowrap" valign="BASELINE">From: </th>
          <td><a class="moz-txt-link-abbreviated" href="mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a></td>
        </tr>
        <tr>
          <th align="RIGHT" nowrap="nowrap" valign="BASELINE">To: </th>
          <td><a class="moz-txt-link-abbreviated" href="mailto:harald@alvestrand.no">harald@alvestrand.no</a></td>
        </tr>
        <tr>
          <th align="RIGHT" nowrap="nowrap" valign="BASELINE">CC: </th>
          <td><a class="moz-txt-link-abbreviated" href="mailto:randell-ietf@jesup.org">randell-ietf@jesup.org</a></td>
        </tr>
      </tbody>
    </table>
    <br>
    <br>
    <pre>A new version of I-D, draft-jesup-rtp-congestion-reqs-00.txt has been successfully submitted by Harald Alvestrand and posted to the IETF repository.

Filename:	 draft-jesup-rtp-congestion-reqs
Revision:	 00
Title:		 Congestion Control Requirements For Real Time Media
Creation date:	 2012-03-04
WG ID:		 Individual Submission
Number of pages: 7

Abstract:
   Congestion control is needed for all data transported across the
   Internet, in order to promote fair usage and prevent congestion
   collapse.  The requirements for interactive, point-to-point real time
   multimedia, which needs by low-delay, semi-reliable data delivery,
   are different from the requirements for bulk transfer like FTP or
   bursty transfers like Web pages, and the TCP algorithms are not
   suitable for this traffic.

   This document attempts to describe a set of requirements that can be
   used to evaluate other congestion control mechanisms in order to
   figure out their fitness for this purpose.


                                                                                  


The IETF Secretariat

</pre>
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--------------070905040602090207080404--

From jmpolk@cisco.com  Sun Mar  4 14:16:43 2012
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To: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
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Subject: Re: [rtcweb] draft-jesup-rtp-congestion-reqs-00.txt: Requirements for RTP congestion control
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Harald

Which of these two is expected to be the conflict with rtp-congestion hour?

TUESDAY, March 27, 2012

1500-1520  Beverage and Snack Break I - Hall Maillot A
1520-1700  Afternoon Session II
252A                    IRTF    iccrg=20
Internet Congestion Control Research Group
Maillot                 RAI     avtext=20
Audio/Video Transport Extensions WG
253                     RAI     vipr=20
Verification Involving PSTN Reachability WG

1710-1810  Afternoon Session III
252A                    IRTF    iccrg=20
Internet Congestion Control Research Group
252B                    RAI     sipcore=20
Session Initiation Protocol Core WG

I vote for Session II, but that's just one voice.

Since most of us aren't part of the ICCRG mailing=20
list, can you ask and pass along which session=20
the rtp-congestion hour long discussion will be in? Thanks

BTW - there already is no TSV conflict during either timeslot.

James

At 03:56 PM 3/4/2012, Harald Alvestrand wrote:
>For the WG's interest.
>This (among other inputs) will be discussed in=20
>the IRTF ICCRG WG on Tuesday in Paris.
>
>=C2 =C2 =C2 =C2 =C2 =C2 =C2 =C2 =C2 =C2 =C2 =C2 =C2  Harald
>
>-------- Original Message --------
>Subject: New Version Notification for=
 draft-jesup-rtp-congestion-reqs-00.txt
>Date: Sun, 04 Mar 2012 06:19:29 -0800
>From: <mailto:internet-drafts@ietf.org>internet-drafts@ietf.org
>To: <mailto:harald@alvestrand.no>harald@alvestrand.no
>CC: <mailto:randell-ietf@jesup.org>randell-ietf@jesup.org
>
>
>
>A new version of I-D,=20
>draft-jesup-rtp-congestion-reqs-00.txt has been=20
>successfully submitted by Harald Alvestrand and posted to the IETF=
 repository.
>
>Filename:        draft-jesup-rtp-congestion-reqs
>Revision:        00
>Title:           Congestion Control Requirements For Real Time Media
>Creation date:   2012-03-04
>WG ID:           Individual Submission
>Number of pages: 7
>
>Abstract:
>    Congestion control is needed for all data transported across the
>    Internet, in order to promote fair usage and prevent congestion
>    collapse.  The requirements for interactive, point-to-point real time
>    multimedia, which needs by low-delay, semi-reliable data delivery,
>    are different from the requirements for bulk transfer like FTP or
>    bursty transfers like Web pages, and the TCP algorithms are not
>    suitable for this traffic.
>
>    This document attempts to describe a set of requirements that can be
>    used to evaluate other congestion control mechanisms in order to
>    figure out their fitness for this purpose.
>
>
>=20
>
>
>
>The IETF Secretariat
>
>
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.orghttps://www.ietf.org/mailman/listinfo/rtcweb


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Subject: Re: [rtcweb] draft-jesup-rtp-congestion-reqs-00.txt: Requirements for RTP congestion control
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On 03/04/2012 11:16 PM, James M. Polk wrote:
> Harald
>
> Which of these two is expected to be the conflict with rtp-congestion 
> hour?
Thanks for the pointer to the conflict - I'll pass along to the ICCRG 
chairs.
>
> TUESDAY, March 27, 2012
>
> 1500-1520  Beverage and Snack Break I - Hall Maillot A
> 1520-1700  Afternoon Session II
> 252A                    IRTF    iccrg Internet Congestion Control 
> Research Group
> Maillot                 RAI     avtext Audio/Video Transport 
> Extensions WG
> 253                     RAI     vipr Verification Involving PSTN 
> Reachability WG
>
> 1710-1810  Afternoon Session III
> 252A                    IRTF    iccrg Internet Congestion Control 
> Research Group
> 252B                    RAI     sipcore Session Initiation Protocol 
> Core WG
>
> I vote for Session II, but that's just one voice.
I think so too. I don't know of RTCWEB related material in either session.

>
> Since most of us aren't part of the ICCRG mailing list, can you ask 
> and pass along which session the rtp-congestion hour long discussion 
> will be in? Thanks

Will do.

>
> BTW - there already is no TSV conflict during either timeslot.
>
> James
>
> At 03:56 PM 3/4/2012, Harald Alvestrand wrote:
>> For the WG's interest.
>> This (among other inputs) will be discussed in the IRTF ICCRG WG on 
>> Tuesday in Paris.
>>
>> Â Â Â Â Â Â Â Â Â Â Â Â Â  Harald
>>
>> -------- Original Message --------
>> Subject: New Version Notification for 
>> draft-jesup-rtp-congestion-reqs-00.txt
>> Date: Sun, 04 Mar 2012 06:19:29 -0800
>> From: <mailto:internet-drafts@ietf.org>internet-drafts@ietf.org
>> To: <mailto:harald@alvestrand.no>harald@alvestrand.no
>> CC: <mailto:randell-ietf@jesup.org>randell-ietf@jesup.org
>>
>>
>>
>> A new version of I-D, draft-jesup-rtp-congestion-reqs-00.txt has been 
>> successfully submitted by Harald Alvestrand and posted to the IETF 
>> repository.
>>
>> Filename:        draft-jesup-rtp-congestion-reqs
>> Revision:        00
>> Title:           Congestion Control Requirements For Real Time Media
>> Creation date:   2012-03-04
>> WG ID:           Individual Submission
>> Number of pages: 7
>>
>> Abstract:
>>    Congestion control is needed for all data transported across the
>>    Internet, in order to promote fair usage and prevent congestion
>>    collapse.  The requirements for interactive, point-to-point real time
>>    multimedia, which needs by low-delay, semi-reliable data delivery,
>>    are different from the requirements for bulk transfer like FTP or
>>    bursty transfers like Web pages, and the TCP algorithms are not
>>    suitable for this traffic.
>>
>>    This document attempts to describe a set of requirements that can be
>>    used to evaluate other congestion control mechanisms in order to
>>    figure out their fitness for this purpose.
>>
>>
>>
>>
>>
>>
>> The IETF Secretariat
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.orghttps://www.ietf.org/mailman/listinfo/rtcweb
>
>


From jmpolk@cisco.com  Sun Mar  4 23:01:30 2012
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At 11:32 PM 3/4/2012, Harald Alvestrand wrote:
>On 03/04/2012 11:16 PM, James M. Polk wrote:
>>Harald
>>
>>Which of these two is expected to be the conflict with rtp-congestion=
 hour?
>Thanks for the pointer to the conflict - I'll pass along to the ICCRG=
 chairs.
>>
>>TUESDAY, March 27, 2012
>>
>>1500-1520  Beverage and Snack Break I - Hall Maillot A
>>1520-1700  Afternoon Session II
>>252A                    IRTF    iccrg Internet=20
>>Congestion Control Research Group
>>Maillot                 RAI     avtext Audio/Video Transport Extensions WG
>>253                     RAI     vipr=20
>>Verification Involving PSTN Reachability WG
>>
>>1710-1810  Afternoon Session III
>>252A                    IRTF    iccrg Internet=20
>>Congestion Control Research Group
>>252B                    RAI     sipcore Session Initiation Protocol Core=
 WG
>>
>>I vote for Session II, but that's just one voice.
>I think so too. I don't know of RTCWEB related material in either session.

fair - but you know there are a LOT of SIP heads in RTCWEB...

James



>>Since most of us aren't part of the ICCRG=20
>>mailing list, can you ask and pass along which=20
>>session the rtp-congestion hour long discussion will be in? Thanks
>
>Will do.
>
>>
>>BTW - there already is no TSV conflict during either timeslot.
>>
>>James
>>
>>At 03:56 PM 3/4/2012, Harald Alvestrand wrote:
>>>For the WG's interest.
>>>This (among other inputs) will be discussed in=20
>>>the IRTF ICCRG WG on Tuesday in Paris.
>>>
>>>=C2 =C2 =C2 =C2 =C2 =C2 =C2 =C2 =C2 =C2 =C2 =C2 =C2  Harald
>>>
>>>-------- Original Message --------
>>>Subject: New Version Notification for=
 draft-jesup-rtp-congestion-reqs-00.txt
>>>Date: Sun, 04 Mar 2012 06:19:29 -0800
>>>From: <mailto:internet-drafts@ietf.org>internet-drafts@ietf.org
>>>To: <mailto:harald@alvestrand.no>harald@alvestrand.no
>>>CC: <mailto:randell-ietf@jesup.org>randell-ietf@jesup.org
>>>
>>>
>>>
>>>A new version of I-D,=20
>>>draft-jesup-rtp-congestion-reqs-00.txt has=20
>>>been successfully submitted by Harald=20
>>>Alvestrand and posted to the IETF repository.
>>>
>>>Filename:        draft-jesup-rtp-congestion-reqs
>>>Revision:        00
>>>Title:           Congestion Control Requirements For Real Time Media
>>>Creation date:   2012-03-04
>>>WG ID:           Individual Submission
>>>Number of pages: 7
>>>
>>>Abstract:
>>>    Congestion control is needed for all data transported across the
>>>    Internet, in order to promote fair usage and prevent congestion
>>>    collapse.  The requirements for interactive, point-to-point real time
>>>    multimedia, which needs by low-delay, semi-reliable data delivery,
>>>    are different from the requirements for bulk transfer like FTP or
>>>    bursty transfers like Web pages, and the TCP algorithms are not
>>>    suitable for this traffic.
>>>
>>>    This document attempts to describe a set of requirements that can be
>>>    used to evaluate other congestion control mechanisms in order to
>>>    figure out their fitness for this purpose.
>>>
>>>
>>>
>>>
>>>
>>>
>>>The IETF Secretariat
>>>
>>>
>>>_______________________________________________
>>>rtcweb mailing list
>>>rtcweb@ietf.orghttps://www.ietf.org/mailman/listinfo/rtcweb
>>


From stefan.lk.hakansson@ericsson.com  Mon Mar  5 06:19:27 2012
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Subject: Re: [rtcweb] Draft Agenda for WebRTC at IETF83
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On 03/01/2012 07:58 PM, Harald Alvestrand wrote:
> Good to get things out early!
>
> Some other things that may want to be on the agenda, either as reporting
> from other groups or as items to discuss whether to take up elsewhere:
>
> - SDP updates: msid and bundle - our use of those functions being
> defined in MMUSIC
> - Use of multiplexing functions described by AVTCORE
> - Congestion control as discussed in the ICCRG meeting
> - RTP feature profiling: What do we use? (mandatory  / permitted /
> recommended-against)
> - SDP feature profiling: What do we use? (same alternatives)

> - Video resolution negotiation per SSRC: What are the alternatives? Can
> we recommend one?
I assume that you include re-negotiation as well (i.e. if the display 
size is changed during the session) in this topic.

I think all the above are important topics, and we could also consider 
touching on:
- Mute/hold signaling (per SSRC)
- How to convey label/id of a MediaStreamTrack over a PeerConnection

I don't know if it is appropriate to use meeting time for this, or if it 
should be done on the list, but these are things we need to sort out.

Stefan
>
> And of course:
>
> - Implementation feedback that should lead to specification updates, if any
>
>
> On 03/01/2012 06:22 PM, Cullen Jennings wrote:
>> Given we are still before the draft deadline, this is a very draft agenda. Roughly speaking the plan is to spend the first session on security and second session on other things.
>>
>> Cullen, Magnus,&   Ted
>>
>>
>>
>> Agenda
>>
>> First Session   -------------------
>>
>> Admin Update - 10 min
>> - update from mmusic, avtcore, and other WG
>>
>> W3C Update - 10 min
>>
>> Identity Proxy  - 50 min
>>
>> What to do about RTP and/or SDES - 80 min
>>
>>
>>
>> Second Session  -----------------------
>>
>> Admin - 10 min
>> - scheduling of next interim meeting
>>
>> Signaling - 60 min
>>
>> Data Channel - 30  min
>>
>> Codec Selection - 50 min
>>
>>
>>
>>
>>
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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This is a multi-part message in MIME format.
--------------030402090305050105020405
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FYI - I just sent the following message to MMUSIC. I have also asked for 
agenda time in MMUSIC in Paris.
Please follow up in that list.

                Harald

-------- Original Message --------
Subject: 	[MMUSIC] Request for MMUSIC review: 
draft-alvestrand-rtcweb-msid-01
Date: 	Mon, 05 Mar 2012 15:29:15 +0100
From: 	Harald Alvestrand <harald@alvestrand.no>
To: 	mmusic <mmusic@ietf.org>



Hello,

this is a request for a cross-group review and evaluation for wider
application:

In the WebRTC effort, the W3C group has identified a preferred
programming model that involves bundling one or more video or audio
tracks together in something called a "MediaStream".

The RTCWEB WG has decided that it is not appropriate to use a CNAME to
group these together, since one identity or synchronization context
(which CNAME is the primary identifier for) may be the source of
multiple MediaStreams.

This means that media flows inside an RTP session have to be associated
with MediaStreams some other way.

One suggestion for this function is described here:

draft-alvestrand-rtcweb-msid-01

which also contains some background and notes about other approaches we
did not pick.

A presentation of the function is available from the minutes of the
RTCWEB interim meeting here:
http://www.ietf.org/proceedings/interim/2012/01/31/rtcweb/proceedings.html

the URL of the presentation is:

http://www.ietf.org/proceedings/interim/2012/01/31/rtcweb/slides/rtcweb-5.pdf

I hope this information is suitable for the MMUSIC group to give
guidance on further processing, including whether this draft needs to be
an MMUSIC draft or an RTCWEB draft.

                             Harald

_______________________________________________
mmusic mailing list
mmusic@ietf.org
https://www.ietf.org/mailman/listinfo/mmusic



--------------030402090305050105020405
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
  <head>

    <meta http-equiv="content-type" content="text/html; charset=ISO-8859-1">
  </head>
  <body bgcolor="#ffffff" text="#000000">
    FYI - I just sent the following message to MMUSIC. I have also asked
    for agenda time in MMUSIC in Paris.<br>
    Please follow up in that list.<br>
    <br>
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Harald<br>
    <br>
    -------- Original Message --------
    <table class="moz-email-headers-table" border="0" cellpadding="0"
      cellspacing="0">
      <tbody>
        <tr>
          <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Subject: </th>
          <td>[MMUSIC] Request for MMUSIC review:
            draft-alvestrand-rtcweb-msid-01</td>
        </tr>
        <tr>
          <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Date: </th>
          <td>Mon, 05 Mar 2012 15:29:15 +0100</td>
        </tr>
        <tr>
          <th align="RIGHT" nowrap="nowrap" valign="BASELINE">From: </th>
          <td>Harald Alvestrand <a class="moz-txt-link-rfc2396E" href="mailto:harald@alvestrand.no">&lt;harald@alvestrand.no&gt;</a></td>
        </tr>
        <tr>
          <th align="RIGHT" nowrap="nowrap" valign="BASELINE">To: </th>
          <td>mmusic <a class="moz-txt-link-rfc2396E" href="mailto:mmusic@ietf.org">&lt;mmusic@ietf.org&gt;</a></td>
        </tr>
      </tbody>
    </table>
    <br>
    <br>
    <pre>Hello,

this is a request for a cross-group review and evaluation for wider 
application:

In the WebRTC effort, the W3C group has identified a preferred 
programming model that involves bundling one or more video or audio 
tracks together in something called a "MediaStream".

The RTCWEB WG has decided that it is not appropriate to use a CNAME to 
group these together, since one identity or synchronization context 
(which CNAME is the primary identifier for) may be the source of 
multiple MediaStreams.

This means that media flows inside an RTP session have to be associated 
with MediaStreams some other way.

One suggestion for this function is described here:

draft-alvestrand-rtcweb-msid-01

which also contains some background and notes about other approaches we 
did not pick.

A presentation of the function is available from the minutes of the 
RTCWEB interim meeting here:
<a class="moz-txt-link-freetext" href="http://www.ietf.org/proceedings/interim/2012/01/31/rtcweb/proceedings.html">http://www.ietf.org/proceedings/interim/2012/01/31/rtcweb/proceedings.html</a>

the URL of the presentation is:

<a class="moz-txt-link-freetext" href="http://www.ietf.org/proceedings/interim/2012/01/31/rtcweb/slides/rtcweb-5.pdf">http://www.ietf.org/proceedings/interim/2012/01/31/rtcweb/slides/rtcweb-5.pdf</a>

I hope this information is suitable for the MMUSIC group to give 
guidance on further processing, including whether this draft needs to be 
an MMUSIC draft or an RTCWEB draft.

                            Harald

_______________________________________________
mmusic mailing list
<a class="moz-txt-link-abbreviated" href="mailto:mmusic@ietf.org">mmusic@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/mmusic">https://www.ietf.org/mailman/listinfo/mmusic</a>

</pre>
  </body>
</html>

--------------030402090305050105020405--

From fluffy@iii.ca  Mon Mar  5 07:29:26 2012
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From: Cullen Jennings <fluffy@iii.ca>
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Another possible topic is the Behave chairs would prefer we deal with =
the STUN and TURN URIs in RTCWeb.=20

On Mar 1, 2012, at 10:22 AM, Cullen Jennings wrote:

>=20
> Given we are still before the draft deadline, this is a very draft =
agenda. Roughly speaking the plan is to spend the first session on =
security and second session on other things.=20
>=20
> Cullen, Magnus, & Ted
>=20
>=20
>=20
> Agenda=20
>=20
> First Session   -------------------
>=20
> Admin Update - 10 min=20
> - update from mmusic, avtcore, and other WG=20
>=20
> W3C Update - 10 min
>=20
> Identity Proxy  - 50 min=20
>=20
> What to do about RTP and/or SDES - 80 min=20
>=20
>=20
>=20
> Second Session  -----------------------
>=20
> Admin - 10 min=20
> - scheduling of next interim meeting=20
>=20
> Signaling - 60 min =20
>=20
> Data Channel - 30  min
>=20
> Codec Selection - 50 min
>=20
>=20
>=20
>=20
>=20
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From igor.faynberg@alcatel-lucent.com  Mon Mar  5 07:43:12 2012
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Subject: Re: [rtcweb] Draft Agenda for WebRTC at IETF83
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One question:  How do the STUN and TURN matters in the RTCWeb 
environment differ from those in non-RTCWeb environments?

In other words, what will RTCWeb do differently?

Igor




On 3/5/2012 10:29 AM, Cullen Jennings wrote:
> Another possible topic is the Behave chairs would prefer we deal with the STUN and TURN URIs in RTCWeb.
>
> On Mar 1, 2012, at 10:22 AM, Cullen Jennings wrote:
>
>> Given we are still before the draft deadline, this is a very draft agenda. Roughly speaking the plan is to spend the first session on security and second session on other things.
>>
>> Cullen, Magnus,&  Ted
>>
>>
>>
>> Agenda
>>
>> First Session   -------------------
>>
>> Admin Update - 10 min
>> - update from mmusic, avtcore, and other WG
>>
>> W3C Update - 10 min
>>
>> Identity Proxy  - 50 min
>>
>> What to do about RTP and/or SDES - 80 min
>>
>>
>>
>> Second Session  -----------------------
>>
>> Admin - 10 min
>> - scheduling of next interim meeting
>>
>> Signaling - 60 min
>>
>> Data Channel - 30  min
>>
>> Codec Selection - 50 min
>>
>>
>>
>>
>>
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From petithug@acm.org  Mon Mar  5 07:59:11 2012
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From: Marc Petit-Huguenin <petithug@acm.org>
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Subject: Re: [rtcweb] Draft Agenda for WebRTC at IETF83
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On 03/05/2012 07:43 AM, Igor Faynberg wrote:
> One question:  How do the STUN and TURN matters in the RTCWeb environment
> differ from those in non-RTCWeb environments?

It is not about STUN/TURN, it is about STUN/TURN *URI*, i.e. how the STUN/TURN
server is provisioned.

> 
> In other words, what will RTCWeb do differently?
> 
> Igor
> 
> 
> 
> 
> On 3/5/2012 10:29 AM, Cullen Jennings wrote:
>> Another possible topic is the Behave chairs would prefer we deal with the
>> STUN and TURN URIs in RTCWeb.
>> 
>> On Mar 1, 2012, at 10:22 AM, Cullen Jennings wrote:
>> 
>>> Given we are still before the draft deadline, this is a very draft
>>> agenda. Roughly speaking the plan is to spend the first session on
>>> security and second session on other things.
>>> 
>>> Cullen, Magnus,&  Ted
>>> 
>>> 
>>> 
>>> Agenda
>>> 
>>> First Session   -------------------
>>> 
>>> Admin Update - 10 min - update from mmusic, avtcore, and other WG
>>> 
>>> W3C Update - 10 min
>>> 
>>> Identity Proxy  - 50 min
>>> 
>>> What to do about RTP and/or SDES - 80 min
>>> 
>>> 
>>> 
>>> Second Session  -----------------------
>>> 
>>> Admin - 10 min - scheduling of next interim meeting
>>> 
>>> Signaling - 60 min
>>> 
>>> Data Channel - 30  min
>>> 
>>> Codec Selection - 50 min
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> _______________________________________________ rtcweb mailing list 
>>> rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb
>> _______________________________________________ rtcweb mailing list 
>> rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb
> _______________________________________________ rtcweb mailing list 
> rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb
> 


- -- 
Marc Petit-Huguenin
Personal email: marc@petit-huguenin.org
Professional email: petithug@acm.org
Blog: http://blog.marc.petit-huguenin.org
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From harald@alvestrand.no  Mon Mar  5 11:05:57 2012
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Subject: [rtcweb] Fwd: Proof of concept: ROAP on top of JSEP - 184 lines and counting
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This might be of interest to people on rtcweb too.
Please keep the discussion on webrtc if at all possible, so that we 
don't do it in both places all the time.

               Harald

-------- Original Message --------
Subject: 	Proof of concept: ROAP on top of JSEP - 184 lines and counting
Resent-Date: 	Mon, 05 Mar 2012 13:02:06 +0000
Resent-From: 	public-webrtc@w3.org
Date: 	Mon, 05 Mar 2012 14:01:25 +0100
From: 	Harald Alvestrand <harald@alvestrand.no>
To: 	public-webrtc@w3.org <public-webrtc@w3.org>



Hi,

my weekend project is now visible to the world:
I implemented a layer of Javascript code that allows you to program
against the ROAP interface while using a JSEP object underneath it.

I have not implemented all aspects, but it's enough to do some testing.
For instance, close() is not implemented :-) - and I also left out
retransmission and actual glare handling.

(Of course, since JSEP isn't implemented yet, I had to create a dummy
JSEP to run my code against... this is also included.)

There are several aspects that show some degree of DOM/JS ignorance, for
instance:

- I don't know how to dispatch a signal from Javascript. Instead, I use
window.setTimeout() to achieve a similar-looking result: Getting
something to happen after the current function finishes.

- I don't know the proper way to generate errors for throwing. So the
code throws text strings when something bad happens.

Still, it's a fairly self-contained piece of code; I think it
demonstrates the order of magnitude of the library that's needed to
allow people to use a ROAP programming model on top of JSEP.

Comments (and patches) welcome!

The code is here:

http://code.google.com/p/webrtc-samples/source/browse/#svn%2Ftrunk%2Froap-jsep

Have fun!

                         Harald






--------------000403020808000607010204
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    <meta http-equiv="content-type" content="text/html; charset=ISO-8859-1">
  </head>
  <body bgcolor="#ffffff" text="#000000">
    This might be of interest to people on rtcweb too.<br>
    Please keep the discussion on webrtc if at all possible, so that we
    don't do it in both places all the time.<br>
    <br>
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Harald<br>
    <br>
    -------- Original Message --------
    <table class="moz-email-headers-table" border="0" cellpadding="0"
      cellspacing="0">
      <tbody>
        <tr>
          <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Subject: </th>
          <td>Proof of concept: ROAP on top of JSEP - 184 lines and
            counting</td>
        </tr>
        <tr>
          <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Resent-Date:
          </th>
          <td>Mon, 05 Mar 2012 13:02:06 +0000</td>
        </tr>
        <tr>
          <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Resent-From:
          </th>
          <td><a class="moz-txt-link-abbreviated" href="mailto:public-webrtc@w3.org">public-webrtc@w3.org</a></td>
        </tr>
        <tr>
          <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Date: </th>
          <td>Mon, 05 Mar 2012 14:01:25 +0100</td>
        </tr>
        <tr>
          <th align="RIGHT" nowrap="nowrap" valign="BASELINE">From: </th>
          <td>Harald Alvestrand <a class="moz-txt-link-rfc2396E" href="mailto:harald@alvestrand.no">&lt;harald@alvestrand.no&gt;</a></td>
        </tr>
        <tr>
          <th align="RIGHT" nowrap="nowrap" valign="BASELINE">To: </th>
          <td><a class="moz-txt-link-abbreviated" href="mailto:public-webrtc@w3.org">public-webrtc@w3.org</a> <a class="moz-txt-link-rfc2396E" href="mailto:public-webrtc@w3.org">&lt;public-webrtc@w3.org&gt;</a></td>
        </tr>
      </tbody>
    </table>
    <br>
    <br>
    <pre>Hi,

my weekend project is now visible to the world:
I implemented a layer of Javascript code that allows you to program 
against the ROAP interface while using a JSEP object underneath it.

I have not implemented all aspects, but it's enough to do some testing. 
For instance, close() is not implemented :-) - and I also left out 
retransmission and actual glare handling.

(Of course, since JSEP isn't implemented yet, I had to create a dummy 
JSEP to run my code against... this is also included.)

There are several aspects that show some degree of DOM/JS ignorance, for 
instance:

- I don't know how to dispatch a signal from Javascript. Instead, I use 
window.setTimeout() to achieve a similar-looking result: Getting 
something to happen after the current function finishes.

- I don't know the proper way to generate errors for throwing. So the 
code throws text strings when something bad happens.

Still, it's a fairly self-contained piece of code; I think it 
demonstrates the order of magnitude of the library that's needed to 
allow people to use a ROAP programming model on top of JSEP.

Comments (and patches) welcome!

The code is here:

<a class="moz-txt-link-freetext" href="http://code.google.com/p/webrtc-samples/source/browse/#svn%2Ftrunk%2Froap-jsep">http://code.google.com/p/webrtc-samples/source/browse/#svn%2Ftrunk%2Froap-jsep</a>

Have fun!

                        Harald




</pre>
  </body>
</html>

--------------000403020808000607010204--

From igor.faynberg@alcatel-lucent.com  Mon Mar  5 14:21:56 2012
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Subject: Re: [rtcweb] Draft Agenda for WebRTC at IETF83
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Ah, I have not followed this discussion on Behave.

I will be interested in learning what has transpired there.

Igor

On 3/5/2012 10:59 AM, Marc Petit-Huguenin wrote:
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA256
>
> On 03/05/2012 07:43 AM, Igor Faynberg wrote:
>> One question:  How do the STUN and TURN matters in the RTCWeb environment
>> differ from those in non-RTCWeb environments?
> It is not about STUN/TURN, it is about STUN/TURN *URI*, i.e. how the STUN/TURN
> server is provisioned.
>
>> In other words, what will RTCWeb do differently?
>>
>> Igor
>>
>>
>>
>>
>> On 3/5/2012 10:29 AM, Cullen Jennings wrote:
>>> Another possible topic is the Behave chairs would prefer we deal with the
>>> STUN and TURN URIs in RTCWeb.
>>>
>>> On Mar 1, 2012, at 10:22 AM, Cullen Jennings wrote:
>>>
>>>> Given we are still before the draft deadline, this is a very draft
>>>> agenda. Roughly speaking the plan is to spend the first session on
>>>> security and second session on other things.
>>>>
>>>> Cullen, Magnus,&   Ted
>>>>
>>>>
>>>>
>>>> Agenda
>>>>
>>>> First Session   -------------------
>>>>
>>>> Admin Update - 10 min - update from mmusic, avtcore, and other WG
>>>>
>>>> W3C Update - 10 min
>>>>
>>>> Identity Proxy  - 50 min
>>>>
>>>> What to do about RTP and/or SDES - 80 min
>>>>
>>>>
>>>>
>>>> Second Session  -----------------------
>>>>
>>>> Admin - 10 min - scheduling of next interim meeting
>>>>
>>>> Signaling - 60 min
>>>>
>>>> Data Channel - 30  min
>>>>
>>>> Codec Selection - 50 min
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________ rtcweb mailing list
>>>> rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb
>>> _______________________________________________ rtcweb mailing list
>>> rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb
>> _______________________________________________ rtcweb mailing list
>> rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb
>>
>
> - -- 
> Marc Petit-Huguenin
> Personal email: marc@petit-huguenin.org
> Professional email: petithug@acm.org
> Blog: http://blog.marc.petit-huguenin.org
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From internet-drafts@ietf.org  Mon Mar  5 15:49:14 2012
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies. This draft is a work item of the Real-Time Communication in WEB-browse=
rs Working Group of the IETF.

	Title           : RTCWeb Datagram Connection
	Author(s)       : Randell Jesup
                          Salvatore Loreto
                          Michael Tuexen
	Filename        : draft-ietf-rtcweb-data-channel-00.txt
	Pages           : 12
	Date            : 2012-03-05

   The Web Real-Time Communication (WebRTC) working group is charged to
   provide protocol support for direct interactive rich communication
   using audio, video, and data between two peers' web-browsers.  This
   document describes the non-media data transport aspects of the WebRTC
   framework.  It provides an architectural overview of how the Stream
   Control Transmission Protocol (SCTP) is used in the WebRTC context as
   a generic transport service allowing Web Browser to exchange generic
   data from peer to peer.


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


From martin.thomson@gmail.com  Mon Mar  5 16:28:28 2012
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On 5 March 2012 07:29, Cullen Jennings <fluffy@iii.ca> wrote:
> Another possible topic is the Behave chairs would prefer we deal with the STUN and TURN URIs in RTCWeb.

That's great, but I don't see how rtcweb can use them.  I suspect that
WebRTC could, maybe some implementations might, but if the idea is to
poke STUN/TURN configuration into the browser, then rtcweb has very
little to say about the matter.

From rachel.huang@huawei.com  Mon Mar  5 17:14:53 2012
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Date: Tue, 06 Mar 2012 01:14:35 +0000
From: "Rachel Huang(yihong)" <rachel.huang@huawei.com>
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Hi all,

We prepared a draft to describe the monitoring usage in RTCWEB. It discusses the current monitoring issues, and gives some guidelines for the selection of RTCP metrics. It is far from complete. Any comments and suggestions are welcomed. 

Best Regards!
Rachel

-----Original Message-----
From: internet-drafts@ietf.org [mailto:internet-drafts@ietf.org] 
Sent: Monday, March 05, 2012 3:09 PM
To: Rachel Huang(yihong)
Cc: Rachel Huang(yihong); roni.even@huawei.com
Subject: New Version Notification for draft-huang-rtcweb-monitoring-00.txt

A new version of I-D, draft-huang-rtcweb-monitoring-00.txt has been successfully submitted by Rachel Huang and posted to the IETF repository.

Filename:	 draft-huang-rtcweb-monitoring
Revision:	 00
Title:		 Web Real-Time Communication (RTCWEB): Monitoring Usage
Creation date:	 2012-03-05
WG ID:		 Individual Submission
Number of pages: 10

Abstract:
   This document describes the monitoring aspect usage in RTCWeb.  It
   also gives some guidelines for which RTCP XR metrics and extentions
   need to be supported.

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

The IETF Secretariat

From randell-ietf@jesup.org  Mon Mar  5 20:29:52 2012
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Subject: [rtcweb] I-D Action: draft-jesup-rtcweb-data-protocoll-00.txt
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You can find the I-D here:
http://www.ietf.org/id/draft-jesup-rtcweb-data-protocol-00.txt

This (minor) protocol provides bidirectional data channels on top of SCTP, per recent discussions, to support the requirements of draft-ietf-rtcweb-data-channel.  This protocol should be applicable to other uses of SCTP as well.

     Randell

A new version of I-D, draft-jesup-rtcweb-data-protocol-00.txt has been successfully submitted by Randell Jesup and posted to the IETF repository.

Filename:	 draft-jesup-rtcweb-data-protocol
Revision:	 00
Title:		 WebRTC Data Channel Protocol
Creation date:	 2012-03-06
WG ID:		 Individual Submission
Number of pages: 9

Abstract:
    The Web Real-Time Communication (WebRTC) working group is charged to
    provide protocols to support for direct interactive rich
    communication using audio, video, and data between two peers&#39; web-
    browsers.  This document specifies an actual (minor) protocol for how
    the JS-layer dataChannel objects provide the data channels between
    the peers.

    This minor protocol has applicability to other bidirectional uses of
    SCTP.

-- 
Randell Jesup
randell-ietf@jesup.org


From stefan.lk.hakansson@ericsson.com  Tue Mar  6 01:28:35 2012
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Subject: [rtcweb] Use-case document
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Hi!

The Use-case document
(<http://datatracker.ietf.org/doc/draft-ietf-rtcweb-use-cases-and-requirements/?include_text=1>)
will expire in early April.

The editors propose to add a new use-case when updating to -07. The new
use-case is about large sessions and is inspired by input made earlier
by Arno Bakker
(http://www.ietf.org/mail-archive/web/rtcweb/current/msg00530.html) and
recently by Harald Alvestrand
(http://lists.w3.org/Archives/Public/public-webrtc/2012Feb/0066.html).

Proposed text for the new use-case:

On-line lecture
===============

A lecture is given on-line by a very popular lecturer. The attendance is
very large. Because of this it is not really feasible to have
direct audiovisual feedback (for e.g. questions) for every participant,
instead participants can request the floor by clicking a button, and
gets a signal when audio and video has been set up (and the lecturer is
ready to take the question). The question is heard by all participants
(not only the lecturer).

Participants can join and leave the lecture at any time (i.e. they do
not need to be there at the start to be able to connect).

The service is operated by the university of the lecturer. It is set up 
such that all participants connect to a central node. The university can 
not invest much in HW, so the processing required (per participant) must 
be minimized.


New requirement:

- It must be possible to set up media streams and encryption in such a
way that processing in a central node is minimized (no transcoding
required, no decryption/re-encryption for every outgoing flow - just
simple forwarding).


Let us know what you think. If there is support we will add this 
use-case, if not we will just make a new revision of the document (with 
no major changes to the content).

Stefan for the editors

From ekr@rtfm.com  Tue Mar  6 05:01:53 2012
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From: Eric Rescorla <ekr@rtfm.com>
Date: Tue, 6 Mar 2012 05:01:12 -0800
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Subject: Re: [rtcweb] Use-case document
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Obviously, this is an interesting use case, but it's not clear to me that
it's something you would want to address entirely via WebRTC

1. It seems likely that the lecturer will be using something
rather more sophisticated than a Web browser in many cases
(often these lecturers are actually joint on-line and in-person
so you need microphone control in the room as well; see for
instance the popular Sandel Justice lecturers out of Harvard).

2. You also probably want to actually support not just real-time
viewing but also podcasting, time-shifting, etc., as well as
support for people who don't have WebRTC.

When you put these together, it might well be natural to have
a one-way stream via Flash, HTML5 video, etc., and then individual
WebRTC uplinks which get mixed into that single stream as necessary.
That makes questions about transcoding, re-encryption, etc. a lot
simpler.

-Ekr


On Tue, Mar 6, 2012 at 1:28 AM, Stefan Hakansson LK
<stefan.lk.hakansson@ericsson.com> wrote:
> Hi!
>
> The Use-case document
> (<http://datatracker.ietf.org/doc/draft-ietf-rtcweb-use-cases-and-requirements/?include_text=1>)
> will expire in early April.
>
> The editors propose to add a new use-case when updating to -07. The new
> use-case is about large sessions and is inspired by input made earlier
> by Arno Bakker
> (http://www.ietf.org/mail-archive/web/rtcweb/current/msg00530.html) and
> recently by Harald Alvestrand
> (http://lists.w3.org/Archives/Public/public-webrtc/2012Feb/0066.html).
>
> Proposed text for the new use-case:
>
> On-line lecture
> ===============
>
> A lecture is given on-line by a very popular lecturer. The attendance is
> very large. Because of this it is not really feasible to have
> direct audiovisual feedback (for e.g. questions) for every participant,
> instead participants can request the floor by clicking a button, and
> gets a signal when audio and video has been set up (and the lecturer is
> ready to take the question). The question is heard by all participants
> (not only the lecturer).
>
> Participants can join and leave the lecture at any time (i.e. they do
> not need to be there at the start to be able to connect).
>
> The service is operated by the university of the lecturer. It is set up such
> that all participants connect to a central node. The university can not
> invest much in HW, so the processing required (per participant) must be
> minimized.
>
>
> New requirement:
>
> - It must be possible to set up media streams and encryption in such a
> way that processing in a central node is minimized (no transcoding
> required, no decryption/re-encryption for every outgoing flow - just
> simple forwarding).
>
>
> Let us know what you think. If there is support we will add this use-case,
> if not we will just make a new revision of the document (with no major
> changes to the content).
>
> Stefan for the editors
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From Ranjit.Avasarala@Polycom.com  Tue Mar  6 22:07:43 2012
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From: "Avasarala, Ranjit" <Ranjit.Avasarala@Polycom.com>
To: Stefan Hakansson LK <stefan.lk.hakansson@ericsson.com>
Date: Wed, 7 Mar 2012 14:07:38 +0800
Thread-Topic: [rtcweb] Use-case document  - additional recording use case
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Subject: Re: [rtcweb] Use-case document  - additional recording use case
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Hi Stefan

Could we add a use case related to session recording? The use case would be

4.2. x Simple Video Recording Service

4.2. x.1 Description

The user who has joined a video communication service wants to record the c=
ommunication data from the web browser.

4.2.x.1.1  Derived requirements

TBD

I also feel there is a need for a API requirement - the Web API should prov=
ide a means to record the incoming media - both audio & video or audio only=
 or video only

Regards
Ranjit

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Stefan Hakansson LK
Sent: Tuesday, March 06, 2012 2:59 PM
To: rtcweb@ietf.org
Subject: [rtcweb] Use-case document

Hi!

The Use-case document
(<http://datatracker.ietf.org/doc/draft-ietf-rtcweb-use-cases-and-requireme=
nts/?include_text=3D1>)
will expire in early April.

The editors propose to add a new use-case when updating to -07. The new use=
-case is about large sessions and is inspired by input made earlier by Arno=
 Bakker
(http://www.ietf.org/mail-archive/web/rtcweb/current/msg00530.html) and rec=
ently by Harald Alvestrand (http://lists.w3.org/Archives/Public/public-webr=
tc/2012Feb/0066.html).

Proposed text for the new use-case:

On-line lecture
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D

A lecture is given on-line by a very popular lecturer. The attendance is ve=
ry large. Because of this it is not really feasible to have direct audiovis=
ual feedback (for e.g. questions) for every participant, instead participan=
ts can request the floor by clicking a button, and gets a signal when audio=
 and video has been set up (and the lecturer is ready to take the question)=
. The question is heard by all participants (not only the lecturer).

Participants can join and leave the lecture at any time (i.e. they do not n=
eed to be there at the start to be able to connect).

The service is operated by the university of the lecturer. It is set up suc=
h that all participants connect to a central node. The university can not i=
nvest much in HW, so the processing required (per participant) must be mini=
mized.


New requirement:

- It must be possible to set up media streams and encryption in such a way =
that processing in a central node is minimized (no transcoding required, no=
 decryption/re-encryption for every outgoing flow - just simple forwarding)=
.


Let us know what you think. If there is support we will add this use-case, =
if not we will just make a new revision of the document (with no major chan=
ges to the content).

Stefan for the editors
_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

From stefan.lk.hakansson@ericsson.com  Wed Mar  7 03:58:58 2012
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Subject: Re: [rtcweb] I-D Action: draft-jesup-rtcweb-data-protocoll-00.txt
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On 03/06/2012 05:27 AM, Randell Jesup wrote:
> You can find the I-D here:
> http://www.ietf.org/id/draft-jesup-rtcweb-data-protocol-00.txt
>
> This (minor) protocol provides bidirectional data channels on top of
> SCTP, per recent discussions, to support the requirements of
> draft-ietf-rtcweb-data-channel.  This protocol should be applicable
> to other uses of SCTP as well.
>
> Randell
>
> A new version of I-D, draft-jesup-rtcweb-data-protocol-00.txt has
> been successfully submitted by Randell Jesup and posted to the IETF
> repository.
>
> Filename:	 draft-jesup-rtcweb-data-protocol Revision:	 00 Title:
> WebRTC Data Channel Protocol Creation date:	 2012-03-06 WG ID:
> Individual Submission Number of pages: 9
>
> Abstract: The Web Real-Time Communication (WebRTC) working group is
> charged to provide protocols to support for direct interactive rich
> communication using audio, video, and data between two peers&#39;
> web- browsers.  This document specifies an actual (minor) protocol
> for how the JS-layer dataChannel objects provide the data channels
> between the peers.
>
> This minor protocol has applicability to other bidirectional uses of
> SCTP.
>

I had a quick glance, and it looks reasonable to me. //Stefan

From harald@alvestrand.no  Wed Mar  7 08:39:29 2012
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Subject: [rtcweb] RTP congestion meeting hosted by ICCRG in Paris
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This should be of interest to several of the participants.
Note that the time slot for ICCRG conflicts with AVTEXT (first half) and 
SIPCORE (second half); we chose the SIPCORE conflict as the lesser evil.

Please keep discussion of the technology on the rtp-congestion list, if 
you have to pick one.

                           Harald

-------- Original Message --------
Subject: 	[Iccrg] Preliminary agenda for ICCRG meeting @ IETF 83
Date: 	Wed, 07 Mar 2012 16:10:23 +0100
From: 	Michael Welzl <michawe@ifi.uio.no>
To: 	iccrg@cs.ucl.ac.uk



Ladies and gentlemen,

Let me present to you the preliminary agenda for the ICCRG meeting at
the upcoming Paris IETF, which is scheduled on Tuesday afternoon, in the
15:20-17:00 and 17:10-18:10 time slots. The second time slot is
dedicated to congestion control for RTP-carried real-time media; this is
an effort that comes out of rtcweb, with its own mailing list, see:
http://www.alvestrand.no/mailman/listinfo/rtp-congestion
The group is considering formation of a new WG, and that's why there is
an item called "a possible WG charter" in the agenda - however, note
that this meeting is not meant as a surrogate for a BoF (which is
clearly not what a RG could or should "host"), so this will be just
informational, not a time slot for wordsmithing. Content-wise,
congestion control for RTP seems to be clearly in scope of ICCRG, and so
I expect this to be an interesting session for all of us.


Time slot 1, 15:20-17:00

* Agenda bashing and update - Michael Welzl (5 min)

* Congestion Control and Notification over Longer Time Scales and at
Higher Layers - Dave McDysan (20 min)
   draft-mcdysan-iccrg-usecases-00

* On the Fairness of Transport Protocols in a Multi-Path Environment -
Hakim Adhari (15 min)
   (based on an upcoming ICC 2012 paper)

* Caching with TCP - Pasi Sarolahti (15 min)

* End-to-End Transmission Control through Inference about the Network -
Keith Winstein (30 min)
   (based on:
http://conferences.sigcomm.org/hotnets/2011/papers/hotnetsX-final100.pdf)


Time slot 2, 17:10-18:10

Topic chair: Harald Alvestrand

     * Introduction: The RTCWEB effort, and the traffic we expect to
generate - Harald (10 min)
           o Doc: draft-ietf-rtcweb-overview-02
           o Doc: draft-jesup-rtp-congestion-reqs-00
     * Nice behaviour for RTP-based services - Varun Singh, Colin
Perkins (10 min)
           o Doc: draft-perkins-avtcore-rtp-circuit-breakers-00
     * A delay based congestion control candidate - Stefan Holmer (15 min)
           o Doc: draft-alvestrand-rtcweb-congetstion-01
     * Feedback mechanisms that might be useful - Harald Alvestrand (5 min)
           o Doc: draft-alvestrand-rmcat-remb-00
     * A possible WG charter: RMCAT (15 min)
           o Doc: http://www.ietf.org/iesg/evaluation/rmcat-charter.txt
     * Summary, conclusions, next steps (5 min)


The times in slot 1 are based on what the presenters requested, but I'd
think that not everyone has planned for the normal (long) duration of
discussions at IETF meetings, and so it's nice to have some headroom.
This being said, I think we can still squeeze a presentation in there if
need be.


Cheers,
Michael


_______________________________________________
Iccrg mailing list
Iccrg@cs.ucl.ac.uk
http://oakham.cs.ucl.ac.uk/mailman/listinfo/iccrg



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    This should be of interest to several of the participants.<br>
    Note that the time slot for ICCRG conflicts with AVTEXT (first half)
    and SIPCORE (second half); we chose the SIPCORE conflict as the
    lesser evil.<br>
    <br>
    Please keep discussion of the technology on the rtp-congestion list,
    if you have to pick one.<br>
    <br>
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Harald<br>
    <br>
    -------- Original Message --------
    <table class="moz-email-headers-table" border="0" cellpadding="0"
      cellspacing="0">
      <tbody>
        <tr>
          <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Subject: </th>
          <td>[Iccrg] Preliminary agenda for ICCRG meeting @ IETF 83</td>
        </tr>
        <tr>
          <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Date: </th>
          <td>Wed, 07 Mar 2012 16:10:23 +0100</td>
        </tr>
        <tr>
          <th align="RIGHT" nowrap="nowrap" valign="BASELINE">From: </th>
          <td>Michael Welzl <a class="moz-txt-link-rfc2396E" href="mailto:michawe@ifi.uio.no">&lt;michawe@ifi.uio.no&gt;</a></td>
        </tr>
        <tr>
          <th align="RIGHT" nowrap="nowrap" valign="BASELINE">To: </th>
          <td><a class="moz-txt-link-abbreviated" href="mailto:iccrg@cs.ucl.ac.uk">iccrg@cs.ucl.ac.uk</a></td>
        </tr>
      </tbody>
    </table>
    <br>
    <br>
    <pre>Ladies and gentlemen,

Let me present to you the preliminary agenda for the ICCRG meeting at 
the upcoming Paris IETF, which is scheduled on Tuesday afternoon, in the 
15:20-17:00 and 17:10-18:10 time slots. The second time slot is 
dedicated to congestion control for RTP-carried real-time media; this is 
an effort that comes out of rtcweb, with its own mailing list, see: 
<a class="moz-txt-link-freetext" href="http://www.alvestrand.no/mailman/listinfo/rtp-congestion">http://www.alvestrand.no/mailman/listinfo/rtp-congestion</a>
The group is considering formation of a new WG, and that's why there is 
an item called "a possible WG charter" in the agenda - however, note 
that this meeting is not meant as a surrogate for a BoF (which is 
clearly not what a RG could or should "host"), so this will be just 
informational, not a time slot for wordsmithing. Content-wise, 
congestion control for RTP seems to be clearly in scope of ICCRG, and so 
I expect this to be an interesting session for all of us.


Time slot 1, 15:20-17:00

* Agenda bashing and update - Michael Welzl (5 min)

* Congestion Control and Notification over Longer Time Scales and at 
Higher Layers - Dave McDysan (20 min)
  draft-mcdysan-iccrg-usecases-00

* On the Fairness of Transport Protocols in a Multi-Path Environment - 
Hakim Adhari (15 min)
  (based on an upcoming ICC 2012 paper)

* Caching with TCP - Pasi Sarolahti (15 min)

* End-to-End Transmission Control through Inference about the Network - 
Keith Winstein (30 min)
  (based on: 
<a class="moz-txt-link-freetext" href="http://conferences.sigcomm.org/hotnets/2011/papers/hotnetsX-final100.pdf">http://conferences.sigcomm.org/hotnets/2011/papers/hotnetsX-final100.pdf</a>)


Time slot 2, 17:10-18:10

Topic chair: Harald Alvestrand

    * Introduction: The RTCWEB effort, and the traffic we expect to 
generate - Harald (10 min)
          o Doc: draft-ietf-rtcweb-overview-02
          o Doc: draft-jesup-rtp-congestion-reqs-00
    * Nice behaviour for RTP-based services - Varun Singh, Colin 
Perkins (10 min)
          o Doc: draft-perkins-avtcore-rtp-circuit-breakers-00
    * A delay based congestion control candidate - Stefan Holmer (15 min)
          o Doc: draft-alvestrand-rtcweb-congetstion-01
    * Feedback mechanisms that might be useful - Harald Alvestrand (5 min)
          o Doc: draft-alvestrand-rmcat-remb-00
    * A possible WG charter: RMCAT (15 min)
          o Doc: <a class="moz-txt-link-freetext" href="http://www.ietf.org/iesg/evaluation/rmcat-charter.txt">http://www.ietf.org/iesg/evaluation/rmcat-charter.txt</a>
    * Summary, conclusions, next steps (5 min)


The times in slot 1 are based on what the presenters requested, but I'd 
think that not everyone has planned for the normal (long) duration of 
discussions at IETF meetings, and so it's nice to have some headroom. 
This being said, I think we can still squeeze a presentation in there if 
need be.


Cheers,
Michael


_______________________________________________
Iccrg mailing list
<a class="moz-txt-link-abbreviated" href="mailto:Iccrg@cs.ucl.ac.uk">Iccrg@cs.ucl.ac.uk</a>
<a class="moz-txt-link-freetext" href="http://oakham.cs.ucl.ac.uk/mailman/listinfo/iccrg">http://oakham.cs.ucl.ac.uk/mailman/listinfo/iccrg</a>

</pre>
  </body>
</html>

--------------010205060602010505030508--

From mkomu@cs.hut.fi  Thu Mar  8 00:51:03 2012
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Dear WG,

I heard that you're making progress and have selected a specific 
protocol approach for rtcweb:

http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel

This is good news indeed. However, we noticed that the original 
comparison was missing few approaches like Teredo (RFC4380) or Host 
Identity Protocol (see e.g. RFC5770):

http://tools.ietf.org/html/draft-jesup-rtcweb-data

Despite the original document missed the two protocols, I'm sure that 
they were was discussed at some point, maybe even just in face-to-face 
conversations. Now, my goal is not advocate either of these protocols as 
a replacement for the WG as you have already consensus, but I'd rather 
like to get feedback why some protocols were excluded.

To be more specific, we're working on a research paper on deployment 
analysis on Host Identity Protocol (HIP) in Aalto University and we'd 
like get your expert opinion on especially why HIP was not the adopted 
in the context of rtcweb. Here's few example considerations:

- Never heard about HIP
- The substitute technologies do their work better
- Lack of implementation maturity or usability
- Not standards track yet; we can build rtcweb out of standards track 
components
- The fatal flaw for HIP in this particular context is X
- HIP is in the wrong layer of the stack?

Besides technical incentives, we're also interested business and 
political incentives. Feel free to respond me directly or via the 
mailing list (unless people think it's too off-the-topic). Your feedback 
will be anonymized in the publication.

Thanks!

From harald@alvestrand.no  Thu Mar  8 02:16:16 2012
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HIP was discussed at a far earlier stage, when talking about security. 
We chose to go with DTLS, SRTP and ICE instead, since these were 
mechanisms in (relatively) common use in this environment.

As far as I understand HIP, it does not contain congestion control 
mechanisms, so its use was not considered when picking a protocol for 
the data channel.

On 03/08/2012 09:51 AM, Miika Komu wrote:
> Dear WG,
>
> I heard that you're making progress and have selected a specific 
> protocol approach for rtcweb:
>
> http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel
>
> This is good news indeed. However, we noticed that the original 
> comparison was missing few approaches like Teredo (RFC4380) or Host 
> Identity Protocol (see e.g. RFC5770):
>
> http://tools.ietf.org/html/draft-jesup-rtcweb-data
>
> Despite the original document missed the two protocols, I'm sure that 
> they were was discussed at some point, maybe even just in face-to-face 
> conversations. Now, my goal is not advocate either of these protocols 
> as a replacement for the WG as you have already consensus, but I'd 
> rather like to get feedback why some protocols were excluded.
>
> To be more specific, we're working on a research paper on deployment 
> analysis on Host Identity Protocol (HIP) in Aalto University and we'd 
> like get your expert opinion on especially why HIP was not the adopted 
> in the context of rtcweb. Here's few example considerations:
>
> - Never heard about HIP
> - The substitute technologies do their work better
> - Lack of implementation maturity or usability
> - Not standards track yet; we can build rtcweb out of standards track 
> components
> - The fatal flaw for HIP in this particular context is X
> - HIP is in the wrong layer of the stack?
>
> Besides technical incentives, we're also interested business and 
> political incentives. Feel free to respond me directly or via the 
> mailing list (unless people think it's too off-the-topic). Your 
> feedback will be anonymized in the publication.
>
> Thanks!
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


From pravindran@sonusnet.com  Thu Mar  8 04:15:43 2012
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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: Justin Uberti <juberti@google.com>, Cullen Jennings <fluffy@iii.ca>
Thread-Topic: [rtcweb] I-D Action: draft-ietf-rtcweb-jsep-00.txt
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Justin/Cullen,

I have noticed that "single offer and multiple answer" callflow is not upda=
ted in the latest draft as discussed in http://www.ietf.org/mail-archive/we=
b/rtcweb/current/msg03507.html, http://www.ietf.org/mail-archive/web/rtcweb=
/current/msg03531.html. Could you please add the callflow in the draft or p=
lease let me know any issues in the mentioned callflow.

Thanks
Partha

>-----Original Message-----
>From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
>Of internet-drafts@ietf.org
>Sent: Sunday, March 04, 2012 10:47 AM
>To: i-d-announce@ietf.org
>Cc: rtcweb@ietf.org
>Subject: [rtcweb] I-D Action: draft-ietf-rtcweb-jsep-00.txt
>
>
>A New Internet-Draft is available from the on-line Internet-Drafts
>directories. This draft is a work item of the Real-Time Communication in
>WEB-browsers Working Group of the IETF.
>
>	Title           : Javascript Session Establishment Protocol
>	Author(s)       : Justin Uberti
>                          Cullen Jennings
>	Filename        : draft-ietf-rtcweb-jsep-00.txt
>	Pages           : 27
>	Date            : 2012-03-03
>
>   This document proposes a mechanism for allowing a Javascript
>   application to fully control the signaling plane of a multimedia
>   session, and discusses how this would work with existing signaling
>   protocols.
>
>   This document is an input document for discussion.  It should be
>   discussed in the RTCWEB WG list, rtcweb@ietf.org.
>
>
>A URL for this Internet-Draft is:
>http://www.ietf.org/internet-drafts/draft-ietf-rtcweb-jsep-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-ietf-rtcweb-jsep-00.txt
>
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb

From magnus.westerlund@ericsson.com  Thu Mar  8 04:55:24 2012
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Subject: [rtcweb] Proposal for media receiver request of codec parameter changes (draft-westerlund-avtext-codec-operation-point-00)
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WG,

As an individual I would like to inform you that my colleagues and I
have submitted a proposal to AVTEXT WG. This proposal is an RTCP
extension that enables RTP session participants to request the media
sender to adjust its media parameters, like video resolution.

The draft can be retrieved here:
https://datatracker.ietf.org/doc/draft-westerlund-avtext-codec-operation-point/

One use case we have designed this for is to enable the dynamic updates
codec parameters as result of WebRTC applications usage of media
streams. If the application changes the rendering layout or the user
changes the browser window size, the size of a video playout can change.
This may happen in a quite dynamic way and often enough that we think
using a mechanism where the receiver can use a request over the media
plane rather than the signalling plane to adjust the parameters will
give a more responsive and satisfying experience. We have also designed
it with centralized conferencing into mind, allowing the central node to
aggregate requests to best suit the whole population.

We request that feedback on the actual extension are directed to the
AVTEXT WG mailing list: avtext@ietf.org

I am happy to discuss the usage and applicability to WebRTC on this list.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


From Jim.Barnett@genesyslab.com  Thu Mar  8 06:37:54 2012
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Trying to tie up some loose ends from earlier discussions:

1.       In the discussion of SDES, we stated that we don't trust the
web server.

2.       In a separate discussion, people  indicated that there were
serious privacy issues (disclosure of location) if ICE started before
the answerer gave consent.

3.       In the JSEP draft,  ICE is started under control of the
JavaScript

4.       which is served by the web server we don't trust.

=20

I may be confused, but it seems to me that 3-4 are in conflict with 1-2.
For example, in the example in Section 8 of the JSEP draft, the offer is
not sent until the user consents, but it looks to me like ICE can start
on the answerer side  before the user gives consent.   In previous
discussions, someone also observed that it also might be possible to set
up a data channel without user consent, since it doesn't require media.


=20

I'm not advocating a particular solution, but would like to hear how the
group intends to restore consistency.  It seems to me that there are 3
choices:

A.      decide we trust the web server after all

B.      decide that the security issues are not so serious after all

C.      set the default so that ICE does not start until the user
consents (this would amount to turning 'startICE' into
'startICEOnceUserConsents').  We might also treat the data channel as a
media channel requiring explicit consent, or say that data channels may
only be created once media channels are already in place. =20

=20

If we choose option C, we should probably run through some examples to
make sure that the potential delay in starting ICE doesn't cause
problems.=20

=20

-          Jim

=20

=20


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trust.<o:p></o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
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to me like ICE can start on the answerer side&nbsp; before the user =
gives consent. &nbsp;&nbsp;In previous discussions, someone also =
observed that it also might be possible to set up a data channel without =
user consent, since it doesn&#8217;t require media.&nbsp; =
<o:p></o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal>I&#8217;m not advocating a particular solution, but =
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</span></span><![endif]>set the default so that ICE does not start until =
the user consents (this would amount to turning &#8216;startICE&#8217; =
into &#8216;startICEOnceUserConsents&#8217;).&nbsp; We might also treat =
the data channel as a media channel requiring explicit consent, or say =
that data channels may only be created once media channels are already =
in place.&nbsp; <o:p></o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>If we choose =
option C, we should probably run through some examples to make sure that =
the potential delay in starting ICE doesn&#8217;t cause problems. =
<o:p></o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l1 level1 =
lfo4'><![if !supportLists]><span style=3D'mso-list:Ignore'>-<span =
style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
</span></span><![endif]>Jim<o:p></o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div></body></html>
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From stefan.lk.hakansson@ericsson.com  Thu Mar  8 07:13:00 2012
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On 03/06/2012 02:01 PM, Eric Rescorla wrote:
> Obviously, this is an interesting use case, but it's not clear to me that
> it's something you would want to address entirely via WebRTC
>
> 1. It seems likely that the lecturer will be using something
> rather more sophisticated than a Web browser in many cases
> (often these lecturers are actually joint on-line and in-person
> so you need microphone control in the room as well; see for
> instance the popular Sandel Justice lecturers out of Harvard).
>
> 2. You also probably want to actually support not just real-time
> viewing but also podcasting, time-shifting, etc., as well as
> support for people who don't have WebRTC.
>
> When you put these together, it might well be natural to have
> a one-way stream via Flash, HTML5 video, etc., and then individual
> WebRTC uplinks which get mixed into that single stream as necessary.
> That makes questions about transcoding, re-encryption, etc. a lot
> simpler.

I agree to that the use-case proposed was perhaps not that well selected 
for the reasons you give above.

What we were really after was to enable multi-party with really low cost 
for the central node (given that you're not doing a mesh). If you have 
to decrypt, alter RTP parameters, and then re-encrypt you quickly get 
into a situation where the central node needs a lot of processing power.

If the packets simply can be forwarded you're much better off - and this 
is what we wanted to capture. It is also a way to in the use-case and 
requirement document reflect a bit of what JSEP enables; if the 
application can *set* certain parts of the media settings used by the 
browser, the app can set it up such that central node would not have to 
alter anything - just forward. We thought it would be nice to have a 
JSEP related req in the document!

Stefan

>
> -Ekr
>
>
> On Tue, Mar 6, 2012 at 1:28 AM, Stefan Hakansson LK
> <stefan.lk.hakansson@ericsson.com>  wrote:
>> Hi!
>>
>> The Use-case document
>> (<http://datatracker.ietf.org/doc/draft-ietf-rtcweb-use-cases-and-requirements/?include_text=1>)
>> will expire in early April.
>>
>> The editors propose to add a new use-case when updating to -07. The new
>> use-case is about large sessions and is inspired by input made earlier
>> by Arno Bakker
>> (http://www.ietf.org/mail-archive/web/rtcweb/current/msg00530.html) and
>> recently by Harald Alvestrand
>> (http://lists.w3.org/Archives/Public/public-webrtc/2012Feb/0066.html).
>>
>> Proposed text for the new use-case:
>>
>> On-line lecture
>> ===============
>>
>> A lecture is given on-line by a very popular lecturer. The attendance is
>> very large. Because of this it is not really feasible to have
>> direct audiovisual feedback (for e.g. questions) for every participant,
>> instead participants can request the floor by clicking a button, and
>> gets a signal when audio and video has been set up (and the lecturer is
>> ready to take the question). The question is heard by all participants
>> (not only the lecturer).
>>
>> Participants can join and leave the lecture at any time (i.e. they do
>> not need to be there at the start to be able to connect).
>>
>> The service is operated by the university of the lecturer. It is set up such
>> that all participants connect to a central node. The university can not
>> invest much in HW, so the processing required (per participant) must be
>> minimized.
>>
>>
>> New requirement:
>>
>> - It must be possible to set up media streams and encryption in such a
>> way that processing in a central node is minimized (no transcoding
>> required, no decryption/re-encryption for every outgoing flow - just
>> simple forwarding).
>>
>>
>> Let us know what you think. If there is support we will add this use-case,
>> if not we will just make a new revision of the document (with no major
>> changes to the content).
>>
>> Stefan for the editors
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb


From harald@alvestrand.no  Thu Mar  8 07:32:19 2012
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Subject: Re: [rtcweb] when should ICE start?
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On 03/08/2012 03:37 PM, Jim Barnett wrote:
>
> Trying to tie up some loose ends from earlier discussions:
>
> 1.In the discussion of SDES, we stated that we don't trust the web server.
>
Well... it's more a question of what we trust the web server with.
>
> 2.In a separate discussion, people  indicated that there were serious 
> privacy issues (disclosure of location) if ICE started before the 
> answerer gave consent.
>
> 3.In the JSEP draft,  ICE is started under control of the JavaScript
>
> 4.which is served by the web server we don't trust.
>
The Javascript is served by a Web server. The Web server already knows 
where I am. So there's no point in trying to guard that information from 
one that already knows.
The one I'm guarding my location from is the person who's calling me, 
not the Web server that serves up my Javascript. In order to do that, I 
need the Web server (who knows where I am) to be on my side.
>
> I may be confused, but it seems to me that 3-4 are in conflict with 
> 1-2.  For example, in the example in Section 8 of the JSEP draft, the 
> offer is not sent until the user consents, but it looks to me like ICE 
> can start on the answerer side  before the user gives consent.   In 
> previous discussions, someone also observed that it also might be 
> possible to set up a data channel without user consent, since it 
> doesn't require media.
>
ICE using relay candidates does not disclose my location. So I can start 
*that* part of ICE.
>
> I'm not advocating a particular solution, but would like to hear how 
> the group intends to restore consistency.  It seems to me that there 
> are 3 choices:
>
> A.decide we trust the web server after all
>
> B.decide that the security issues are not so serious after all
>
> C.set the default so that ICE does not start until the user consents 
> (this would amount to turning 'startICE' into 
> 'startICEOnceUserConsents').  We might also treat the data channel as 
> a media channel requiring explicit consent, or say that data channels 
> may only be created once media channels are already in place.
>
> If we choose option C, we should probably run through some examples to 
> make sure that the potential delay in starting ICE doesn't cause 
> problems.
>
See above. I don't think those are the only options.

> -Jim
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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    On 03/08/2012 03:37 PM, Jim Barnett wrote:
    <blockquote
cite="mid:E17CAD772E76C742B645BD4DC602CD8105DFC884@NAHALD.us.int.genesyslab.com"
      type="cite">
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      <div class="WordSection1">
        <p class="MsoNormal">Trying to tie up some loose ends from
          earlier discussions:<o:p></o:p></p>
        <p class="MsoListParagraph" style="text-indent: -0.25in;"><!--[if !supportLists]--><span
            style="">1.<span style="font: 7pt &quot;Times New
              Roman&quot;;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span></span><!--[endif]-->In the
          discussion of SDES, we stated that we don&#8217;t trust the web
          server.</p>
      </div>
    </blockquote>
    Well... it's more a question of what we trust the web server with.<br>
    <blockquote
cite="mid:E17CAD772E76C742B645BD4DC602CD8105DFC884@NAHALD.us.int.genesyslab.com"
      type="cite">
      <div class="WordSection1">
        <p class="MsoListParagraph" style="text-indent: -0.25in;"><o:p></o:p></p>
        <p class="MsoListParagraph" style="text-indent: -0.25in;"><!--[if !supportLists]--><span
            style="">2.<span style="font: 7pt &quot;Times New
              Roman&quot;;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span></span><!--[endif]-->In a
          separate discussion, people&nbsp; indicated that there were serious
          privacy issues (disclosure of location) if ICE started before
          the answerer gave consent.<o:p></o:p></p>
        <p class="MsoListParagraph" style="text-indent: -0.25in;"><!--[if !supportLists]--><span
            style="">3.<span style="font: 7pt &quot;Times New
              Roman&quot;;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span></span><!--[endif]-->In the
          JSEP draft,&nbsp; ICE is started under control of the JavaScript<o:p></o:p></p>
        <p class="MsoListParagraph" style="text-indent: -0.25in;"><!--[if !supportLists]--><span
            style="">4.<span style="font: 7pt &quot;Times New
              Roman&quot;;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span></span><!--[endif]-->which is
          served by the web server we don&#8217;t trust.</p>
      </div>
    </blockquote>
    The Javascript is served by a Web server. The Web server already
    knows where I am. So there's no point in trying to guard that
    information from one that already knows.<br>
    The one I'm guarding my location from is the person who's calling
    me, not the Web server that serves up my Javascript. In order to do
    that, I need the Web server (who knows where I am) to be on my side.<br>
    <blockquote
cite="mid:E17CAD772E76C742B645BD4DC602CD8105DFC884@NAHALD.us.int.genesyslab.com"
      type="cite">
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        <p class="MsoListParagraph" style="text-indent: -0.25in;"><o:p></o:p></p>
        <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
        <p class="MsoNormal">I may be confused, but it seems to me that
          3-4 are in conflict with 1-2.&nbsp; For example, in the example in
          Section 8 of the JSEP draft, the offer is not sent until the
          user consents, but it looks to me like ICE can start on the
          answerer side&nbsp; before the user gives consent. &nbsp;&nbsp;In previous
          discussions, someone also observed that it also might be
          possible to set up a data channel without user consent, since
          it doesn&#8217;t require media.&nbsp; </p>
      </div>
    </blockquote>
    ICE using relay candidates does not disclose my location. So I can
    start *that* part of ICE.<br>
    <blockquote
cite="mid:E17CAD772E76C742B645BD4DC602CD8105DFC884@NAHALD.us.int.genesyslab.com"
      type="cite">
      <div class="WordSection1">
        <p class="MsoNormal"><o:p></o:p></p>
        <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
        <p class="MsoNormal">I&#8217;m not advocating a particular solution,
          but would like to hear how the group intends to restore
          consistency.&nbsp; It seems to me that there are 3 choices:<o:p></o:p></p>
        <p class="MsoListParagraph" style="text-indent: -0.25in;"><!--[if !supportLists]--><span
            style="">A.<span style="font: 7pt &quot;Times New
              Roman&quot;;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span></span><!--[endif]-->decide we
          trust the web server after all<o:p></o:p></p>
        <p class="MsoListParagraph" style="text-indent: -0.25in;"><!--[if !supportLists]--><span
            style="">B.<span style="font: 7pt &quot;Times New
              Roman&quot;;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span></span><!--[endif]-->decide
          that the security issues are not so serious after all<o:p></o:p></p>
        <p class="MsoListParagraph" style="text-indent: -0.25in;"><!--[if !supportLists]--><span
            style="">C.<span style="font: 7pt &quot;Times New
              Roman&quot;;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span></span><!--[endif]-->set the
          default so that ICE does not start until the user consents
          (this would amount to turning &#8216;startICE&#8217; into
          &#8216;startICEOnceUserConsents&#8217;).&nbsp; We might also treat the data
          channel as a media channel requiring explicit consent, or say
          that data channels may only be created once media channels are
          already in place.&nbsp; <o:p></o:p></p>
        <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
        <p class="MsoNormal">If we choose option C, we should probably
          run through some examples to make sure that the potential
          delay in starting ICE doesn&#8217;t cause problems. </p>
      </div>
    </blockquote>
    See above. I don't think those are the only options.<br>
    <br>
    <blockquote
cite="mid:E17CAD772E76C742B645BD4DC602CD8105DFC884@NAHALD.us.int.genesyslab.com"
      type="cite">
      <div class="WordSection1">
        <p class="MsoNormal"><o:p></o:p></p>
        <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
        <p class="MsoListParagraph" style="text-indent: -0.25in;"><!--[if !supportLists]--><span
            style="">-<span style="font: 7pt &quot;Times New
              Roman&quot;;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span></span><!--[endif]-->Jim<o:p></o:p></p>
        <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
        <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
      </div>
      <pre wrap="">
<fieldset class="mimeAttachmentHeader"></fieldset>
_______________________________________________
rtcweb mailing list
<a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
    </blockquote>
    <br>
  </body>
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From Jim.Barnett@genesyslab.com  Thu Mar  8 07:39:13 2012
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Harald,

You're right that the web server already knows where I am.  I'm worried
about it serving me malicious/careless JS that reveals my location to
caller (Syrian secret police, etc.) before I'm aware that I'm being
called.  That's the sense in which I don't trust it.  However you are
right that there are more options than I outlined, and starting ICE
using only relay candidates is one of them.

=20

-          Jim

=20

From: Harald Alvestrand [mailto:harald@alvestrand.no]=20
Sent: Thursday, March 08, 2012 10:32 AM
To: Jim Barnett
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] when should ICE start?

=20

On 03/08/2012 03:37 PM, Jim Barnett wrote:=20

Trying to tie up some loose ends from earlier discussions:

In the discussion of SDES, we stated that we don't trust the web server.

Well... it's more a question of what we trust the web server with.



In a separate discussion, people  indicated that there were serious
privacy issues (disclosure of location) if ICE started before the
answerer gave consent.

In the JSEP draft,  ICE is started under control of the JavaScript

which is served by the web server we don't trust.

The Javascript is served by a Web server. The Web server already knows
where I am. So there's no point in trying to guard that information from
one that already knows.
The one I'm guarding my location from is the person who's calling me,
not the Web server that serves up my Javascript. In order to do that, I
need the Web server (who knows where I am) to be on my side.



=20

I may be confused, but it seems to me that 3-4 are in conflict with 1-2.
For example, in the example in Section 8 of the JSEP draft, the offer is
not sent until the user consents, but it looks to me like ICE can start
on the answerer side  before the user gives consent.   In previous
discussions, someone also observed that it also might be possible to set
up a data channel without user consent, since it doesn't require media.


ICE using relay candidates does not disclose my location. So I can start
*that* part of ICE.



=20

I'm not advocating a particular solution, but would like to hear how the
group intends to restore consistency.  It seems to me that there are 3
choices:

decide we trust the web server after all

decide that the security issues are not so serious after all

set the default so that ICE does not start until the user consents (this
would amount to turning 'startICE' into 'startICEOnceUserConsents').  We
might also treat the data channel as a media channel requiring explicit
consent, or say that data channels may only be created once media
channels are already in place. =20

=20

If we choose option C, we should probably run through some examples to
make sure that the potential delay in starting ICE doesn't cause
problems.=20

See above. I don't think those are the only options.




=20

Jim

=20

=20

=20
=20
_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

=20


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<o:idmap v:ext=3D"edit" data=3D"1" />
</o:shapelayout></xml><![endif]--></head><body bgcolor=3Dwhite =
lang=3DEN-US link=3Dblue vlink=3Dpurple><div class=3DWordSection1><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'>Harald,<o:p></o:p></span></p><p =
class=3DMsoNormal style=3D'text-indent:4.5pt'><span =
style=3D'color:#1F497D'>You&#8217;re right that the web server already =
knows where I am. &nbsp;I&#8217;m worried about it serving me =
malicious/careless JS that reveals my location to caller (Syrian secret =
police, etc.) before I&#8217;m aware that I&#8217;m being called.&nbsp; =
That&#8217;s the sense in which I don&#8217;t trust it.&nbsp; However =
you are right that there are more options than I outlined, and starting =
ICE using only relay candidates is one of them.<o:p></o:p></span></p><p =
class=3DMsoNormal style=3D'text-indent:4.5pt'><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoListParagraph =
style=3D'margin-left:22.5pt;text-indent:-.25in;mso-list:l0 level1 =
lfo1'><![if !supportLists]><span style=3D'color:#1F497D'><span =
style=3D'mso-list:Ignore'>-<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
</span></span></span><![endif]><span =
style=3D'color:#1F497D'>Jim<o:p></o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><div><div =
style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in'><p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowt=
ext'>From:</span></b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowt=
ext'> Harald Alvestrand [mailto:harald@alvestrand.no] <br><b>Sent:</b> =
Thursday, March 08, 2012 10:32 AM<br><b>To:</b> Jim =
Barnett<br><b>Cc:</b> rtcweb@ietf.org<br><b>Subject:</b> Re: [rtcweb] =
when should ICE start?<o:p></o:p></span></p></div></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>On =
03/08/2012 03:37 PM, Jim Barnett wrote: <o:p></o:p></p><p =
class=3DMsoNormal>Trying to tie up some loose ends from earlier =
discussions:<o:p></o:p></p><p class=3DMsoListParagraph =
style=3D'text-indent:-.25in'>In the discussion of SDES, we stated that =
we don&#8217;t trust the web server.<o:p></o:p></p><p =
class=3DMsoNormal><span style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif"'>Well... it's more a question of what we trust the web =
server with.<br><br><o:p></o:p></span></p><p class=3DMsoListParagraph =
style=3D'text-indent:-.25in'>In a separate discussion, people&nbsp; =
indicated that there were serious privacy issues (disclosure of =
location) if ICE started before the answerer gave =
consent.<o:p></o:p></p><p class=3DMsoListParagraph =
style=3D'text-indent:-.25in'>In the JSEP draft,&nbsp; ICE is started =
under control of the JavaScript<o:p></o:p></p><p =
class=3DMsoListParagraph style=3D'text-indent:-.25in'>which is served by =
the web server we don&#8217;t trust.<o:p></o:p></p><p =
class=3DMsoNormal><span style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif"'>The Javascript is served by a Web server. The Web server =
already knows where I am. So there's no point in trying to guard that =
information from one that already knows.<br>The one I'm guarding my =
location from is the person who's calling me, not the Web server that =
serves up my Javascript. In order to do that, I need the Web server (who =
knows where I am) to be on my side.<br><br><o:p></o:p></span></p><p =
class=3DMsoNormal>&nbsp;<o:p></o:p></p><p class=3DMsoNormal>I may be =
confused, but it seems to me that 3-4 are in conflict with 1-2.&nbsp; =
For example, in the example in Section 8 of the JSEP draft, the offer is =
not sent until the user consents, but it looks to me like ICE can start =
on the answerer side&nbsp; before the user gives consent. &nbsp;&nbsp;In =
previous discussions, someone also observed that it also might be =
possible to set up a data channel without user consent, since it =
doesn&#8217;t require media.&nbsp; <o:p></o:p></p><p =
class=3DMsoNormal><span style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif"'>ICE using relay candidates does not disclose my =
location. So I can start *that* part of =
ICE.<br><br><o:p></o:p></span></p><p =
class=3DMsoNormal>&nbsp;<o:p></o:p></p><p class=3DMsoNormal>I&#8217;m =
not advocating a particular solution, but would like to hear how the =
group intends to restore consistency.&nbsp; It seems to me that there =
are 3 choices:<o:p></o:p></p><p class=3DMsoListParagraph =
style=3D'text-indent:-.25in'>decide we trust the web server after =
all<o:p></o:p></p><p class=3DMsoListParagraph =
style=3D'text-indent:-.25in'>decide that the security issues are not so =
serious after all<o:p></o:p></p><p class=3DMsoListParagraph =
style=3D'text-indent:-.25in'>set the default so that ICE does not start =
until the user consents (this would amount to turning =
&#8216;startICE&#8217; into =
&#8216;startICEOnceUserConsents&#8217;).&nbsp; We might also treat the =
data channel as a media channel requiring explicit consent, or say that =
data channels may only be created once media channels are already in =
place.&nbsp; <o:p></o:p></p><p class=3DMsoNormal>&nbsp;<o:p></o:p></p><p =
class=3DMsoNormal>If we choose option C, we should probably run through =
some examples to make sure that the potential delay in starting ICE =
doesn&#8217;t cause problems. <o:p></o:p></p><p class=3DMsoNormal><span =
style=3D'font-size:12.0pt;font-family:"Times New Roman","serif"'>See =
above. I don't think those are the only =
options.<br><br><br><o:p></o:p></span></p><p =
class=3DMsoNormal>&nbsp;<o:p></o:p></p><p class=3DMsoListParagraph =
style=3D'text-indent:-.25in'>Jim<o:p></o:p></p><p =
class=3DMsoNormal>&nbsp;<o:p></o:p></p><p =
class=3DMsoNormal>&nbsp;<o:p></o:p></p><pre><o:p>&nbsp;</o:p></pre><pre><=
o:p>&nbsp;</o:p></pre><pre>______________________________________________=
_<o:p></o:p></pre><pre>rtcweb mailing list<o:p></o:p></pre><pre><a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><o:p></o:p></pre><pre>=
<a =
href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.or=
g/mailman/listinfo/rtcweb</a><o:p></o:p></pre><p class=3DMsoNormal><span =
style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif"'><o:p>&nbsp;</o:p></span></p></div></body></html>
------_=_NextPart_001_01CCFD41.96C5465E--

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Subject: Re: [rtcweb] when should ICE start?
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On Thu, Mar 8, 2012 at 7:38 AM, Jim Barnett <Jim.Barnett@genesyslab.com> wr=
ote:
> Harald,
>
> You=92re right that the web server already knows where I am. =A0I=92m wor=
ried
> about it serving me malicious/careless JS that reveals my location to cal=
ler
> (Syrian secret police, etc.) before I=92m aware that I=92m being called.=
=A0 That=92s
> the sense in which I don=92t trust it.

Right but in the most common case, as soon as you connect to the
server, the server knows your IP, so there's nothing we can do about
this.

-Ekr

From Jim.Barnett@genesyslab.com  Thu Mar  8 08:01:47 2012
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From: "Jim Barnett" <Jim.Barnett@genesyslab.com>
To: "Eric Rescorla" <ekr@rtfm.com>
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Subject: Re: [rtcweb] when should ICE start?
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Yes, certainly if the Syrian secret police controls the server, I'm in =
trouble no matter what.  But isn't there a separate case where a =
third-party server, not under police control, serves up careless JS that =
reveals my location to the caller before I'm aware I'm being called?  =
This was the case we were worried about in the earlier discussion - that =
the caller could discover the answerer's  location even if the answerer =
rejected the call.  (I think that the example in Section 8 will permit =
this.)  So the trust question is:  do we let the server decide whether =
to allow this, or do we put defaults in the browser to protect location =
information even if the server is carelessly coded?

- Jim=20

-----Original Message-----
From: Eric Rescorla [mailto:ekr@rtfm.com]=20
Sent: Thursday, March 08, 2012 10:54 AM
To: Jim Barnett
Cc: Harald Alvestrand; rtcweb@ietf.org
Subject: Re: [rtcweb] when should ICE start?

On Thu, Mar 8, 2012 at 7:38 AM, Jim Barnett <Jim.Barnett@genesyslab.com> =
wrote:
> Harald,
>
> You're right that the web server already knows where I am. =A0I'm=20
> worried about it serving me malicious/careless JS that reveals my=20
> location to caller (Syrian secret police, etc.) before I'm aware that=20
> I'm being called.=A0 That's the sense in which I don't trust it.

Right but in the most common case, as soon as you connect to the server, =
the server knows your IP, so there's nothing we can do about this.

-Ekr

From ted.ietf@gmail.com  Thu Mar  8 08:27:48 2012
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Subject: Re: [rtcweb] I'd like to discuss my proposal (jingle-web) at next WG session
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Hi Suzuki-san,

Given that this apparently revisits some decisions that were already
made, the chairs would like to see working group discussion of the
draft before allocating agenda time to it for the upcoming meeting.

best regards,

Ted Hardie for the Chairs

On Wed, Feb 29, 2012 at 7:34 PM, HIro Suzuki
<hiro.suzuki@d3communications.jp> wrote:
> Hello RTCWeb WG members,
>
>
> It's my first post to this ML, and at Feb 28th, I submitted my ID
> from IETF ID submit tool. This ID's file name is
> "draft-suzuki-rtcweb-jingle-web-00.txt" and this ID provides my proposal
> to realize one of solutions of RTCWeb by using XMPP Jingle as a signaling
> protocol.
>
> Following paragraph is abstract of my ID, and I'd like to discuss my
> proposal at RTCWeb WG session in next Paris meeting.
>
>
> ---
> Abstract:
> =A0XMPP Jingle specification defines an XMPP protocol extension for
> =A0managing peer-to-peer media sessions between two users. And XMPP
> =A0Jingle can manage multi contents simultaneously in one Jingle
> =A0stream, but in the XMPP world there is no common way to layout
> =A0variable multi contents on each display. In this document, a
> =A0solution to this issue is provided by using Web browser's layout
> =A0function.
>
> =A0This document describes a new concept to realize one of solutions
> =A0of RTCWeb. Of course, &quot;Web Browser&quot; is used for Web applicat=
ion's
> =A0frontend for real-time communication, and XMPP Jingle specification
> =A0(XEP-166) is used as signaling protocol. And a simple mapping
> =A0manner between jingle contents and HTML graphical elements enables
> =A0to unite Web browser&#39;s layout function and Jingle&#39;s media cont=
ent
> =A0management function to realize RTCWeb functions.
> ---
>
>
> Best Regards,
> Yoshihiro Suzuki
> (hiro.suzuki@d3communications.jp)
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From ekr@rtfm.com  Thu Mar  8 08:30:25 2012
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Subject: Re: [rtcweb] when should ICE start?
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On Thu, Mar 8, 2012 at 8:01 AM, Jim Barnett <Jim.Barnett@genesyslab.com> wr=
ote:
> Yes, certainly if the Syrian secret police controls the server, I'm in tr=
ouble no matter what. =A0But isn't there a separate case where a third-part=
y server, not under police control, serves up careless JS that reveals my l=
ocation to the caller before I'm aware I'm being called? =A0This was the ca=
se we were worried about in the earlier discussion - that the caller could =
discover the answerer's =A0location even if the answerer rejected the call.=
 =A0(I think that the example in Section 8 will permit this.) =A0So the tru=
st question is: =A0do we let the server decide whether to allow this, or do=
 we put defaults in the browser to protect location information even if the=
 server is carelessly coded?
>

I'm not sure that ICE and WebRTC create much of a new risk here. If the ser=
ver
is badly coded then there are a lot of ways to leak the user's location. To=
 give
one, say I use targeted ad buys to get the server to show you JS that inclu=
des
(a) cookies and (b) calls back to my server to reveal location information.

-Ekr

From mkomu@cs.hut.fi  Thu Mar  8 09:08:24 2012
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From: Miika Komu <mkomu@cs.hut.fi>
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Subject: Re: [rtcweb] host identity protocol
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Hi Harald,

thanks for the quick response. You're absolutely right, HIP does not 
support congestion control. It actually even shouldn't support it 
because it's not a transport-layer protocol. Typically people use 
HIP/IPsec for end-to-end tunneling of TCP or UDP. The tunnel is 
encrusted in an extra UDP header to make HIP/ESP pass NAT boxes which 
actually makes it a nice fit for NAT penetration. The data-plane can be 
negotiated during the key exchange process and e.g. S-RTP has been 
proposed to replace IPsec.

Are you sure about the reasoning - has it been a misunderstanding or 
maybe some other reasoning fits better? Thanks for your feedback!

On 03/08/2012 12:16 PM, Harald Alvestrand wrote:
> HIP was discussed at a far earlier stage, when talking about security.
> We chose to go with DTLS, SRTP and ICE instead, since these were
> mechanisms in (relatively) common use in this environment.
>
> As far as I understand HIP, it does not contain congestion control
> mechanisms, so its use was not considered when picking a protocol for
> the data channel.
>
> On 03/08/2012 09:51 AM, Miika Komu wrote:
>> Dear WG,
>>
>> I heard that you're making progress and have selected a specific
>> protocol approach for rtcweb:
>>
>> http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel
>>
>> This is good news indeed. However, we noticed that the original
>> comparison was missing few approaches like Teredo (RFC4380) or Host
>> Identity Protocol (see e.g. RFC5770):
>>
>> http://tools.ietf.org/html/draft-jesup-rtcweb-data
>>
>> Despite the original document missed the two protocols, I'm sure that
>> they were was discussed at some point, maybe even just in face-to-face
>> conversations. Now, my goal is not advocate either of these protocols
>> as a replacement for the WG as you have already consensus, but I'd
>> rather like to get feedback why some protocols were excluded.
>>
>> To be more specific, we're working on a research paper on deployment
>> analysis on Host Identity Protocol (HIP) in Aalto University and we'd
>> like get your expert opinion on especially why HIP was not the adopted
>> in the context of rtcweb. Here's few example considerations:
>>
>> - Never heard about HIP
>> - The substitute technologies do their work better
>> - Lack of implementation maturity or usability
>> - Not standards track yet; we can build rtcweb out of standards track
>> components
>> - The fatal flaw for HIP in this particular context is X
>> - HIP is in the wrong layer of the stack?
>>
>> Besides technical incentives, we're also interested business and
>> political incentives. Feel free to respond me directly or via the
>> mailing list (unless people think it's too off-the-topic). Your
>> feedback will be anonymized in the publication.
>>
>> Thanks!
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>


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Subject: [rtcweb] Draft Agenda for RTCWeb at IETF83 v2
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Here is an updated draft agenda. Please let me know what internet drafts =
you think are mandatory reading material for each section.=20


Draft Agenda for RTCWeb at IETF83=20
(Version 2 of agenda done March 8)


First Session   -------------------

Admin Update (Chairs)  - 10 min=20
- update from mmusic, avtcore, ICCRG and other WG=20

W3C Update (Harald) - 10 min

Identity Proxy (Eric Rescorla)  - 50 min=20

What to do about RTP and/or SDES (Dan Wing) - 80 min=20



Second Session  -----------------------

Admin (Chairs) - 10 min=20
- scheduling the next interim meeting=20

Signaling (Justin Uberti) - 60 min =20
draft-ietf-rtcweb-jsep

Data Channel (Salvatore Loreto) - 30  min
draft-ietf-rtcweb-data-channel

Codec Selection (Adam Roach) - 50 min




There are some things that we did not put onto the agenda that I would =
to comment on.=20

The following items have time in other meetings and the chairs will =
provide summary information on what happened

- SDP updates: msid and bundle - our use of those functions being =
defined in MMUSIC
- Use of multiplexing functions described by AVTCORE
- Congestion control as discussed in the ICCRG meeting

The following items are clearly areas where we need more work but, given =
the time limits of the meeting and the list discussion to date, the =
chairs hope to make these to be big topics at our next interim meeting.=20=


- RTP feature profiling: What do we use? (mandatory  / permitted / =
recommended-against)
- SDP feature profiling: What do we use? (same alternatives)

There is also work that needs discussion but overlaps into the work =
being done in  MMUSIC and other place including:

- Video resolution negotiation per SSRC: What are the alternatives? Can =
we recommend one?
- Mute/hold signaling (per SSRC)
- How to convey label/id of a MediaStreamTrack over a PeerConnection

The final topic which we believe can simply progress with mailing list =
discussion is=20

- defining URIs for STUN and TURN servers=20


Cullen on behalf of the chairs








From harald@alvestrand.no  Thu Mar  8 10:07:43 2012
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On 03/08/2012 06:08 PM, Miika Komu wrote:
> Hi Harald,
>
> thanks for the quick response. You're absolutely right, HIP does not 
> support congestion control. It actually even shouldn't support it 
> because it's not a transport-layer protocol. Typically people use 
> HIP/IPsec for end-to-end tunneling of TCP or UDP. The tunnel is 
> encrusted in an extra UDP header to make HIP/ESP pass NAT boxes which 
> actually makes it a nice fit for NAT penetration. The data-plane can 
> be negotiated during the key exchange process and e.g. S-RTP has been 
> proposed to replace IPsec.
>
> Are you sure about the reasoning - has it been a misunderstanding or 
> maybe some other reasoning fits better? Thanks for your feedback!
The last discussion was in september/october 2011. Please review the 
archives.

>
> On 03/08/2012 12:16 PM, Harald Alvestrand wrote:
>> HIP was discussed at a far earlier stage, when talking about security.
>> We chose to go with DTLS, SRTP and ICE instead, since these were
>> mechanisms in (relatively) common use in this environment.
>>
>> As far as I understand HIP, it does not contain congestion control
>> mechanisms, so its use was not considered when picking a protocol for
>> the data channel.
>>
>> On 03/08/2012 09:51 AM, Miika Komu wrote:
>>> Dear WG,
>>>
>>> I heard that you're making progress and have selected a specific
>>> protocol approach for rtcweb:
>>>
>>> http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel
>>>
>>> This is good news indeed. However, we noticed that the original
>>> comparison was missing few approaches like Teredo (RFC4380) or Host
>>> Identity Protocol (see e.g. RFC5770):
>>>
>>> http://tools.ietf.org/html/draft-jesup-rtcweb-data
>>>
>>> Despite the original document missed the two protocols, I'm sure that
>>> they were was discussed at some point, maybe even just in face-to-face
>>> conversations. Now, my goal is not advocate either of these protocols
>>> as a replacement for the WG as you have already consensus, but I'd
>>> rather like to get feedback why some protocols were excluded.
>>>
>>> To be more specific, we're working on a research paper on deployment
>>> analysis on Host Identity Protocol (HIP) in Aalto University and we'd
>>> like get your expert opinion on especially why HIP was not the adopted
>>> in the context of rtcweb. Here's few example considerations:
>>>
>>> - Never heard about HIP
>>> - The substitute technologies do their work better
>>> - Lack of implementation maturity or usability
>>> - Not standards track yet; we can build rtcweb out of standards track
>>> components
>>> - The fatal flaw for HIP in this particular context is X
>>> - HIP is in the wrong layer of the stack?
>>>
>>> Besides technical incentives, we're also interested business and
>>> political incentives. Feel free to respond me directly or via the
>>> mailing list (unless people think it's too off-the-topic). Your
>>> feedback will be anonymized in the publication.
>>>
>>> Thanks!
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>
>>
>
>


From mkomu@cs.hut.fi  Thu Mar  8 10:50:53 2012
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From: Miika Komu <mkomu@cs.hut.fi>
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To: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] host identity protocol
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Hi Harald,

On 03/08/2012 08:07 PM, Harald Alvestrand wrote:
> On 03/08/2012 06:08 PM, Miika Komu wrote:
>> Hi Harald,
>>
>> thanks for the quick response. You're absolutely right, HIP does not
>> support congestion control. It actually even shouldn't support it
>> because it's not a transport-layer protocol. Typically people use
>> HIP/IPsec for end-to-end tunneling of TCP or UDP. The tunnel is
>> encrusted in an extra UDP header to make HIP/ESP pass NAT boxes which
>> actually makes it a nice fit for NAT penetration. The data-plane can
>> be negotiated during the key exchange process and e.g. S-RTP has been
>> proposed to replace IPsec.
>>
>> Are you sure about the reasoning - has it been a misunderstanding or
>> maybe some other reasoning fits better? Thanks for your feedback!
> The last discussion was in september/october 2011. Please review the
> archives.

I went through the archives and it appears that there was no technical 
reason for excluding HIP:

http://www.ietf.org/mail-archive/web/rtcweb/current/msg01679.html

So, it just appears that there was only one person suggesting HIP, i.e., 
the reason was the lack of contributors. Technically, it appears that 
the WG decision was based on an incomplete analysis of the solution 
space though (but, that's life).

Nevertheless, I'm sure that you'll find inventive uses for SCTP 
associations in the context web - I wish you the very best luck in your 
endeavors!

>> On 03/08/2012 12:16 PM, Harald Alvestrand wrote:
>>> HIP was discussed at a far earlier stage, when talking about security.
>>> We chose to go with DTLS, SRTP and ICE instead, since these were
>>> mechanisms in (relatively) common use in this environment.
>>>
>>> As far as I understand HIP, it does not contain congestion control
>>> mechanisms, so its use was not considered when picking a protocol for
>>> the data channel.
>>>
>>> On 03/08/2012 09:51 AM, Miika Komu wrote:
>>>> Dear WG,
>>>>
>>>> I heard that you're making progress and have selected a specific
>>>> protocol approach for rtcweb:
>>>>
>>>> http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel
>>>>
>>>> This is good news indeed. However, we noticed that the original
>>>> comparison was missing few approaches like Teredo (RFC4380) or Host
>>>> Identity Protocol (see e.g. RFC5770):
>>>>
>>>> http://tools.ietf.org/html/draft-jesup-rtcweb-data
>>>>
>>>> Despite the original document missed the two protocols, I'm sure that
>>>> they were was discussed at some point, maybe even just in face-to-face
>>>> conversations. Now, my goal is not advocate either of these protocols
>>>> as a replacement for the WG as you have already consensus, but I'd
>>>> rather like to get feedback why some protocols were excluded.
>>>>
>>>> To be more specific, we're working on a research paper on deployment
>>>> analysis on Host Identity Protocol (HIP) in Aalto University and we'd
>>>> like get your expert opinion on especially why HIP was not the adopted
>>>> in the context of rtcweb. Here's few example considerations:
>>>>
>>>> - Never heard about HIP
>>>> - The substitute technologies do their work better
>>>> - Lack of implementation maturity or usability
>>>> - Not standards track yet; we can build rtcweb out of standards track
>>>> components
>>>> - The fatal flaw for HIP in this particular context is X
>>>> - HIP is in the wrong layer of the stack?
>>>>
>>>> Besides technical incentives, we're also interested business and
>>>> political incentives. Feel free to respond me directly or via the
>>>> mailing list (unless people think it's too off-the-topic). Your
>>>> feedback will be anonymized in the publication.
>>>>
>>>> Thanks!
>>>> _______________________________________________
>>>> rtcweb mailing list
>>>> rtcweb@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>>
>>>
>>
>>
>


From Jim.Barnett@genesyslab.com  Thu Mar  8 10:53:35 2012
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Thread-Topic: [rtcweb] when should ICE start?
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References: <E17CAD772E76C742B645BD4DC602CD8105DFC884@NAHALD.us.int.genesyslab.com> <4F58D0FE.9060909@alvestrand.no> <E17CAD772E76C742B645BD4DC602CD8105DFC8A6@NAHALD.us.int.genesyslab.com> <CABcZeBPB65cg6ZdC8rmLAD7QgdhBo+52QQ158s-CYYC5V9WTdA@mail.gmail.com> <E17CAD772E76C742B645BD4DC602CD8105DFC8AF@NAHALD.us.int.genesyslab.com> <CABcZeBNpFuGOoFG0jjgvc_08bJFhRFN8hgzBHwHu=fYmEw1ENw@mail.gmail.com>
From: "Jim Barnett" <Jim.Barnett@genesyslab.com>
To: "Eric Rescorla" <ekr@rtfm.com>
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] when should ICE start?
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Eric,
  I guess this amounts to my original option B - decide that the problem =
isn't as serious as we thought.  That's not what it sounded like during =
the first discussion of the issue, but I'm fine with it others agree.

- Jim

-----Original Message-----
From: Eric Rescorla [mailto:ekr@rtfm.com]=20
Sent: Thursday, March 08, 2012 11:30 AM
To: Jim Barnett
Cc: Harald Alvestrand; rtcweb@ietf.org
Subject: Re: [rtcweb] when should ICE start?

On Thu, Mar 8, 2012 at 8:01 AM, Jim Barnett <Jim.Barnett@genesyslab.com> =
wrote:
> Yes, certainly if the Syrian secret police controls the server, I'm in =
trouble no matter what. =A0But isn't there a separate case where a =
third-party server, not under police control, serves up careless JS that =
reveals my location to the caller before I'm aware I'm being called? =
=A0This was the case we were worried about in the earlier discussion - =
that the caller could discover the answerer's =A0location even if the =
answerer rejected the call. =A0(I think that the example in Section 8 =
will permit this.) =A0So the trust question is: =A0do we let the server =
decide whether to allow this, or do we put defaults in the browser to =
protect location information even if the server is carelessly coded?
>

I'm not sure that ICE and WebRTC create much of a new risk here. If the =
server is badly coded then there are a lot of ways to leak the user's =
location. To give one, say I use targeted ad buys to get the server to =
show you JS that includes
(a) cookies and (b) calls back to my server to reveal location =
information.

-Ekr

From harald@alvestrand.no  Thu Mar  8 11:43:30 2012
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On 03/08/2012 07:53 PM, Jim Barnett wrote:
> Eric,
>    I guess this amounts to my original option B - decide that the problem isn't as serious as we thought.  That's not what it sounded like during the first discussion of the issue, but I'm fine with it others agree.
It's strongly application dependent. Fedex' answering machine, or a call 
center switchboard, will take great pain to reply as quickly as 
possibly, and has no expectation of privacy; a subscriber to "anonymous 
flirting page" might have completely different expectations.

When something is application dependent, I don't see that it's possible 
to make rules that allow "only one way to do it".


> - Jim
>
> -----Original Message-----
> From: Eric Rescorla [mailto:ekr@rtfm.com]
> Sent: Thursday, March 08, 2012 11:30 AM
> To: Jim Barnett
> Cc: Harald Alvestrand; rtcweb@ietf.org
> Subject: Re: [rtcweb] when should ICE start?
>
> On Thu, Mar 8, 2012 at 8:01 AM, Jim Barnett<Jim.Barnett@genesyslab.com>  wrote:
>> Yes, certainly if the Syrian secret police controls the server, I'm in trouble no matter what.  But isn't there a separate case where a third-party server, not under police control, serves up careless JS that reveals my location to the caller before I'm aware I'm being called?  This was the case we were worried about in the earlier discussion - that the caller could discover the answerer's  location even if the answerer rejected the call.  (I think that the example in Section 8 will permit this.)  So the trust question is:  do we let the server decide whether to allow this, or do we put defaults in the browser to protect location information even if the server is carelessly coded?
>>
> I'm not sure that ICE and WebRTC create much of a new risk here. If the server is badly coded then there are a lot of ways to leak the user's location. To give one, say I use targeted ad buys to get the server to show you JS that includes
> (a) cookies and (b) calls back to my server to reveal location information.
>
> -Ekr
>


From igor.faynberg@alcatel-lucent.com  Thu Mar  8 12:35:06 2012
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Subject: Re: [rtcweb] when should ICE start?
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Just following up on your original question: Can we " put defaults in 
the browser to protect location information"?

Igor


On 3/8/2012 1:53 PM, Jim Barnett wrote:
> Eric,
>    I guess this amounts to my original option B - decide that the problem isn't as serious as we thought.  That's not what it sounded like during the first discussion of the issue, but I'm fine with it others agree.
>
> - Jim
>
> -----Original Message-----
> From: Eric Rescorla [mailto:ekr@rtfm.com]
> Sent: Thursday, March 08, 2012 11:30 AM
> To: Jim Barnett
> Cc: Harald Alvestrand; rtcweb@ietf.org
> Subject: Re: [rtcweb] when should ICE start?
>
> On Thu, Mar 8, 2012 at 8:01 AM, Jim Barnett<Jim.Barnett@genesyslab.com>  wrote:
>> Yes, certainly if the Syrian secret police controls the server, I'm in trouble no matter what.  But isn't there a separate case where a third-party server, not under police control, serves up careless JS that reveals my location to the caller before I'm aware I'm being called?  This was the case we were worried about in the earlier discussion - that the caller could discover the answerer's  location even if the answerer rejected the call.  (I think that the example in Section 8 will permit this.)  So the trust question is:  do we let the server decide whether to allow this, or do we put defaults in the browser to protect location information even if the server is carelessly coded?
>>
> I'm not sure that ICE and WebRTC create much of a new risk here. If the server is badly coded then there are a lot of ways to leak the user's location. To give one, say I use targeted ad buys to get the server to show you JS that includes
> (a) cookies and (b) calls back to my server to reveal location information.
>
> -Ekr
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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From: Bernard Aboba <bernard_aboba@hotmail.com>
To: <igor.faynberg@alcatel-lucent.com>, <rtcweb@ietf.org>
Date: Thu, 8 Mar 2012 14:37:58 -0800
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References: <E17CAD772E76C742B645BD4DC602CD8105DFC884@NAHALD.us.int.genesyslab.com> <4F58D0FE.9060909@alvestrand.no> <E17CAD772E76C742B645BD4DC602CD8105DFC8A6@NAHALD.us.int.genesyslab.com> <CABcZeBPB65cg6ZdC8rmLAD7QgdhBo+52QQ158s-CYYC5V9WTdA@mail.gmail.com> <E17CAD772E76C742B645BD4DC602CD8105DFC8AF@NAHALD.us.int.genesyslab.com> <CABcZeBNpFuGOoFG0jjgvc_08bJFhRFN8hgzBHwHu=fYmEw1ENw@mail.gmail.com>, <E17CAD772E76C742B645BD4DC602CD8105DFC96E@NAHALD.us.int.genesyslab.com>, <4F5917F7.8090601@alcatel-lucent.com>
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Subject: Re: [rtcweb] Location/Privacy protection
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Protection of location (and privacy in general) depends on a number of indi=
vidual decisions=2C not just use of relays.=20

For example=2C use of an FQDN in the CNAME can leak location information.=20

My recommendation would be to focus on the media plane rather than signalin=
g=2C since
there are a lot of ways to leak location in SIP (or XMPP=2C for that matter=
). =20

 		 	   		  =

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<body class=3D'hmmessage'><div dir=3D'ltr'>
Protection of location (and privacy in general) depends on a number of indi=
vidual decisions=2C not just use of relays. <br><br>For example=2C use of a=
n FQDN in the CNAME can leak location information. <br><br>My recommendatio=
n would be to focus on the media plane rather than signaling=2C since<br>th=
ere are a lot of ways to leak location in SIP (or XMPP=2C for that matter).=
&nbsp=3B <br><div><div id=3D"SkyDrivePlaceholder"></div><br></div> 		 	   	=
	  </div></body>
</html>=

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From bernard_aboba@hotmail.com  Thu Mar  8 14:48:08 2012
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Subject: Re: [rtcweb] I'd like to discuss my proposal (jingle-web) at next WG session
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In general=2C Jingle extensions are within the purview of the XSF=2C rather=
 than IETF and the DOM is owned by W3C.=20

So other than the question of mandatory signaling implementation in browser=
s (which as Ted noted=2C has already been discussed)=2C it seems to me that=
 much of this document is out of scope not only of RTCWEB=2C but the IETF. =
=20



> Date: Thu=2C 8 Mar 2012 08:27:47 -0800
> From: ted.ietf@gmail.com
> To: hiro.suzuki@d3communications.jp
> CC: rtcweb@ietf.org
> Subject: Re: [rtcweb] I'd like to discuss my proposal (jingle-web) at nex=
t	WG session
>=20
> Hi Suzuki-san=2C
>=20
> Given that this apparently revisits some decisions that were already
> made=2C the chairs would like to see working group discussion of the
> draft before allocating agenda time to it for the upcoming meeting.
>=20
> best regards=2C
>=20
> Ted Hardie for the Chairs
>=20
> On Wed=2C Feb 29=2C 2012 at 7:34 PM=2C HIro Suzuki
> <hiro.suzuki@d3communications.jp> wrote:
> > Hello RTCWeb WG members=2C
> >
> >
> > It's my first post to this ML=2C and at Feb 28th=2C I submitted my ID
> > from IETF ID submit tool. This ID's file name is
> > "draft-suzuki-rtcweb-jingle-web-00.txt" and this ID provides my proposa=
l
> > to realize one of solutions of RTCWeb by using XMPP Jingle as a signali=
ng
> > protocol.
> >
> > Following paragraph is abstract of my ID=2C and I'd like to discuss my
> > proposal at RTCWeb WG session in next Paris meeting.
> >
> >
> > ---
> > Abstract:
> >  XMPP Jingle specification defines an XMPP protocol extension for
> >  managing peer-to-peer media sessions between two users. And XMPP
> >  Jingle can manage multi contents simultaneously in one Jingle
> >  stream=2C but in the XMPP world there is no common way to layout
> >  variable multi contents on each display. In this document=2C a
> >  solution to this issue is provided by using Web browser's layout
> >  function.
> >
> >  This document describes a new concept to realize one of solutions
> >  of RTCWeb. Of course=2C &quot=3BWeb Browser&quot=3B is used for Web ap=
plication's
> >  frontend for real-time communication=2C and XMPP Jingle specification
> >  (XEP-166) is used as signaling protocol. And a simple mapping
> >  manner between jingle contents and HTML graphical elements enables
> >  to unite Web browser&#39=3Bs layout function and Jingle&#39=3Bs media =
content
> >  management function to realize RTCWeb functions.
> > ---
> >
> >
> > Best Regards=2C
> > Yoshihiro Suzuki
> > (hiro.suzuki@d3communications.jp)
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
 		 	   		  =

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<body class=3D'hmmessage'><div dir=3D'ltr'>
In general=2C Jingle extensions are within the purview of the XSF=2C rather=
 than IETF and the DOM is owned by W3C. <br><br>So other than the question =
of mandatory signaling implementation in browsers (which as Ted noted=2C ha=
s already been discussed)=2C it seems to me that much of this document is o=
ut of scope not only of RTCWEB=2C but the IETF.&nbsp=3B <br><br><br><br><di=
v><div id=3D"SkyDrivePlaceholder"></div>&gt=3B Date: Thu=2C 8 Mar 2012 08:2=
7:47 -0800<br>&gt=3B From: ted.ietf@gmail.com<br>&gt=3B To: hiro.suzuki@d3c=
ommunications.jp<br>&gt=3B CC: rtcweb@ietf.org<br>&gt=3B Subject: Re: [rtcw=
eb] I'd like to discuss my proposal (jingle-web) at next	WG session<br>&gt=
=3B <br>&gt=3B Hi Suzuki-san=2C<br>&gt=3B <br>&gt=3B Given that this appare=
ntly revisits some decisions that were already<br>&gt=3B made=2C the chairs=
 would like to see working group discussion of the<br>&gt=3B draft before a=
llocating agenda time to it for the upcoming meeting.<br>&gt=3B <br>&gt=3B =
best regards=2C<br>&gt=3B <br>&gt=3B Ted Hardie for the Chairs<br>&gt=3B <b=
r>&gt=3B On Wed=2C Feb 29=2C 2012 at 7:34 PM=2C HIro Suzuki<br>&gt=3B &lt=
=3Bhiro.suzuki@d3communications.jp&gt=3B wrote:<br>&gt=3B &gt=3B Hello RTCW=
eb WG members=2C<br>&gt=3B &gt=3B<br>&gt=3B &gt=3B<br>&gt=3B &gt=3B It's my=
 first post to this ML=2C and at Feb 28th=2C I submitted my ID<br>&gt=3B &g=
t=3B from IETF ID submit tool. This ID's file name is<br>&gt=3B &gt=3B "dra=
ft-suzuki-rtcweb-jingle-web-00.txt" and this ID provides my proposal<br>&gt=
=3B &gt=3B to realize one of solutions of RTCWeb by using XMPP Jingle as a =
signaling<br>&gt=3B &gt=3B protocol.<br>&gt=3B &gt=3B<br>&gt=3B &gt=3B Foll=
owing paragraph is abstract of my ID=2C and I'd like to discuss my<br>&gt=
=3B &gt=3B proposal at RTCWeb WG session in next Paris meeting.<br>&gt=3B &=
gt=3B<br>&gt=3B &gt=3B<br>&gt=3B &gt=3B ---<br>&gt=3B &gt=3B Abstract:<br>&=
gt=3B &gt=3B &nbsp=3BXMPP Jingle specification defines an XMPP protocol ext=
ension for<br>&gt=3B &gt=3B &nbsp=3Bmanaging peer-to-peer media sessions be=
tween two users. And XMPP<br>&gt=3B &gt=3B &nbsp=3BJingle can manage multi =
contents simultaneously in one Jingle<br>&gt=3B &gt=3B &nbsp=3Bstream=2C bu=
t in the XMPP world there is no common way to layout<br>&gt=3B &gt=3B &nbsp=
=3Bvariable multi contents on each display. In this document=2C a<br>&gt=3B=
 &gt=3B &nbsp=3Bsolution to this issue is provided by using Web browser's l=
ayout<br>&gt=3B &gt=3B &nbsp=3Bfunction.<br>&gt=3B &gt=3B<br>&gt=3B &gt=3B =
&nbsp=3BThis document describes a new concept to realize one of solutions<b=
r>&gt=3B &gt=3B &nbsp=3Bof RTCWeb. Of course=2C &amp=3Bquot=3BWeb Browser&a=
mp=3Bquot=3B is used for Web application's<br>&gt=3B &gt=3B &nbsp=3Bfronten=
d for real-time communication=2C and XMPP Jingle specification<br>&gt=3B &g=
t=3B &nbsp=3B(XEP-166) is used as signaling protocol. And a simple mapping<=
br>&gt=3B &gt=3B &nbsp=3Bmanner between jingle contents and HTML graphical =
elements enables<br>&gt=3B &gt=3B &nbsp=3Bto unite Web browser&amp=3B#39=3B=
s layout function and Jingle&amp=3B#39=3Bs media content<br>&gt=3B &gt=3B &=
nbsp=3Bmanagement function to realize RTCWeb functions.<br>&gt=3B &gt=3B --=
-<br>&gt=3B &gt=3B<br>&gt=3B &gt=3B<br>&gt=3B &gt=3B Best Regards=2C<br>&gt=
=3B &gt=3B Yoshihiro Suzuki<br>&gt=3B &gt=3B (hiro.suzuki@d3communications.=
jp)<br>&gt=3B &gt=3B _______________________________________________<br>&gt=
=3B &gt=3B rtcweb mailing list<br>&gt=3B &gt=3B rtcweb@ietf.org<br>&gt=3B &=
gt=3B https://www.ietf.org/mailman/listinfo/rtcweb<br>&gt=3B ______________=
_________________________________<br>&gt=3B rtcweb mailing list<br>&gt=3B r=
tcweb@ietf.org<br>&gt=3B https://www.ietf.org/mailman/listinfo/rtcweb<br></=
div> 		 	   		  </div></body>
</html>=

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References: <E17CAD772E76C742B645BD4DC602CD8105DFC884@NAHALD.us.int.genesyslab.com> <4F58D0FE.9060909@alvestrand.no> <E17CAD772E76C742B645BD4DC602CD8105DFC8A6@NAHALD.us.int.genesyslab.com> <CABcZeBPB65cg6ZdC8rmLAD7QgdhBo+52QQ158s-CYYC5V9WTdA@mail.gmail.com> <E17CAD772E76C742B645BD4DC602CD8105DFC8AF@NAHALD.us.int.genesyslab.com> <CABcZeBNpFuGOoFG0jjgvc_08bJFhRFN8hgzBHwHu=fYmEw1ENw@mail.gmail.com> <E17CAD772E76C742B645BD4DC602CD8105DFC96E@NAHALD.us.int.genesyslab.com> <4F590BDF.7050004@alvestrand.no> <E17CAD772E76C742B645BD4DC602CD8105DFCA24@NAHALD.us.int.genesyslab.com>
From: Eric Rescorla <ekr@rtfm.com>
Date: Thu, 8 Mar 2012 18:23:11 -0800
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Subject: Re: [rtcweb] when should ICE start?
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On Thu, Mar 8, 2012 at 6:16 PM, Jim Barnett <Jim.Barnett@genesyslab.com> wrote:
> Yes, but if we can trust the anonymous flirting site to protect our
> privacy, why can't we trust it not to record our calls if we use SDES?
> Can we trust the web server to do what we expect, or should we be
> systematically untrusting?

Is this a trick question?

1. There's a pretty big difference between knowing where I am and
being able to listen to my calls.
2. The IP-location privacy threat is pre-existing and not fundamentally
a WebRTC problem.
3. Nevertheless, people are concerned about it, which is why things like
Tor exist.

-Ekr

From magnus.westerlund@ericsson.com  Fri Mar  9 02:50:51 2012
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Subject: Re: [rtcweb] Proposal for media receiver request of codec parameter changes (draft-westerlund-avtext-codec-operation-point-00)
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Hi,

To make it clear. There is an IPR disclosure on this draft from Ericsson.
https://datatracker.ietf.org/ipr/1701/

Regards

Magnus

On 2012-03-08 13:55, Magnus Westerlund wrote:
> WG,
> 
> As an individual I would like to inform you that my colleagues and I
> have submitted a proposal to AVTEXT WG. This proposal is an RTCP
> extension that enables RTP session participants to request the media
> sender to adjust its media parameters, like video resolution.
> 
> The draft can be retrieved here:
> https://datatracker.ietf.org/doc/draft-westerlund-avtext-codec-operation-point/
> 
> One use case we have designed this for is to enable the dynamic updates
> codec parameters as result of WebRTC applications usage of media
> streams. If the application changes the rendering layout or the user
> changes the browser window size, the size of a video playout can change.
> This may happen in a quite dynamic way and often enough that we think
> using a mechanism where the receiver can use a request over the media
> plane rather than the signalling plane to adjust the parameters will
> give a more responsive and satisfying experience. We have also designed
> it with centralized conferencing into mind, allowing the central node to
> aggregate requests to best suit the whole population.
> 
> We request that feedback on the actual extension are directed to the
> AVTEXT WG mailing list: avtext@ietf.org
> 
> I am happy to discuss the usage and applicability to WebRTC on this list.
> 
> Cheers
> 
> Magnus Westerlund
> 
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> Färögatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
> 
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> 


-- 

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


From zs68j2ee@gmail.com  Fri Mar  9 09:40:45 2012
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To: rtcweb@ietf.org
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Subject: [rtcweb] some thoughts on an alternative WebRTC approach: iWebPP - instant web p2p technology
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--f46d043893ff65d1f004bad2e45b
Content-Type: text/plain; charset=UTF-8

Hi,


  WebRTC intends to setup P2P communication between browsers, then carry
video/audio/text media, etc.

  For example, in the conference call, WebRTC takes four tasks:
1. pick up local medias: video, voice, etc
2. encoding local medias
3. streaming medias via p2p channel securely
4. decoding medias

for item 1, it need to export OS level media interfaces to browser via JS.
for item 3, it should implement Streaming Server in browser with html5
techs: web workers, etc
for item 2 and 4, it should implement a lot of algorithms on video and
audio. the most of them is redundant works against gstreamer/ffmpeg.

AFAIK there are two reference implementations to WebRTC: Google's and
Ericsson Lab's. Google is doing all stuff, while Ericsson reuses gstreamer.
Item1 above has no many stories. But, item2,4 and item3 may have another
stories.

Why we need WebRTC? Firstly, Web is the most popular network app, secondly,
video/voice brings the best user experience.
But, the problem is that HTTP runs upon TCP by now, while P2P runs upon UDP
normally.

Suppose both web browser and server can run HTTP upon UDP(the protocol
schema as HTTPP), what happen?
Firstly, Web app developers can program HTTPP like HTTP, secondly, P2P
traffic can be carried on HTTPP.

Basically iWebPP consists of two parts: HTTPP-enabled web browser and web
server.
We have implemented a Firefox extension and Node.js variant  to run HTTP
upon UDP. They are the first step and will be open source soon.

Any thoughts? thanks.

Best regards
   Tom

--f46d043893ff65d1f004bad2e45b
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div class=3D"gmail_quote"><div class=3D"gmail_quote">Hi,<div><div><br><br>=
=C2=A0 WebRTC intends to setup P2P communication between browsers, then car=
ry video/audio/text media, etc.<br><br><div>=C2=A0 For example, in the conf=
erence call, WebRTC takes four tasks:<br>

1. pick up local medias: video, voice, etc<br>


2. encoding local medias<br>3. streaming medias via p2p channel securely<br=
>4. decoding medias<br><br>for item 1, it need to export OS level media int=
erfaces to browser via JS.<br>for item 3, it should implement Streaming Ser=
ver in browser with html5 techs: web workers, etc<br>




for item 2 and 4, it should implement a lot of algorithms on video and audi=
o. the most of them is redundant works against gstreamer/ffmpeg.<br><br><di=
v>AFAIK there are two reference implementations to WebRTC: Google&#39;s and=
 Ericsson Lab&#39;s. Google is doing all stuff, while Ericsson reuses gstre=
amer.</div>



<div>Item1 above has no many stories. But, item2,4 and item3 may have anoth=
er stories.=C2=A0</div><div><br></div><div>Why we need WebRTC? Firstly, Web=
 is the most popular network app, secondly, video/voice brings the best use=
r experience.</div>



<div>But, the problem is that HTTP runs upon TCP by now, while P2P runs upo=
n UDP normally.=C2=A0</div><div><br></div><div>Suppose both web browser and=
 server can run HTTP upon UDP(the protocol schema as HTTPP), what happen?</=
div>



<div>Firstly, Web app developers can program HTTPP like HTTP, secondly, P2P=
 traffic can be carried on HTTPP.=C2=A0</div><div><br></div><div>Basically =
iWebPP consists of two parts: HTTPP-enabled web browser and web server.</di=
v>



<div>We have implemented a Firefox extension and Node.js variant =C2=A0to r=
un HTTP upon UDP. They are the first step and will be open source soon.</di=
v><div><br></div><div>Any thoughts? thanks.</div><div><br></div><div>Best r=
egards</div>


<span><font color=3D"#888888">
<div>=C2=A0 Tom<br><br>=C2=A0<br><br><br><br><br>
</div></font></span></div>
</div></div></div><br>
</div><br>

--f46d043893ff65d1f004bad2e45b--

From salvatore.loreto@ericsson.com  Mon Mar 12 00:36:04 2012
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Date: Mon, 12 Mar 2012 09:36:00 +0200
From: Salvatore Loreto <salvatore.loreto@ericsson.com>
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Subject: [rtcweb] draft-ietf-mmusic-sctp-sdp-01
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Hi there,

I have submitted a new version of the draft-ietf-mmusic-sctp-sdp,
that other then fix some little mistakes
defines the *DTLS/SCTP* protocol identifier
to allow the usage of SCTP on top of the Datagram Transport Layer
Security (DTLS) protocol,
as defined in [I-D.tuexen-tsvwg-sctp-dtls-encaps], using SDP.

SCTP over DTLS is used by the RTCWeb protocol suite for transporting
non-media data between browsers.

regards
Salvatore

-- 
Salvatore Loreto
www.sloreto.com



On 3/12/12 9:03 AM, internet-drafts@ietf.org wrote:
>  A New Internet-Draft is available from the on-line Internet-Drafts directories. This draft is a work item of the Multiparty Multimedia Session Control Working Group of the IETF.
>
>  	Title           : Stream Control Transmission Protocol (SCTP)-Based Media Transport in the Session Description Protocol (SDP)
>  	Author(s)       : Salvatore Loreto
>                             Gonzalo Camarillo
>  	Filename        : draft-ietf-mmusic-sctp-sdp-01.txt
>  	Pages           : 10
>  	Date            : 2012-03-12
>
>      SCTP (Stream Control Transmission Protocol) is a transport protocol
>      used to establish associations between two endpoints.  This document
>      describes how to express media transport over SCTP in SDP (Session
>      Description Protocol).  This document defines the 'SCTP', 'SCTP/DTLS'
>      and 'DTLS/SCTP' protocol identifiers for SDP.
>
>
>  A URL for this Internet-Draft is:
>  http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sctp-sdp-01.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-ietf-mmusic-sctp-sdp-01.txt
>
>  _______________________________________________
>  mmusic mailing list
>  mmusic@ietf.org
>  https://www.ietf.org/mailman/listinfo/mmusic
>


From internet-drafts@ietf.org  Mon Mar 12 14:24:17 2012
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies. This draft is a work item of the Real-Time Communication in WEB-browse=
rs Working Group of the IETF.

	Title           : Web Real-Time Communication (WebRTC): Media Transport an=
d Use of RTP
	Author(s)       : Colin Perkins
                          Magnus Westerlund
                          Joerg Ott
	Filename        : draft-ietf-rtcweb-rtp-usage-02.txt
	Pages           : 27
	Date            : 2012-03-12

   The Web Real-Time Communication (WebRTC) framework provides support
   for direct interactive rich communication using audio, video, text,
   collaboration, games, etc. between two peers' web-browsers.  This
   memo describes the media transport aspects of the WebRTC framework.
   It specifies how the Real-time Transport Protocol (RTP) is used in
   the WebRTC context, and gives requirements for which RTP features,
   profiles, and extensions need to be supported.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-rtcweb-rtp-usage-02.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-ietf-rtcweb-rtp-usage-02.txt


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From: Eric Rescorla <ekr@rtfm.com>
Date: Mon, 12 Mar 2012 14:24:55 -0700
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] DoS attack despite ICE/STUN consent verification
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I want to make sure I understand this attack correctly.

Alice is making a call to Zed and either Zed is the attacker or the attacke=
r
somehow captures Alice's media parameters. The attacker then uses
Alice's offer (and her parameters) to start ICE handshaking on a pile
of Web browsers that he controls. They all start ICE and since the
checks with Alice will succeed, then the attacker has apparent consent
to send as much traffic as he wants to Alice from each machine.
Do I have this right?

If I understand this correctly, then the severity of the attack seems to
be limited since Alice has to actually be calling someone that is
malicious or that has very tight control of the network, and soon
after Alice hangs up the phone, the problem goes away. (Assuming
consent freshness).

As Dan notes, the interaction with consent freshness seems like a key
mitigation here. Moreover, I think this suggests that even if integrity
is not used, as in draft-muthu, the ufrags should both be present
(and unmodifiable from the JS). This would allow a client to discard
ICE associations with ufrags that do not match whichever ICE
association got selected for the call.

-Ekr


On Mon, Feb 6, 2012 at 6:27 AM, Magnus Westerlund
<magnus.westerlund@ericsson.com> wrote:
> On 2012-02-06 11:19, Oscar Ohlsson wrote:
>>
>>
>>> -----Original Message-----
>>> From: rtcweb-bounces@ietf.org
>>> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Magnus Westerlund
>>> Sent: den 1 februari 2012 23:31
>>> To: rtcweb@ietf.org
>>> Subject: [rtcweb] DoS attack despite ICE/STUN consent verification
>>>
>>> Hi,
>>>
>>> As I brought up in yesterdays meeting I think there exist an
>>> attack that should be considered. I don't think it is super
>>> strong, but not without issues.
>>>
>>> Setting:
>>>
>>> Webserver is evil and have a target Alice. Alice can be a
>>> legacy device, like a SIP/SDP video phone. It knows the
>>> target is doing ICE and have Alice's ufrag and password and
>>> candidates. The server would like to hammer this target by
>>> utilizing its set of connected WebRTC clients (bob, charlie
>>> etc.). Thus it would like to ensure that it gets the browser
>>> to start sending RTP media when creating a PeerConnection in
>>> the clients.
>>>
>>> So the server gets the client to open PeerConnections towards
>>> the ICE parameters from the target (Alice).
>>>
>>> In default ICE behavior there is an optimization to allow the Offerer
>>> (Alice) to respond to STUN binding requests from Bob without
>>> having received an SDP answer from Bob. The STUN binding
>>> request must only have the Alice's ufrag and its password.
>>> This enables that Webrtc clients can get passed the data
>>> consent step with ICE capable end-point for which the ICE
>>> parameters can be retrieved.
>>>
>>> So from Alice perspective this attack will look like its
>>> offer got forked to many peers based on the many different
>>> peers that send STUN messages. The webserver could even
>>> provide SIP/SDP answers to alice with the webclients. But
>>> that is not necessary but is one variation of this.
>>
>>
>> Correct me if I'm wrong, but I believe the webserver must provide the
> SIP/SDP answers. As far as I can tell there are always two connectivity
> checks, one for each direction (Section 2.2 RFC 5245):
>
>> =A0 =A0L =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0R
>> =A0 =A0- =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0-
>> =A0 =A0STUN request -> =A0 =A0 =A0 =A0 =A0 =A0 \ =A0L's
>> =A0 =A0 =A0 =A0 =A0 =A0 =A0<- STUN response =A0/ =A0check
>>
>> =A0 =A0 =A0 =A0 =A0 =A0 =A0 <- STUN request =A0\ =A0R's
>> =A0 =A0STUN response -> =A0 =A0 =A0 =A0 =A0 =A0/ =A0check
>>
>> Alice would automatically respond to the remote side's check.
>> However, in order to genereate her own check (R's check in the
>> diagram) she would need to have received the remote side's password.
>> According to Section 7.1.2.3. in RFC 5245:
>>
>> "A connectivity check from R to L utilizes the username LFRAG:RFRAG
>> and a password of LPASS."
>>
>> If this interpretation is correct then counter-measure A below would
>> always be a possibility.
>>
>
> Oscar,
>
> From L's perspective, the process of nomination and validation can
> complete without R having succeeded. Thus L (WebRTC Browser) can start
> send media to R without R having successfully validated any return path
> from R to L.
>
> Thus, I don't see how A could be made to reliably work as a mitigation.
>
> In addition are there anyone who actually have this type of attack
> protection in their SIP stacks or SIP proxies?
>
> Cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Phone =A0+46 10 7148287
> F=E4r=F6gatan 6 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From internet-drafts@ietf.org  Mon Mar 12 14:30:07 2012
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Subject: [rtcweb] I-D Action: draft-ietf-rtcweb-overview-03.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies. This draft is a work item of the Real-Time Communication in WEB-browse=
rs Working Group of the IETF.

	Title           : Overview: Real Time Protocols for Brower-based Applicati=
ons
	Author(s)       : Harald T. Alvestrand
	Filename        : draft-ietf-rtcweb-overview-03.txt
	Pages           : 20
	Date            : 2012-03-12

   This document gives an overview and context of a protocol suite
   intended for use with real-time applications that can be deployed in
   browsers - "real time communication on the Web".

   It intends to serve as a starting and coordination point to make sure
   all the parts that are needed to achieve this goal are findable, and
   that the parts that belong in the Internet protocol suite are fully
   specified and on the right publication track.

   This work is an attempt to synthesize the input of many people, but
   makes no claims to fully represent the views of any of them.  All
   parts of the document should be regarded as open for discussion,
   unless the RTCWEB chairs have declared consensus on an item.

   This document is a work item of the RTCWEB working group.



A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-rtcweb-overview-03.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-ietf-rtcweb-overview-03.txt


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From: Eric Rescorla <ekr@rtfm.com>
Date: Mon, 12 Mar 2012 14:30:12 -0700
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] DoS attack despite ICE/STUN consent verification
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I should also mention that the users will of course have to have
given the attacker's site media access--at least for legacy SIP
clients, the attempt by the zombies to form SCTP-over-DTLS
associations will fail and therefore the data channel cannot
be used here.

-Ekr

On Mon, Mar 12, 2012 at 2:24 PM, Eric Rescorla <ekr@rtfm.com> wrote:
> I want to make sure I understand this attack correctly.
>
> Alice is making a call to Zed and either Zed is the attacker or the attac=
ker
> somehow captures Alice's media parameters. The attacker then uses
> Alice's offer (and her parameters) to start ICE handshaking on a pile
> of Web browsers that he controls. They all start ICE and since the
> checks with Alice will succeed, then the attacker has apparent consent
> to send as much traffic as he wants to Alice from each machine.
> Do I have this right?
>
> If I understand this correctly, then the severity of the attack seems to
> be limited since Alice has to actually be calling someone that is
> malicious or that has very tight control of the network, and soon
> after Alice hangs up the phone, the problem goes away. (Assuming
> consent freshness).
>
> As Dan notes, the interaction with consent freshness seems like a key
> mitigation here. Moreover, I think this suggests that even if integrity
> is not used, as in draft-muthu, the ufrags should both be present
> (and unmodifiable from the JS). This would allow a client to discard
> ICE associations with ufrags that do not match whichever ICE
> association got selected for the call.
>
> -Ekr
>
>
> On Mon, Feb 6, 2012 at 6:27 AM, Magnus Westerlund
> <magnus.westerlund@ericsson.com> wrote:
>> On 2012-02-06 11:19, Oscar Ohlsson wrote:
>>>
>>>
>>>> -----Original Message-----
>>>> From: rtcweb-bounces@ietf.org
>>>> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Magnus Westerlund
>>>> Sent: den 1 februari 2012 23:31
>>>> To: rtcweb@ietf.org
>>>> Subject: [rtcweb] DoS attack despite ICE/STUN consent verification
>>>>
>>>> Hi,
>>>>
>>>> As I brought up in yesterdays meeting I think there exist an
>>>> attack that should be considered. I don't think it is super
>>>> strong, but not without issues.
>>>>
>>>> Setting:
>>>>
>>>> Webserver is evil and have a target Alice. Alice can be a
>>>> legacy device, like a SIP/SDP video phone. It knows the
>>>> target is doing ICE and have Alice's ufrag and password and
>>>> candidates. The server would like to hammer this target by
>>>> utilizing its set of connected WebRTC clients (bob, charlie
>>>> etc.). Thus it would like to ensure that it gets the browser
>>>> to start sending RTP media when creating a PeerConnection in
>>>> the clients.
>>>>
>>>> So the server gets the client to open PeerConnections towards
>>>> the ICE parameters from the target (Alice).
>>>>
>>>> In default ICE behavior there is an optimization to allow the Offerer
>>>> (Alice) to respond to STUN binding requests from Bob without
>>>> having received an SDP answer from Bob. The STUN binding
>>>> request must only have the Alice's ufrag and its password.
>>>> This enables that Webrtc clients can get passed the data
>>>> consent step with ICE capable end-point for which the ICE
>>>> parameters can be retrieved.
>>>>
>>>> So from Alice perspective this attack will look like its
>>>> offer got forked to many peers based on the many different
>>>> peers that send STUN messages. The webserver could even
>>>> provide SIP/SDP answers to alice with the webclients. But
>>>> that is not necessary but is one variation of this.
>>>
>>>
>>> Correct me if I'm wrong, but I believe the webserver must provide the
>> SIP/SDP answers. As far as I can tell there are always two connectivity
>> checks, one for each direction (Section 2.2 RFC 5245):
>>
>>> =A0 =A0L =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0R
>>> =A0 =A0- =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0-
>>> =A0 =A0STUN request -> =A0 =A0 =A0 =A0 =A0 =A0 \ =A0L's
>>> =A0 =A0 =A0 =A0 =A0 =A0 =A0<- STUN response =A0/ =A0check
>>>
>>> =A0 =A0 =A0 =A0 =A0 =A0 =A0 <- STUN request =A0\ =A0R's
>>> =A0 =A0STUN response -> =A0 =A0 =A0 =A0 =A0 =A0/ =A0check
>>>
>>> Alice would automatically respond to the remote side's check.
>>> However, in order to genereate her own check (R's check in the
>>> diagram) she would need to have received the remote side's password.
>>> According to Section 7.1.2.3. in RFC 5245:
>>>
>>> "A connectivity check from R to L utilizes the username LFRAG:RFRAG
>>> and a password of LPASS."
>>>
>>> If this interpretation is correct then counter-measure A below would
>>> always be a possibility.
>>>
>>
>> Oscar,
>>
>> From L's perspective, the process of nomination and validation can
>> complete without R having succeeded. Thus L (WebRTC Browser) can start
>> send media to R without R having successfully validated any return path
>> from R to L.
>>
>> Thus, I don't see how A could be made to reliably work as a mitigation.
>>
>> In addition are there anyone who actually have this type of attack
>> protection in their SIP stacks or SIP proxies?
>>
>> Cheers
>>
>> Magnus Westerlund
>>
>> ----------------------------------------------------------------------
>> Multimedia Technologies, Ericsson Research EAB/TVM
>> ----------------------------------------------------------------------
>> Ericsson AB =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Phone =A0+46 10 7148287
>> F=E4r=F6gatan 6 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Mobile +46 73 0949079
>> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
>> ----------------------------------------------------------------------
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb

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FYI. Not much has changed, but a number of new drafts got referenced.

-------- Original Message --------
Subject: 	New Version Notification for draft-ietf-rtcweb-overview-03.txt
Date: 	Mon, 12 Mar 2012 14:30:00 -0700
From: 	internet-drafts@ietf.org
To: 	harald@alvestrand.no



A new version of I-D, draft-ietf-rtcweb-overview-03.txt has been successfully submitted by Harald T. Alvestrand and posted to the IETF repository.

Filename:	 draft-ietf-rtcweb-overview
Revision:	 03
Title:		 Overview: Real Time Protocols for Brower-based Applications
Creation date:	 2012-03-12
WG ID:		 rtcweb
Number of pages: 20

Abstract:
    This document gives an overview and context of a protocol suite
    intended for use with real-time applications that can be deployed in
    browsers -&quot;real time communication on the Web&quot;.

    It intends to serve as a starting and coordination point to make sure
    all the parts that are needed to achieve this goal are findable, and
    that the parts that belong in the Internet protocol suite are fully
    specified and on the right publication track.

    This work is an attempt to synthesize the input of many people, but
    makes no claims to fully represent the views of any of them.  All
    parts of the document should be regarded as open for discussion,
    unless the RTCWEB chairs have declared consensus on an item.

    This document is a work item of the RTCWEB working group.





The IETF Secretariat



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    FYI. Not much has changed, but a number of new drafts got
    referenced.<br>
    <br>
    -------- Original Message --------
    <table class="moz-email-headers-table" border="0" cellpadding="0"
      cellspacing="0">
      <tbody>
        <tr>
          <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Subject: </th>
          <td>New Version Notification for
            draft-ietf-rtcweb-overview-03.txt</td>
        </tr>
        <tr>
          <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Date: </th>
          <td>Mon, 12 Mar 2012 14:30:00 -0700</td>
        </tr>
        <tr>
          <th align="RIGHT" nowrap="nowrap" valign="BASELINE">From: </th>
          <td><a class="moz-txt-link-abbreviated" href="mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a></td>
        </tr>
        <tr>
          <th align="RIGHT" nowrap="nowrap" valign="BASELINE">To: </th>
          <td><a class="moz-txt-link-abbreviated" href="mailto:harald@alvestrand.no">harald@alvestrand.no</a></td>
        </tr>
      </tbody>
    </table>
    <br>
    <br>
    <pre>A new version of I-D, draft-ietf-rtcweb-overview-03.txt has been successfully submitted by Harald T. Alvestrand and posted to the IETF repository.

Filename:	 draft-ietf-rtcweb-overview
Revision:	 03
Title:		 Overview: Real Time Protocols for Brower-based Applications
Creation date:	 2012-03-12
WG ID:		 rtcweb
Number of pages: 20

Abstract:
   This document gives an overview and context of a protocol suite
   intended for use with real-time applications that can be deployed in
   browsers - &amp;quot;real time communication on the Web&amp;quot;.

   It intends to serve as a starting and coordination point to make sure
   all the parts that are needed to achieve this goal are findable, and
   that the parts that belong in the Internet protocol suite are fully
   specified and on the right publication track.

   This work is an attempt to synthesize the input of many people, but
   makes no claims to fully represent the views of any of them.  All
   parts of the document should be regarded as open for discussion,
   unless the RTCWEB chairs have declared consensus on an item.

   This document is a work item of the RTCWEB working group.


                                                                                  


The IETF Secretariat

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--------------070804010801000409060708--

From csp@csperkins.org  Mon Mar 12 14:58:53 2012
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We've just posted an update to our draft on the use of RTP in the RTCWeb =
context. This is a minor update, consisting almost entirely of editorial =
cleanups. The only technical changes are to make RFC 6222 support =
REQUIRED rather than RECOMMENDED, and to replace the discussion of =
congestion control requirements with references to =
draft-jesup-rtp-congestion-reqs-00 and =
draft-perkins-avtcore-rtp-circuit-breakers-00.

Colin


Begin forwarded message:
> From: internet-drafts@ietf.org
> Subject: I-D Action: draft-ietf-rtcweb-rtp-usage-02.txt
> Date: 12 March 2012 21:24:16 GMT
> To: i-d-announce@ietf.org
> Cc: rtcweb@ietf.org
> Reply-To: internet-drafts@ietf.org
>=20
>=20
> A New Internet-Draft is available from the on-line Internet-Drafts =
directories. This draft is a work item of the Real-Time Communication in =
WEB-browsers Working Group of the IETF.
>=20
> 	Title           : Web Real-Time Communication (WebRTC): Media =
Transport and Use of RTP
> 	Author(s)       : Colin Perkins
>                          Magnus Westerlund
>                          Joerg Ott
> 	Filename        : draft-ietf-rtcweb-rtp-usage-02.txt
> 	Pages           : 27
> 	Date            : 2012-03-12
>=20
>   The Web Real-Time Communication (WebRTC) framework provides support
>   for direct interactive rich communication using audio, video, text,
>   collaboration, games, etc. between two peers' web-browsers.  This
>   memo describes the media transport aspects of the WebRTC framework.
>   It specifies how the Real-time Transport Protocol (RTP) is used in
>   the WebRTC context, and gives requirements for which RTP features,
>   profiles, and extensions need to be supported.
>=20
>=20
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-rtcweb-rtp-usage-02.txt

--=20
Colin Perkins
http://csperkins.org/




From internet-drafts@ietf.org  Mon Mar 12 15:01:56 2012
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies. This draft is a work item of the Real-Time Communication in WEB-browse=
rs Working Group of the IETF.

	Title           : RTCWEB Security Architecture
	Author(s)       : Eric Rescorla
	Filename        : draft-ietf-rtcweb-security-arch-01.txt
	Pages           : 20
	Date            : 2012-03-12

   The Real-Time Communications on the Web (RTCWEB) working group is
   tasked with standardizing protocols for real-time communications
   between Web browsers.  The major use cases for RTCWEB technology are
   real-time audio and/or video calls, Web conferencing, and direct data
   transfer.  Unlike most conventional real-time systems (e.g., SIP-
   based soft phones) RTCWEB communications are directly controlled by
   some Web server, which poses new security challenges.  For instance,
   a Web browser might expose a JavaScript API which allows a server to
   place a video call.  Unrestricted access to such an API would allow
   any site which a user visited to "bug" a user's computer, capturing
   any activity which passed in front of their camera.  [I-D.ietf-
   rtcweb-security] defines the RTCWEB threat model.  This document
   defines an architecture which provides security within that threat
   model.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-rtcweb-security-arch-01.txt

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ftp://ftp.ietf.org/internet-drafts/

This Internet-Draft can be retrieved at:
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From internet-drafts@ietf.org  Mon Mar 12 15:28:54 2012
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies. This draft is a work item of the Real-Time Communication in WEB-browse=
rs Working Group of the IETF.

	Title           : Security Considerations for RTC-Web
	Author(s)       : Eric Rescorla
	Filename        : draft-ietf-rtcweb-security-02.txt
	Pages           : 21
	Date            : 2012-03-12

   The Real-Time Communications on the Web (RTC-Web) working group is
   tasked with standardizing protocols for real-time communications
   between Web browsers.  The major use cases for RTC-Web technology are
   real-time audio and/or video calls, Web conferencing, and direct data
   transfer.  Unlike most conventional real-time systems (e.g., SIP-
   based soft phones) RTC-Web communications are directly controlled by
   some Web server, which poses new security challenges.  For instance,
   a Web browser might expose a JavaScript API which allows a server to
   place a video call.  Unrestricted access to such an API would allow
   any site which a user visited to "bug" a user's computer, capturing
   any activity which passed in front of their camera.  This document
   defines the RTC-Web threat model and defines an architecture which
   provides security within that threat model.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-rtcweb-security-02.txt

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From magnus.westerlund@ericsson.com  Mon Mar 12 15:42:05 2012
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Hi,

Yes, your understanding appears correct. I was maybe more considering of
using this attack against SIP clients, especially the kind that can't
move out of the way easily. Like call centers etc. In that context being
able to direct a 1 mbps video stream onto a audio receiver works
excellent if your goal is to just DoS the path.

But I do agree that consent freshness appears to be an important method
for mitigating this. It also seems to create a requirement on that
mechanism. It can't be something that the far end can trick the target
end-point to generate just by default. It needs to be something where
the end-point takes a decision to respond to each.

Cheers

Magnus

On 2012-03-12 22:24, Eric Rescorla wrote:
> I want to make sure I understand this attack correctly.
> 
> Alice is making a call to Zed and either Zed is the attacker or the attacker
> somehow captures Alice's media parameters. The attacker then uses
> Alice's offer (and her parameters) to start ICE handshaking on a pile
> of Web browsers that he controls. They all start ICE and since the
> checks with Alice will succeed, then the attacker has apparent consent
> to send as much traffic as he wants to Alice from each machine.
> Do I have this right?
> 
> If I understand this correctly, then the severity of the attack seems to
> be limited since Alice has to actually be calling someone that is
> malicious or that has very tight control of the network, and soon
> after Alice hangs up the phone, the problem goes away. (Assuming
> consent freshness).
> 
> As Dan notes, the interaction with consent freshness seems like a key
> mitigation here. Moreover, I think this suggests that even if integrity
> is not used, as in draft-muthu, the ufrags should both be present
> (and unmodifiable from the JS). This would allow a client to discard
> ICE associations with ufrags that do not match whichever ICE
> association got selected for the call.
> 
> -Ekr
> 
> 
> On Mon, Feb 6, 2012 at 6:27 AM, Magnus Westerlund
> <magnus.westerlund@ericsson.com> wrote:
>> On 2012-02-06 11:19, Oscar Ohlsson wrote:
>>>
>>>
>>>> -----Original Message-----
>>>> From: rtcweb-bounces@ietf.org
>>>> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Magnus Westerlund
>>>> Sent: den 1 februari 2012 23:31
>>>> To: rtcweb@ietf.org
>>>> Subject: [rtcweb] DoS attack despite ICE/STUN consent verification
>>>>
>>>> Hi,
>>>>
>>>> As I brought up in yesterdays meeting I think there exist an
>>>> attack that should be considered. I don't think it is super
>>>> strong, but not without issues.
>>>>
>>>> Setting:
>>>>
>>>> Webserver is evil and have a target Alice. Alice can be a
>>>> legacy device, like a SIP/SDP video phone. It knows the
>>>> target is doing ICE and have Alice's ufrag and password and
>>>> candidates. The server would like to hammer this target by
>>>> utilizing its set of connected WebRTC clients (bob, charlie
>>>> etc.). Thus it would like to ensure that it gets the browser
>>>> to start sending RTP media when creating a PeerConnection in
>>>> the clients.
>>>>
>>>> So the server gets the client to open PeerConnections towards
>>>> the ICE parameters from the target (Alice).
>>>>
>>>> In default ICE behavior there is an optimization to allow the Offerer
>>>> (Alice) to respond to STUN binding requests from Bob without
>>>> having received an SDP answer from Bob. The STUN binding
>>>> request must only have the Alice's ufrag and its password.
>>>> This enables that Webrtc clients can get passed the data
>>>> consent step with ICE capable end-point for which the ICE
>>>> parameters can be retrieved.
>>>>
>>>> So from Alice perspective this attack will look like its
>>>> offer got forked to many peers based on the many different
>>>> peers that send STUN messages. The webserver could even
>>>> provide SIP/SDP answers to alice with the webclients. But
>>>> that is not necessary but is one variation of this.
>>>
>>>
>>> Correct me if I'm wrong, but I believe the webserver must provide the
>> SIP/SDP answers. As far as I can tell there are always two connectivity
>> checks, one for each direction (Section 2.2 RFC 5245):
>>
>>>    L                        R
>>>    -                        -
>>>    STUN request ->             \  L's
>>>              <- STUN response  /  check
>>>
>>>               <- STUN request  \  R's
>>>    STUN response ->            /  check
>>>
>>> Alice would automatically respond to the remote side's check.
>>> However, in order to genereate her own check (R's check in the
>>> diagram) she would need to have received the remote side's password.
>>> According to Section 7.1.2.3. in RFC 5245:
>>>
>>> "A connectivity check from R to L utilizes the username LFRAG:RFRAG
>>> and a password of LPASS."
>>>
>>> If this interpretation is correct then counter-measure A below would
>>> always be a possibility.
>>>
>>
>> Oscar,
>>
>> From L's perspective, the process of nomination and validation can
>> complete without R having succeeded. Thus L (WebRTC Browser) can start
>> send media to R without R having successfully validated any return path
>> from R to L.
>>
>> Thus, I don't see how A could be made to reliably work as a mitigation.
>>
>> In addition are there anyone who actually have this type of attack
>> protection in their SIP stacks or SIP proxies?
>>
>> Cheers
>>
>> Magnus Westerlund
>>
>> ----------------------------------------------------------------------
>> Multimedia Technologies, Ericsson Research EAB/TVM
>> ----------------------------------------------------------------------
>> Ericsson AB                | Phone  +46 10 7148287
>> Färögatan 6                | Mobile +46 73 0949079
>> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
>> ----------------------------------------------------------------------
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
> 


-- 

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


From ekr@rtfm.com  Mon Mar 12 15:44:40 2012
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From: Eric Rescorla <ekr@rtfm.com>
Date: Mon, 12 Mar 2012 15:43:58 -0700
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Subject: Re: [rtcweb] DoS attack despite ICE/STUN consent verification
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On Mon, Mar 12, 2012 at 3:41 PM, Magnus Westerlund
<magnus.westerlund@ericsson.com> wrote:
> Hi,
>
> Yes, your understanding appears correct. I was maybe more considering of
> using this attack against SIP clients, especially the kind that can't
> move out of the way easily. Like call centers etc. In that context being
> able to direct a 1 mbps video stream onto a audio receiver works
> excellent if your goal is to just DoS the path.

Yep. Though of course in the specific case of a call center, since it
auto-answers you can just mount the attack with independent calls....


> But I do agree that consent freshness appears to be an important method
> for mitigating this. It also seems to create a requirement on that
> mechanism. It can't be something that the far end can trick the target
> end-point to generate just by default. It needs to be something where
> the end-point takes a decision to respond to each.

Yeah. :(
Maybe you, I, Dan, and anyone else who is interested can brainstorm
on this a bit in Paris.

-Ekr

> Cheers
>
> Magnus
>
> On 2012-03-12 22:24, Eric Rescorla wrote:
>> I want to make sure I understand this attack correctly.
>>
>> Alice is making a call to Zed and either Zed is the attacker or the atta=
cker
>> somehow captures Alice's media parameters. The attacker then uses
>> Alice's offer (and her parameters) to start ICE handshaking on a pile
>> of Web browsers that he controls. They all start ICE and since the
>> checks with Alice will succeed, then the attacker has apparent consent
>> to send as much traffic as he wants to Alice from each machine.
>> Do I have this right?
>>
>> If I understand this correctly, then the severity of the attack seems to
>> be limited since Alice has to actually be calling someone that is
>> malicious or that has very tight control of the network, and soon
>> after Alice hangs up the phone, the problem goes away. (Assuming
>> consent freshness).
>>
>> As Dan notes, the interaction with consent freshness seems like a key
>> mitigation here. Moreover, I think this suggests that even if integrity
>> is not used, as in draft-muthu, the ufrags should both be present
>> (and unmodifiable from the JS). This would allow a client to discard
>> ICE associations with ufrags that do not match whichever ICE
>> association got selected for the call.
>>
>> -Ekr
>>
>>
>> On Mon, Feb 6, 2012 at 6:27 AM, Magnus Westerlund
>> <magnus.westerlund@ericsson.com> wrote:
>>> On 2012-02-06 11:19, Oscar Ohlsson wrote:
>>>>
>>>>
>>>>> -----Original Message-----
>>>>> From: rtcweb-bounces@ietf.org
>>>>> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Magnus Westerlund
>>>>> Sent: den 1 februari 2012 23:31
>>>>> To: rtcweb@ietf.org
>>>>> Subject: [rtcweb] DoS attack despite ICE/STUN consent verification
>>>>>
>>>>> Hi,
>>>>>
>>>>> As I brought up in yesterdays meeting I think there exist an
>>>>> attack that should be considered. I don't think it is super
>>>>> strong, but not without issues.
>>>>>
>>>>> Setting:
>>>>>
>>>>> Webserver is evil and have a target Alice. Alice can be a
>>>>> legacy device, like a SIP/SDP video phone. It knows the
>>>>> target is doing ICE and have Alice's ufrag and password and
>>>>> candidates. The server would like to hammer this target by
>>>>> utilizing its set of connected WebRTC clients (bob, charlie
>>>>> etc.). Thus it would like to ensure that it gets the browser
>>>>> to start sending RTP media when creating a PeerConnection in
>>>>> the clients.
>>>>>
>>>>> So the server gets the client to open PeerConnections towards
>>>>> the ICE parameters from the target (Alice).
>>>>>
>>>>> In default ICE behavior there is an optimization to allow the Offerer
>>>>> (Alice) to respond to STUN binding requests from Bob without
>>>>> having received an SDP answer from Bob. The STUN binding
>>>>> request must only have the Alice's ufrag and its password.
>>>>> This enables that Webrtc clients can get passed the data
>>>>> consent step with ICE capable end-point for which the ICE
>>>>> parameters can be retrieved.
>>>>>
>>>>> So from Alice perspective this attack will look like its
>>>>> offer got forked to many peers based on the many different
>>>>> peers that send STUN messages. The webserver could even
>>>>> provide SIP/SDP answers to alice with the webclients. But
>>>>> that is not necessary but is one variation of this.
>>>>
>>>>
>>>> Correct me if I'm wrong, but I believe the webserver must provide the
>>> SIP/SDP answers. As far as I can tell there are always two connectivity
>>> checks, one for each direction (Section 2.2 RFC 5245):
>>>
>>>> =A0 =A0L =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0R
>>>> =A0 =A0- =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0-
>>>> =A0 =A0STUN request -> =A0 =A0 =A0 =A0 =A0 =A0 \ =A0L's
>>>> =A0 =A0 =A0 =A0 =A0 =A0 =A0<- STUN response =A0/ =A0check
>>>>
>>>> =A0 =A0 =A0 =A0 =A0 =A0 =A0 <- STUN request =A0\ =A0R's
>>>> =A0 =A0STUN response -> =A0 =A0 =A0 =A0 =A0 =A0/ =A0check
>>>>
>>>> Alice would automatically respond to the remote side's check.
>>>> However, in order to genereate her own check (R's check in the
>>>> diagram) she would need to have received the remote side's password.
>>>> According to Section 7.1.2.3. in RFC 5245:
>>>>
>>>> "A connectivity check from R to L utilizes the username LFRAG:RFRAG
>>>> and a password of LPASS."
>>>>
>>>> If this interpretation is correct then counter-measure A below would
>>>> always be a possibility.
>>>>
>>>
>>> Oscar,
>>>
>>> From L's perspective, the process of nomination and validation can
>>> complete without R having succeeded. Thus L (WebRTC Browser) can start
>>> send media to R without R having successfully validated any return path
>>> from R to L.
>>>
>>> Thus, I don't see how A could be made to reliably work as a mitigation.
>>>
>>> In addition are there anyone who actually have this type of attack
>>> protection in their SIP stacks or SIP proxies?
>>>
>>> Cheers
>>>
>>> Magnus Westerlund
>>>
>>> ----------------------------------------------------------------------
>>> Multimedia Technologies, Ericsson Research EAB/TVM
>>> ----------------------------------------------------------------------
>>> Ericsson AB =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Phone =A0+46 10 7148287
>>> F=E4r=F6gatan 6 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Mobile +46 73 0949079
>>> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
>>> ----------------------------------------------------------------------
>>>
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>
>
> --
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Phone =A0+46 10 7148287
> F=E4r=F6gatan 6 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>

From ekr@rtfm.com  Mon Mar 12 16:06:44 2012
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From: Eric Rescorla <ekr@rtfm.com>
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Subject: [rtcweb] Updated security documents
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I have submitted new versions of:

* Security Considerations for RTC-Web
  http://tools.ietf.org/html/draft-ietf-rtcweb-security-02

  Primary change was to remove Appendix A (now mostly
  draft-ietf-rtcweb-security-arch-01) and to harmonize
  with that document.


* RTCWEB Security Architecture
  http://tools.ietf.org/html/draft-ietf-rtcweb-security-arch-01

  Updated to reflect the decisions made in the interim. Note
  that I did not try to specify a new STUN consent check b/c
  Perumal, Kaplan, and Wing got there first. If we have
  comments about that mechanism we should probably work them
  into that draft.


* RTCWEB Generic Identity Provider Interface
  http://tools.ietf.org/html/draft-rescorla-rtcweb-generic-idp-00

  Minor editorial updates.

-Ekr

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Temporary Maximum Media Stream Bit Rate Request (TMMBR, RFC5104)
should address the concerns with bandwidth exceeding expected limits.
That is, if this can be mandated (see below).

Rate limiting ICE nominations should limit the number of hosts that
can send media.  That is, you might have to limit the rate at which
your peer can appear to "move".  Then you only expose yourself to
something like ceil((time between "moves" + 1) / (maximum time between
consent checks, including timeout)) times the bandwidth that a host
can generate in your direction.  In order to do so, you have to fail
to respond to (nominating) connectivity checks under certain
conditions.

All this gets much more sticky and disgusting if you can't reasonably
mandate TMMBR and especially consent freshness checking.  If you are
expecting a SIP end host to be able to take media directly from
browsers - without supporting any of these features - then let the
brainstorming begin.

--Martin

On 12 March 2012 15:43, Eric Rescorla <ekr@rtfm.com> wrote:
> On Mon, Mar 12, 2012 at 3:41 PM, Magnus Westerlund
> <magnus.westerlund@ericsson.com> wrote:
>> Hi,
>>
>> Yes, your understanding appears correct. I was maybe more considering of
>> using this attack against SIP clients, especially the kind that can't
>> move out of the way easily. Like call centers etc. In that context being
>> able to direct a 1 mbps video stream onto a audio receiver works
>> excellent if your goal is to just DoS the path.
>
> Yep. Though of course in the specific case of a call center, since it
> auto-answers you can just mount the attack with independent calls....
>
>
>> But I do agree that consent freshness appears to be an important method
>> for mitigating this. It also seems to create a requirement on that
>> mechanism. It can't be something that the far end can trick the target
>> end-point to generate just by default. It needs to be something where
>> the end-point takes a decision to respond to each.
>
> Yeah. :(
> Maybe you, I, Dan, and anyone else who is interested can brainstorm
> on this a bit in Paris.
>
> -Ekr
>
>> Cheers
>>
>> Magnus
>>
>> On 2012-03-12 22:24, Eric Rescorla wrote:
>>> I want to make sure I understand this attack correctly.
>>>
>>> Alice is making a call to Zed and either Zed is the attacker or the att=
acker
>>> somehow captures Alice's media parameters. The attacker then uses
>>> Alice's offer (and her parameters) to start ICE handshaking on a pile
>>> of Web browsers that he controls. They all start ICE and since the
>>> checks with Alice will succeed, then the attacker has apparent consent
>>> to send as much traffic as he wants to Alice from each machine.
>>> Do I have this right?
>>>
>>> If I understand this correctly, then the severity of the attack seems t=
o
>>> be limited since Alice has to actually be calling someone that is
>>> malicious or that has very tight control of the network, and soon
>>> after Alice hangs up the phone, the problem goes away. (Assuming
>>> consent freshness).
>>>
>>> As Dan notes, the interaction with consent freshness seems like a key
>>> mitigation here. Moreover, I think this suggests that even if integrity
>>> is not used, as in draft-muthu, the ufrags should both be present
>>> (and unmodifiable from the JS). This would allow a client to discard
>>> ICE associations with ufrags that do not match whichever ICE
>>> association got selected for the call.
>>>
>>> -Ekr
>>>
>>>
>>> On Mon, Feb 6, 2012 at 6:27 AM, Magnus Westerlund
>>> <magnus.westerlund@ericsson.com> wrote:
>>>> On 2012-02-06 11:19, Oscar Ohlsson wrote:
>>>>>
>>>>>
>>>>>> -----Original Message-----
>>>>>> From: rtcweb-bounces@ietf.org
>>>>>> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Magnus Westerlund
>>>>>> Sent: den 1 februari 2012 23:31
>>>>>> To: rtcweb@ietf.org
>>>>>> Subject: [rtcweb] DoS attack despite ICE/STUN consent verification
>>>>>>
>>>>>> Hi,
>>>>>>
>>>>>> As I brought up in yesterdays meeting I think there exist an
>>>>>> attack that should be considered. I don't think it is super
>>>>>> strong, but not without issues.
>>>>>>
>>>>>> Setting:
>>>>>>
>>>>>> Webserver is evil and have a target Alice. Alice can be a
>>>>>> legacy device, like a SIP/SDP video phone. It knows the
>>>>>> target is doing ICE and have Alice's ufrag and password and
>>>>>> candidates. The server would like to hammer this target by
>>>>>> utilizing its set of connected WebRTC clients (bob, charlie
>>>>>> etc.). Thus it would like to ensure that it gets the browser
>>>>>> to start sending RTP media when creating a PeerConnection in
>>>>>> the clients.
>>>>>>
>>>>>> So the server gets the client to open PeerConnections towards
>>>>>> the ICE parameters from the target (Alice).
>>>>>>
>>>>>> In default ICE behavior there is an optimization to allow the Offere=
r
>>>>>> (Alice) to respond to STUN binding requests from Bob without
>>>>>> having received an SDP answer from Bob. The STUN binding
>>>>>> request must only have the Alice's ufrag and its password.
>>>>>> This enables that Webrtc clients can get passed the data
>>>>>> consent step with ICE capable end-point for which the ICE
>>>>>> parameters can be retrieved.
>>>>>>
>>>>>> So from Alice perspective this attack will look like its
>>>>>> offer got forked to many peers based on the many different
>>>>>> peers that send STUN messages. The webserver could even
>>>>>> provide SIP/SDP answers to alice with the webclients. But
>>>>>> that is not necessary but is one variation of this.
>>>>>
>>>>>
>>>>> Correct me if I'm wrong, but I believe the webserver must provide the
>>>> SIP/SDP answers. As far as I can tell there are always two connectivit=
y
>>>> checks, one for each direction (Section 2.2 RFC 5245):
>>>>
>>>>> =C2=A0 =C2=A0L =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0R
>>>>> =C2=A0 =C2=A0- =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0-
>>>>> =C2=A0 =C2=A0STUN request -> =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 \ =C2=A0L's
>>>>> =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0<- STUN response =C2=
=A0/ =C2=A0check
>>>>>
>>>>> =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 <- STUN request =C2=
=A0\ =C2=A0R's
>>>>> =C2=A0 =C2=A0STUN response -> =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0/ =C2=A0check
>>>>>
>>>>> Alice would automatically respond to the remote side's check.
>>>>> However, in order to genereate her own check (R's check in the
>>>>> diagram) she would need to have received the remote side's password.
>>>>> According to Section 7.1.2.3. in RFC 5245:
>>>>>
>>>>> "A connectivity check from R to L utilizes the username LFRAG:RFRAG
>>>>> and a password of LPASS."
>>>>>
>>>>> If this interpretation is correct then counter-measure A below would
>>>>> always be a possibility.
>>>>>
>>>>
>>>> Oscar,
>>>>
>>>> From L's perspective, the process of nomination and validation can
>>>> complete without R having succeeded. Thus L (WebRTC Browser) can start
>>>> send media to R without R having successfully validated any return pat=
h
>>>> from R to L.
>>>>
>>>> Thus, I don't see how A could be made to reliably work as a mitigation=
.
>>>>
>>>> In addition are there anyone who actually have this type of attack
>>>> protection in their SIP stacks or SIP proxies?
>>>>
>>>> Cheers
>>>>
>>>> Magnus Westerlund
>>>>
>>>> ----------------------------------------------------------------------
>>>> Multimedia Technologies, Ericsson Research EAB/TVM
>>>> ----------------------------------------------------------------------
>>>> Ericsson AB =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0| P=
hone =C2=A0+46 10 7148287
>>>> F=C3=A4r=C3=B6gatan 6 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0=
 =C2=A0| Mobile +46 73 0949079
>>>> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
>>>> ----------------------------------------------------------------------
>>>>
>>>> _______________________________________________
>>>> rtcweb mailing list
>>>> rtcweb@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>
>>
>>
>> --
>>
>> Magnus Westerlund
>>
>> ----------------------------------------------------------------------
>> Multimedia Technologies, Ericsson Research EAB/TVM
>> ----------------------------------------------------------------------
>> Ericsson AB =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0| Pho=
ne =C2=A0+46 10 7148287
>> F=C3=A4r=C3=B6gatan 6 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0| Mobile +46 73 0949079
>> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
>> ----------------------------------------------------------------------
>>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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Subject: Re: [rtcweb] DoS attack despite ICE/STUN consent verification
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My major question on this thread, which I have been watching with interest:

Why do we care about a DoS attack  against a single client?  (I mean 
specifically DoS, not any other security attack.)

It appears to me that there are somewhat simpler and more effective DoS 
methods for this case: 1) cut the wire to the client's device or 2) jam 
WiFi or other wireless signal.

Or did I misunderstand the problem?

Igor

On 3/12/2012 6:41 PM, Magnus Westerlund wrote:
> Hi,
>
> Yes, your understanding appears correct. I was maybe more considering of
> using this attack against SIP clients, especially the kind that can't
> move out of the way easily. Like call centers etc. In that context being
> able to direct a 1 mbps video stream onto a audio receiver works
> excellent if your goal is to just DoS the path.
>
> But I do agree that consent freshness appears to be an important method
> for mitigating this. It also seems to create a requirement on that
> mechanism. It can't be something that the far end can trick the target
> end-point to generate just by default. It needs to be something where
> the end-point takes a decision to respond to each.
>
> Cheers
>
> Magnus
>
> On 2012-03-12 22:24, Eric Rescorla wrote:
>> I want to make sure I understand this attack correctly.
>>
>> Alice is making a call to Zed and either Zed is the attacker or the attacker
>> somehow captures Alice's media parameters. The attacker then uses
>> Alice's offer (and her parameters) to start ICE handshaking on a pile
>> of Web browsers that he controls. They all start ICE and since the
>> checks with Alice will succeed, then the attacker has apparent consent
>> to send as much traffic as he wants to Alice from each machine.
>> Do I have this right?
>>
>> If I understand this correctly, then the severity of the attack seems to
>> be limited since Alice has to actually be calling someone that is
>> malicious or that has very tight control of the network, and soon
>> after Alice hangs up the phone, the problem goes away. (Assuming
>> consent freshness).
>>
>> As Dan notes, the interaction with consent freshness seems like a key
>> mitigation here. Moreover, I think this suggests that even if integrity
>> is not used, as in draft-muthu, the ufrags should both be present
>> (and unmodifiable from the JS). This would allow a client to discard
>> ICE associations with ufrags that do not match whichever ICE
>> association got selected for the call.
>>
>> -Ekr
>>
>>
>> On Mon, Feb 6, 2012 at 6:27 AM, Magnus Westerlund
>> <magnus.westerlund@ericsson.com>  wrote:
>>> On 2012-02-06 11:19, Oscar Ohlsson wrote:
>>>>
>>>>> -----Original Message-----
>>>>> From: rtcweb-bounces@ietf.org
>>>>> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Magnus Westerlund
>>>>> Sent: den 1 februari 2012 23:31
>>>>> To: rtcweb@ietf.org
>>>>> Subject: [rtcweb] DoS attack despite ICE/STUN consent verification
>>>>>
>>>>> Hi,
>>>>>
>>>>> As I brought up in yesterdays meeting I think there exist an
>>>>> attack that should be considered. I don't think it is super
>>>>> strong, but not without issues.
>>>>>
>>>>> Setting:
>>>>>
>>>>> Webserver is evil and have a target Alice. Alice can be a
>>>>> legacy device, like a SIP/SDP video phone. It knows the
>>>>> target is doing ICE and have Alice's ufrag and password and
>>>>> candidates. The server would like to hammer this target by
>>>>> utilizing its set of connected WebRTC clients (bob, charlie
>>>>> etc.). Thus it would like to ensure that it gets the browser
>>>>> to start sending RTP media when creating a PeerConnection in
>>>>> the clients.
>>>>>
>>>>> So the server gets the client to open PeerConnections towards
>>>>> the ICE parameters from the target (Alice).
>>>>>
>>>>> In default ICE behavior there is an optimization to allow the Offerer
>>>>> (Alice) to respond to STUN binding requests from Bob without
>>>>> having received an SDP answer from Bob. The STUN binding
>>>>> request must only have the Alice's ufrag and its password.
>>>>> This enables that Webrtc clients can get passed the data
>>>>> consent step with ICE capable end-point for which the ICE
>>>>> parameters can be retrieved.
>>>>>
>>>>> So from Alice perspective this attack will look like its
>>>>> offer got forked to many peers based on the many different
>>>>> peers that send STUN messages. The webserver could even
>>>>> provide SIP/SDP answers to alice with the webclients. But
>>>>> that is not necessary but is one variation of this.
>>>>
>>>> Correct me if I'm wrong, but I believe the webserver must provide the
>>> SIP/SDP answers. As far as I can tell there are always two connectivity
>>> checks, one for each direction (Section 2.2 RFC 5245):
>>>
>>>>     L                        R
>>>>     -                        -
>>>>     STUN request ->              \  L's
>>>>               <- STUN response  /  check
>>>>
>>>>                <- STUN request  \  R's
>>>>     STUN response ->             /  check
>>>>
>>>> Alice would automatically respond to the remote side's check.
>>>> However, in order to genereate her own check (R's check in the
>>>> diagram) she would need to have received the remote side's password.
>>>> According to Section 7.1.2.3. in RFC 5245:
>>>>
>>>> "A connectivity check from R to L utilizes the username LFRAG:RFRAG
>>>> and a password of LPASS."
>>>>
>>>> If this interpretation is correct then counter-measure A below would
>>>> always be a possibility.
>>>>
>>> Oscar,
>>>
>>>  From L's perspective, the process of nomination and validation can
>>> complete without R having succeeded. Thus L (WebRTC Browser) can start
>>> send media to R without R having successfully validated any return path
>>> from R to L.
>>>
>>> Thus, I don't see how A could be made to reliably work as a mitigation.
>>>
>>> In addition are there anyone who actually have this type of attack
>>> protection in their SIP stacks or SIP proxies?
>>>
>>> Cheers
>>>
>>> Magnus Westerlund
>>>
>>> ----------------------------------------------------------------------
>>> Multimedia Technologies, Ericsson Research EAB/TVM
>>> ----------------------------------------------------------------------
>>> Ericsson AB                | Phone  +46 10 7148287
>>> Färögatan 6                | Mobile +46 73 0949079
>>> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
>>> ----------------------------------------------------------------------
>>>
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>

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Subject: Re: [rtcweb] DoS attack despite ICE/STUN consent verification
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On 13 March 2012 09:46, Igor Faynberg <igor.faynberg@alcatel-lucent.com> wr=
ote:
> Why do we care about a DoS attack =C2=A0against a single client? =C2=A0(I=
 mean
> specifically DoS, not any other security attack.)

The concern here is that it isn't that hard to get someone to visit
your website.  By the same token, it should be easy to get someone to
call (you).

Once that "call" is up, you can direct the media anywhere.  However,
if the source of the media has some incentive to not send media - and
browsers certainly do - then they want to check that their media is
acceptable before sending it.  Hence the consent mechanisms.

That adds an another challenge: how do you get the target to consent
to receipt of media from your drones?  Obviously, if they consent to
receiving one call, and you can use that to launch the media from
multiple calls toward them, then you have an attack.

> It appears to me that there are somewhat simpler and more effective DoS
> methods for this case: 1) cut the wire to the client's device or 2) jam W=
iFi
> or other wireless signal.

Stabbing your target with a knife also serves as a reasonable DoS
method too.  It's all about finding an appropriate defense for
plausible threats.

From martin.thomson@gmail.com  Tue Mar 13 11:30:56 2012
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One concern that I have with rtcweb is the amount of time it takes to
start a session.  Speaking as an Internet user well accustomed to long
round trip times, the fact that it could take up to 7 to get media
bothers me a little.  It turns out that this is unnecessary, with a
few tweaks.

I've written a draft on the subject:

  https://docs.google.com/document/d/1UZjCnqDmHcAOsETLZ4CRTmpHkm97PXMEebiRRqe8nXE/edit

I don't want to add to the reviewing burden or distract folks from
more important tasks.  But I've had some interest.  I plan to submit
this once draft submissions reopen.

--Martin

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On 3/13/2012 2:02 PM, Martin Thomson wrote:
> ...
> Stabbing your target with a knife also serves as a reasonable DoS
> method too.  It's all about finding an appropriate defense for
> plausible threats.


Well, this was one option I decided not to list... :)

But, really, I just don't see a reason why anyone would want to waste 
resources on a DoS attack against a /person/.
I am sure I am missing a bigger picture.  I am also sure, I will start 
understanding these things better after a drink or two, or three.

To this end, Eric, please count me among the interested parties  for the 
off-line discussion in Paris!

Igor



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    On 3/13/2012 2:02 PM, Martin Thomson wrote:
    <blockquote
cite="mid:CABkgnnU5GAE4aXrJn_PY0SfAUnjp38OUHR31LC7ntibwG1+pOg@mail.gmail.com"
      type="cite">...<br>
      <pre wrap="">Stabbing your target with a knife also serves as a reasonable DoS
method too.  It's all about finding an appropriate defense for
plausible threats.</pre>
    </blockquote>
    <br>
    <br>
    Well, this was one option I decided not to list... :)<br>
    <br>
    But, really, I just don't see a reason why anyone would want to
    waste resources on a DoS attack against a <i>person</i>.Â  <br>
    I am sure I am missing a bigger picture.Â  I am also sure, I will
    start understanding these things better after a drink or two, or
    three.<br>
    <br>
    To this end, Eric, please count me among the interested partiesÂ  for
    the off-line discussion in Paris!<br>
    <br>
    Igor<br>
    <br>
    <br>
  </body>
</html>

--------------000409030007070702020908--

From dwing@cisco.com  Wed Mar 14 09:29:56 2012
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From: "Dan Wing" <dwing@cisco.com>
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> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Martin Thomson
> Sent: Tuesday, March 13, 2012 11:31 AM
> To: rtcweb@ietf.org
> Subject: [rtcweb] Using DTLS for ICE
> 
> One concern that I have with rtcweb is the amount of time it takes to
> start a session.  Speaking as an Internet user well accustomed to long
> round trip times, the fact that it could take up to 7 to get media
> bothers me a little.

As EKR commented in your googledocs, the DTLS cookie exchange isn't
necessary (or useful) if you're already doing ICE, so the un-optimized
case is at least one round-trip less than that.

> It turns out that this is unnecessary, with a
> few tweaks.
> 
> I've written a draft on the subject:
> 
> 
> https://docs.google.com/document/d/1UZjCnqDmHcAOsETLZ4CRTmpHkm97PXMEebi
> RRqe8nXE/edit

It seems like a good idea worth considering.

-d


> I don't want to add to the reviewing burden or distract folks from
> more important tasks.  But I've had some interest.  I plan to submit
> this once draft submissions reopen.
> 
> --Martin
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From martin.thomson@gmail.com  Wed Mar 14 13:33:48 2012
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On 14 March 2012 09:29, Dan Wing <dwing@cisco.com> wrote:
> As EKR commented in your googledocs, the DTLS cookie exchange isn't
> necessary (or useful) if you're already doing ICE, so the un-optimized
> case is at least one round-trip less than that.

Not you too.  As I told EKR, that's mostly just there for effect.  And
now you've gone an spoiled the effect.

But I did think that there is a window of opportunity for DoS if you
allow for the possibility of spoofed source IP+port.  I mean, that's
the scenario that the cookie was invented for, right?  After all, you
have to store all those ClientHello messages, valid or not.  Without
the cookie, you have no assurance that they are valid.  The STUN
messages have MESSAGE-INTEGRITY, but the ClientHello could have come
from anyone.

I guess that checking source IP+port would be sufficient for most
folks.  That does mean that you effectively force the attacker to
obtain the candidate pair details before he can DoS you, and the
window of opportunity is fairly narrow.

--Martin

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From: "Dan Wing" <dwing@cisco.com>
To: "'Martin Thomson'" <martin.thomson@gmail.com>
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Subject: Re: [rtcweb] Using DTLS for ICE
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> -----Original Message-----
> From: Martin Thomson [mailto:martin.thomson@gmail.com]
> Sent: Wednesday, March 14, 2012 1:34 PM
> To: Dan Wing
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Using DTLS for ICE
> 
> On 14 March 2012 09:29, Dan Wing <dwing@cisco.com> wrote:
> > As EKR commented in your googledocs, the DTLS cookie exchange isn't
> > necessary (or useful) if you're already doing ICE, so the un-
> optimized
> > case is at least one round-trip less than that.
> 
> Not you too.  As I told EKR, that's mostly just there for effect.  And
> now you've gone an spoiled the effect.
> 
> But I did think that there is a window of opportunity for DoS if you
> allow for the possibility of spoofed source IP+port.  I mean, that's
> the scenario that the cookie was invented for, right?  After all, you
> have to store all those ClientHello messages, valid or not.  Without
> the cookie, you have no assurance that they are valid.  The STUN
> messages have MESSAGE-INTEGRITY, but the ClientHello could have come
> from anyone.
>
> I guess that checking source IP+port would be sufficient for most
> folks.

Exactly.  After doing ICE with that IP+port, then DTLS handshakes
can happen with that IP+port.  If ICE hasn't been done with that 
IP+port, DTLS handshakes should not happen with that IP+port.

But it's a nit.  I'm not worried about that nit of how ICE and
DTLS can work best together.  That should be written down somewhere,
but it is just background in your document.  I don't want to
concentrate on that.

>  That does mean that you effectively force the attacker to
> obtain the candidate pair details before he can DoS you, and the
> window of opportunity is fairly narrow.

You have proposed a useful optimization that 
reduces the round trips by, essentially, using the initial DTLS 
handshake messages in lieu of the ICE handshake messages.
What you are doing is collapsing the two.  I think that is a good
optimization, and I would like to pursue it.   Another approach, 
which I would like to discuss/compare with your proposal, is 
to put the initial DTLS handshake messages into the ICE 
messages themselves, as new STUN attributes.  This changes how
DTLS does its handshake, but your proposal changes how ICE
does its handshake.  There are pros/cons of either change.
Might be worth discussing over cookies.  EKR likes cookies.

-d




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Subject: [rtcweb] "PeerConnection" API would manage mapping information between media streams and DOM elements (Re: I'd like to discuss my proposal (jingle-web) at next WG session)
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Hi Bernard,

Jingle extensions are discussed in XSF, and DOM specification would be =
discussed in
W3C, as you says.

But if there are any numbers of media streams between peers, and if =
there are any
numbers of "<video/>" tags or "<div/>" tags with <video/> tag, (both =
case is possible
in Jingle spec. and HTML spec.) how can both browsers map between any =
numbers
of media/RTP streams and "<div/>" elements in DOM in consistent manner. =
I think
mapping information between media/RTP streams and HTML/DOM elements =
should
be signaled.

Current JSEP draft (draft-ietf-rtcweb-jsep-00) specified to make =
abstracted signaling
by JavaScript between both peers, and in this specification signaling =
procedures and
signaling state machine is well defined.  But the current draft =
specifies these mapping
information?  When set of media/RTP streams changes, correspondent =
mapping
information should be updated and signaled at the same time, I think.

This issue is more important in the use-case of multi user conference, =
to keep same
view on all participants' web browser, mapping information between =
media/RTP
streams and HTML/DOM elements should keep consistency between all =
participant.

Maybe how to store this mapping information is specified in =
PeerConnection API, and
how to get media/RTP stream identification information, how to set this =
identification
information to DOM elements in DOM tree, and vice versa.

My I-D describes how to signal mapping information in Jingle, and =
proposes two
mapping manner, and this proposal can be applied to multi user =
conference too.
I think JSEP draft can expand to manage this mapping information.

Cheers,
Hiro

On 2012/03/09, at 7:48, Bernard Aboba wrote:

> In general, Jingle extensions are within the purview of the XSF, =
rather than IETF and the DOM is owned by W3C.=20
>=20
> So other than the question of mandatory signaling implementation in =
browsers (which as Ted noted, has already been discussed), it seems to =
me that much of this document is out of scope not only of RTCWEB, but =
the IETF. =20
>=20
>=20
>=20
> > Date: Thu, 8 Mar 2012 08:27:47 -0800
> > From: ted.ietf@gmail.com
> > To: hiro.suzuki@d3communications.jp
> > CC: rtcweb@ietf.org
> > Subject: Re: [rtcweb] I'd like to discuss my proposal (jingle-web) =
at next	WG session
> >=20
> > Hi Suzuki-san,
> >=20
> > Given that this apparently revisits some decisions that were already
> > made, the chairs would like to see working group discussion of the
> > draft before allocating agenda time to it for the upcoming meeting.
> >=20
> > best regards,
> >=20
> > Ted Hardie for the Chairs
> >=20
> > On Wed, Feb 29, 2012 at 7:34 PM, HIro Suzuki
> > <hiro.suzuki@d3communications.jp> wrote:
> > > Hello RTCWeb WG members,
> > >
> > >
> > > It's my first post to this ML, and at Feb 28th, I submitted my ID
> > > from IETF ID submit tool. This ID's file name is
> > > "draft-suzuki-rtcweb-jingle-web-00.txt" and this ID provides my =
proposal
> > > to realize one of solutions of RTCWeb by using XMPP Jingle as a =
signaling
> > > protocol.
> > >
> > > Following paragraph is abstract of my ID, and I'd like to discuss =
my
> > > proposal at RTCWeb WG session in next Paris meeting.
> > >
> > >
> > > ---
> > > Abstract:
> > >  XMPP Jingle specification defines an XMPP protocol extension for
> > >  managing peer-to-peer media sessions between two users. And XMPP
> > >  Jingle can manage multi contents simultaneously in one Jingle
> > >  stream, but in the XMPP world there is no common way to layout
> > >  variable multi contents on each display. In this document, a
> > >  solution to this issue is provided by using Web browser's layout
> > >  function.
> > >
> > >  This document describes a new concept to realize one of solutions
> > >  of RTCWeb. Of course, &quot;Web Browser&quot; is used for Web =
application's
> > >  frontend for real-time communication, and XMPP Jingle =
specification
> > >  (XEP-166) is used as signaling protocol. And a simple mapping
> > >  manner between jingle contents and HTML graphical elements =
enables
> > >  to unite Web browser&#39;s layout function and Jingle&#39;s media =
content
> > >  management function to realize RTCWeb functions.
> > > ---
> > >
> > >
> > > Best Regards,
> > > Yoshihiro Suzuki
> > > (hiro.suzuki@d3communications.jp)
> > > _______________________________________________
> > > rtcweb mailing list
> > > rtcweb@ietf.org
> > > https://www.ietf.org/mailman/listinfo/rtcweb
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb


--Apple-Mail=_06057822-AB92-4FA6-8963-5D47C125D05E
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<html><head><base href=3D"x-msg://335/"></head><body style=3D"word-wrap: =
break-word; -webkit-nbsp-mode: space; -webkit-line-break: =
after-white-space; "><div>Hi Bernard,</div><div><br></div><div>Jingle =
extensions are discussed in XSF, and DOM specification would be =
discussed in</div><div>W3C, as you says.</div><div><br></div><div>But if =
there are any numbers of media streams between peers,&nbsp;and if there =
are&nbsp;any</div><div>numbers of "&lt;video/&gt;" tags or =
"&lt;div/&gt;" tags&nbsp;with &lt;video/&gt; tag, (both case is =
possible</div><div>in Jingle spec. and HTML spec.)&nbsp;how can both =
browsers&nbsp;map between any numbers</div><div>of media/RTP streams and =
"&lt;div/&gt;" elements in DOM in&nbsp;consistent manner. I =
think</div><div>mapping information between media/RTP streams and =
HTML/DOM elements should</div><div>be =
signaled.</div><div><br></div><div>Current JSEP draft =
(draft-ietf-rtcweb-jsep-00) specified to make abstracted =
signaling</div><div>by JavaScript&nbsp;between both peers, and in this =
specification signaling procedures and</div><div>signaling state machine =
is well defined. &nbsp;But the current draft specifies these =
mapping</div><div>information? &nbsp;When set of media/RTP streams =
changes, correspondent mapping</div><div>information should be updated =
and signaled at the same time, I think.</div><div><br></div><div>This =
issue is more important in the use-case of multi user conference, to =
keep same</div><div>view on all participants' web browser, =
mapping&nbsp;information between media/RTP</div><div>streams =
and&nbsp;HTML/DOM elements should keep&nbsp;consistency between all =
participant.</div><div><br></div><div>Maybe how to store this mapping =
information is specified in PeerConnection API, and</div><div>how to get =
media/RTP stream identification information, how to set this =
identification</div><div>information to DOM elements in DOM tree, and =
vice versa.</div><div><br></div><div>My I-D describes how to signal =
mapping information in Jingle, and proposes =
two</div><div>mapping&nbsp;manner, and this proposal can be applied to =
multi user conference too.</div><div>I think JSEP draft can expand to =
manage this mapping =
information.</div><div><br></div><div>Cheers,</div><div>Hiro</div><br><div=
><div>On 2012/03/09, at 7:48, Bernard Aboba wrote:</div><br =
class=3D"Apple-interchange-newline"><blockquote type=3D"cite"><span =
class=3D"Apple-style-span" style=3D"border-collapse: separate; =
font-family: Helvetica; font-style: normal; font-variant: normal; =
font-weight: normal; letter-spacing: normal; line-height: normal; =
orphans: 2; text-align: -webkit-auto; text-indent: 0px; text-transform: =
none; white-space: normal; widows: 2; word-spacing: 0px; =
-webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; font-size: medium; "><div =
class=3D"hmmessage" style=3D"font-size: 10pt; font-family: Tahoma; =
"><div dir=3D"ltr">In general, Jingle extensions are within the purview =
of the XSF, rather than IETF and the DOM is owned by W3C.<span =
class=3D"Apple-converted-space">&nbsp;</span><br><br>So other than the =
question of mandatory signaling implementation in browsers (which as Ted =
noted, has already been discussed), it seems to me that much of this =
document is out of scope not only of RTCWEB, but the IETF.&nbsp;<span =
class=3D"Apple-converted-space">&nbsp;</span><br><br><br><br><div><div =
id=3D"SkyDrivePlaceholder"></div>&gt; Date: Thu, 8 Mar 2012 08:27:47 =
-0800<br>&gt; From:<span class=3D"Apple-converted-space">&nbsp;</span><a =
href=3D"mailto:ted.ietf@gmail.com">ted.ietf@gmail.com</a><br>&gt; =
To:<span class=3D"Apple-converted-space">&nbsp;</span><a =
href=3D"mailto:hiro.suzuki@d3communications.jp">hiro.suzuki@d3communicatio=
ns.jp</a><br>&gt; CC:<span class=3D"Apple-converted-space">&nbsp;</span><a=
 href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>&gt; Subject: =
Re: [rtcweb] I'd like to discuss my proposal (jingle-web) at next	=
WG session<br>&gt;<span =
class=3D"Apple-converted-space">&nbsp;</span><br>&gt; Hi =
Suzuki-san,<br>&gt;<span =
class=3D"Apple-converted-space">&nbsp;</span><br>&gt; Given that this =
apparently revisits some decisions that were already<br>&gt; made, the =
chairs would like to see working group discussion of the<br>&gt; draft =
before allocating agenda time to it for the upcoming =
meeting.<br>&gt;<span =
class=3D"Apple-converted-space">&nbsp;</span><br>&gt; best =
regards,<br>&gt;<span =
class=3D"Apple-converted-space">&nbsp;</span><br>&gt; Ted Hardie for the =
Chairs<br>&gt;<span class=3D"Apple-converted-space">&nbsp;</span><br>&gt; =
On Wed, Feb 29, 2012 at 7:34 PM, HIro Suzuki<br>&gt; &lt;<a =
href=3D"mailto:hiro.suzuki@d3communications.jp">hiro.suzuki@d3communicatio=
ns.jp</a>&gt; wrote:<br>&gt; &gt; Hello RTCWeb WG members,<br>&gt; =
&gt;<br>&gt; &gt;<br>&gt; &gt; It's my first post to this ML, and at Feb =
28th, I submitted my ID<br>&gt; &gt; from IETF ID submit tool. This ID's =
file name is<br>&gt; &gt; "draft-suzuki-rtcweb-jingle-web-00.txt" and =
this ID provides my proposal<br>&gt; &gt; to realize one of solutions of =
RTCWeb by using XMPP Jingle as a signaling<br>&gt; &gt; =
protocol.<br>&gt; &gt;<br>&gt; &gt; Following paragraph is abstract of =
my ID, and I'd like to discuss my<br>&gt; &gt; proposal at RTCWeb WG =
session in next Paris meeting.<br>&gt; &gt;<br>&gt; &gt;<br>&gt; &gt; =
---<br>&gt; &gt; Abstract:<br>&gt; &gt; &nbsp;XMPP Jingle specification =
defines an XMPP protocol extension for<br>&gt; &gt; &nbsp;managing =
peer-to-peer media sessions between two users. And XMPP<br>&gt; &gt; =
&nbsp;Jingle can manage multi contents simultaneously in one =
Jingle<br>&gt; &gt; &nbsp;stream, but in the XMPP world there is no =
common way to layout<br>&gt; &gt; &nbsp;variable multi contents on each =
display. In this document, a<br>&gt; &gt; &nbsp;solution to this issue =
is provided by using Web browser's layout<br>&gt; &gt; =
&nbsp;function.<br>&gt; &gt;<br>&gt; &gt; &nbsp;This document describes =
a new concept to realize one of solutions<br>&gt; &gt; &nbsp;of RTCWeb. =
Of course, &amp;quot;Web Browser&amp;quot; is used for Web =
application's<br>&gt; &gt; &nbsp;frontend for real-time communication, =
and XMPP Jingle specification<br>&gt; &gt; &nbsp;(XEP-166) is used as =
signaling protocol. And a simple mapping<br>&gt; &gt; &nbsp;manner =
between jingle contents and HTML graphical elements enables<br>&gt; &gt; =
&nbsp;to unite Web browser&amp;#39;s layout function and =
Jingle&amp;#39;s media content<br>&gt; &gt; &nbsp;management function to =
realize RTCWeb functions.<br>&gt; &gt; ---<br>&gt; &gt;<br>&gt; =
&gt;<br>&gt; &gt; Best Regards,<br>&gt; &gt; Yoshihiro Suzuki<br>&gt; =
&gt; (<a =
href=3D"mailto:hiro.suzuki@d3communications.jp">hiro.suzuki@d3communicatio=
ns.jp</a>)<br>&gt; &gt; =
_______________________________________________<br>&gt; &gt; rtcweb =
mailing list<br>&gt; &gt;<span =
class=3D"Apple-converted-space">&nbsp;</span><a =
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From christer.holmberg@ericsson.com  Thu Mar 15 06:04:40 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Eric Rescorla <ekr@rtfm.com>, Magnus Westerlund <magnus.westerlund@ericsson.com>
Date: Thu, 15 Mar 2012 14:04:35 +0100
Thread-Topic: [rtcweb] DoS attack despite ICE/STUN consent verification
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Subject: Re: [rtcweb] DoS attack despite ICE/STUN consent verification
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Hi,

>> Yes, your understanding appears correct. I was maybe more considering=20
>> of using this attack against SIP clients, especially the kind that=20
>> can't move out of the way easily. Like call centers etc. In that=20
>> context being able to direct a 1 mbps video stream onto a audio=20
>> receiver works excellent if your goal is to just DoS the path.
>
> Yep. Though of course in the specific case of a call center, since it aut=
o-answers you can just mount the attack with independent calls....
>
>> But I do agree that consent freshness appears to be an important=20
>> method for mitigating this. It also seems to create a requirement on=20
>> that mechanism. It can't be something that the far end can trick the=20
>> target end-point to generate just by default. It needs to be something=20
> where the end-point takes a decision to respond to each.
>
> Yeah. :(
> Maybe you, I, Dan, and anyone else who is interested can brainstorm on th=
is a bit in Paris.

I would also be interested to participate in such brainstorming.

Regards,

Christer


> Cheers
>
> Magnus
>
> On 2012-03-12 22:24, Eric Rescorla wrote:
>> I want to make sure I understand this attack correctly.
>>
>> Alice is making a call to Zed and either Zed is the attacker or the=20
>> attacker somehow captures Alice's media parameters. The attacker then=20
>> uses Alice's offer (and her parameters) to start ICE handshaking on a=20
>> pile of Web browsers that he controls. They all start ICE and since=20
>> the checks with Alice will succeed, then the attacker has apparent=20
>> consent to send as much traffic as he wants to Alice from each machine.
>> Do I have this right?
>>
>> If I understand this correctly, then the severity of the attack seems=20
>> to be limited since Alice has to actually be calling someone that is=20
>> malicious or that has very tight control of the network, and soon=20
>> after Alice hangs up the phone, the problem goes away. (Assuming=20
>> consent freshness).
>>
>> As Dan notes, the interaction with consent freshness seems like a key=20
>> mitigation here. Moreover, I think this suggests that even if=20
>> integrity is not used, as in draft-muthu, the ufrags should both be=20
>> present (and unmodifiable from the JS). This would allow a client to=20
>> discard ICE associations with ufrags that do not match whichever ICE=20
>> association got selected for the call.
>>
>> -Ekr
>>
>>
>> On Mon, Feb 6, 2012 at 6:27 AM, Magnus Westerlund=20
>> <magnus.westerlund@ericsson.com> wrote:
>>> On 2012-02-06 11:19, Oscar Ohlsson wrote:
>>>>
>>>>
>>>>> -----Original Message-----
>>>>> From: rtcweb-bounces@ietf.org
>>>>> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Magnus Westerlund
>>>>> Sent: den 1 februari 2012 23:31
>>>>> To: rtcweb@ietf.org
>>>>> Subject: [rtcweb] DoS attack despite ICE/STUN consent verification
>>>>>
>>>>> Hi,
>>>>>
>>>>> As I brought up in yesterdays meeting I think there exist an=20
>>>>> attack that should be considered. I don't think it is super=20
>>>>> strong, but not without issues.
>>>>>
>>>>> Setting:
>>>>>
>>>>> Webserver is evil and have a target Alice. Alice can be a legacy=20
>>>>> device, like a SIP/SDP video phone. It knows the target is doing=20
>>>>> ICE and have Alice's ufrag and password and candidates. The server=20
>>>>> would like to hammer this target by utilizing its set of connected=20
>>>>> WebRTC clients (bob, charlie etc.). Thus it would like to ensure=20
>>>>> that it gets the browser to start sending RTP media when creating=20
>>>>> a PeerConnection in the clients.
>>>>>
>>>>> So the server gets the client to open PeerConnections towards the=20
>>>>> ICE parameters from the target (Alice).
>>>>>
>>>>> In default ICE behavior there is an optimization to allow the=20
>>>>> Offerer
>>>>> (Alice) to respond to STUN binding requests from Bob without=20
>>>>> having received an SDP answer from Bob. The STUN binding request=20
>>>>> must only have the Alice's ufrag and its password.
>>>>> This enables that Webrtc clients can get passed the data consent=20
>>>>> step with ICE capable end-point for which the ICE parameters can=20
>>>>> be retrieved.
>>>>>
>>>>> So from Alice perspective this attack will look like its offer got=20
>>>>> forked to many peers based on the many different peers that send=20
>>>>> STUN messages. The webserver could even provide SIP/SDP answers to=20
>>>>> alice with the webclients. But that is not necessary but is one=20
>>>>> variation of this.
>>>>
>>>>
>>>> Correct me if I'm wrong, but I believe the webserver must provide=20
>>>> the
>>> SIP/SDP answers. As far as I can tell there are always two=20
>>> connectivity checks, one for each direction (Section 2.2 RFC 5245):
>>>
>>>> =A0 =A0L =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0R
>>>> =A0 =A0- =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0-
>>>> =A0 =A0STUN request -> =A0 =A0 =A0 =A0 =A0 =A0 \ =A0L's
>>>> =A0 =A0 =A0 =A0 =A0 =A0 =A0<- STUN response =A0/ =A0check
>>>>
>>>> =A0 =A0 =A0 =A0 =A0 =A0 =A0 <- STUN request =A0\ =A0R's
>>>> =A0 =A0STUN response -> =A0 =A0 =A0 =A0 =A0 =A0/ =A0check
>>>>
>>>> Alice would automatically respond to the remote side's check.
>>>> However, in order to genereate her own check (R's check in the
>>>> diagram) she would need to have received the remote side's password.
>>>> According to Section 7.1.2.3. in RFC 5245:
>>>>
>>>> "A connectivity check from R to L utilizes the username LFRAG:RFRAG=20
>>>> and a password of LPASS."
>>>>
>>>> If this interpretation is correct then counter-measure A below=20
>>>> would always be a possibility.
>>>>
>>>
>>> Oscar,
>>>
>>> From L's perspective, the process of nomination and validation can=20
>>> complete without R having succeeded. Thus L (WebRTC Browser) can=20
>>> start send media to R without R having successfully validated any=20
>>> return path from R to L.
>>>
>>> Thus, I don't see how A could be made to reliably work as a mitigation.
>>>
>>> In addition are there anyone who actually have this type of attack=20
>>> protection in their SIP stacks or SIP proxies?
>>>
>>> Cheers
>>>
>>> Magnus Westerlund
>>>
>>> --------------------------------------------------------------------
>>> -- Multimedia Technologies, Ericsson Research EAB/TVM
>>> --------------------------------------------------------------------
>>> -- Ericsson AB =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Phone =A0+46 10 7148287=
 F=E4r=F6gatan 6 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0
>>> | Mobile +46 73 0949079
>>> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
>>> --------------------------------------------------------------------
>>> --
>>>
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>
>
> --
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Phone =A0+46 10 7148287 F=E4=
r=F6gatan 6 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0
> | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>
_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

From oscar.ohlsson@ericsson.com  Thu Mar 15 08:12:20 2012
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From: Oscar Ohlsson <oscar.ohlsson@ericsson.com>
To: Eric Rescorla <ekr@rtfm.com>, Magnus Westerlund <magnus.westerlund@ericsson.com>
Date: Thu, 15 Mar 2012 16:12:15 +0100
Thread-Topic: [rtcweb] DoS attack despite ICE/STUN consent verification
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] DoS attack despite ICE/STUN consent verification
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> Hi,
>=20
> >> Yes, your understanding appears correct. I was maybe more=20
> considering=20
> >> of using this attack against SIP clients, especially the kind that=20
> >> can't move out of the way easily. Like call centers etc. In that=20
> >> context being able to direct a 1 mbps video stream onto a audio=20
> >> receiver works excellent if your goal is to just DoS the path.
> >
> > Yep. Though of course in the specific case of a call=20
> center, since it auto-answers you can just mount the attack=20
> with independent calls....
> >
> >> But I do agree that consent freshness appears to be an important=20
> >> method for mitigating this. It also seems to create a=20
> requirement on=20
> >> that mechanism. It can't be something that the far end can=20
> trick the=20
> >> target end-point to generate just by default. It needs to be=20
> >> something
> > where the end-point takes a decision to respond to each.
> >
> > Yeah. :(
> > Maybe you, I, Dan, and anyone else who is interested can=20
> brainstorm on this a bit in Paris.
>=20
> I would also be interested to participate in such brainstorming.
>=20
> Regards,
>=20
> Christer

I would also like to join.

Regards,

Oscar Ohlsson


>=20
>=20
> > Cheers
> >
> > Magnus
> >
> > On 2012-03-12 22:24, Eric Rescorla wrote:
> >> I want to make sure I understand this attack correctly.
> >>
> >> Alice is making a call to Zed and either Zed is the=20
> attacker or the=20
> >> attacker somehow captures Alice's media parameters. The=20
> attacker then=20
> >> uses Alice's offer (and her parameters) to start ICE=20
> handshaking on a=20
> >> pile of Web browsers that he controls. They all start ICE=20
> and since=20
> >> the checks with Alice will succeed, then the attacker has apparent=20
> >> consent to send as much traffic as he wants to Alice from=20
> each machine.
> >> Do I have this right?
> >>
> >> If I understand this correctly, then the severity of the=20
> attack seems=20
> >> to be limited since Alice has to actually be calling=20
> someone that is=20
> >> malicious or that has very tight control of the network, and soon=20
> >> after Alice hangs up the phone, the problem goes away. (Assuming=20
> >> consent freshness).
> >>
> >> As Dan notes, the interaction with consent freshness seems=20
> like a key=20
> >> mitigation here. Moreover, I think this suggests that even if=20
> >> integrity is not used, as in draft-muthu, the ufrags=20
> should both be=20
> >> present (and unmodifiable from the JS). This would allow a=20
> client to=20
> >> discard ICE associations with ufrags that do not match=20
> whichever ICE=20
> >> association got selected for the call.
> >>
> >> -Ekr
> >>
> >>
> >> On Mon, Feb 6, 2012 at 6:27 AM, Magnus Westerlund=20
> >> <magnus.westerlund@ericsson.com> wrote:
> >>> On 2012-02-06 11:19, Oscar Ohlsson wrote:
> >>>>
> >>>>
> >>>>> -----Original Message-----
> >>>>> From: rtcweb-bounces@ietf.org
> >>>>> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Magnus Westerlund
> >>>>> Sent: den 1 februari 2012 23:31
> >>>>> To: rtcweb@ietf.org
> >>>>> Subject: [rtcweb] DoS attack despite ICE/STUN consent=20
> verification
> >>>>>
> >>>>> Hi,
> >>>>>
> >>>>> As I brought up in yesterdays meeting I think there exist an=20
> >>>>> attack that should be considered. I don't think it is super=20
> >>>>> strong, but not without issues.
> >>>>>
> >>>>> Setting:
> >>>>>
> >>>>> Webserver is evil and have a target Alice. Alice can be=20
> a legacy=20
> >>>>> device, like a SIP/SDP video phone. It knows the target=20
> is doing=20
> >>>>> ICE and have Alice's ufrag and password and candidates.=20
> The server=20
> >>>>> would like to hammer this target by utilizing its set=20
> of connected=20
> >>>>> WebRTC clients (bob, charlie etc.). Thus it would like=20
> to ensure=20
> >>>>> that it gets the browser to start sending RTP media=20
> when creating=20
> >>>>> a PeerConnection in the clients.
> >>>>>
> >>>>> So the server gets the client to open PeerConnections=20
> towards the=20
> >>>>> ICE parameters from the target (Alice).
> >>>>>
> >>>>> In default ICE behavior there is an optimization to allow the=20
> >>>>> Offerer
> >>>>> (Alice) to respond to STUN binding requests from Bob without=20
> >>>>> having received an SDP answer from Bob. The STUN=20
> binding request=20
> >>>>> must only have the Alice's ufrag and its password.
> >>>>> This enables that Webrtc clients can get passed the=20
> data consent=20
> >>>>> step with ICE capable end-point for which the ICE=20
> parameters can=20
> >>>>> be retrieved.
> >>>>>
> >>>>> So from Alice perspective this attack will look like=20
> its offer got=20
> >>>>> forked to many peers based on the many different peers=20
> that send=20
> >>>>> STUN messages. The webserver could even provide SIP/SDP=20
> answers to=20
> >>>>> alice with the webclients. But that is not necessary but is one=20
> >>>>> variation of this.
> >>>>
> >>>>
> >>>> Correct me if I'm wrong, but I believe the webserver=20
> must provide=20
> >>>> the
> >>> SIP/SDP answers. As far as I can tell there are always two=20
> >>> connectivity checks, one for each direction (Section 2.2=20
> RFC 5245):
> >>>
> >>>> =A0 =A0L =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0R
> >>>> =A0 =A0- =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0-
> >>>> =A0 =A0STUN request -> =A0 =A0 =A0 =A0 =A0 =A0 \ =A0L's
> >>>> =A0 =A0 =A0 =A0 =A0 =A0 =A0<- STUN response =A0/ =A0check
> >>>>
> >>>> =A0 =A0 =A0 =A0 =A0 =A0 =A0 <- STUN request =A0\ =A0R's
> >>>> =A0 =A0STUN response -> =A0 =A0 =A0 =A0 =A0 =A0/ =A0check
> >>>>
> >>>> Alice would automatically respond to the remote side's check.
> >>>> However, in order to genereate her own check (R's check in the
> >>>> diagram) she would need to have received the remote=20
> side's password.
> >>>> According to Section 7.1.2.3. in RFC 5245:
> >>>>
> >>>> "A connectivity check from R to L utilizes the username=20
> LFRAG:RFRAG=20
> >>>> and a password of LPASS."
> >>>>
> >>>> If this interpretation is correct then counter-measure A below=20
> >>>> would always be a possibility.
> >>>>
> >>>
> >>> Oscar,
> >>>
> >>> From L's perspective, the process of nomination and=20
> validation can=20
> >>> complete without R having succeeded. Thus L (WebRTC Browser) can=20
> >>> start send media to R without R having successfully validated any=20
> >>> return path from R to L.
> >>>
> >>> Thus, I don't see how A could be made to reliably work as=20
> a mitigation.
> >>>
> >>> In addition are there anyone who actually have this type=20
> of attack=20
> >>> protection in their SIP stacks or SIP proxies?
> >>>
> >>> Cheers
> >>>
> >>> Magnus Westerlund
> >>>
> >>>=20
> --------------------------------------------------------------------
> >>> -- Multimedia Technologies, Ericsson Research EAB/TVM
> >>>=20
> --------------------------------------------------------------------
> >>> -- Ericsson AB =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Phone =A0+46 10 71482=
87 F=E4r=F6gatan 6
> >>> | Mobile +46 73 0949079
> >>> SE-164 80 Stockholm, Sweden| mailto:=20
> magnus.westerlund@ericsson.com
> >>>=20
> --------------------------------------------------------------------
> >>> --
> >>>
> >>> _______________________________________________
> >>> rtcweb mailing list
> >>> rtcweb@ietf.org
> >>> https://www.ietf.org/mailman/listinfo/rtcweb
> >>
> >
> >
> > --
> >
> > Magnus Westerlund
> >
> >=20
> ----------------------------------------------------------------------
> > Multimedia Technologies, Ericsson Research EAB/TVM
> >=20
> ----------------------------------------------------------------------
> > Ericsson AB =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Phone =A0+46 10 7148287 F=
=E4r=F6gatan 6
> > | Mobile +46 73 0949079
> > SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> >=20
> ----------------------------------------------------------------------
> >
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> =

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Thread-Topic: [rtcweb] DoS attack despite ICE/STUN consent verification
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From: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
To: "Eric Rescorla" <ekr@rtfm.com>, "Magnus Westerlund" <magnus.westerlund@ericsson.com>
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Subject: Re: [rtcweb] DoS attack despite ICE/STUN consent verification
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|Maybe you, I, Dan, and anyone else who is interested can=20
|brainstorm on this a bit in Paris.

I would be interested..please count me in.

Muthu

|-----Original Message-----
|From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On =
Behalf Of Eric Rescorla
|Sent: Tuesday, March 13, 2012 4:14 AM
|To: Magnus Westerlund
|Cc: rtcweb@ietf.org
|Subject: Re: [rtcweb] DoS attack despite ICE/STUN consent verification
|
|On Mon, Mar 12, 2012 at 3:41 PM, Magnus Westerlund
|<magnus.westerlund@ericsson.com> wrote:
|> Hi,
|>
|> Yes, your understanding appears correct. I was maybe more considering =
of
|> using this attack against SIP clients, especially the kind that can't
|> move out of the way easily. Like call centers etc. In that context =
being
|> able to direct a 1 mbps video stream onto a audio receiver works
|> excellent if your goal is to just DoS the path.
|
|Yep. Though of course in the specific case of a call center, since it
|auto-answers you can just mount the attack with independent calls....
|
|
|> But I do agree that consent freshness appears to be an important =
method
|> for mitigating this. It also seems to create a requirement on that
|> mechanism. It can't be something that the far end can trick the =
target
|> end-point to generate just by default. It needs to be something where
|> the end-point takes a decision to respond to each.
|
|Yeah. :(
|Maybe you, I, Dan, and anyone else who is interested can brainstorm
|on this a bit in Paris.
|
|-Ekr
|
|> Cheers
|>
|> Magnus
|>
|> On 2012-03-12 22:24, Eric Rescorla wrote:
|>> I want to make sure I understand this attack correctly.
|>>
|>> Alice is making a call to Zed and either Zed is the attacker or the =
attacker
|>> somehow captures Alice's media parameters. The attacker then uses
|>> Alice's offer (and her parameters) to start ICE handshaking on a =
pile
|>> of Web browsers that he controls. They all start ICE and since the
|>> checks with Alice will succeed, then the attacker has apparent =
consent
|>> to send as much traffic as he wants to Alice from each machine.
|>> Do I have this right?
|>>
|>> If I understand this correctly, then the severity of the attack =
seems to
|>> be limited since Alice has to actually be calling someone that is
|>> malicious or that has very tight control of the network, and soon
|>> after Alice hangs up the phone, the problem goes away. (Assuming
|>> consent freshness).
|>>
|>> As Dan notes, the interaction with consent freshness seems like a =
key
|>> mitigation here. Moreover, I think this suggests that even if =
integrity
|>> is not used, as in draft-muthu, the ufrags should both be present
|>> (and unmodifiable from the JS). This would allow a client to discard
|>> ICE associations with ufrags that do not match whichever ICE
|>> association got selected for the call.
|>>
|>> -Ekr
|>>
|>>
|>> On Mon, Feb 6, 2012 at 6:27 AM, Magnus Westerlund
|>> <magnus.westerlund@ericsson.com> wrote:
|>>> On 2012-02-06 11:19, Oscar Ohlsson wrote:
|>>>>
|>>>>
|>>>>> -----Original Message-----
|>>>>> From: rtcweb-bounces@ietf.org
|>>>>> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Magnus Westerlund
|>>>>> Sent: den 1 februari 2012 23:31
|>>>>> To: rtcweb@ietf.org
|>>>>> Subject: [rtcweb] DoS attack despite ICE/STUN consent =
verification
|>>>>>
|>>>>> Hi,
|>>>>>
|>>>>> As I brought up in yesterdays meeting I think there exist an
|>>>>> attack that should be considered. I don't think it is super
|>>>>> strong, but not without issues.
|>>>>>
|>>>>> Setting:
|>>>>>
|>>>>> Webserver is evil and have a target Alice. Alice can be a
|>>>>> legacy device, like a SIP/SDP video phone. It knows the
|>>>>> target is doing ICE and have Alice's ufrag and password and
|>>>>> candidates. The server would like to hammer this target by
|>>>>> utilizing its set of connected WebRTC clients (bob, charlie
|>>>>> etc.). Thus it would like to ensure that it gets the browser
|>>>>> to start sending RTP media when creating a PeerConnection in
|>>>>> the clients.
|>>>>>
|>>>>> So the server gets the client to open PeerConnections towards
|>>>>> the ICE parameters from the target (Alice).
|>>>>>
|>>>>> In default ICE behavior there is an optimization to allow the =
Offerer
|>>>>> (Alice) to respond to STUN binding requests from Bob without
|>>>>> having received an SDP answer from Bob. The STUN binding
|>>>>> request must only have the Alice's ufrag and its password.
|>>>>> This enables that Webrtc clients can get passed the data
|>>>>> consent step with ICE capable end-point for which the ICE
|>>>>> parameters can be retrieved.
|>>>>>
|>>>>> So from Alice perspective this attack will look like its
|>>>>> offer got forked to many peers based on the many different
|>>>>> peers that send STUN messages. The webserver could even
|>>>>> provide SIP/SDP answers to alice with the webclients. But
|>>>>> that is not necessary but is one variation of this.
|>>>>
|>>>>
|>>>> Correct me if I'm wrong, but I believe the webserver must provide =
the
|>>> SIP/SDP answers. As far as I can tell there are always two =
connectivity
|>>> checks, one for each direction (Section 2.2 RFC 5245):
|>>>
|>>>> =A0 =A0L =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0R
|>>>> =A0 =A0- =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0-
|>>>> =A0 =A0STUN request -> =A0 =A0 =A0 =A0 =A0 =A0 \ =A0L's
|>>>> =A0 =A0 =A0 =A0 =A0 =A0 =A0<- STUN response =A0/ =A0check
|>>>>
|>>>> =A0 =A0 =A0 =A0 =A0 =A0 =A0 <- STUN request =A0\ =A0R's
|>>>> =A0 =A0STUN response -> =A0 =A0 =A0 =A0 =A0 =A0/ =A0check
|>>>>
|>>>> Alice would automatically respond to the remote side's check.
|>>>> However, in order to genereate her own check (R's check in the
|>>>> diagram) she would need to have received the remote side's =
password.
|>>>> According to Section 7.1.2.3. in RFC 5245:
|>>>>
|>>>> "A connectivity check from R to L utilizes the username =
LFRAG:RFRAG
|>>>> and a password of LPASS."
|>>>>
|>>>> If this interpretation is correct then counter-measure A below =
would
|>>>> always be a possibility.
|>>>>
|>>>
|>>> Oscar,
|>>>
|>>> From L's perspective, the process of nomination and validation can
|>>> complete without R having succeeded. Thus L (WebRTC Browser) can =
start
|>>> send media to R without R having successfully validated any return =
path
|>>> from R to L.
|>>>
|>>> Thus, I don't see how A could be made to reliably work as a =
mitigation.
|>>>
|>>> In addition are there anyone who actually have this type of attack
|>>> protection in their SIP stacks or SIP proxies?
|>>>
|>>> Cheers
|>>>
|>>> Magnus Westerlund
|>>>
|>>> =
----------------------------------------------------------------------
|>>> Multimedia Technologies, Ericsson Research EAB/TVM
|>>> =
----------------------------------------------------------------------
|>>> Ericsson AB =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Phone =A0+46 10 =
7148287
|>>> F=E4r=F6gatan 6 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Mobile +46 73 =
0949079
|>>> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
|>>> =
----------------------------------------------------------------------
|>>>
|>>> _______________________________________________
|>>> rtcweb mailing list
|>>> rtcweb@ietf.org
|>>> https://www.ietf.org/mailman/listinfo/rtcweb
|>>
|>
|>
|> --
|>
|> Magnus Westerlund
|>
|> =
----------------------------------------------------------------------
|> Multimedia Technologies, Ericsson Research EAB/TVM
|> =
----------------------------------------------------------------------
|> Ericsson AB =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Phone =A0+46 10 7148287
|> F=E4r=F6gatan 6 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Mobile +46 73 =
0949079
|> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
|> =
----------------------------------------------------------------------
|>
|_______________________________________________
|rtcweb mailing list
|rtcweb@ietf.org
|https://www.ietf.org/mailman/listinfo/rtcweb

From HKaplan@acmepacket.com  Thu Mar 15 10:32:13 2012
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: "<igor.faynberg@alcatel-lucent.com>" <igor.faynberg@alcatel-lucent.com>
Thread-Topic: [rtcweb] DoS attack despite ICE/STUN consent verification
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Subject: Re: [rtcweb] DoS attack despite ICE/STUN consent verification
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Because although we describe things in terms of a single user (Alice in thi=
s case), that single "user" could represent a whole Enterprise call-center/=
IVR system, or an emergency service center (ie, PSAPs), etc.

Of course most such things receive calls not generate them, as in this atta=
ck scenario; so for this attack scenario imagine Alice is the Emergency Ale=
rt System, or NAWAS, or IPAWS, or whatever.

-hadriel
p.s. not implying that I think this particular attack-type is important, ju=
st sayin' why we care about poor old Alice.


On Mar 13, 2012, at 12:46 PM, Igor Faynberg wrote:

> My major question on this thread, which I have been watching with interes=
t:
>=20
> Why do we care about a DoS attack  against a single client?  (I mean spec=
ifically DoS, not any other security attack.)
>=20
> It appears to me that there are somewhat simpler and more effective DoS m=
ethods for this case: 1) cut the wire to the client's device or 2) jam WiFi=
 or other wireless signal.
>=20
> Or did I misunderstand the problem?
>=20
> Igor
>=20


From igor.faynberg@alcatel-lucent.com  Thu Mar 15 13:34:56 2012
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Hadriel, thanks!

My on-going confusion was caused by the (myopic, I admit) interpretation 
of a browser=ONE person on the receiving end.  I never thought of the 
call center, let alone EAS or NAWAS.

Perhaps one thing that I missed was that the situation applies outside 
of a browser-to-browser scenario.

But what you said starts to make a lot of sense to me.  I think you are 
saying that a media gateway itself can be a subject to the attack, 
right?  I had never thought about that before...

Igor

On 3/15/2012 1:32 PM, Hadriel Kaplan wrote:
> Because although we describe things in terms of a single user (Alice in this case), that single "user" could represent a whole Enterprise call-center/IVR system, or an emergency service center (ie, PSAPs), etc.
>
> Of course most such things receive calls not generate them, as in this attack scenario; so for this attack scenario imagine Alice is the Emergency Alert System, or NAWAS, or IPAWS, or whatever.
>
> -hadriel
> p.s. not implying that I think this particular attack-type is important, just sayin' why we care about poor old Alice.
>
>
> On Mar 13, 2012, at 12:46 PM, Igor Faynberg wrote:
>
>> My major question on this thread, which I have been watching with interest:
>>
>> Why do we care about a DoS attack  against a single client?  (I mean specifically DoS, not any other security attack.)
>>
>> It appears to me that there are somewhat simpler and more effective DoS methods for this case: 1) cut the wire to the client's device or 2) jam WiFi or other wireless signal.
>>
>> Or did I misunderstand the problem?
>>
>> Igor
>>

From oscar.ohlsson@ericsson.com  Fri Mar 16 08:05:01 2012
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From: Oscar Ohlsson <oscar.ohlsson@ericsson.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Fri, 16 Mar 2012 16:04:54 +0100
Thread-Topic: SDES vs DTLS-SRTP revisited
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Subject: [rtcweb] SDES vs DTLS-SRTP revisited
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Hi,

There were a couple of things that I didn't explain properly in my SDES vs =
DTLS-SRTP presentation at the interim meeting. Since time will be short at =
the next meeting as well, I thought I would try to explain myself in an ema=
il.

I noticed that the discussion quickly turned into a direct comparison of th=
e two protocols. When this happens DTLS-SRTP definitely comes out as the wi=
nner and trying to defend SDES just seems foolish. But this is not the righ=
t focus of the discussion IMO. We should be looking at the system as a whol=
e and determine more exactly the scenarios where SDES poses a security risk=
.

The first major argument against SDES is that it doesn't offer forward secr=
ecy. Obviously, if the remote endpoint does not support DTLS-SRTP you have =
to switch to SDES or plain RTP at some point. And from that point and onwar=
ds you will never have forward secrecy. The only difference is that the con=
version point is moved slightly to the left (one step?) when SDES is suppor=
ted. However, if web developers starts using SDES without reason (i.e. when=
 the remote endpoint is a browser) then I agree that the lack of forward se=
crecy is a problem. Do people on this list believe this will be the case? C=
an it be prevented somehow?

The second major argument is that SDES makes it possible for the service pr=
ovider to intercept calls undetectable. I've been thinking about this for s=
ome time but still haven't understood the problem completely. Say that you'=
re on a call and want to determine if the service provider is listening in:

Case 1: Both endpoints support DTLS-SRTP (can be determined by asking the o=
ther party)

If SDES is used you directly conclude that something fishy is going on and =
hang up. If DTLS-SRTP is used you compare fingerprints and hang up if they =
don't match. (I've assumed that a user can view detailed security informati=
on in the browser).

Case 2: The other endpoint does not support DTLS-SRTP

Regardless of whether SDES is allowed or not you will never be able to tell=
 if the call is being intercepted. The best thing to do in this case is pro=
bably to simply hang up.

Does the fingerprint mismatch in case 1 somehow serve as a stronger indicat=
ion that the call is being intercepted? What action would a suspicious user=
 take besides hanging up? I would be happy if someone could explain this to=
 me.

Regards,

Oscar Ohlsson=

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Subject: Re: [rtcweb] SDES vs DTLS-SRTP revisited
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Oscar,

I am afraid I missed the point here.

How can one determine that the call is being listened to?   How can the 
browser know whom the RTP stream is passing through?

If you know the SRTP key and filter the flow directed at me, you will be 
on the conversation, right?

Igor

On 3/16/2012 11:04 AM, Oscar Ohlsson wrote:
> ... Say that you're on a call and want to determine if the service provider is listening in:
>
> Case 1: Both endpoints support DTLS-SRTP (can be determined by asking the other party)
>
> If SDES is used you directly conclude that something fishy is going on and hang up. If DTLS-SRTP is used you compare fingerprints and hang up if they don't match. (I've assumed that a user can view detailed security information in the browser).
>
> Case 2: The other endpoint does not support DTLS-SRTP
...
>

From pravindran@sonusnet.com  Fri Mar 16 12:25:52 2012
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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Resolving RTP/SDES question in Paris
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Magnus,

As I mentioned in earlier mail thread (http://www.ietf.org/mail-archive/web=
/rtcweb/current/msg03148.html), RTP has to be supported by RTCWeb client wi=
th local user consent.

To clarify why RTP is not harmful in some usage, VPN access of intranet RTC=
Web client in public internet usecase is as follows:=20

Usecase details: Enterprise employee Alice access enterprise (HTTP & WebRTC=
 compliant) intranet with VPN connection using Airport WiFi internet connec=
tion, Alice will have the mechanism to call another employee Bob in the sam=
e enterprise using WebRTC session and the session between Alice and bob wil=
l be plain RTP in VPN network.

Security aspect: Here, accessing (HTTP) intranet browsing today is safe eve=
n though Alice is using Airport Wifi internet connection as he has VPN conn=
ection (IPSec) and no dependency on HTTPS. In the same way, WebRTC mechanis=
m should not mandate for SRTP-DTLS in media and MUST allow RTP.  The local =
user consent (configuration) is required to restrict this usage by Dr Evil =
website.

Thanks
Partha



>-----Original Message-----
>From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
>Of Magnus Westerlund
>Sent: Friday, February 24, 2012 3:05 PM
>To: rtcweb@ietf.org
>Subject: [rtcweb] Resolving RTP/SDES question in Paris
>
>WG,
>
>There has been extensive discussion at WG meetings at whether we need to
>support SDES and RTP in addition to DTLS-SRTP. The architecture draft
>currently has a giant "Open Issue" text around the question of the
>appropriate level of support for modes less secure that DTLS-SRTP.
>The draft currently implicitly allows but does not require these modes:
>
>  Implementations MUST implement DTLS and DTLS-SRTP.  All data channels
>  MUST be secured via DTLS.  DTLS-SRTP MUST be offered for every media
>  channel and MUST be the default; i.e., if an implementation receives
>  an offer for DTLS-SRTP and SDES and/or plain RTP, DTLS-SRTP MUST be
>  selected.
>
>The chairs intend to use significant meeting time in Paris with an aim
>to reach consensus around this issue. Please consider this thread an
>invitation to discuss this topic on the list. In Paris, we intend to
>have a presentation summarizing the discussion so far, digging into the
>details of different possible outcomes and of the arguments have been
>raised in order to set the stage for a discussion and a consensus call.
>
>Please come prepared to have that discussion in Paris.
>
>Magnus Westerlund
>(For the WG chairs)
>
>----------------------------------------------------------------------
>Multimedia Technologies, Ericsson Research EAB/TVM
>----------------------------------------------------------------------
>Ericsson AB                | Phone  +46 10 7148287
>F=E4r=F6gatan 6                | Mobile +46 73 0949079
>SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
>----------------------------------------------------------------------
>
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb

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On 3/16/2012 3:25 PM, Ravindran, Parthasarathi wrote:
> Magnus,
>
> As I mentioned in earlier mail thread (http://www.ietf.org/mail-archive/web/rtcweb/current/msg03148.html), RTP has to be supported by RTCWeb client with local user consent.

That is your stipulation.  Thus far this does not have consensus.

> To clarify why RTP is not harmful in some usage, VPN access of intranet RTCWeb client in public internet usecase is as follows:
>
> Usecase details: Enterprise employee Alice access enterprise (HTTP&  WebRTC compliant) intranet with VPN connection using Airport WiFi internet connection, Alice will have the mechanism to call another employee Bob in the same enterprise using WebRTC session and the session between Alice and bob will be plain RTP in VPN network.
>
> Security aspect: Here, accessing (HTTP) intranet browsing today is safe even though Alice is using Airport Wifi internet connection as he has VPN connection (IPSec) and no dependency on HTTPS. In the same way, WebRTC mechanism should not mandate for SRTP-DTLS in media and MUST allow RTP.  The local user consent (configuration) is required to restrict this usage by Dr Evil website.

This says the media is in theory 'safe' as RTP over a VPN.  Similar 
arguments were used for "inside the corporate network" in previous 
discussions.  That doesn't mean that the spec should allow for RTP; for 
example it might create bid-down attack possibilities - and the 
application has lots of trouble knowing if hte link really is secure 
(and doesn't transition off security anywhere after leaving the machine).

-- 
Randell Jesup
randell-ietf@jesup.org


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From: Eric Rescorla <ekr@rtfm.com>
Date: Fri, 16 Mar 2012 15:32:05 -0700
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] SDES vs DTLS-SRTP revisited
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On Fri, Mar 16, 2012 at 8:04 AM, Oscar Ohlsson
<oscar.ohlsson@ericsson.com> wrote:
> Hi,
>
> There were a couple of things that I didn't explain properly in my SDES v=
s DTLS-SRTP presentation at the interim meeting. Since time will be short a=
t the next meeting as well, I thought I would try to explain myself in an e=
mail.
>
> I noticed that the discussion quickly turned into a direct comparison of =
the two protocols. When this happens DTLS-SRTP definitely comes out as the =
winner and trying to defend SDES just seems foolish. But this is not the ri=
ght focus of the discussion IMO. We should be looking at the system as a wh=
ole and determine more exactly the scenarios where SDES poses a security ri=
sk.
>
> The first major argument against SDES is that it doesn't offer forward se=
crecy. Obviously, if the remote endpoint does not support DTLS-SRTP you hav=
e to switch to SDES or plain RTP at some point. And from that point and onw=
ards you will never have forward secrecy. The only difference is that the c=
onversion point is moved slightly to the left (one step?) when SDES is supp=
orted. However, if web developers starts using SDES without reason (i.e. wh=
en the remote endpoint is a browser) then I agree that the lack of forward =
secrecy is a problem. Do people on this list believe this will be the case?=
 Can it be prevented somehow?
>
> The second major argument is that SDES makes it possible for the service =
provider to intercept calls undetectable. I've been thinking about this for=
 some time but still haven't understood the problem completely. Say that yo=
u're on a call and want to determine if the service provider is listening i=
n:
>
> Case 1: Both endpoints support DTLS-SRTP (can be determined by asking the=
 other party)
>
> If SDES is used you directly conclude that something fishy is going on an=
d hang up. If DTLS-SRTP is used you compare fingerprints and hang up if the=
y don't match. (I've assumed that a user can view detailed security informa=
tion in the browser).
>
> Case 2: The other endpoint does not support DTLS-SRTP
>
> Regardless of whether SDES is allowed or not you will never be able to te=
ll if the call is being intercepted. The best thing to do in this case is p=
robably to simply hang up.
>
> Does the fingerprint mismatch in case 1 somehow serve as a stronger indic=
ation that the call is being intercepted? What action would a suspicious us=
er take besides hanging up? I would be happy if someone could explain this =
to me.

Hi Oscar,

Yes, I think the fingerprint mismatch in case 1 serves as a stronger indica=
tion
that something was wrong, because there is no technical reason for the
fingerprint not to match if both sides believe they are doing DTLS. I.e.,
it always indicates either a software defect or a man in the middle.
And the suspicious user of course should hang up, but they should also
post pictures to /., which, I would think, serves
as a pretty significant deterrent to providers tampering with people's call=
s!
By contrast, with SDES, the attack scenario and the legitimate scenario
look the same.

We actually have a worked example of this with Diginotar, where a mechanica=
l
system for detecting MITM attacks lead to a report to Google of such an
attack and, IIRC, Diginotar being removed from the root list.

Best,
-Ekr

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Fri, 16 Mar 2012 23:55:00 +0100
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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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2012/3/16 Ravindran, Parthasarathi <pravindran@sonusnet.com>:
> As I mentioned in earlier mail thread (http://www.ietf.org/mail-archive/w=
eb/rtcweb/current/msg03148.html), RTP has to be supported by RTCWeb client =
with local user consent.

Well, that's your opinnion, not a MUST (or not yet).



> To clarify why RTP is not harmful in some usage, VPN access of intranet R=
TCWeb client in public internet usecase is as follows:
>
> Usecase details: Enterprise employee Alice access enterprise (HTTP & WebR=
TC compliant) intranet with VPN connection using Airport WiFi internet conn=
ection, Alice will have the mechanism to call another employee Bob in the s=
ame enterprise using WebRTC session and the session between Alice and bob w=
ill be plain RTP in VPN network.
>
> Security aspect: Here, accessing (HTTP) intranet browsing today is safe e=
ven though Alice is using Airport Wifi internet connection as he has VPN co=
nnection (IPSec) and no dependency on HTTPS. In the same way, WebRTC mechan=
ism should not mandate for SRTP-DTLS in media and MUST allow RTP. =C2=A0The=
 local user consent (configuration) is required to restrict this usage by D=
r Evil website.


To clarify why RTP *is* harmful in *common* usages:

Usecase details: An user in an airport using an open WiFi connection
and accesing to a social web page implementing WebRTC for calls
between web users and between web users and PSTN numbers. And not, the
user does not use a VPN to connect to a social web. The user calls to
a PSTN number (since the contact he wants to call to is not currently
online in the web). The call is established with a SIP PSTN gateway
which does not implement SRTP but just RTP.

Security aspect: Plain RTP is negotiated so anyone in the airport can
spoof the media. A good reason to state "WebRTC mechanism MUST mandate
for SRTP and MUST NOT allow plain RTP, since the security aspect
depends on the service provider (and there are tons of irresponsible
service providers out there in Internet).



> The local user consent (configuration) is required to restrict this usage=
 by Dr Evil website.

Giving the security responsability to end users as a configurable
option is not a good idea. The world would be much better if the
browsers would not allow plain HTTP, but it's not possible to change
the WWW requeriments and mandate HTTPS given the ammount of plain HTTP
deployments. In contrast, WebRTC does not exist yet so *now* is the
moment to mandate security.


Regards.



--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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--e89a8ffba8a5b28c4704bb64a8c6
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On Fri, Mar 16, 2012 at 6:55 PM, I=F1aki Baz Castillo <ibc@aliax.net> wrote=
:

> Giving the security responsability to end users as a configurable
> option is not a good idea. The world would be much better if the
> browsers would not allow plain HTTP, but it's not possible to change
> the WWW requeriments and mandate HTTPS given the ammount of plain HTTP
> deployments. In contrast, WebRTC does not exist yet so *now* is the
> moment to mandate security.
>
>
And you imply that the fact that security is most of the times unnecessary
and often prohibited, has absolutely nothing to do with using HTTP over
HTTPS. If you think people in the military or in prisons do not use web
browsers, think again. Requiring secure communications will prevent them
from using WebRTC.

Furthermore comparison between HTTP/HTTPS and RTP/SRTP in not entirely
appropriate. HTTPS communications are reasonably secure. SRTP calls, unless
we make identity management a requirement for WebRTC, are only giving
people a false sense of security. The fact that browser uses SRTP does not,
in any way, imply that the communication is secure. If the key was
delivered over HTTP, anybody intercepting your IP traffic can listen to it.
If we are dealing with the server based attack then all hell breaks loose.
So, unless we requiring that only connections are over DTLS-SRTP using an
SDP signed by a verified remote party are allowed, we allow unsecure
connections. We need to show an appropriate warning for all of them, and
warning RTP vs SDES-SRTP vs unsigned DTLS-SRTP should not be very
different. It should say "You are trying to communicate over unsecure
peer--to-peer channel with an unknown party. Continue?" There is no, "you
are trying to use slightly more secure channel. Is this ok?".
_____________
Roman Shpount

--e89a8ffba8a5b28c4704bb64a8c6
Content-Type: text/html; charset=ISO-8859-1
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<div class=3D"gmail_quote">On Fri, Mar 16, 2012 at 6:55 PM, I=F1aki Baz Cas=
tillo <span dir=3D"ltr">&lt;<a href=3D"mailto:ibc@aliax.net">ibc@aliax.net<=
/a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:=
0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Giving the security responsability to end users as a configurable<br>
option is not a good idea. The world would be much better if the<br>
browsers would not allow plain HTTP, but it&#39;s not possible to change<br=
>
the WWW requeriments and mandate HTTPS given the ammount of plain HTTP<br>
deployments. In contrast, WebRTC does not exist yet so *now* is the<br>
moment to mandate security.<br>
<br></blockquote></div><br>And you imply that the fact that security is mos=
t of the times unnecessary and often prohibited, has absolutely nothing to =
do with using HTTP over HTTPS. If you think people in the military or in pr=
isons do not use web browsers, think again. Requiring secure communications=
 will prevent them from using WebRTC.<br>
<br>Furthermore comparison between HTTP/HTTPS and RTP/SRTP in not entirely =
appropriate. HTTPS communications are reasonably secure. SRTP calls, unless=
 we make identity management a requirement for WebRTC, are only giving peop=
le a false sense of security. The fact that browser uses SRTP does not, in =
any way, imply that the communication is secure. If the key was delivered o=
ver HTTP, anybody intercepting your IP traffic can listen to it. If we are =
dealing with the server based attack then all hell breaks loose. So, unless=
 we requiring that only connections are over DTLS-SRTP using an SDP signed =
by a verified remote party are allowed, we allow unsecure connections. We n=
eed to show an appropriate warning for all of them, and warning RTP vs SDES=
-SRTP vs unsigned DTLS-SRTP should not be very different. It should say &qu=
ot;You are trying to communicate over unsecure peer--to-peer channel with a=
n unknown party. Continue?&quot; There is no, &quot;you are trying to use s=
lightly more secure channel. Is this ok?&quot;.<br clear=3D"all">
_____________<br>Roman Shpount<br>
<br>

--e89a8ffba8a5b28c4704bb64a8c6--

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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Eric Rescorla <ekr@rtfm.com>
Thread-Topic: [rtcweb] SDES vs DTLS-SRTP revisited
Thread-Index: AQHNA+xjU0SbLPnT3kyGR6DTEQ1R9g==
Date: Sat, 17 Mar 2012 03:16:48 +0000
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Subject: Re: [rtcweb] SDES vs DTLS-SRTP revisited
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On Mar 16, 2012, at 6:32 PM, Eric Rescorla wrote:

> On Fri, Mar 16, 2012 at 8:04 AM, Oscar Ohlsson
> <oscar.ohlsson@ericsson.com> wrote:
>> Case 1: Both endpoints support DTLS-SRTP (can be determined by asking th=
e other party)
>>=20
>> If SDES is used you directly conclude that something fishy is going on a=
nd hang up. If DTLS-SRTP is used you compare fingerprints and hang up if th=
ey don't match. (I've assumed that a user can view detailed security inform=
ation in the browser).
>>=20
>> Case 2: The other endpoint does not support DTLS-SRTP
>>=20
>> Regardless of whether SDES is allowed or not you will never be able to t=
ell if the call is being intercepted. The best thing to do in this case is =
probably to simply hang up.
>>=20
>> Does the fingerprint mismatch in case 1 somehow serve as a stronger indi=
cation that the call is being intercepted? What action would a suspicious u=
ser take besides hanging up? I would be happy if someone could explain this=
 to me.
>=20
> Yes, I think the fingerprint mismatch in case 1 serves as a stronger indi=
cation
> that something was wrong, because there is no technical reason for the
> fingerprint not to match if both sides believe they are doing DTLS. I.e.,
> it always indicates either a software defect or a man in the middle.

Absolutely, if DTLS-SRTP is used AND both sides verify the same fingerprint=
s, you know it's either end-to-end or not. (you can't actually jummp to the=
 conclusion of a software defect or MitM, but I'll respond to that in a sep=
arate email) =20

But that's not really the point.  The point is the browser can't show a loc=
k-icon regardless - it needs the user to do something more, or else it has =
no more meaning of security than SDES.  The DTLS connection *might* be end-=
to-end, or it might not, and the only way to know is for the user to actual=
ly check.  So the browser cannot show a lock-icon, and will have to say som=
ething like this in the info box of advanced details: "The media might be s=
ecure end-to-end, but only if the following string matches what the far-end=
 user sees=85".

So imagine if we allow SDES, only if the far-end did not offer DTLS-SRTP, a=
nd only if the browser's connection to the web-server was HTTPS.  Then if t=
he user goes to check that advanced details box, we can mandate in the RFC =
that they will be shown that there is no way to verify it is end-to-end: so=
mething like "the media cannot be verified to be secure end-to-end".  So wh=
y would this be useful?  Because it (1) is actually more accurate and less =
misleading in cases when talking through gateways to/from the PSTN, since e=
ven if the gateway did DTLS-SRTP it would be no more knowably secure than u=
sing SDES to begin with, and (2) avoids the problem at hand, which is preve=
nting eavesdropping and injection over WiFi or other insecure access medium=
s.

Even if we had an Identity Provider model to use so we could avoid users ch=
ecking fingerprints (which we don't have)... as far as I know it's still re=
lying on a third-party (the identity provider).  And since you mentioned Di=
ginotar, I'd note it's an example that such providers cannot, in fact, be w=
holly trusted either.

-hadriel


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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Eric Rescorla <ekr@rtfm.com>
Thread-Topic: [rtcweb] SDES vs DTLS-SRTP revisited
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Subject: Re: [rtcweb] SDES vs DTLS-SRTP revisited
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On Mar 16, 2012, at 6:32 PM, Eric Rescorla wrote:
>=20
> Yes, I think the fingerprint mismatch in case 1 serves as a stronger indi=
cation
> that something was wrong, because there is no technical reason for the
> fingerprint not to match if both sides believe they are doing DTLS. I.e.,
> it always indicates either a software defect or a man in the middle.
> And the suspicious user of course should hang up, but they should also
> post pictures to /., which, I would think, serves
> as a pretty significant deterrent to providers tampering with people's ca=
lls!
> By contrast, with SDES, the attack scenario and the legitimate scenario
> look the same.

I don't think you can jump to the conclusion that there's either a software=
 defect or a MitM.  Imagine if WebRTC user Alice calls Bob through provider=
-A, using Bob's telephone number.  Provider-A supports DTLS-SRTP on its PST=
N gateway, so it handles that and generates a call into the PSTN for Bob, w=
ho happens to use Provider-B.  Bob is currently on a WebRTC client connecte=
d to Provider-B, and Provider-B happens to also support DTLS-SRTP on its PS=
TN gateway.  So both Alice and Bob see "DTLS-SRTP", but different fingerpri=
nts.  It's not a software defect.  You could argue it's a MitM (the PSTN ga=
teways doing what they do), and certainly there may *be* someone recording =
the call in the middle for all you know - but you really don't know either =
way.  Alice dialed Bob, and got the best case scenario given the situation.

The good news is the media is secure from the browsers to the PSTN, which i=
s not an insignificant accomplishment.  If we're worried about WiFi - and I=
 am, for at least marketing/perception reasons if nothing else - then we've=
 alleviated that concern. =20

So then there's slashdot.  If Alice or Bob posts to slashdot, it's bad news=
.  But we can hardly blame Provider-A or Provider-B, since they had no alte=
rnative.  Ironically, if WebRTC supported SDES, then Provider-A and B could=
 do that instead, and make it clear/explicit no end-to-end guarantee can be=
 made for the call - while *still* avoiding the main concern, which is the =
WiFi problem.  In some sense *we*, the IETF/W3C, would be more honest if we=
 allowed SDES (over HTTPS) for gateway cases, since that's actually the lev=
el of security being provided.  Mandating DTLS-SRTP provides a false sense =
of security, and false sense of insecurity if it fails.

-hadriel


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From: Eric Rescorla <ekr@rtfm.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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On Fri, Mar 16, 2012 at 4:35 PM, Roman Shpount <roman@telurix.com> wrote:
> On Fri, Mar 16, 2012 at 6:55 PM, I=F1aki Baz Castillo <ibc@aliax.net> wro=
te:
>>
>> Giving the security responsability to end users as a configurable
>> option is not a good idea. The world would be much better if the
>> browsers would not allow plain HTTP, but it's not possible to change
>> the WWW requeriments and mandate HTTPS given the ammount of plain HTTP
>> deployments. In contrast, WebRTC does not exist yet so *now* is the
>> moment to mandate security.
>>
>
> And you imply that the fact that security is most of the times unnecessar=
y
> and often prohibited, has absolutely nothing to do with using HTTP over
> HTTPS. If you think people in the military or in prisons do not use web
> browsers, think again. Requiring secure communications will prevent them
> from using WebRTC.

This doesn't seem obvious at all. Rather it will require the people who wis=
h
to access their communications to do so explicitly in the same way as
enterprises who wish to monitor HTTPS do so now.

-Ekr

From pravindran@sonusnet.com  Fri Mar 16 21:13:39 2012
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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: Randell Jesup <randell-ietf@jesup.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Resolving RTP/SDES question in Paris
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Randell,

In my usecase, the application will not be able to access the website witho=
ut VPN connection . Please explain your bid-down attack in my usecase.=20

You can speculate that all intranet in the world shall be broken in a momen=
t if they use HTTP but IMO, it is just speculation. IETF specification has =
to be generic enough to serve all the deployment rather than handling some =
specific mechanism or application only.

Thanks
Partha
>-----Original Message-----
>From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
>Of Randell Jesup
>Sent: Saturday, March 17, 2012 3:40 AM
>To: rtcweb@ietf.org
>Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
>
>On 3/16/2012 3:25 PM, Ravindran, Parthasarathi wrote:
>> Magnus,
>>
>> As I mentioned in earlier mail thread (http://www.ietf.org/mail-
>archive/web/rtcweb/current/msg03148.html), RTP has to be supported by
>RTCWeb client with local user consent.
>
>That is your stipulation.  Thus far this does not have consensus.
>
>> To clarify why RTP is not harmful in some usage, VPN access of
>intranet RTCWeb client in public internet usecase is as follows:
>>
>> Usecase details: Enterprise employee Alice access enterprise (HTTP&
>WebRTC compliant) intranet with VPN connection using Airport WiFi
>internet connection, Alice will have the mechanism to call another
>employee Bob in the same enterprise using WebRTC session and the session
>between Alice and bob will be plain RTP in VPN network.
>>
>> Security aspect: Here, accessing (HTTP) intranet browsing today is
>safe even though Alice is using Airport Wifi internet connection as he
>has VPN connection (IPSec) and no dependency on HTTPS. In the same way,
>WebRTC mechanism should not mandate for SRTP-DTLS in media and MUST
>allow RTP.  The local user consent (configuration) is required to
>restrict this usage by Dr Evil website.
>
>This says the media is in theory 'safe' as RTP over a VPN.  Similar
>arguments were used for "inside the corporate network" in previous
>discussions.  That doesn't mean that the spec should allow for RTP; for
>example it might create bid-down attack possibilities - and the
>application has lots of trouble knowing if hte link really is secure
>(and doesn't transition off security anywhere after leaving the
>machine).
>
>--
>Randell Jesup
>randell-ietf@jesup.org
>
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb

From pravindran@sonusnet.com  Fri Mar 16 21:32:01 2012
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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: Eric Rescorla <ekr@rtfm.com>, Roman Shpount <roman@telurix.com>
Thread-Topic: [rtcweb] Resolving RTP/SDES question in Paris
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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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Ekr,

As Roman mentioned for specific military & prison deployment, some of the E=
nterprise will also restrict public internet HTTPS usage as it may risk Ent=
erprise business.

Thanks
Partha

>-----Original Message-----
>From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
>Of Eric Rescorla
>Sent: Saturday, March 17, 2012 9:28 AM
>To: Roman Shpount
>Cc: rtcweb@ietf.org
>Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
>
>On Fri, Mar 16, 2012 at 4:35 PM, Roman Shpount <roman@telurix.com>
>wrote:
>> On Fri, Mar 16, 2012 at 6:55 PM, I=F1aki Baz Castillo <ibc@aliax.net>
>wrote:
>>>
>>> Giving the security responsability to end users as a configurable
>>> option is not a good idea. The world would be much better if the
>>> browsers would not allow plain HTTP, but it's not possible to change
>>> the WWW requeriments and mandate HTTPS given the ammount of plain
>>> HTTP deployments. In contrast, WebRTC does not exist yet so *now* is
>>> the moment to mandate security.
>>>
>>
>> And you imply that the fact that security is most of the times
>> unnecessary and often prohibited, has absolutely nothing to do with
>> using HTTP over HTTPS. If you think people in the military or in
>> prisons do not use web browsers, think again. Requiring secure
>> communications will prevent them from using WebRTC.
>
>This doesn't seem obvious at all. Rather it will require the people who
>wish to access their communications to do so explicitly in the same way
>as enterprises who wish to monitor HTTPS do so now.
>
>-Ekr
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb

From randell-ietf@jesup.org  Fri Mar 16 22:02:21 2012
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Subject: Re: [rtcweb] SDES vs DTLS-SRTP revisited
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On 3/16/2012 11:04 AM, Oscar Ohlsson wrote:
> Hi,
>
> There were a couple of things that I didn't explain properly in my SDES vs DTLS-SRTP presentation at the interim meeting. Since time will be short at the next meeting as well, I thought I would try to explain myself in an email.
>
> I noticed that the discussion quickly turned into a direct comparison of the two protocols. When this happens DTLS-SRTP definitely comes out as the winner and trying to defend SDES just seems foolish. But this is not the right focus of the discussion IMO. We should be looking at the system as a whole and determine more exactly the scenarios where SDES poses a security risk.
>
> The first major argument against SDES is that it doesn't offer forward secrecy. Obviously, if the remote endpoint does not support DTLS-SRTP you have to switch to SDES or plain RTP at some point. And from that point and onwards you will never have forward secrecy. The only difference is that the conversion point is moved slightly to the left (one step?) when SDES is supported. However, if web developers starts using SDES without reason (i.e. when the remote endpoint is a browser) then I agree that the lack of forward secrecy is a problem. Do people on this list believe this will be the case? Can it be prevented somehow?
>
> The second major argument is that SDES makes it possible for the service provider to intercept calls undetectable. I've been thinking about this for some time but still haven't understood the problem completely. Say that you're on a call and want to determine if the service provider is listening in:

It also lets the service provider record the key and provide it later to 
someone who separately had recorded the packets on the media path, 
without requiring a true realtime MITM with packet forwarding and 
realtime decryption.  Note that packet capturing is a technology that's 
in use.

>
> Case 1: Both endpoints support DTLS-SRTP (can be determined by asking the other party)
>
> If SDES is used you directly conclude that something fishy is going on and hang up. If DTLS-SRTP is used you compare fingerprints and hang up if they don't match. (I've assumed that a user can view detailed security information in the browser).

As has been discussed, "comparing fingerprints" is beyond 99.9% of users 
- not from ability, but from understanding what they are and what the 
connotations are.  It's way, way too geeky for even most 
privacy-sensitive people to understand, let along go through to 
considerable hassle to do.  The same argument applied to PGPhone, and a 
bunch of other protocols.  Even SSH's "unknown host" warning is rarely 
ever given a second thought before hitting "ok" - I've seen exactly 1 
user actually check the host key - and that was a security standards 
person.  (Yes, I know there are some people with very specific security 
requirements who I'm sure check, but they're beyond just being the 
exception to the rule.)

> Case 2: The other endpoint does not support DTLS-SRTP
>
> Regardless of whether SDES is allowed or not you will never be able to tell if the call is being intercepted. The best thing to do in this case is probably to simply hang up.
>
> Does the fingerprint mismatch in case 1 somehow serve as a stronger indication that the call is being intercepted? What action would a suspicious user take besides hanging up? I would be happy if someone could explain this to me.

You can be MITM'd with DTLS-SRTP without identity validation, though in 
theory reading fingerprints would catch it.

This is part of the point of the Identity mechanisms proposed by EKR 
(using the BrowserId example); to verify the identity of the other party 
in the call in a way that's more usable than reading fingerprints (and 
identifies you as the owner of a particular email address, which is 
useful if you don't personally know the person you're calling.)

Also realize that exposing the key to the JS app also enables it to 
provide that key directly to other parties, and to mirror traffic to a 
3rd-party who would not have to be in the media path.  It could in 
theory decode the data itself and process it or forward it.

None of these are really surprising, and the big issue is "how much 
security do you give up (and to whom)" and "how easy is it to use"?

-- 
Randell Jesup
randell-ietf@jesup.org


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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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On Fri, Mar 16, 2012 at 9:31 PM, Ravindran, Parthasarathi
<pravindran@sonusnet.com> wrote:
> Ekr,
>
> As Roman mentioned for specific military & prison deployment, some of the Enterprise will also restrict public internet HTTPS usage as it may risk Enterprise business.

Generally the way this is done is by deploying MITM proxies.


-Ekr

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Cc: Randell Jesup <randell-ietf@jesup.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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On 3/17/2012 12:13 AM, Ravindran, Parthasarathi wrote:
> Randell,
>
> In my usecase, the application will not be able to access the website without VPN connection . Please explain your bid-down attack in my usecase.

It may not in your particular case.  But the application would have to 
know the connection was over VPN and somehow verify that (how?)  Careful 
about saying "the server can't be accessed unless the VPN is up" because 
traffic can be spoofed (and not just DNS traffic) if the attacker 
controls the first router/etc.

The solution has to be solid without the user having to verify things 
and such that there isn't a high likelihood of minor mistakes 
compromising the user's security.

> You can speculate that all intranet in the world shall be broken in a moment if they use HTTP but IMO, it is just speculation. IETF specification has to be generic enough to serve all the deployment rather than handling some specific mechanism or application only.

Well, no, IETF specs do not have to serve "all" deployments or 
applications.  That's why we try to agree to use-cases and have goals.  
Often trying to solve *all* variations or cases leads to a 
never-finished, too-complex design.  It can at times be better to focus 
on a few related or similar cases that we know there's a need for, and 
later specs can extend it or provide alternatives.

Of course, we try to include as wide a range of uses as possible, and so 
your use-case may be included, but I don't think we included it in our 
use-case document last fall.

-- 
Randell Jesup
randell-ietf@jesup.org


From HKaplan@acmepacket.com  Fri Mar 16 23:17:44 2012
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Randell Jesup <randell-ietf@jesup.org>
Thread-Topic: [rtcweb] SDES vs DTLS-SRTP revisited
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Subject: Re: [rtcweb] SDES vs DTLS-SRTP revisited
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On Mar 17, 2012, at 12:59 AM, Randell Jesup wrote:

>> The second major argument is that SDES makes it possible for the service=
 provider to intercept calls undetectable. I've been thinking about this fo=
r some time but still haven't understood the problem completely. Say that y=
ou're on a call and want to determine if the service provider is listening =
in:
>=20
> It also lets the service provider record the key and provide it later to =
someone who separately had recorded the packets on the media path, without =
requiring a true realtime MITM with packet forwarding and realtime decrypti=
on.  Note that packet capturing is a technology that's in use.

And the exact same thing can be accomplished with DTLS-SRTP.  Imagine if DT=
LS-SRTP is mandated, and PSTN gateways go do DTLS-SRTP.  Alice makes a call=
 to Bob.  The call goes through the gateway.  At that point the gateway has=
 the actual keys used for encrypting/decrypting the SRTP.  At any future ti=
me, if the gateway stored that key anywhere (such as in a log), the key can=
 be retrieved and wireshark'ed captures of the encrypted SRTP can be decryp=
ted. =20

So then someone might say "sure but at least the web-server won't be able t=
o decrypt it", so just imagine the web-server routes the call through such =
a gateway it owns.  Same problem.  If they want to do it, they can do it - =
i.e., if they're malicious, they'll be malicious.

The only thing that makes DTLS-SRTP any "better", is that you can go and ve=
rify the fingerprint with the far-end human (or at least you hope it's the =
far-end human).  If *and only if* you perform that step, is anything knowin=
gly "better".  Of course as you've pointed out earlier, no one will perform=
 that step, except for folks who probably wouldn't trust this whole thing t=
o begin with.  So unless the identity provider thing happens and is mandate=
d - and I don't hear anyone suggesting that be mandatory to use in order fo=
r a call to succeed - then we're arguing about the thickness of the Emperor=
's clothes, when he's not wearing any.  I'd be happy if we could at least g=
et him a T-shirt.  :)

-hadriel


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Subject: Re: [rtcweb] SDES vs DTLS-SRTP revisited
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On 3/17/2012 2:17 AM, Hadriel Kaplan wrote:
> On Mar 17, 2012, at 12:59 AM, Randell Jesup wrote:
>
>>> The second major argument is that SDES makes it possible for the service provider to intercept calls undetectable. I've been thinking about this for some time but still haven't understood the problem completely. Say that you're on a call and want to determine if the service provider is listening in:
>> It also lets the service provider record the key and provide it later to someone who separately had recorded the packets on the media path, without requiring a true realtime MITM with packet forwarding and realtime decryption.  Note that packet capturing is a technology that's in use.
> And the exact same thing can be accomplished with DTLS-SRTP.  Imagine if DTLS-SRTP is mandated, and PSTN gateways go do DTLS-SRTP.  Alice makes a call to Bob.  The call goes through the gateway.  At that point the gateway has the actual keys used for encrypting/decrypting the SRTP.  At any future time, if the gateway stored that key anywhere (such as in a log), the key can be retrieved and wireshark'ed captures of the encrypted SRTP can be decrypted.

Of course - gateways can tap off anything.  No shock there.

> The only thing that makes DTLS-SRTP any "better", is that you can go and verify the fingerprint with the far-end human (or at least you hope it's the far-end human).  If *and only if* you perform that step, is anything knowingly "better".  Of course as you've pointed out earlier, no one will perform that step, except for folks who probably wouldn't trust this whole thing to begin with.  So unless the identity provider thing happens and is mandated - and I don't hear anyone suggesting that be mandatory to use in order for a call to succeed - then we're arguing about the thickness of the Emperor's clothes, when he's not wearing any.  I'd be happy if we could at least get him a T-shirt.  :)

Well imagine if the user's identity provider was Google, and it keys off 
your normal google login.  When when you call anyone, you have a 
verified identity.  Or if you're even a bit privacy-aware, you may want 
to know it's joe and not MITM-man you're talking to, and if the UI is 
arranged well, it's fairly painless.  It doesn't have to be mandatory to 
be useful.

And for some of the distributed-hash uses we've discussed, effectively 
mandating identity would make sense, but that could be at the 
application level.

-- 
Randell Jesup
randell-ietf@jesup.org


From tim@phonefromhere.com  Sat Mar 17 04:52:42 2012
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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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On 17 Mar 2012, at 04:13, Ravindran, Parthasarathi wrote:

> Randell,
>=20
> In my usecase, the application will not be able to access the website =
without VPN connection . Please explain your bid-down attack in my =
usecase.=20

I think there probably is an vulnerability if Bob goes on line (to =
upgrade the call to video say) and happens to be on the
same wifi ISP as Alice. The ICE probes may find a route between them =
that does not involve Alice's corporate VPN, but rather
over the non-routable ISP's /16 network. This would result in RTP in the =
clear being visible on both parties local wifi.

It's a slightly contrived case, but not impossible.=20

Tim.=

From ibc@aliax.net  Sat Mar 17 09:24:37 2012
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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2012/3/17 Roman Shpount <roman@telurix.com>:
> Furthermore comparison between HTTP/HTTPS and RTP/SRTP in not entirely
> appropriate. HTTPS communications are reasonably secure. SRTP calls, unle=
ss
> we make identity management a requirement for WebRTC, are only giving peo=
ple
> a false sense of security. The fact that browser uses SRTP does not, in a=
ny
> way, imply that the communication is secure. If the key was delivered ove=
r
> HTTP, anybody intercepting your IP traffic can listen to it. If we are
> dealing with the server based attack then all hell breaks loose. So, unle=
ss
> we requiring that only connections are over DTLS-SRTP using an SDP signed=
 by
> a verified remote party are allowed, we allow unsecure connections.

Well, I just said that in order to get secure communications WebRTC
has to mandate SRTP and disallow plain RTP. I didn't enter in details
but of course I agree with you. However to accomplish with your
points, my statements (SRTP is a MUST and RTP should not be allowed)
are valid.


> We need
> to show an appropriate warning for all of them, and warning RTP vs SDES-S=
RTP
> vs unsigned DTLS-SRTP should not be very different. It should say "You ar=
e
> trying to communicate over unsecure peer--to-peer channel with an unknown
> party. Continue?" There is no, "you are trying to use slightly more secur=
e
> channel. Is this ok?".

I don't entirely agree. In my example (the airport with open WiFi) the
signaling could be secured via HTTPS and the media secured with
SDES-SRTP. In this way other users in same open WiFi network cannot
monitor the signaling so they cannot get the SDES key, and therefore
they cannot monitor the SDES-SRTP stream. This is really better than
allowing plain RTP in an open WiFi network, am I wrong?


Regards.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: Tim Panton <tim@phonefromhere.com>
Thread-Topic: [rtcweb] Resolving RTP/SDES question in Paris
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Cc: Randell Jesup <randell-ietf@jesup.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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Tim,

I think that you miss the point of RTP and ICE are (IPSec) encrypted betwee=
n (RTCWeb client) endpoint and Enterprise during VPN connection. So, RTP & =
ICE packets from endpoint are routed in WiFi ISP as IP packet with encrypte=
d payload and no security issues.

Thanks
Partha

>-----Original Message-----
>From: Tim Panton [mailto:tim@phonefromhere.com]
>Sent: Saturday, March 17, 2012 5:23 PM
>To: Ravindran, Parthasarathi
>Cc: Randell Jesup; rtcweb@ietf.org
>Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
>
>
>On 17 Mar 2012, at 04:13, Ravindran, Parthasarathi wrote:
>
>> Randell,
>>
>> In my usecase, the application will not be able to access the website
>without VPN connection . Please explain your bid-down attack in my
>usecase.
>
>I think there probably is an vulnerability if Bob goes on line (to
>upgrade the call to video say) and happens to be on the same wifi ISP as
>Alice. The ICE probes may find a route between them that does not
>involve Alice's corporate VPN, but rather over the non-routable ISP's
>/16 network. This would result in RTP in the clear being visible on both
>parties local wifi.
>
>It's a slightly contrived case, but not impossible.
>
>Tim.

From ibc@aliax.net  Sat Mar 17 09:33:44 2012
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Sat, 17 Mar 2012 17:33:21 +0100
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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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2012/3/17 Ravindran, Parthasarathi <pravindran@sonusnet.com>:
> In my usecase, the application will not be able to access the website wit=
hout VPN connection . Please explain your bid-down attack in my usecase.

IMHO the topic here is not about finding a usecase in which using
plain RTP is secure (as in your example involving a VPN). Rather, the
topic is about finding common usecases in which the end user joins
unsecure media communications without the proper knowledge of "what
that means" so it accepts it. If the security depends on the end user
approval ("this web site does not provide secure audio/video
communications, do you accept it?") the we are lost (IMHO).

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From ekr@rtfm.com  Sat Mar 17 09:38:40 2012
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From: Eric Rescorla <ekr@rtfm.com>
Date: Sat, 17 Mar 2012 09:37:58 -0700
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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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On Sat, Mar 17, 2012 at 9:33 AM, Ravindran, Parthasarathi
<pravindran@sonusnet.com> wrote:
> Tim,
>
> I think that you miss the point of RTP and ICE are (IPSec) encrypted betw=
een (RTCWeb client) endpoint and Enterprise during VPN connection. So, RTP =
& ICE packets from endpoint are routed in WiFi ISP as IP packet with encryp=
ted payload and no security issues.

Partha,

I don't find the scenario you suggest particularly compelling.

Yes, it's true that it's more secure to run your communications over
a VPN than not, but it's not obviously the case that you necessarily
trust everyone who is on the VPN (after all, this is why companies
with VPNs run internal access controls on their systems). And it's
of course yet more secure to run point-to-point encryption even
when you are operating on a VPN.

With that said, even if we were to stipulate that the VPN case
is safe, you haven't explained why it's desirable not to cryptographically
protect the traffic in that case. Since it's easier and safer to
simply have only a secure mode of operation, I don't see
that this scenario is an argument in favor of RTP.

-Ekr

From roman@telurix.com  Sat Mar 17 09:40:07 2012
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--e89a8ff1c35e28c12104bb72fa3a
Content-Type: text/plain; charset=ISO-8859-1

On Fri, Mar 16, 2012 at 11:58 PM, Eric Rescorla <ekr@rtfm.com> wrote:

> On Fri, Mar 16, 2012 at 4:35 PM, Roman Shpount <roman@telurix.com> wrote:
> >
> > And you imply that the fact that security is most of the times
> unnecessary
> > and often prohibited, has absolutely nothing to do with using HTTP over
> > HTTPS. If you think people in the military or in prisons do not use web
> > browsers, think again. Requiring secure communications will prevent them
> > from using WebRTC.
>
> This doesn't seem obvious at all. Rather it will require the people who
> wish
> to access their communications to do so explicitly in the same way as
> enterprises who wish to monitor HTTPS do so now.
>
> -Ekr
>

Do you expect this done through some sort of desktop monitoring software or
does it mean that we need to design a network protocol for proxying WebRTC
media traffic? In case we need a recording proxy, we will need a protocol,
that on one hand, would allow the intermediary proxy to decode and monitor
all the traffic, and on the other hand would continue to work when
encryption and identity management is enabled. I am not sure something like
this is available, but, at the very least, it should include the mechanism
for a client to share the keys with the proxy.
_____________
Roman Shpount

--e89a8ff1c35e28c12104bb72fa3a
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<br clear=3D"all"><div class=3D"gmail_quote">On Fri, Mar 16, 2012 at 11:58 =
PM, Eric Rescorla <span dir=3D"ltr">&lt;<a href=3D"mailto:ekr@rtfm.com">ekr=
@rtfm.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=
=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div class=3D"im">On Fri, Mar 16, 2012 at 4:35 PM, Roman Shpount &lt;<a hre=
f=3D"mailto:roman@telurix.com">roman@telurix.com</a>&gt; wrote:<br>
&gt;<br>
&gt; And you imply that the fact that security is most of the times unneces=
sary<br>
&gt; and often prohibited, has absolutely nothing to do with using HTTP ove=
r<br>
&gt; HTTPS. If you think people in the military or in prisons do not use we=
b<br>
&gt; browsers, think again. Requiring secure communications will prevent th=
em<br>
&gt; from using WebRTC.<br>
<br>
</div>This doesn&#39;t seem obvious at all. Rather it will require the peop=
le who wish<br>
to access their communications to do so explicitly in the same way as<br>
enterprises who wish to monitor HTTPS do so now.<br>
<br>
-Ekr<br></blockquote><div><br></div></div>Do you expect this done through s=
ome sort of desktop monitoring software or does it mean that we need to des=
ign a network protocol for proxying WebRTC media traffic? In case we need a=
 recording proxy, we will need a protocol, that on one hand, would allow th=
e intermediary proxy to decode and monitor all the traffic, and on the othe=
r hand would continue to work when encryption and identity management is en=
abled. I am not sure something like this is available, but, at the very lea=
st, it should include the mechanism for a client to share the keys with the=
 proxy.<br>
_____________<br>Roman Shpount<br>
<br>

--e89a8ff1c35e28c12104bb72fa3a--

From ibc@aliax.net  Sat Mar 17 09:41:10 2012
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Sat, 17 Mar 2012 17:40:43 +0100
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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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2012/3/17 Ravindran, Parthasarathi <pravindran@sonusnet.com>:
> I think that you miss the point of RTP and ICE are (IPSec) encrypted betw=
een (RTCWeb client) endpoint and Enterprise during VPN connection. So, RTP =
& ICE packets from endpoint are routed in WiFi ISP as IP packet with encryp=
ted payload and no security issues.

Wrong. During ICE processing the WebRTC will try to find out all its
local network interfaces for the ICE offer, and these include the VPN
interface along with the "normal" interface used for the default
gateway 0.0.0.0 / 0::0. So in case Bob is in the same open WiFi
network, ICE would success and plain RTP would be used in a network
than anyone can monitor.

However I don't understand what are you trying to say. Are you looking
for a single usecase in which plain RTP is secure? Then let's be easy
and explain an example in which a user has no Internet and home and
uses his laptop to access a Web/WebRTC server also at home. Should
WebRTC allow plain RTP just for that case?

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From HKaplan@acmepacket.com  Sat Mar 17 10:29:47 2012
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Randell Jesup <randell-ietf@jesup.org>
Thread-Topic: [rtcweb] SDES vs DTLS-SRTP revisited
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Subject: Re: [rtcweb] SDES vs DTLS-SRTP revisited
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On Mar 17, 2012, at 2:34 AM, Randell Jesup wrote:

> Of course - gateways can tap off anything.  No shock there.

Exactly. (and yes I know you knew that already :)
So my point is that DTLS-SRTP doesn't give you much more than SDES over HTT=
PS, because any number of DTLS-b2bua gateways can be along the path.  Obvio=
usly it's *easier* for a malicious or rogue server to read your media when =
using SDES, because it doesn't have to do be a full media-b2bua to do so, b=
ut it's still possible either way.  Until and unless you verify the fingerp=
rint.


>> The only thing that makes DTLS-SRTP any "better", is that you can go and=
 verify the fingerprint with the far-end human (or at least you hope it's t=
he far-end human).  If *and only if* you perform that step, is anything kno=
wingly "better".  Of course as you've pointed out earlier, no one will perf=
orm that step, except for folks who probably wouldn't trust this whole thin=
g to begin with.  So unless the identity provider thing happens and is mand=
ated - and I don't hear anyone suggesting that be mandatory to use in order=
 for a call to succeed - then we're arguing about the thickness of the Empe=
ror's clothes, when he's not wearing any.  I'd be happy if we could at leas=
t get him a T-shirt.  :)
>=20
> Well imagine if the user's identity provider was Google, and it keys off =
your normal google login.  When when you call anyone, you have a verified i=
dentity. =20

Not really, as far as I can tell.  You have an identity Google claims is va=
lid.  If you trust Google not to monitor your conversation, and to have goo=
d identity validation, then sure you can trust the identity is what Google =
claims it to be.  And that would NOT be the case for "when you call anyone"=
 - only when you call other Google users and they are reached by their WebR=
TC connection to Google.  If, for example, you called another Google user w=
ho's only on the PSTN at that time and Google offered call-forwarding, then=
 it would go through their DTLS-gateway and we're back to square one.

Furthermore, if you trusted Google that much already, it's not a huge leap =
to trust that your SDES over HTTPS to Google is not getting sent in the cle=
ar elsewhere by Google, and is not getting recorded by Google.  It is no do=
ubt easier for them to screw that up, but it's also not the scenario we hav=
e to worry about - because in the Google case for user-to-user WebRTC-to-We=
bRTC calls, we'd get DTLS-SRTP *anyway*.  No one's suggesting we don't mand=
ate DTLS-SRTP also be implemented by browsers, and clearly we'd mandate tha=
t if the browser receives an offer with DTLS-SRTP, it must use it.  So we'd=
 get DTLS-SRTP within a WebRTC domain.  That's not the problem scenario at =
hand.

The problem scenario is what to do about the cases when the call goes to/fr=
om the PSTN, or to/from an interworking gateway to another protocol/domain =
that does not have support for end-to-end DTLS-SRTP.  We can either mandate=
 those gateways/IWF implement DTLS-SRTP, or minimally just SDES.  The advan=
tages of DTLS-SRTP over SDES in those scenarios are far less clear, while t=
he advantages of SDES over DTLS in those scenarios are clear.

-hadriel


From martin.thomson@gmail.com  Sat Mar 17 13:45:20 2012
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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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On 17 March 2012 09:40, Roman Shpount <roman@telurix.com> wrote:
> Do you expect this done through some sort of desktop monitoring software or
> does it mean that we need to design a network protocol for proxying WebRTC
> media traffic? In case we need a recording proxy, we will need a protocol,
> that on one hand, would allow the intermediary proxy to decode and monitor
> all the traffic, and on the other hand would continue to work when
> encryption and identity management is enabled. I am not sure something like
> this is available, but, at the very least, it should include the mechanism
> for a client to share the keys with the proxy.

This isn't that hard.  I run an intercepting proxy that offers its own
certificate.  Then I explicitly place a trust anchor for that
certificate in your browser.  The proxy is then able to intercept any
encrypted traffic it chooses to.  The proxy does two lots of crypto
work, but that's not as much of a big deal as you might think.

No protocol changes at all.  The proxy might need to do more work to
convince you that the identity hasn't been replaced (swap itself in as
the identity provider, for example), but anything is possible once it
is "trusted".

--Martin

From igor.faynberg@alcatel-lucent.com  Sat Mar 17 14:14:00 2012
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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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..

On 3/17/2012 4:45 PM, Martin Thomson wrote:
> ... Then I explicitly place a trust anchor for that
> certificate in your browser.

How?  I thought the browser would never allow that to happen. (I assumed 
it would come provisioned with anchors,  or would allow anchor provision 
through some different channel.)

Igor

From bernard_aboba@hotmail.com  Sat Mar 17 14:17:18 2012
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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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EKR said:=20

> Partha=2C
>=20
> I don't find the scenario you suggest particularly compelling.
>=20
> Yes=2C it's true that it's more secure to run your communications over
> a VPN than not=2C but it's not obviously the case that you necessarily
> trust everyone who is on the VPN (after all=2C this is why companies
> with VPNs run internal access controls on their systems).=20

[BA] The NSA has made exactly this point:  it is not sufficient to run
unencrypted RTP over a VPN=3B  the signaling MUST be secured
(e.g. via TLS) and SRTP MUST also be used.=20

I would also note that NSA allows use of SDES/SRTP over a VPN=2C since
this is widely support=2C even though this was not their originally favored=
 solution=20

For details=2C see:
http://www.theverge.com/2012/3/2/2838729/nsa-project-fishbowl-secure-androi=
d-devices-network




 		 	   		  =

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<html>
<head>
<style><!--
.hmmessage P
{
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}
body.hmmessage
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font-size: 10pt=3B
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<body class=3D'hmmessage'><div dir=3D'ltr'>
EKR said: <br><br><div>&gt=3B Partha=2C<br>&gt=3B <br>&gt=3B I don't find t=
he scenario you suggest particularly compelling.<br>&gt=3B <br>&gt=3B Yes=
=2C it's true that it's more secure to run your communications over<br>&gt=
=3B a VPN than not=2C but it's not obviously the case that you necessarily<=
br>&gt=3B trust everyone who is on the VPN (after all=2C this is why compan=
ies<br>&gt=3B with VPNs run internal access controls on their systems). <br=
><br>[BA] The NSA has made exactly this point:&nbsp=3B it is not sufficient=
 to run<br>unencrypted RTP over a VPN=3B&nbsp=3B the signaling MUST be secu=
red<br>(e.g. via TLS) and SRTP MUST also be used. <br><br>I would also note=
 that NSA allows use of SDES/SRTP over a VPN=2C since<br>this is widely sup=
port=2C even though this was not their originally favored solution <br><br>=
For details=2C see:<br>http://www.theverge.com/2012/3/2/2838729/nsa-project=
-fishbowl-secure-android-devices-network<br><br><br><br><br></div> 		 	   	=
	  </div></body>
</html>=

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On 17 March 2012 14:14, Igor Faynberg <igor.faynberg@alcatel-lucent.com> wr=
ote:
> On 3/17/2012 4:45 PM, Martin Thomson wrote:
>> ... Then I explicitly place a trust anchor for that
>> certificate in your browser.
>
> How? =C2=A0I thought the browser would never allow that to happen. (I ass=
umed it
> would come provisioned with anchors, =C2=A0or would allow anchor provisio=
n
> through some different channel.)

I own your computer.  Or I make it a condition of you adding your
computer to my network.

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Ah, so this is the enterprise scenario.  I get it. Thanks!

This case though would not worry me much because when you own my 
computer you make me confirm that I understand that I should not expect 
any privacy whatsoever.

Igor

On 3/17/2012 5:31 PM, Martin Thomson wrote:
> On 17 March 2012 14:14, Igor Faynberg<igor.faynberg@alcatel-lucent.com>  wrote:
>> On 3/17/2012 4:45 PM, Martin Thomson wrote:
>>> ... Then I explicitly place a trust anchor for that
>>> certificate in your browser.
>> How?  I thought the browser would never allow that to happen. (I assumed it
>> would come provisioned with anchors,  or would allow anchor provision
>> through some different channel.)
> I own your computer.  Or I make it a condition of you adding your
> computer to my network.

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On Sat, Mar 17, 2012 at 4:45 PM, Martin Thomson <martin.thomson@gmail.com>wrote:

> This isn't that hard.  I run an intercepting proxy that offers its own
> certificate.  Then I explicitly place a trust anchor for that
> certificate in your browser.  The proxy is then able to intercept any
> encrypted traffic it chooses to.  The proxy does two lots of crypto
> work, but that's not as much of a big deal as you might think.
>
> No protocol changes at all.  The proxy might need to do more work to
> convince you that the identity hasn't been replaced (swap itself in as
> the identity provider, for example), but anything is possible once it
> is "trusted".
>
>
I understand how this will work with HTTPS, but how would this work with
WebRTC? Signaling is traveling over some separate path withing a websocket
or some other HTTP-based encapsulating protocol. Keys/Key signatures are
part of this signaling, which is almost impossible to intercept, even if
you have access to all HTTP(S) traffic, since each application can define
the protocol being used. Intercepting proxy needs access to this info, or
it cannot decode the media or spoof the signatures. Unless I am missing
something, client needs to send the SDP to the proxy and get the modified
SDP back, which probably requires some sort of network protocol.
_____________
Roman Shpount

--047d7b15aee928ae3e04bb8f3448
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<br clear=3D"all"><div class=3D"gmail_quote">On Sat, Mar 17, 2012 at 4:45 P=
M, Martin Thomson <span dir=3D"ltr">&lt;<a href=3D"mailto:martin.thomson@gm=
ail.com">martin.thomson@gmail.com</a>&gt;</span> wrote:<br><blockquote clas=
s=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;pad=
ding-left:1ex">
This isn&#39;t that hard. =A0I run an intercepting proxy that offers its ow=
n<br>
certificate. =A0Then I explicitly place a trust anchor for that<br>
certificate in your browser. =A0The proxy is then able to intercept any<br>
encrypted traffic it chooses to. =A0The proxy does two lots of crypto<br>
work, but that&#39;s not as much of a big deal as you might think.<br>
<br>
No protocol changes at all. =A0The proxy might need to do more work to<br>
convince you that the identity hasn&#39;t been replaced (swap itself in as<=
br>
the identity provider, for example), but anything is possible once it<br>
is &quot;trusted&quot;.<br>
<font color=3D"#888888"></font><br></blockquote></div><br>I understand how =
this will work with HTTPS, but how would this work with WebRTC? Signaling i=
s traveling over some separate path withing a websocket or some other HTTP-=
based encapsulating protocol. Keys/Key signatures are part of this signalin=
g, which is almost impossible to intercept, even if you have access to all =
HTTP(S) traffic, since each application can define the protocol being used.=
 Intercepting proxy needs access to this info, or it cannot decode the medi=
a or spoof the signatures. Unless I am missing something, client needs to s=
end the SDP to the proxy and get the modified SDP back, which probably requ=
ires some sort of network protocol.<br>
_____________<br>Roman Shpount<br>
<br><br>

--047d7b15aee928ae3e04bb8f3448--

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From: "Ejzak, Richard P (Richard)" <richard.ejzak@alcatel-lucent.com>
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Date: Sun, 18 Mar 2012 23:10:03 -0500
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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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I would like to argue two points: 1) DTLS-SRTP with IdP is the only media s=
ecurity scheme that rtcweb should support; and 2) rtcweb should also suppor=
t RTP (no media security) with a clear requirement on the browser UI to war=
n the user of the complete lack of media security.

DTLS-SRTP

Some of the arguments made against DTLS-SRTP in recent emails are valid unl=
ess it is integrated with trusted IdP that is sufficient to verify the iden=
tity of the remote party.  With this one caveat, DTLS-SRTP is equal to or (=
mostly) better than SDES-SRTP in almost all respects (not to be repeated he=
re).  Even if the identity is suspect, the IdP can at least provide some de=
gree of assurance and accountability when an issue arises.  Without that, a=
 user has no recourse when a problem develops.

The key is that the browser UI must make clear to the user the identity of =
the remote party with whom media security has been established (even when i=
t is a network gateway, in which case the identify of the gateway operator)=
.  It should also be possible for the end user to mandate that a remote par=
ty identity is always verified before communication is allowed (perhaps eve=
n make this a requirement when media security is selected).  This is not to=
o burdensome a requirement if established up front to help manage SPIT.

When end-to-end security is possible, then Alice knows that she's talking t=
o Bob and vice versa.

Even when Alice is talking to someone reached via the PSTN, she can know th=
at she has a secure media connection to a gateway owned by a particular (th=
e identified) operator, thus establishing a significant part of the chain o=
f accountability.  We can't improve on the security available through the P=
STN, but we can improve on the security available to the point of interconn=
ection with the identified operator (which may be different from the owner =
of the rtcweb application).  With DTLS-SRTP the user can have some basis fo=
r deciding whether to trust the identified operator; else (even with SDES-S=
RTP) you must completely trust the owner of the rtcweb application and ther=
e is no way to identify or verify any security attributes associated with t=
he media.

Finally, there are multiple reasons to avoid supporting multiple media secu=
rity schemes.  It is difficult enough to expect the end user to make sure t=
hey are comfortable with the identity of the remote party of a secure media=
 connection, but to ask if they are comfortable with an alternative media s=
ecurity scheme is simply too much to expect.  Bid-down attacks are just too=
 easy and it is impossible to verify any remote party identity with SDES-SR=
TP.

RTP

There are too many reasonable use cases for unsecured media for us to ignor=
e them.  The browser should be very clear when media security is unavailabl=
e and make it easy to prevent communication without media security (perhaps=
 as a default setting), but there are legitimate use cases in various priva=
te environments (e.g., enterprise, jail) where media security may not be al=
lowed (or must somehow be compromised), or is just not available, that we s=
hould just support straight RTP.  As long as the user can control the condi=
tions under which media security is disabled, I do not see a problem.

Richard Ejzak



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<div class=3DSection1>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>I would like to argue two points: 1)
DTLS-SRTP with IdP is the only media security scheme that rtcweb should
support; and 2) rtcweb should also support RTP (no media security) with a c=
lear
requirement on the browser UI to warn the user of the complete lack of medi=
a
security.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>DTLS-SRTP<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Some of the arguments made against
DTLS-SRTP in recent emails are valid unless it is integrated with trusted I=
dP
that is sufficient to verify the identity of the remote party. &nbsp;With t=
his
one caveat, DTLS-SRTP is equal to or (mostly) better than SDES-SRTP in almo=
st
all respects (not to be repeated here). &nbsp;Even if the identity is suspe=
ct,
the IdP can at least provide some degree of assurance and accountability wh=
en
an issue arises. &nbsp;Without that, a user has no recourse when a problem
develops. &nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>The key is that the browser UI must ma=
ke
clear to the user the identity of the remote party with whom media security=
 has
been established (even when it is a network gateway, in which case the iden=
tify
of the gateway operator). &nbsp;It should also be possible for the end user=
 to
mandate that a remote party identity is always verified before communicatio=
n is
allowed (perhaps even make this a requirement when media security is select=
ed).
&nbsp;This is not too burdensome a requirement if established up front to h=
elp manage
SPIT. &nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>When end-to-end security is possible, =
then
<st1:City w:st=3D"on"><st1:place w:st=3D"on">Alice</st1:place></st1:City> k=
nows
that she&#8217;s talking to Bob and vice versa. &nbsp;<o:p></o:p></span></f=
ont></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Even when <st1:City w:st=3D"on"><st1:p=
lace
 w:st=3D"on">Alice</st1:place></st1:City> is talking to someone reached via=
 the
PSTN, she can know that she has a secure media connection to a gateway owne=
d by
a particular (the identified) operator, thus establishing a significant par=
t of
the chain of accountability. &nbsp;We can&#8217;t improve on the security
available through the PSTN, but we can improve on the security available to=
 the
point of interconnection with the identified operator (which may be differe=
nt
from the owner of the rtcweb application).&nbsp; With DTLS-SRTP the user ca=
n
have some basis for deciding whether to trust the identified operator; else
(even with SDES-SRTP) you must completely trust the owner of the rtcweb
application and there is no way to identify or verify any security attribut=
es
associated with the media.&nbsp; <o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Finally, there are multiple reasons to
avoid supporting multiple media security schemes.&nbsp; It is difficult eno=
ugh
to expect the end user to make sure they are comfortable with the identity =
of
the remote party of a secure media connection, but to ask if they are
comfortable with an alternative media security scheme is simply too much to
expect. &nbsp;Bid-down attacks are just too easy and it is impossible to ve=
rify
any remote party identity with SDES-SRTP.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>RTP<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>There are too many reasonable use case=
s
for unsecured media for us to ignore them. &nbsp;The browser should be very
clear when media security is unavailable and make it easy to prevent
communication without media security (perhaps as a default setting), but th=
ere
are legitimate use cases in various private environments (e.g., enterprise,
jail) where media security may not be allowed (or must somehow be compromis=
ed),
or is just not available, that we should just support straight RTP. &nbsp;A=
s
long as the user can control the conditions under which media security is
disabled, I do not see a problem.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Richard Ejzak<o:p></o:p></span></font>=
</p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

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Date: Mon, 19 Mar 2012 00:21:28 -0400
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From: Roman Shpount <roman@telurix.com>
To: "Ejzak, Richard P (Richard)" <richard.ejzak@alcatel-lucent.com>
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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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+1
_____________
Roman Shpount


On Mon, Mar 19, 2012 at 12:10 AM, Ejzak, Richard P (Richard) <
richard.ejzak@alcatel-lucent.com> wrote:

> ** **
>
> I would like to argue two points: 1) DTLS-SRTP with IdP is the only media
> security scheme that rtcweb should support; and 2) rtcweb should also
> support RTP (no media security) with a clear requirement on the browser U=
I
> to warn the user of the complete lack of media security.****
>
> ** **
>
> DTLS-SRTP****
>
> ** **
>
> Some of the arguments made against DTLS-SRTP in recent emails are valid
> unless it is integrated with trusted IdP that is sufficient to verify the
> identity of the remote party.  With this one caveat, DTLS-SRTP is equal t=
o
> or (mostly) better than SDES-SRTP in almost all respects (not to be
> repeated here).  Even if the identity is suspect, the IdP can at least
> provide some degree of assurance and accountability when an issue arises.
>  Without that, a user has no recourse when a problem develops.  ****
>
> ** **
>
> The key is that the browser UI must make clear to the user the identity o=
f
> the remote party with whom media security has been established (even when
> it is a network gateway, in which case the identify of the gateway
> operator).  It should also be possible for the end user to mandate that a
> remote party identity is always verified before communication is allowed
> (perhaps even make this a requirement when media security is selected).
>  This is not too burdensome a requirement if established up front to help
> manage SPIT.  ****
>
> ** **
>
> When end-to-end security is possible, then ****Alice**** knows that she=
=92s
> talking to Bob and vice versa.  ****
>
> ** **
>
> Even when ****Alice**** is talking to someone reached via the PSTN, she
> can know that she has a secure media connection to a gateway owned by a
> particular (the identified) operator, thus establishing a significant par=
t
> of the chain of accountability.  We can=92t improve on the security avail=
able
> through the PSTN, but we can improve on the security available to the poi=
nt
> of interconnection with the identified operator (which may be different
> from the owner of the rtcweb application).  With DTLS-SRTP the user can
> have some basis for deciding whether to trust the identified operator; el=
se
> (even with SDES-SRTP) you must completely trust the owner of the rtcweb
> application and there is no way to identify or verify any security
> attributes associated with the media.  ****
>
> ** **
>
> Finally, there are multiple reasons to avoid supporting multiple media
> security schemes.  It is difficult enough to expect the end user to make
> sure they are comfortable with the identity of the remote party of a secu=
re
> media connection, but to ask if they are comfortable with an alternative
> media security scheme is simply too much to expect.  Bid-down attacks are
> just too easy and it is impossible to verify any remote party identity wi=
th
> SDES-SRTP.****
>
> ** **
>
> RTP****
>
> ** **
>
> There are too many reasonable use cases for unsecured media for us to
> ignore them.  The browser should be very clear when media security is
> unavailable and make it easy to prevent communication without media
> security (perhaps as a default setting), but there are legitimate use cas=
es
> in various private environments (e.g., enterprise, jail) where media
> security may not be allowed (or must somehow be compromised), or is just
> not available, that we should just support straight RTP.  As long as the
> user can control the conditions under which media security is disabled, I
> do not see a problem.****
>
> ** **
>
> Richard Ejzak****
>
> ** **
>
> ** **
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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+1<br clear=3D"all">_____________<br>Roman Shpount<br>
<br><br><div class=3D"gmail_quote">On Mon, Mar 19, 2012 at 12:10 AM, Ejzak,=
 Richard P (Richard) <span dir=3D"ltr">&lt;<a href=3D"mailto:richard.ejzak@=
alcatel-lucent.com">richard.ejzak@alcatel-lucent.com</a>&gt;</span> wrote:<=
br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">




<u></u>
<u></u>





<div link=3D"blue" vlink=3D"blue" lang=3D"EN-US">

<div>

<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial"><span style=3D"f=
ont-size:10.0pt;font-family:Arial;color:navy">I would like to argue two poi=
nts: 1)
DTLS-SRTP with IdP is the only media security scheme that rtcweb should
support; and 2) rtcweb should also support RTP (no media security) with a c=
lear
requirement on the browser UI to warn the user of the complete lack of medi=
a
security.<u></u><u></u></span></font></p>

<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial"><span style=3D"f=
ont-size:10.0pt;font-family:Arial;color:navy"><u></u>=A0<u></u></span></fon=
t></p>

<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial"><span style=3D"f=
ont-size:10.0pt;font-family:Arial;color:navy">DTLS-SRTP<u></u><u></u></span=
></font></p>

<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial"><span style=3D"f=
ont-size:10.0pt;font-family:Arial;color:navy"><u></u>=A0<u></u></span></fon=
t></p>

<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial"><span style=3D"f=
ont-size:10.0pt;font-family:Arial;color:navy">Some of the arguments made ag=
ainst
DTLS-SRTP in recent emails are valid unless it is integrated with trusted I=
dP
that is sufficient to verify the identity of the remote party. =A0With this
one caveat, DTLS-SRTP is equal to or (mostly) better than SDES-SRTP in almo=
st
all respects (not to be repeated here). =A0Even if the identity is suspect,
the IdP can at least provide some degree of assurance and accountability wh=
en
an issue arises. =A0Without that, a user has no recourse when a problem
develops. =A0<u></u><u></u></span></font></p>

<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial"><span style=3D"f=
ont-size:10.0pt;font-family:Arial;color:navy"><u></u>=A0<u></u></span></fon=
t></p>

<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial"><span style=3D"f=
ont-size:10.0pt;font-family:Arial;color:navy">The key is that the browser U=
I must make
clear to the user the identity of the remote party with whom media security=
 has
been established (even when it is a network gateway, in which case the iden=
tify
of the gateway operator). =A0It should also be possible for the end user to
mandate that a remote party identity is always verified before communicatio=
n is
allowed (perhaps even make this a requirement when media security is select=
ed).
=A0This is not too burdensome a requirement if established up front to help=
 manage
SPIT. =A0<u></u><u></u></span></font></p>

<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial"><span style=3D"f=
ont-size:10.0pt;font-family:Arial;color:navy"><u></u>=A0<u></u></span></fon=
t></p>

<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial"><span style=3D"f=
ont-size:10.0pt;font-family:Arial;color:navy">When end-to-end security is p=
ossible, then
<u></u><u></u>Alice<u></u><u></u> knows
that she=92s talking to Bob and vice versa. =A0<u></u><u></u></span></font>=
</p>

<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial"><span style=3D"f=
ont-size:10.0pt;font-family:Arial;color:navy"><u></u>=A0<u></u></span></fon=
t></p>

<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial"><span style=3D"f=
ont-size:10.0pt;font-family:Arial;color:navy">Even when <u></u><u></u>Alice=
<u></u><u></u> is talking to someone reached via the
PSTN, she can know that she has a secure media connection to a gateway owne=
d by
a particular (the identified) operator, thus establishing a significant par=
t of
the chain of accountability. =A0We can=92t improve on the security
available through the PSTN, but we can improve on the security available to=
 the
point of interconnection with the identified operator (which may be differe=
nt
from the owner of the rtcweb application).=A0 With DTLS-SRTP the user can
have some basis for deciding whether to trust the identified operator; else
(even with SDES-SRTP) you must completely trust the owner of the rtcweb
application and there is no way to identify or verify any security attribut=
es
associated with the media.=A0 <u></u><u></u></span></font></p>

<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial"><span style=3D"f=
ont-size:10.0pt;font-family:Arial;color:navy"><u></u>=A0<u></u></span></fon=
t></p>

<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial"><span style=3D"f=
ont-size:10.0pt;font-family:Arial;color:navy">Finally, there are multiple r=
easons to
avoid supporting multiple media security schemes.=A0 It is difficult enough
to expect the end user to make sure they are comfortable with the identity =
of
the remote party of a secure media connection, but to ask if they are
comfortable with an alternative media security scheme is simply too much to
expect. =A0Bid-down attacks are just too easy and it is impossible to verif=
y
any remote party identity with SDES-SRTP.<u></u><u></u></span></font></p>

<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial"><span style=3D"f=
ont-size:10.0pt;font-family:Arial;color:navy"><u></u>=A0<u></u></span></fon=
t></p>

<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial"><span style=3D"f=
ont-size:10.0pt;font-family:Arial;color:navy">RTP<u></u><u></u></span></fon=
t></p>

<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial"><span style=3D"f=
ont-size:10.0pt;font-family:Arial;color:navy"><u></u>=A0<u></u></span></fon=
t></p>

<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial"><span style=3D"f=
ont-size:10.0pt;font-family:Arial;color:navy">There are too many reasonable=
 use cases
for unsecured media for us to ignore them. =A0The browser should be very
clear when media security is unavailable and make it easy to prevent
communication without media security (perhaps as a default setting), but th=
ere
are legitimate use cases in various private environments (e.g., enterprise,
jail) where media security may not be allowed (or must somehow be compromis=
ed),
or is just not available, that we should just support straight RTP. =A0As
long as the user can control the conditions under which media security is
disabled, I do not see a problem.<u></u><u></u></span></font></p>

<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial"><span style=3D"f=
ont-size:10.0pt;font-family:Arial;color:navy"><u></u>=A0<u></u></span></fon=
t></p><font color=3D"#888888">

<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial"><span style=3D"f=
ont-size:10.0pt;font-family:Arial;color:navy">Richard Ejzak<u></u><u></u></=
span></font></p>

<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial"><span style=3D"f=
ont-size:10.0pt;font-family:Arial;color:navy"><u></u>=A0<u></u></span></fon=
t></p>

<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial"><span style=3D"f=
ont-size:10.0pt;font-family:Arial;color:navy"><u></u>=A0<u></u></span></fon=
t></p>

</font></div>

</div>


<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br>

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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: Roman Shpount <roman@telurix.com>, "Ejzak, Richard P (Richard)" <richard.ejzak@alcatel-lucent.com>
Thread-Topic: [rtcweb] Resolving RTP/SDES question in Paris
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+1

I'll add more usecase to strengthen RTP usage in WebRTC.

From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Roman Shpount
Sent: Monday, March 19, 2012 9:51 AM
To: Ejzak, Richard P (Richard)
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris

+1
_____________
Roman Shpount

On Mon, Mar 19, 2012 at 12:10 AM, Ejzak, Richard P (Richard) <richard.ejzak=
@alcatel-lucent.com<mailto:richard.ejzak@alcatel-lucent.com>> wrote:
I would like to argue two points: 1) DTLS-SRTP with IdP is the only media s=
ecurity scheme that rtcweb should support; and 2) rtcweb should also suppor=
t RTP (no media security) with a clear requirement on the browser UI to war=
n the user of the complete lack of media security.

DTLS-SRTP

Some of the arguments made against DTLS-SRTP in recent emails are valid unl=
ess it is integrated with trusted IdP that is sufficient to verify the iden=
tity of the remote party.  With this one caveat, DTLS-SRTP is equal to or (=
mostly) better than SDES-SRTP in almost all respects (not to be repeated he=
re).  Even if the identity is suspect, the IdP can at least provide some de=
gree of assurance and accountability when an issue arises.  Without that, a=
 user has no recourse when a problem develops.

The key is that the browser UI must make clear to the user the identity of =
the remote party with whom media security has been established (even when i=
t is a network gateway, in which case the identify of the gateway operator)=
.  It should also be possible for the end user to mandate that a remote par=
ty identity is always verified before communication is allowed (perhaps eve=
n make this a requirement when media security is selected).  This is not to=
o burdensome a requirement if established up front to help manage SPIT.

When end-to-end security is possible, then Alice knows that she's talking t=
o Bob and vice versa.

Even when Alice is talking to someone reached via the PSTN, she can know th=
at she has a secure media connection to a gateway owned by a particular (th=
e identified) operator, thus establishing a significant part of the chain o=
f accountability.  We can't improve on the security available through the P=
STN, but we can improve on the security available to the point of interconn=
ection with the identified operator (which may be different from the owner =
of the rtcweb application).  With DTLS-SRTP the user can have some basis fo=
r deciding whether to trust the identified operator; else (even with SDES-S=
RTP) you must completely trust the owner of the rtcweb application and ther=
e is no way to identify or verify any security attributes associated with t=
he media.

Finally, there are multiple reasons to avoid supporting multiple media secu=
rity schemes.  It is difficult enough to expect the end user to make sure t=
hey are comfortable with the identity of the remote party of a secure media=
 connection, but to ask if they are comfortable with an alternative media s=
ecurity scheme is simply too much to expect.  Bid-down attacks are just too=
 easy and it is impossible to verify any remote party identity with SDES-SR=
TP.

RTP

There are too many reasonable use cases for unsecured media for us to ignor=
e them.  The browser should be very clear when media security is unavailabl=
e and make it easy to prevent communication without media security (perhaps=
 as a default setting), but there are legitimate use cases in various priva=
te environments (e.g., enterprise, jail) where media security may not be al=
lowed (or must somehow be compromised), or is just not available, that we s=
hould just support straight RTP.  As long as the user can control the condi=
tions under which media security is disabled, I do not see a problem.

Richard Ejzak



_______________________________________________
rtcweb mailing list
rtcweb@ietf.org<mailto:rtcweb@ietf.org>
https://www.ietf.org/mailman/listinfo/rtcweb


--_000_387F9047F55E8C42850AD6B3A7A03C6C0E1FF09Binbamail01sonus_
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<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">&#43;1
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">I&#8217;ll add more useca=
se to strengthen RTP usage in WebRTC.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> rtcweb-b=
ounces@ietf.org [mailto:rtcweb-bounces@ietf.org]
<b>On Behalf Of </b>Roman Shpount<br>
<b>Sent:</b> Monday, March 19, 2012 9:51 AM<br>
<b>To:</b> Ejzak, Richard P (Richard)<br>
<b>Cc:</b> rtcweb@ietf.org<br>
<b>Subject:</b> Re: [rtcweb] Resolving RTP/SDES question in Paris<o:p></o:p=
></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">&#43;1<br clear=3D"al=
l">
_____________<br>
Roman Shpount<br>
<br>
<o:p></o:p></p>
<div>
<p class=3D"MsoNormal">On Mon, Mar 19, 2012 at 12:10 AM, Ejzak, Richard P (=
Richard) &lt;<a href=3D"mailto:richard.ejzak@alcatel-lucent.com">richard.ej=
zak@alcatel-lucent.com</a>&gt; wrote:<o:p></o:p></p>
<div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quo=
t;sans-serif&quot;;color:navy">I would like to argue two points: 1) DTLS-SR=
TP with IdP is the only media security scheme that rtcweb
 should support; and 2) rtcweb should also support RTP (no media security) =
with a clear requirement on the browser UI to warn the user of the complete=
 lack of media security.</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quo=
t;sans-serif&quot;;color:navy">&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quo=
t;sans-serif&quot;;color:navy">DTLS-SRTP</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quo=
t;sans-serif&quot;;color:navy">&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quo=
t;sans-serif&quot;;color:navy">Some of the arguments made against DTLS-SRTP=
 in recent emails are valid unless it is integrated with trusted
 IdP that is sufficient to verify the identity of the remote party. &nbsp;W=
ith this one caveat, DTLS-SRTP is equal to or (mostly) better than SDES-SRT=
P in almost all respects (not to be repeated here). &nbsp;Even if the ident=
ity is suspect, the IdP can at least provide
 some degree of assurance and accountability when an issue arises. &nbsp;Wi=
thout that, a user has no recourse when a problem develops. &nbsp;</span><o=
:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quo=
t;sans-serif&quot;;color:navy">&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quo=
t;sans-serif&quot;;color:navy">The key is that the browser UI must make cle=
ar to the user the identity of the remote party with whom
 media security has been established (even when it is a network gateway, in=
 which case the identify of the gateway operator). &nbsp;It should also be =
possible for the end user to mandate that a remote party identity is always=
 verified before communication is allowed
 (perhaps even make this a requirement when media security is selected). &n=
bsp;This is not too burdensome a requirement if established up front to hel=
p manage SPIT. &nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quo=
t;sans-serif&quot;;color:navy">&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quo=
t;sans-serif&quot;;color:navy">When end-to-end security is possible, then A=
lice knows that she&#8217;s talking to Bob and vice versa. &nbsp;</span><o:=
p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quo=
t;sans-serif&quot;;color:navy">&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quo=
t;sans-serif&quot;;color:navy">Even when Alice is talking to someone reache=
d via the PSTN, she can know that she has a secure media connection
 to a gateway owned by a particular (the identified) operator, thus establi=
shing a significant part of the chain of accountability. &nbsp;We can&#8217=
;t improve on the security available through the PSTN, but we can improve o=
n the security available to the point of interconnection
 with the identified operator (which may be different from the owner of the=
 rtcweb application).&nbsp; With DTLS-SRTP the user can have some basis for=
 deciding whether to trust the identified operator; else (even with SDES-SR=
TP) you must completely trust the owner
 of the rtcweb application and there is no way to identify or verify any se=
curity attributes associated with the media.&nbsp;
</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quo=
t;sans-serif&quot;;color:navy">&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quo=
t;sans-serif&quot;;color:navy">Finally, there are multiple reasons to avoid=
 supporting multiple media security schemes.&nbsp; It is difficult
 enough to expect the end user to make sure they are comfortable with the i=
dentity of the remote party of a secure media connection, but to ask if the=
y are comfortable with an alternative media security scheme is simply too m=
uch to expect. &nbsp;Bid-down attacks
 are just too easy and it is impossible to verify any remote party identity=
 with SDES-SRTP.</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quo=
t;sans-serif&quot;;color:navy">&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quo=
t;sans-serif&quot;;color:navy">RTP</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quo=
t;sans-serif&quot;;color:navy">&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quo=
t;sans-serif&quot;;color:navy">There are too many reasonable use cases for =
unsecured media for us to ignore them. &nbsp;The browser should
 be very clear when media security is unavailable and make it easy to preve=
nt communication without media security (perhaps as a default setting), but=
 there are legitimate use cases in various private environments (e.g., ente=
rprise, jail) where media security
 may not be allowed (or must somehow be compromised), or is just not availa=
ble, that we should just support straight RTP. &nbsp;As long as the user ca=
n control the conditions under which media security is disabled, I do not s=
ee a problem.</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quo=
t;sans-serif&quot;;color:navy">&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quo=
t;sans-serif&quot;;color:navy">Richard Ejzak</span><span style=3D"color:#88=
8888"><o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quo=
t;sans-serif&quot;;color:navy">&nbsp;</span><span style=3D"color:#888888"><=
o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quo=
t;sans-serif&quot;;color:navy">&nbsp;</span><span style=3D"color:#888888"><=
o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><o:p></o:p></p>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
</body>
</html>

--_000_387F9047F55E8C42850AD6B3A7A03C6C0E1FF09Binbamail01sonus_--

From oscar.ohlsson@ericsson.com  Mon Mar 19 00:52:43 2012
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From: Oscar Ohlsson <oscar.ohlsson@ericsson.com>
To: "igor.faynberg@alcatel-lucent.com" <igor.faynberg@alcatel-lucent.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Mon, 19 Mar 2012 08:52:38 +0100
Thread-Topic: [rtcweb] SDES vs DTLS-SRTP revisited
Thread-Index: Ac0DmTIq4s4gJHtgRKWyBUthos4tLQCCVoiQ
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Subject: Re: [rtcweb] SDES vs DTLS-SRTP revisited
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=20

> -----Original Message-----
> From: rtcweb-bounces@ietf.org=20
> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Igor Faynberg
> Sent: den 16 mars 2012 18:21
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] SDES vs DTLS-SRTP revisited
>=20
> Oscar,
>=20
> I am afraid I missed the point here.
>=20
> How can one determine that the call is being listened to?  =20
> How can the=20
> browser know whom the RTP stream is passing through?
>=20
> If you know the SRTP key and filter the flow directed at me,=20
> you will be on the conversation, right?
>=20
> Igor

Hi Igor,

No you can't know for sure that the call is being listened to, but the beha=
viour is very suscpicious. If both parties support DTLS-SRTP then why did t=
he web application switch from default DTLS-SRTP to SDES? There are not tha=
t many reasonable answers:

- interception/recording
- transcoding
- something else?

That's why I was wondering whether DTLS-SRTP fingerprint mismatch really is=
 a stronger indication of interception. To me it's not an obvious question.

Regards,

Oscar



>=20
> On 3/16/2012 11:04 AM, Oscar Ohlsson wrote:
> > ... Say that you're on a call and want to determine if the=20
> service provider is listening in:
> >
> > Case 1: Both endpoints support DTLS-SRTP (can be determined=20
> by asking=20
> > the other party)
> >
> > If SDES is used you directly conclude that something fishy=20
> is going on and hang up. If DTLS-SRTP is used you compare=20
> fingerprints and hang up if they don't match. (I've assumed=20
> that a user can view detailed security information in the browser).
> >
> > Case 2: The other endpoint does not support DTLS-SRTP
> ...
> >
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> =

From oscar.ohlsson@ericsson.com  Mon Mar 19 01:20:56 2012
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From: Oscar Ohlsson <oscar.ohlsson@ericsson.com>
To: Eric Rescorla <ekr@rtfm.com>
Date: Mon, 19 Mar 2012 09:20:53 +0100
Thread-Topic: [rtcweb] SDES vs DTLS-SRTP revisited
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Subject: Re: [rtcweb] SDES vs DTLS-SRTP revisited
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=20

> -----Original Message-----
> From: Eric Rescorla [mailto:ekr@rtfm.com]=20
> Sent: den 16 mars 2012 23:32
> To: Oscar Ohlsson
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] SDES vs DTLS-SRTP revisited
>=20
> On Fri, Mar 16, 2012 at 8:04 AM, Oscar Ohlsson=20
> <oscar.ohlsson@ericsson.com> wrote:
> > Hi,
> >
> > There were a couple of things that I didn't explain=20
> properly in my SDES vs DTLS-SRTP presentation at the interim=20
> meeting. Since time will be short at the next meeting as=20
> well, I thought I would try to explain myself in an email.
> >
> > I noticed that the discussion quickly turned into a direct=20
> comparison of the two protocols. When this happens DTLS-SRTP=20
> definitely comes out as the winner and trying to defend SDES=20
> just seems foolish. But this is not the right focus of the=20
> discussion IMO. We should be looking at the system as a whole=20
> and determine more exactly the scenarios where SDES poses a=20
> security risk.
> >
> > The first major argument against SDES is that it doesn't=20
> offer forward secrecy. Obviously, if the remote endpoint does=20
> not support DTLS-SRTP you have to switch to SDES or plain RTP=20
> at some point. And from that point and onwards you will never=20
> have forward secrecy. The only difference is that the=20
> conversion point is moved slightly to the left (one step?)=20
> when SDES is supported. However, if web developers starts=20
> using SDES without reason (i.e. when the remote endpoint is a=20
> browser) then I agree that the lack of forward secrecy is a=20
> problem. Do people on this list believe this will be the=20
> case? Can it be prevented somehow?
> >
> > The second major argument is that SDES makes it possible=20
> for the service provider to intercept calls undetectable.=20
> I've been thinking about this for some time but still haven't=20
> understood the problem completely. Say that you're on a call=20
> and want to determine if the service provider is listening in:
> >
> > Case 1: Both endpoints support DTLS-SRTP (can be determined=20
> by asking=20
> > the other party)
> >
> > If SDES is used you directly conclude that something fishy=20
> is going on and hang up. If DTLS-SRTP is used you compare=20
> fingerprints and hang up if they don't match. (I've assumed=20
> that a user can view detailed security information in the browser).
> >
> > Case 2: The other endpoint does not support DTLS-SRTP
> >
> > Regardless of whether SDES is allowed or not you will never=20
> be able to tell if the call is being intercepted. The best=20
> thing to do in this case is probably to simply hang up.
> >
> > Does the fingerprint mismatch in case 1 somehow serve as a=20
> stronger indication that the call is being intercepted? What=20
> action would a suspicious user take besides hanging up? I=20
> would be happy if someone could explain this to me.
>=20
> Hi Oscar,
>=20
> Yes, I think the fingerprint mismatch in case 1 serves as a=20
> stronger indication that something was wrong, because there=20
> is no technical reason for the fingerprint not to match if=20
> both sides believe they are doing DTLS. I.e., it always=20
> indicates either a software defect or a man in the middle.
> And the suspicious user of course should hang up, but they=20
> should also post pictures to /., which, I would think, serves=20
> as a pretty significant deterrent to providers tampering with=20
> people's calls!
> By contrast, with SDES, the attack scenario and the=20
> legitimate scenario look the same.

Hi Eric,

What is the legitimate scenario you're speaking of above? The only legitima=
te scenario I can come up with is transcoding and that is applicable to DTL=
S-SRTP as well.

I'm having hard time seeing why DTLS-SRTP shouldn't be used when both parti=
es support it. That's why I consider SDES downgrading and fingerprint repla=
cement equally suspicious.

Regards,

Oscar





>=20
> We actually have a worked example of this with Diginotar,=20
> where a mechanical system for detecting MITM attacks lead to=20
> a report to Google of such an attack and, IIRC, Diginotar=20
> being removed from the root list.
>=20
> Best,
> -Ekr
> =

From randell-ietf@jesup.org  Mon Mar 19 01:43:26 2012
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Subject: Re: [rtcweb] SDES vs DTLS-SRTP revisited
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On 3/17/2012 1:29 PM, Hadriel Kaplan wrote:
> On Mar 17, 2012, at 2:34 AM, Randell Jesup wrote:
>
>> Of course - gateways can tap off anything.  No shock there.
> Exactly. (and yes I know you knew that already :)
> So my point is that DTLS-SRTP doesn't give you much more than SDES over HTTPS, because any number of DTLS-b2bua gateways can be along the path.  Obviously it's *easier* for a malicious or rogue server to read your media when using SDES, because it doesn't have to do be a full media-b2bua to do so, but it's still possible either way.  Until and unless you verify the fingerprint.

Right, a gateway is nothing more than a MITM from a security perspective.

>>> The only thing that makes DTLS-SRTP any "better", is that you can go and verify the fingerprint with the far-end human (or at least you hope it's the far-end human).  If *and only if* you perform that step, is anything knowingly "better".  Of course as you've pointed out earlier, no one will perform that step, except for folks who probably wouldn't trust this whole thing to begin with.  So unless the identity provider thing happens and is mandated - and I don't hear anyone suggesting that be mandatory to use in order for a call to succeed - then we're arguing about the thickness of the Emperor's clothes, when he's not wearing any.  I'd be happy if we could at least get him a T-shirt.  :)
>> Well imagine if the user's identity provider was Google, and it keys off your normal google login.  When when you call anyone, you have a verified identity.
> Not really, as far as I can tell.  You have an identity Google claims is valid.  If you trust Google not to monitor your conversation, and to have good identity validation, then sure you can trust the identity is what Google claims it to be.  And that would NOT be the case for "when you call anyone" - only when you call other Google users and they are reached by their WebRTC connection to Google.

Actually, identity ala BrowserID would work even if neither of you were 
using Google to make the call - google would only be involved indirectly 
as the identity provider.  Google can (I believe) lie and say you're the 
owner of foo@gmail.com when you're not - that's the case with any 
PKI-ish system that doesn't have users publishing their own keys (and 
they of course you have to trust the manner in which you got the keys, 
which basically this is).

The advantage of a BrowserID-like system here is that the asserted 
identity level is closely tied to the source of the trust in the 
identity - you're trusting the person you're talking to to be 
foo@gmail.com, and it's gmail.com that you're trusting to verify that.  
The trust in the provider of the identity is closely tied to the 
identity itself, and exposed (in that a BrowserID-like system exposes 
that you're talking to foo@gmail.com, not "Eric Rescorla").

This is not absolute trust nor absolute identification; BrowserID is 
mean to be an alternative (one far less trackable and intrusive) to 
systems like "Facebook Connect"/"Login with Facebook".  However, within 
these limits it *does* provide a way to give users a way to detect MITMs 
and verify who they're talking to (owner of "foo@gmail.com"), and to do 
so without impossible-in-practice reading of fingerprints, etc.

>    If, for example, you called another Google user who's only on the PSTN at that time and Google offered call-forwarding, then it would go through their DTLS-gateway and we're back to square one.
>
> Furthermore, if you trusted Google that much already, it's not a huge leap to trust that your SDES over HTTPS to Google is not getting sent in the clear elsewhere by Google, and is not getting recorded by Google.

The missing part here is that you need to trust google that the other 
person is foo@gmail.com, but that this trust occurs even if the call 
never traverses google (and even the trust validation may not touch 
google).    If the call does go through google as the service provider 
(or any call where the service provider is also the identity host),  
then it is correct (and has been discussed) that you are trusting google 
not to MITM you.  (BrowserID schemes may be able to provide some 
additional level of assurance in this case, but not as much as the use 
of a third-party IdP (Identity Provider).)

>    It is no doubt easier for them to screw that up, but it's also not the scenario we have to worry about - because in the Google case for user-to-user WebRTC-to-WebRTC calls, we'd get DTLS-SRTP *anyway*.  No one's suggesting we don't mandate DTLS-SRTP also be implemented by browsers, and clearly we'd mandate that if the browser receives an offer with DTLS-SRTP, it must use it.  So we'd get DTLS-SRTP within a WebRTC domain.  That's not the problem scenario at hand.
>
> The problem scenario is what to do about the cases when the call goes to/from the PSTN, or to/from an interworking gateway to another protocol/domain that does not have support for end-to-end DTLS-SRTP.  We can either mandate those gateways/IWF implement DTLS-SRTP, or minimally just SDES.  The advantages of DTLS-SRTP over SDES in those scenarios are far less clear, while the advantages of SDES over DTLS in those scenarios are clear.

As mentioned elsewhere, for a PSTN gateway I'd use DTLS-SRTP and (as the 
terminating WebRTC context) identify the gateway as the remote party, 
which makes clear the call is secure up to that point only.

-- 
Randell Jesup
randell-ietf@jesup.org


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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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On 3/19/2012 12:10 AM, Ejzak, Richard P (Richard) wrote:
> I would like to argue two points: 1) DTLS-SRTP with IdP is the only
> media security scheme that rtcweb should support; and 2) rtcweb should
> also support RTP (no media security) with a clear requirement on the
> browser UI to warn the user of the complete lack of media security.
>
> DTLS-SRTP

[SNIP]

I generally agree.

> RTP
>
> There are too many reasonable use cases for unsecured media for us to
> ignore them. The browser should be very clear when media security is
> unavailable and make it easy to prevent communication without media
> security (perhaps as a default setting), but there are legitimate use
> cases in various private environments (e.g., enterprise, jail) where
> media security may not be allowed (or must somehow be compromised), or
> is just not available, that we should just support straight RTP. As long
> as the user can control the conditions under which media security is
> disabled, I do not see a problem.

The issue with "legacy" RTP is more one of controlling 'consent', such 
that we can avoid a WebRTC endpoint being used as a DDoS client.  I 
spent many hours trying to find a way around the issue, and only found 
ones that involve some level of modification to the target RTP client or 
its network (and still have serious issues).

For your case (not necessarily 'legacy', but needs-to-be-in-the-clear), 
perhaps there are other solutions.

One would be the command-line switch to force a NULL SRTP cipher.

An enterprise could use a WebRTC gateway, which in the context of WebRTC 
the user could provide their IdP credentials to (or the enterprise could 
be the IdP itself); or just act as a WebRTC MITM without bothering with 
IdP support.  Certainly there are usecases for keeping media traffic 
within an enterprise encrypted - I remember getting router-traffic dumps 
at an old company and being able to replay RTP conversations, including 
sensitive ones with external council, board members, HR, etc.  Modern 
switched setups make this tougher for J-random-employee to do, but far 
from impossible, and trivial for anyone working in networking IT or with 
physical access to switches or wiring.

-- 
Randell Jesup
randell-ietf@jesup.org

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Mon, 19 Mar 2012 12:38:01 +0100
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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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2012/3/19 Ejzak, Richard P (Richard) <richard.ejzak@alcatel-lucent.com>:
> RTP
>
> There are too many reasonable use cases for unsecured media for us to ign=
ore
> them.

Which ones? I just see one: interoperability with legacy SIP devices.
The question is: does it make sense to create a new specification
(WebRTC) less secure that it can be instead of forcing SIP vendors to
implement SRTP in their devices? If SIP vendors assumed that their
devices would just work in trusted networks (or wallen gardens) that
should not affect the security aspects of a new specification made for
*Internet*. That's their problem (SRTP was initially defined for SIP,
so they must do their homeworks).

Is plain RTP an advantage for WebRTC? Not at all since WebRTC
implementations will support SRTP by design so downgrading to plain
RTP is never better than using SRTP. Even if you use an VPN for your
enterprise WebRTC application there is NO problem at all in using SRTP
over the VPN.

So instead of anecdotal usecases in which plain RTP is "good enough" I
would prefer to read the truth: "I am a telco/vendor and I want WebRTC
(a specification originally designed for Internet) to support plain
RTP so I can make business with my legacy and not secure SIP devices,
those that don't implement SRTP regardless SRTP was designed for SIP".



>=C2=A0The browser should be very clear when media security is unavailable
> and make it easy to prevent communication without media security (perhaps=
 as
> a default setting), but there are legitimate use cases in various private
> environments (e.g., enterprise, jail) where media security may not be
> allowed (or must somehow be compromised), or is just not available, that =
we
> should just support straight RTP. =C2=A0As long as the user can control t=
he
> conditions under which media security is disabled, I do not see a problem=
.

I assume that finally plain RTP will be allowed in WebRTC, too much
interest in interoperability with legacy and non secure SIP devices.
However, adding more and more "security approvals" for end users in
their web browsers is much worse than mandating security by design.


Regards.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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> Which ones? I just see one: interoperability with legacy SIP devices.

[BA] Agreed.  And as we've already seen=2C SDES/SRTP support is prevalent e=
nough among legacy devices that the legacy use case does not require RTP.=20

> Even if you use an VPN for your enterprise WebRTC application there is NO=
 problem at all in using SRTP
> over the VPN.

[BA] And in fact=2C this is exactly the NSA "Secure VoIP" architecture (wit=
h SDES/SRTP=2C by the way).=20

> I can make business with my legacy and not secure SIP devices=2C
> those that don't implement SRTP regardless SRTP was designed for SIP".

[BA] At this point=2C support for SRTP is an expected feature on legacy equ=
ipment.  For
example=2C all the leading PSTN gateway vendors support SRTP already.  By t=
he time
RTCWEB specs are final=2C SRTP support will be very prevalent.=20









 		 	   		  =

--_5be95578-009e-496a-90af-aa19ae6908ef_
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<html>
<head>
<style><!--
.hmmessage P
{
margin:0px=3B
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{
font-size: 10pt=3B
font-family:Tahoma
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<body class=3D'hmmessage'><div dir=3D'ltr'>
&gt=3B Which ones? I just see one: interoperability with legacy SIP devices=
.<br><br>[BA] Agreed.&nbsp=3B And as we've already seen=2C SDES/SRTP suppor=
t is prevalent enough among legacy devices that the legacy use case does no=
t require RTP. <br><br>&gt=3B Even if you use an VPN for your enterprise We=
bRTC application there is NO problem at all in using SRTP<br><div>&gt=3B ov=
er the VPN.<br><br>[BA] And in fact=2C this is exactly the NSA "Secure VoIP=
" architecture (with SDES/SRTP=2C by the way). <br><br>&gt=3B I can make bu=
siness with my legacy and not secure SIP devices=2C<br>&gt=3B those that do=
n't implement SRTP regardless SRTP was designed for SIP".<br><br>[BA] At th=
is point=2C support for SRTP is an expected feature on legacy equipment.&nb=
sp=3B For<br>example=2C all the leading PSTN gateway vendors support SRTP a=
lready.&nbsp=3B By the time<br>RTCWEB specs are final=2C SRTP support will =
be very prevalent. <br><br><br><br><br><br><br><br><br><br></div> 		 	   		=
  </div></body>
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2012/3/19 Bernard Aboba <bernard_aboba@hotmail.com>:
> At this point, support for SRTP is an expected feature on legacy equipmen=
t.
> For example, all the leading PSTN gateway vendors support SRTP already.=
=C2=A0 By the time RTCWEB specs are final, SRTP support will be very preval=
ent.


And if they don't support SRTP then bad luck for them. WebRTC cannot
be less secure and worse just because some SIP legacy equipments don't
implement a specification from 2004 (RFC 3711 - SRTP) !!

Please, make WebRTC as secure as possible for common usages in the
open Internet, rather than decreasing the security just to get
interoperability with telcos non supporting SRTP.

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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2012/3/19 Ravindran, Parthasarathi <pravindran@sonusnet.com>:
> I=E2=80=99ll add more usecase to strengthen RTP usage in WebRTC.

Partha, instead of showing usecases in which SRTP is not fully
required (due to VPN existence or just localhost with no Internet
connection), please identify just a single usecase in which using SRTP
is *bad* or *worse* than using plain RTP. Just one.

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From fluffy@iii.ca  Mon Mar 19 07:00:57 2012
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I like draft-ietf-mmusic-sctp-sdp but I'd like to see it have more stuff =
in it. Specifically I would like it to=20

1) clearly define how to do the SDP for use of SCTP over DTLS over UPD =
with ICE. This is the use case we need to solve=20

2) make sure it works with bundle=20

3) SCTP has sub addresses (SCTP Channels), I think these need to be =
negotiated as well in some cases so I would like to see an optional way =
of negotiating and signaling the channels including correlation with =
some sort of application defined label=20

Cullen=20



From ekr@rtfm.com  Mon Mar 19 07:08:03 2012
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From: Eric Rescorla <ekr@rtfm.com>
Date: Mon, 19 Mar 2012 07:01:45 -0700
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I think I may have misunderstood you. I thought the SDES case (#2)
was one where
the other side actually didn't speak DTLS and that's what I was trying
to analyze.

I agree that if the other side supports DTLS and you get SDES, that's bad.

-Ekr


On Mon, Mar 19, 2012 at 1:20 AM, Oscar Ohlsson
<oscar.ohlsson@ericsson.com> wrote:
>
>
>> -----Original Message-----
>> From: Eric Rescorla [mailto:ekr@rtfm.com]
>> Sent: den 16 mars 2012 23:32
>> To: Oscar Ohlsson
>> Cc: rtcweb@ietf.org
>> Subject: Re: [rtcweb] SDES vs DTLS-SRTP revisited
>>
>> On Fri, Mar 16, 2012 at 8:04 AM, Oscar Ohlsson
>> <oscar.ohlsson@ericsson.com> wrote:
>> > Hi,
>> >
>> > There were a couple of things that I didn't explain
>> properly in my SDES vs DTLS-SRTP presentation at the interim
>> meeting. Since time will be short at the next meeting as
>> well, I thought I would try to explain myself in an email.
>> >
>> > I noticed that the discussion quickly turned into a direct
>> comparison of the two protocols. When this happens DTLS-SRTP
>> definitely comes out as the winner and trying to defend SDES
>> just seems foolish. But this is not the right focus of the
>> discussion IMO. We should be looking at the system as a whole
>> and determine more exactly the scenarios where SDES poses a
>> security risk.
>> >
>> > The first major argument against SDES is that it doesn't
>> offer forward secrecy. Obviously, if the remote endpoint does
>> not support DTLS-SRTP you have to switch to SDES or plain RTP
>> at some point. And from that point and onwards you will never
>> have forward secrecy. The only difference is that the
>> conversion point is moved slightly to the left (one step?)
>> when SDES is supported. However, if web developers starts
>> using SDES without reason (i.e. when the remote endpoint is a
>> browser) then I agree that the lack of forward secrecy is a
>> problem. Do people on this list believe this will be the
>> case? Can it be prevented somehow?
>> >
>> > The second major argument is that SDES makes it possible
>> for the service provider to intercept calls undetectable.
>> I've been thinking about this for some time but still haven't
>> understood the problem completely. Say that you're on a call
>> and want to determine if the service provider is listening in:
>> >
>> > Case 1: Both endpoints support DTLS-SRTP (can be determined
>> by asking
>> > the other party)
>> >
>> > If SDES is used you directly conclude that something fishy
>> is going on and hang up. If DTLS-SRTP is used you compare
>> fingerprints and hang up if they don't match. (I've assumed
>> that a user can view detailed security information in the browser).
>> >
>> > Case 2: The other endpoint does not support DTLS-SRTP
>> >
>> > Regardless of whether SDES is allowed or not you will never
>> be able to tell if the call is being intercepted. The best
>> thing to do in this case is probably to simply hang up.
>> >
>> > Does the fingerprint mismatch in case 1 somehow serve as a
>> stronger indication that the call is being intercepted? What
>> action would a suspicious user take besides hanging up? I
>> would be happy if someone could explain this to me.
>>
>> Hi Oscar,
>>
>> Yes, I think the fingerprint mismatch in case 1 serves as a
>> stronger indication that something was wrong, because there
>> is no technical reason for the fingerprint not to match if
>> both sides believe they are doing DTLS. I.e., it always
>> indicates either a software defect or a man in the middle.
>> And the suspicious user of course should hang up, but they
>> should also post pictures to /., which, I would think, serves
>> as a pretty significant deterrent to providers tampering with
>> people's calls!
>> By contrast, with SDES, the attack scenario and the
>> legitimate scenario look the same.
>
> Hi Eric,
>
> What is the legitimate scenario you're speaking of above? The only legitimate scenario I can come up with is transcoding and that is applicable to DTLS-SRTP as well.
>
> I'm having hard time seeing why DTLS-SRTP shouldn't be used when both parties support it. That's why I consider SDES downgrading and fingerprint replacement equally suspicious.
>
> Regards,
>
> Oscar
>
>
>
>
>
>>
>> We actually have a worked example of this with Diginotar,
>> where a mechanical system for detecting MITM attacks lead to
>> a report to Google of such an attack and, IIRC, Diginotar
>> being removed from the root list.
>>
>> Best,
>> -Ekr
>>

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On Sat, Mar 17, 2012 at 4:52 AM, Tim Panton <tim@phonefromhere.com> wrote:
>

> I think there probably is an vulnerability if Bob goes on line (to upgrade the call to video say) and happens to be on the
> same wifi ISP as Alice. The ICE probes may find a route between them that does not involve Alice's corporate VPN, but rather
> over the non-routable ISP's /16 network. This would result in RTP in the clear being visible on both parties local wifi.
>
> It's a slightly contrived case, but not impossible.
>
> Tim.
I don't think it's all that contrived, as I think we could see at an
IETF meeting pretty easily.  Alice and Bob go to their companies
intranet site to start their call, but ICE puts them both on the local
IETF lan, so RTP flows directly there.  Note that this could be the
case if they are simply screen sharing and typing in a chat--something
that they might be doing in order to have a back channel discussion
before a presentation.  I've certainly seen that happen at the IETF on
more than one occasion.

Ted

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On Sun, Mar 18, 2012 at 9:10 PM, Ejzak, Richard P (Richard)
<richard.ejzak@alcatel-lucent.com> wrote:
> I would like to argue two points: 1) DTLS-SRTP with IdP is the only media
> security scheme that rtcweb should support; and 2) rtcweb should also
> support RTP (no media security) with a clear requirement on the browser UI
> to warn the user of the complete lack of media security.
>

I'd like to comment on the "browser UI" issue here.  As we have
discussed in the past, the users of this technology may very well
include applications which are not browsers.  In the "pokerstars"
example, a mobile device may very well be using a purpose-built
application instead of the mobile browser in order to reach the site.
These mobile apps may interact with web-browser based clients during
the same hand of poker.   If the web-browser based client wants to use
an insecure media channel and can get consent, there is still no
guarantee that other clients can do the same using the same UI
feature.

Web technology != browser.  Let's be careful not to build the security
story at the protocol level on UI features we can neither predict nor
control.  We can choose not to have certain levels of security, but
hand-waving that decision away because we believe it can be managed at
some other layer is a very risky proposition.

regards,

Ted

From ibc@aliax.net  Mon Mar 19 07:57:24 2012
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Date: Mon, 19 Mar 2012 15:57:22 +0100
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
To: Ted Hardie <ted.ietf@gmail.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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+2 (I don't like "+1" since Google has decided to own it)
El 19/03/2012 15:50, "Ted Hardie" <ted.ietf@gmail.com> escribi=C3=B3:

> On Sun, Mar 18, 2012 at 9:10 PM, Ejzak, Richard P (Richard)
> <richard.ejzak@alcatel-lucent.com> wrote:
> > I would like to argue two points: 1) DTLS-SRTP with IdP is the only med=
ia
> > security scheme that rtcweb should support; and 2) rtcweb should also
> > support RTP (no media security) with a clear requirement on the browser
> UI
> > to warn the user of the complete lack of media security.
> >
>
> I'd like to comment on the "browser UI" issue here.  As we have
> discussed in the past, the users of this technology may very well
> include applications which are not browsers.  In the "pokerstars"
> example, a mobile device may very well be using a purpose-built
> application instead of the mobile browser in order to reach the site.
> These mobile apps may interact with web-browser based clients during
> the same hand of poker.   If the web-browser based client wants to use
> an insecure media channel and can get consent, there is still no
> guarantee that other clients can do the same using the same UI
> feature.
>
> Web technology !=3D browser.  Let's be careful not to build the security
> story at the protocol level on UI features we can neither predict nor
> control.  We can choose not to have certain levels of security, but
> hand-waving that decision away because we believe it can be managed at
> some other layer is a very risky proposition.
>
> regards,
>
> Ted
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<p>+2 (I don&#39;t like &quot;+1&quot; since Google has decided to own it)<=
/p>
<div class=3D"gmail_quote">El 19/03/2012 15:50, &quot;Ted Hardie&quot; &lt;=
<a href=3D"mailto:ted.ietf@gmail.com">ted.ietf@gmail.com</a>&gt; escribi=C3=
=B3:<br type=3D"attribution"><blockquote class=3D"gmail_quote" style=3D"mar=
gin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
On Sun, Mar 18, 2012 at 9:10 PM, Ejzak, Richard P (Richard)<br>
&lt;<a href=3D"mailto:richard.ejzak@alcatel-lucent.com">richard.ejzak@alcat=
el-lucent.com</a>&gt; wrote:<br>
&gt; I would like to argue two points: 1) DTLS-SRTP with IdP is the only me=
dia<br>
&gt; security scheme that rtcweb should support; and 2) rtcweb should also<=
br>
&gt; support RTP (no media security) with a clear requirement on the browse=
r UI<br>
&gt; to warn the user of the complete lack of media security.<br>
&gt;<br>
<br>
I&#39;d like to comment on the &quot;browser UI&quot; issue here. =C2=A0As =
we have<br>
discussed in the past, the users of this technology may very well<br>
include applications which are not browsers. =C2=A0In the &quot;pokerstars&=
quot;<br>
example, a mobile device may very well be using a purpose-built<br>
application instead of the mobile browser in order to reach the site.<br>
These mobile apps may interact with web-browser based clients during<br>
the same hand of poker. =C2=A0 If the web-browser based client wants to use=
<br>
an insecure media channel and can get consent, there is still no<br>
guarantee that other clients can do the same using the same UI<br>
feature.<br>
<br>
Web technology !=3D browser. =C2=A0Let&#39;s be careful not to build the se=
curity<br>
story at the protocol level on UI features we can neither predict nor<br>
control. =C2=A0We can choose not to have certain levels of security, but<br=
>
hand-waving that decision away because we believe it can be managed at<br>
some other layer is a very risky proposition.<br>
<br>
regards,<br>
<br>
Ted<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div>

--f46d0434bf42934a3004bb99c6d9--

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Date: Mon, 19 Mar 2012 08:19:09 -0700
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From: Cameron Byrne <cb.list6@gmail.com>
To: =?ISO-8859-1?Q?I=F1aki_Baz_Castillo?= <ibc@aliax.net>
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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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On Mar 19, 2012 5:25 AM, "I=F1aki Baz Castillo" <ibc@aliax.net> wrote:
>
> 2012/3/19 Bernard Aboba <bernard_aboba@hotmail.com>:
> > At this point, support for SRTP is an expected feature on legacy
equipment.
> > For example, all the leading PSTN gateway vendors support SRTP
already.  By the time RTCWEB specs are final, SRTP support will be very
prevalent.
>
>
> And if they don't support SRTP then bad luck for them. WebRTC cannot
> be less secure and worse just because some SIP legacy equipments don't
> implement a specification from 2004 (RFC 3711 - SRTP) !!
>
> Please, make WebRTC as secure as possible for common usages in the
> open Internet, rather than decreasing the security just to get
> interoperability with telcos non supporting SRTP.
>

+1, srtp is required

And, it blows my mind this discussion is still going on.

If srtp is not mandatory, it creates a great deal more work for me (a telco
who has customers that expect privacy)

Cb
> --
> I=F1aki Baz Castillo
> <ibc@aliax.net>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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<p><br>
On Mar 19, 2012 5:25 AM, &quot;I=F1aki Baz Castillo&quot; &lt;<a href=3D"ma=
ilto:ibc@aliax.net">ibc@aliax.net</a>&gt; wrote:<br>
&gt;<br>
&gt; 2012/3/19 Bernard Aboba &lt;<a href=3D"mailto:bernard_aboba@hotmail.co=
m">bernard_aboba@hotmail.com</a>&gt;:<br>
&gt; &gt; At this point, support for SRTP is an expected feature on legacy =
equipment.<br>
&gt; &gt; For example, all the leading PSTN gateway vendors support SRTP al=
ready.=A0 By the time RTCWEB specs are final, SRTP support will be very pre=
valent.<br>
&gt;<br>
&gt;<br>
&gt; And if they don&#39;t support SRTP then bad luck for them. WebRTC cann=
ot<br>
&gt; be less secure and worse just because some SIP legacy equipments don&#=
39;t<br>
&gt; implement a specification from 2004 (RFC 3711 - SRTP) !!<br>
&gt;<br>
&gt; Please, make WebRTC as secure as possible for common usages in the<br>
&gt; open Internet, rather than decreasing the security just to get<br>
&gt; interoperability with telcos non supporting SRTP.<br>
&gt;</p>
<p>+1, srtp is required</p>
<p>And, it blows my mind this discussion is still going on. </p>
<p>If srtp is not mandatory, it creates a great deal more work for me (a te=
lco who has customers that expect privacy)</p>
<p>Cb<br>
&gt; --<br>
&gt; I=F1aki Baz Castillo<br>
&gt; &lt;<a href=3D"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt;<br>
&gt; _______________________________________________<br>
&gt; rtcweb mailing list<br>
&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.i=
etf.org/mailman/listinfo/rtcweb</a><br>
</p>

--e89a8ff255ae7f9c7a04bb9a1443--

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From: "Ejzak, Richard P (Richard)" <richard.ejzak@alcatel-lucent.com>
To: Ted Hardie <ted.ietf@gmail.com>
Date: Mon, 19 Mar 2012 11:00:59 -0500
Thread-Topic: [rtcweb] Resolving RTP/SDES question in Paris
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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Ted,
I simply do not buy your argument.

It doesn't matter what security scheme is used if you download a dedicated =
communications client that you don't trust.  It can play whatever games it =
wants with the UI and the security configuration and you are none the bette=
r off.  You have to trust the application.

It is only within the context of a trusted browser (and its UI) that any of=
 this security work makes sense.  As long as you can trust the browser to t=
ell you what's going on, you don't have to trust the application.

You might also have a trusted dedicated application but it better follow th=
e same guidelines if it expects to remain trusted.

Richard

-----Original Message-----
From: Ted Hardie [mailto:ted.ietf@gmail.com]=20
Sent: Monday, March 19, 2012 9:51 AM
To: Ejzak, Richard P (Richard)
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris

On Sun, Mar 18, 2012 at 9:10 PM, Ejzak, Richard P (Richard)
<richard.ejzak@alcatel-lucent.com> wrote:
> I would like to argue two points: 1) DTLS-SRTP with IdP is the only media
> security scheme that rtcweb should support; and 2) rtcweb should also
> support RTP (no media security) with a clear requirement on the browser U=
I
> to warn the user of the complete lack of media security.
>

I'd like to comment on the "browser UI" issue here.  As we have
discussed in the past, the users of this technology may very well
include applications which are not browsers.  In the "pokerstars"
example, a mobile device may very well be using a purpose-built
application instead of the mobile browser in order to reach the site.
These mobile apps may interact with web-browser based clients during
the same hand of poker.   If the web-browser based client wants to use
an insecure media channel and can get consent, there is still no
guarantee that other clients can do the same using the same UI
feature.

Web technology !=3D browser.  Let's be careful not to build the security
story at the protocol level on UI features we can neither predict nor
control.  We can choose not to have certain levels of security, but
hand-waving that decision away because we believe it can be managed at
some other layer is a very risky proposition.

regards,

Ted

From richard.ejzak@alcatel-lucent.com  Mon Mar 19 09:03:14 2012
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From: "Ejzak, Richard P (Richard)" <richard.ejzak@alcatel-lucent.com>
To: Cameron Byrne <cb.list6@gmail.com>, =?iso-8859-1?Q?I=F1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Mon, 19 Mar 2012 11:03:09 -0500
Thread-Topic: [rtcweb] Resolving RTP/SDES question in Paris
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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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SRTP by itself guarantees nothing.  What is the point of insisting that the=
 browser encrypt media if you know nothing about the other endpoint of the =
encrypted media or even whether anyone else has keys?

Richard

________________________________
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Cameron Byrne
Sent: Monday, March 19, 2012 10:19 AM
To: I=F1aki Baz Castillo
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris


On Mar 19, 2012 5:25 AM, "I=F1aki Baz Castillo" <ibc@aliax.net<mailto:ibc@a=
liax.net>> wrote:
>
> 2012/3/19 Bernard Aboba <bernard_aboba@hotmail.com<mailto:bernard_aboba@h=
otmail.com>>:
> > At this point, support for SRTP is an expected feature on legacy equipm=
ent.
> > For example, all the leading PSTN gateway vendors support SRTP already.=
  By the time RTCWEB specs are final, SRTP support will be very prevalent.
>
>
> And if they don't support SRTP then bad luck for them. WebRTC cannot
> be less secure and worse just because some SIP legacy equipments don't
> implement a specification from 2004 (RFC 3711 - SRTP) !!
>
> Please, make WebRTC as secure as possible for common usages in the
> open Internet, rather than decreasing the security just to get
> interoperability with telcos non supporting SRTP.
>

+1, srtp is required

And, it blows my mind this discussion is still going on.

If srtp is not mandatory, it creates a great deal more work for me (a telco=
 who has customers that expect privacy)

Cb
> --
> I=F1aki Baz Castillo
> <ibc@aliax.net<mailto:ibc@aliax.net>>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org<mailto:rtcweb@ietf.org>
> https://www.ietf.org/mailman/listinfo/rtcweb

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<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>SRTP by itself guarantees nothing. =A0=
What
is the point of insisting that the browser encrypt media if you know nothin=
g
about the other endpoint of the encrypted media or even whether anyone else=
 has
keys?<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Richard<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font siz=
e=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span style=3D'font-si=
ze:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=3D2
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'>
rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] <b><span
style=3D'font-weight:bold'>On Behalf Of </span></b>Cameron Byrne<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Monday, March 19, 2012=
 10:19
AM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> I=F1aki Baz Castillo<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> rtcweb@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [rtcweb] Resolv=
ing
RTP/SDES question in <st1:City w:st=3D"on"><st1:place w:st=3D"on">Paris</st=
1:place></st1:City></span></font><o:p></o:p></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p><font size=3D3 face=3D"Times New Roman"><span style=3D'font-size:12.0pt'=
><br>
On Mar 19, 2012 5:25 AM, &quot;I=F1aki Baz Castillo&quot; &lt;<a
href=3D"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt; wrote:<br>
&gt;<br>
&gt; 2012/3/19 Bernard Aboba &lt;<a href=3D"mailto:bernard_aboba@hotmail.co=
m">bernard_aboba@hotmail.com</a>&gt;:<br>
&gt; &gt; At this point, support for SRTP is an expected feature on legacy
equipment.<br>
&gt; &gt; For example, all the leading PSTN gateway vendors support SRTP
already.&nbsp; By the time RTCWEB specs are final, SRTP support will be ver=
y
prevalent.<br>
&gt;<br>
&gt;<br>
&gt; And if they don't support SRTP then bad luck for them. WebRTC cannot<b=
r>
&gt; be less secure and worse just because some SIP legacy equipments don't=
<br>
&gt; implement a specification from 2004 (RFC 3711 - SRTP) !!<br>
&gt;<br>
&gt; Please, make WebRTC as secure as possible for common usages in the<br>
&gt; open Internet, rather than decreasing the security just to get<br>
&gt; interoperability with telcos non supporting SRTP.<br>
&gt;<o:p></o:p></span></font></p>

<p><font size=3D3 face=3D"Times New Roman"><span style=3D'font-size:12.0pt'=
>+1, srtp
is required<o:p></o:p></span></font></p>

<p><font size=3D3 face=3D"Times New Roman"><span style=3D'font-size:12.0pt'=
>And, it
blows my mind this discussion is still going on. <o:p></o:p></span></font><=
/p>

<p><font size=3D3 face=3D"Times New Roman"><span style=3D'font-size:12.0pt'=
>If srtp
is not mandatory, it creates a great deal more work for me (a telco who has
customers that expect privacy)<o:p></o:p></span></font></p>

<p><font size=3D3 face=3D"Times New Roman"><span style=3D'font-size:12.0pt'=
>Cb<br>
&gt; --<br>
&gt; I=F1aki Baz Castillo<br>
&gt; &lt;<a href=3D"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt;<br>
&gt; _______________________________________________<br>
&gt; rtcweb mailing list<br>
&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.i=
etf.org/mailman/listinfo/rtcweb</a><o:p></o:p></span></font></p>

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Mon, 19 Mar 2012 17:10:48 +0100
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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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2012/3/19 Ejzak, Richard P (Richard) <richard.ejzak@alcatel-lucent.com>:
> SRTP by itself guarantees nothing. =C2=A0What is the point of insisting t=
hat the
> browser encrypt media if you know nothing about the other endpoint of the
> encrypted media or even whether anyone else has keys?

If I am at the airport using an open WiFi connection, I visit a web
page using HTTPS and my browser validates the server TLS certificate,
neither I can be sure that the server has not been hacked by
attackers. But at least I know that nobody in the airport can monitor
my HTTPS traffic.

Indeed SRTP by itself guarantees nothing, but if the signaling path is
secured (HTTPS or WebSocket over TLS, so SRTP-SDES becomes a secure
solution) nobody in my network can intercept my media communication.
IMHO that's much better than nothing (plain RTP).

Regards.

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From igor.faynberg@alcatel-lucent.com  Mon Mar 19 09:21:38 2012
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Date: Mon, 19 Mar 2012 12:21:46 -0400
From: Igor Faynberg <igor.faynberg@alcatel-lucent.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] SDES vs DTLS-SRTP revisited
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In complete agreement.

Igor

On 3/19/2012 3:52 AM, Oscar Ohlsson wrote:
>
>
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org
>> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Igor Faynberg
>> Sent: den 16 mars 2012 18:21
>> To: rtcweb@ietf.org
>> Subject: Re: [rtcweb] SDES vs DTLS-SRTP revisited
>>
>> Oscar,
>>
>> I am afraid I missed the point here.
>>
>> How can one determine that the call is being listened to?
>> How can the
>> browser know whom the RTP stream is passing through?
>>
>> If you know the SRTP key and filter the flow directed at me,
>> you will be on the conversation, right?
>>
>> Igor
> Hi Igor,
>
> No you can't know for sure that the call is being listened to, but the behaviour is very suscpicious. If both parties support DTLS-SRTP then why did the web application switch from default DTLS-SRTP to SDES? There are not that many reasonable answers:
>
> - interception/recording
> - transcoding
> - something else?
>
> That's why I was wondering whether DTLS-SRTP fingerprint mismatch really is a stronger indication of interception. To me it's not an obvious question.
>
> Regards,
>
> Oscar
>
>
>
>> On 3/16/2012 11:04 AM, Oscar Ohlsson wrote:
>>> ... Say that you're on a call and want to determine if the
>> service provider is listening in:
>>> Case 1: Both endpoints support DTLS-SRTP (can be determined
>> by asking
>>> the other party)
>>>
>>> If SDES is used you directly conclude that something fishy
>> is going on and hang up. If DTLS-SRTP is used you compare
>> fingerprints and hang up if they don't match. (I've assumed
>> that a user can view detailed security information in the browser).
>>> Case 2: The other endpoint does not support DTLS-SRTP
>> ...
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb

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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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On Mar 19, 2012 9:11 AM, "I=F1aki Baz Castillo" <ibc@aliax.net> wrote:
>
> 2012/3/19 Ejzak, Richard P (Richard) <richard.ejzak@alcatel-lucent.com>:
> > SRTP by itself guarantees nothing.  What is the point of insisting that
the
> > browser encrypt media if you know nothing about the other endpoint of
the
> > encrypted media or even whether anyone else has keys?
>
> If I am at the airport using an open WiFi connection, I visit a web
> page using HTTPS and my browser validates the server TLS certificate,
> neither I can be sure that the server has not been hacked by
> attackers. But at least I know that nobody in the airport can monitor
> my HTTPS traffic.
>
> Indeed SRTP by itself guarantees nothing, but if the signaling path is
> secured (HTTPS or WebSocket over TLS, so SRTP-SDES becomes a secure
> solution) nobody in my network can intercept my media communication.
> IMHO that's much better than nothing (plain RTP).
>

No security tool ever guarantees anything.

They just increase the cost/time of hacking interception.

Cb
> Regards.
>
> --
> I=F1aki Baz Castillo
> <ibc@aliax.net>

--047d7b10d0f57cd8ea04bb9b06c9
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<p><br>
On Mar 19, 2012 9:11 AM, &quot;I=F1aki Baz Castillo&quot; &lt;<a href=3D"ma=
ilto:ibc@aliax.net">ibc@aliax.net</a>&gt; wrote:<br>
&gt;<br>
&gt; 2012/3/19 Ejzak, Richard P (Richard) &lt;<a href=3D"mailto:richard.ejz=
ak@alcatel-lucent.com">richard.ejzak@alcatel-lucent.com</a>&gt;:<br>
&gt; &gt; SRTP by itself guarantees nothing. =A0What is the point of insist=
ing that the<br>
&gt; &gt; browser encrypt media if you know nothing about the other endpoin=
t of the<br>
&gt; &gt; encrypted media or even whether anyone else has keys?<br>
&gt;<br>
&gt; If I am at the airport using an open WiFi connection, I visit a web<br=
>
&gt; page using HTTPS and my browser validates the server TLS certificate,<=
br>
&gt; neither I can be sure that the server has not been hacked by<br>
&gt; attackers. But at least I know that nobody in the airport can monitor<=
br>
&gt; my HTTPS traffic.<br>
&gt;<br>
&gt; Indeed SRTP by itself guarantees nothing, but if the signaling path is=
<br>
&gt; secured (HTTPS or WebSocket over TLS, so SRTP-SDES becomes a secure<br=
>
&gt; solution) nobody in my network can intercept my media communication.<b=
r>
&gt; IMHO that&#39;s much better than nothing (plain RTP).<br>
&gt;</p>
<p>No security tool ever guarantees anything. </p>
<p>They just increase the cost/time of hacking interception. <br></p>
<p>Cb<br>
&gt; Regards.<br>
&gt;<br>
&gt; --<br>
&gt; I=F1aki Baz Castillo<br>
&gt; &lt;<a href=3D"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt;<br>
</p>

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To: "Ejzak, Richard P (Richard)" <richard.ejzak@alcatel-lucent.com>
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On 19 Mar 2012, at 16:03, Ejzak, Richard P (Richard) wrote:

> SRTP by itself guarantees nothing.  What is the point of insisting =
that the browser encrypt media if you know nothing about the other =
endpoint of the encrypted media or even whether anyone else has keys?

If you are in an airport lounge, it should mean the script kiddie at the =
next gate over has a harder time reading your screenshare.
Admittedly you are only incrementally securing the first hop, but often =
- to coin a phrase - "the first hop is the weakest" ;-)

Tim.=

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<html><head></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; "><br><div><div>On 19 Mar 2012, at 16:03, Ejzak, Richard P (Richard) wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite">


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<div class="Section1"><p class="MsoNormal"><font size="2" color="navy" face="Arial"><span style="font-size:
10.0pt;font-family:Arial;color:navy">SRTP by itself guarantees nothing. &nbsp;What
is the point of insisting that the browser encrypt media if you know nothing
about the other endpoint of the encrypted media or even whether anyone else has
keys?</span></font></p></div></div></o:smarttagtype></o:smarttagtype></blockquote><div><br></div><div>If you are in an airport lounge, it should mean the script kiddie at the next gate over has a harder time reading your screenshare.</div><div>Admittedly you are only incrementally securing the first hop, but often - to coin a phrase - "the first hop is the weakest" ;-)</div><div><br></div><div>Tim.</div></div></body></html>
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From ted.ietf@gmail.com  Mon Mar 19 10:08:53 2012
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Date: Mon, 19 Mar 2012 10:08:51 -0700
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From: Ted Hardie <ted.ietf@gmail.com>
To: "Ejzak, Richard P (Richard)" <richard.ejzak@alcatel-lucent.com>
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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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On Mon, Mar 19, 2012 at 9:00 AM, Ejzak, Richard P (Richard)
<richard.ejzak@alcatel-lucent.com> wrote:
> Ted,
> I simply do not buy your argument.
>
> It doesn't matter what security scheme is used if you download a dedicate=
d
> communications client that you don't trust. =A0It can play whatever games=
 it
> wants with the UI and the security configuration and you are none the
> better off. =A0You have to trust the application.

Obviously, I have not expressed myself well.  If I have a downloaded
application for playing poker which includes WebRTC, I have a fairly fixed =
UI.
Advice given to browsers on how to reflect that a particular
connection has lower than usual security characteristics is likely
useless to that application UI designer, and the likelihood of a
consistent UI across multiple downloaded applications for "the other
endpoint wants unprotected media" is pretty close to zero.

In cases where you have both browsers and downloaded apps, you cannot
allow browsers to set the ravish-me bits for the flow because you
believe they gave consent *without creating a problem for the
downloaded apps and their users*.  Each one of the downloaded apps has
to invent a UI marker for this, and the user has to understand what
each marker means in its context for this to be successful.  That's
not nearly as tractable as getting Chrome, Safari, Mozilla, IE, Opera,
and friends to agree what belongs in browser chrome.

regards,

Ted

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From: Roman Shpount <roman@telurix.com>
To: Tim Panton <tim@phonefromhere.com>
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On Mon, Mar 19, 2012 at 12:53 PM, Tim Panton <tim@phonefromhere.com> wrote:

>
> On 19 Mar 2012, at 16:03, Ejzak, Richard P (Richard) wrote:
>
> ** **
>
> SRTP by itself guarantees nothing.  What is the point of insisting that
> the browser encrypt media if you know nothing about the other endpoint of
> the encrypted media or even whether anyone else has keys?
> ****
>
>
> If you are in an airport lounge, it should mean the script kiddie at the
> next gate over has a harder time reading your screenshare.
> Admittedly you are only incrementally securing the first hop, but often -
> to coin a phrase - "the first hop is the weakest" ;-)
>
> Tim.
>
>
In this case the simplest way to compromise your security is to seat next
to you. Your first hop is between your mouth and the microphone. Or between
your screen and your face.
_____________
Roman Shpount

--047d7b10cde3daf94504bb9bacfc
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<br clear=3D"all"><div class=3D"gmail_quote">On Mon, Mar 19, 2012 at 12:53 =
PM, Tim Panton <span dir=3D"ltr">&lt;<a href=3D"mailto:tim@phonefromhere.co=
m">tim@phonefromhere.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmai=
l_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left=
:1ex">
<div style=3D"word-wrap:break-word"><br><div><div class=3D"im"><div>On 19 M=
ar 2012, at 16:03, Ejzak, Richard P (Richard) wrote:</div><br><blockquote t=
ype=3D"cite">




<u></u>
<u></u>





<div link=3D"blue" vlink=3D"blue" lang=3D"EN-US">

<div><p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial"><span style=
=3D"font-size:10.0pt;font-family:Arial;color:navy">SRTP by itself guarantee=
s nothing. =A0What
is the point of insisting that the browser encrypt media if you know nothin=
g
about the other endpoint of the encrypted media or even whether anyone else=
 has
keys?</span></font></p></div></div><u></u><u></u></blockquote><div><br></di=
v></div><div>If you are in an airport lounge, it should mean the script kid=
die at the next gate over has a harder time reading your screenshare.</div>
<div>Admittedly you are only incrementally securing the first hop, but ofte=
n - to coin a phrase - &quot;the first hop is the weakest&quot; ;-)</div><d=
iv><br></div><font color=3D"#888888"><div>Tim.</div></font></div></div><br>
</blockquote></div><br>In this case the simplest way to compromise your sec=
urity is to seat next to you. Your first hop is between your mouth and the =
microphone. Or between your screen and your face.<br>_____________<br>Roman=
 Shpount<br>

<br><br>

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Mon, 19 Mar 2012 18:20:12 +0100
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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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2012/3/19 Roman Shpount <roman@telurix.com>:
> In this case the simplest way to compromise your security is to seat next=
 to
> you. Your first hop is between your mouth and the microphone. Or between
> your screen and your face.

You can control that (you know whether there is people hearing you or
not, or at least you can check it). But if you are using plain RTP in
an open WiFi connection you have no way to determine whether somebody
in the floor above/below is intercepting your RTP.

If your argument would be valid, then HTTPS is useless since somebody
could seat near you and watch your screen. Ok, let's focus the real
discussion...

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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From: Roman Shpount <roman@telurix.com>
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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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I guess my question is, when are we going to tell the user that connection
is "secure"? What I really do not want is making the distinction between
secured and unsecured connection so muddy, that a normal user will never be
able to understand the difference and will agree to anything. Because of
this, I would like to see that if connection is over DTLS-SRTP with a
remove party verified by an identity provider, user would be asked if they
would like to communicate with remote party XYZ. If this is anything else,
(DTLS-SRTP with no identity, SDES-SRTP, or even plain RTP), user should be
asked either to allow unsecure communication with an unknown party, or,
completely prohibited to do so.

I do not see the difference, as far as security is concerned, between
SDES-SRTP and plain RTP. If we allow SDES-SRTP for compatibility reasons,
we might as well allow plain RTP. It would be compatible with more devices
and as insecure as SDES-SRTP. There is no requirement that WebRTC
application must use HTTPS for signaling. If the application is using HTTP,
all the "sitting in the airport" examples are as unsecured with SDES-SRTP
as they are with plain RTP.

I do not think we would be able to force WebRTC application developers to
create a secure application unless they would decide to create care and
effort to develop it as such. Our goal should be to provide application
developers with tools that would allow them to create secure applications
and with mechanisms that would allow users to identify when their
communications are actually secure. Everything else is just adding random
requirements to the standard.
_____________
Roman Shpount

--047d7b15aee970c29f04bb9c8b4f
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I guess my question is, when are we going to tell the user that connection =
is &quot;secure&quot;? What I really do not want is making the distinction =
between secured and unsecured connection so muddy, that a normal user will =
never be able to understand the difference and will agree to anything. Beca=
use of this, I would like to see that if connection is over DTLS-SRTP with =
a remove party verified by an identity provider, user would be asked if the=
y would like to communicate with remote party XYZ. If this is anything else=
, (DTLS-SRTP with no identity, SDES-SRTP, or even plain RTP), user should b=
e asked either to allow unsecure communication with an unknown party, or, c=
ompletely prohibited to do so. <br>
<br>I do not see the difference, as far as security is concerned, between S=
DES-SRTP and plain RTP. If we allow SDES-SRTP for compatibility reasons, we=
 might as well allow plain RTP. It would be compatible with more devices an=
d as insecure as SDES-SRTP. There is no requirement that WebRTC application=
 must use HTTPS for signaling. If the application is using HTTP, all the &q=
uot;sitting in the airport&quot; examples are as unsecured with SDES-SRTP a=
s they are with plain RTP.<br>
<br>I do not think we would be able to force WebRTC application developers =
to create a secure application unless they would decide to create care and =
effort to develop it as such. Our goal should be to provide application dev=
elopers with tools that would allow them to create secure applications and =
with mechanisms that would allow users to identify when their communication=
s are actually secure. Everything else is just adding random requirements t=
o the standard.<br clear=3D"all">
_____________<br>Roman Shpount<br>
<br><br>

--047d7b15aee970c29f04bb9c8b4f--

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On 19 Mar 2012, at 18:15, Roman Shpount wrote:

> I do not see the difference, as far as security is concerned, between =
SDES-SRTP and plain RTP. If we allow SDES-SRTP for compatibility =
reasons, we might as well allow plain RTP. It would be compatible with =
more devices and as insecure as SDES-SRTP. There is no requirement that =
WebRTC application must use HTTPS for signaling. If the application is =
using HTTP, all the "sitting in the airport" examples are as unsecured =
with SDES-SRTP as they are with plain RTP.


How many of these plain RTP only legacy devices support password =
verified ICE correctly? - I'd be shocked if you found _any_.=20

With JSEP there is nothing to stop the application from encrypting the =
SDP blob in javascript before forwarding it to the far end=20
over HTTP - not my preferred option, but technically possible, and it =
would definitely make a firesheep style attack a bit harder to=20
pull off.

Tim.=

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<html><head></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
"><br><div><div><div><div>On 19 Mar 2012, at 18:15, Roman Shpount =
wrote:</div><br class=3D"Apple-interchange-newline"><blockquote =
type=3D"cite"><span class=3D"Apple-style-span" style=3D"border-collapse: =
separate; font-family: Helvetica; font-style: normal; font-variant: =
normal; font-weight: normal; letter-spacing: normal; line-height: =
normal; orphans: 2; text-align: -webkit-auto; text-indent: 0px; =
text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; =
-webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; font-size: medium; ">I do not see =
the difference, as far as security is concerned, between SDES-SRTP and =
plain RTP. If we allow SDES-SRTP for compatibility reasons, we might as =
well allow plain RTP. It would be compatible with more devices and as =
insecure as SDES-SRTP. There is no requirement that WebRTC application =
must use HTTPS for signaling. If the application is using HTTP, all the =
"sitting in the airport" examples are as unsecured with SDES-SRTP as =
they are with plain =
RTP.</span></blockquote></div><div><br></div><div>How many of these =
plain RTP only legacy devices support password verified ICE correctly? - =
I'd be shocked if you found _any_.&nbsp;</div><div><br></div><div>With =
JSEP there is nothing to stop the application from encrypting the SDP =
blob in javascript before forwarding it to the far =
end&nbsp;</div><div>over HTTP - not my preferred option, but technically =
possible, and it would definitely make a firesheep style attack a bit =
harder to&nbsp;</div><div>pull =
off.</div><div><br></div><div>Tim.</div></div></div></body></html>=

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--047d7b10cde315a66804bb9cf221
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On Mon, Mar 19, 2012 at 2:35 PM, Tim Panton <tim@phonefromhere.com> wrote:

> How many of these plain RTP only legacy devices support password verified
> ICE correctly? - I'd be shocked if you found _any_.
>
>
And how many of the SDES-SRTP devices support password verified ICE? I
doubt you would find any of them either. So, why is SDES-SRTP is OK and RTP
is not?



> With JSEP there is nothing to stop the application from encrypting the SDP
> blob in javascript before forwarding it to the far end
> over HTTP - not my preferred option, but technically possible, and it
> would definitely make a firesheep style attack a bit harder to pull off.
>
>
Once again, my point was that application developer would need to properly
develop signaling application (ie deliver it over HTTPS; don't put
encryption keys into something that can be easily accessable, etc). Unless
a lot of care is taken, SDES-SRTP is not secure.

Why do we need to support SDES-SRTP at all? I do not understand the
arguments that say RTP is bad, but SDES-SRTP is ok. If you need interop,
RTP is your best option. If you need security DTLS-SRTP is your answer.
SDES-SRTP does not serve either purpose well.
_____________
Roman Shpount

--047d7b10cde315a66804bb9cf221
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<br clear=3D"all"><br><div class=3D"gmail_quote">On Mon, Mar 19, 2012 at 2:=
35 PM, Tim Panton <span dir=3D"ltr">&lt;<a href=3D"mailto:tim@phonefromhere=
.com">tim@phonefromhere.com</a>&gt;</span> wrote:<br><blockquote class=3D"g=
mail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-l=
eft:1ex">
<div style=3D"word-wrap:break-word">How many of these plain RTP only legacy=
 devices support password verified ICE correctly? - I&#39;d be shocked if y=
ou found _any_.=A0<div><div><div class=3D"im"><div><br></div></div></div></=
div>
</div></blockquote><div><br>And how many of the SDES-SRTP devices support p=
assword verified ICE? I doubt you would find any of them either. So, why is=
 SDES-SRTP is OK and RTP is not?<br><br>=A0</div><blockquote class=3D"gmail=
_quote" style=3D"margin:0pt 0pt 0pt 0.8ex;border-left:1px solid rgb(204,204=
,204);padding-left:1ex">
<div style=3D"word-wrap:break-word"><div><div><div class=3D"im"><div></div>=
<div>With JSEP there is nothing to stop the application from encrypting the=
 SDP blob in javascript before forwarding it to the far end=A0</div><div>ov=
er HTTP - not my preferred option, but technically possible, and it would d=
efinitely make a firesheep style attack a bit harder to pull off.</div>
<div><br></div></div></div></div></div></blockquote></div><br>Once again, m=
y point was that application developer would need to properly develop signa=
ling application (ie deliver it over HTTPS; don&#39;t put encryption keys i=
nto something that can be easily accessable, etc). Unless a lot of care is =
taken, SDES-SRTP is not secure. <br>
<br>Why do we need to support SDES-SRTP at all? I do not understand the arg=
uments that say RTP is bad, but SDES-SRTP is ok. If you need interop, RTP i=
s your best option. If you need security DTLS-SRTP is your answer. SDES-SRT=
P does not serve either purpose well.<br>
_____________<br>
Roman Shpount<br>

<br>

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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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This is the question that I have been asking for a while...  I don't 
expect a complete fireproof answer, of course, and I also understand 
that the browser today is telling me a few things about the security of 
a site and warns me when "the site is trying to access the data it 
should not be accessing."

But I  also imagine that a rogue site could display a message mimicking 
the security assurance as though it comes from the browser.

So it would be good to have a very clear idea when  the determination 
about the security of the connection and such is made and how the end 
user can verify that it actually comes from the browser.

(To this end, the user MUST trust the browser, of course.)

Igor

On 3/19/2012 2:15 PM, Roman Shpount wrote:
> I guess my question is, when are we going to tell the user that 
> connection is "secure"?...

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Mon, 19 Mar 2012 20:47:41 +0100
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To: Roman Shpount <roman@telurix.com>
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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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2012/3/19 Roman Shpount <roman@telurix.com>:
> Once again, my point was that application developer would need to properl=
y
> develop signaling application (ie deliver it over HTTPS; don't put
> encryption keys into something that can be easily accessable, etc). Unles=
s a
> lot of care is taken, SDES-SRTP is not secure.

Again in the airport with open WiFi:

- If you use SDES-SRTP and HTTPS (or secure WebSocket), so SDES keys
are not interceptable in the WiFi network, you can be sure that no one
in the airport can monitor your media (I assume the server is not
hacked by a person in the same airport!).

- If you use plain RTP you know that every one in the airport can
monitor your media (regardless there is TLS in the signaling path or
not).



> Why do we need to support SDES-SRTP at all?

In conjunction with a secure signaling path (HTTPS or secure
WebSocket) IMHO it provides a reasonable security level for avoiding
media interception.



> I do not understand the
> arguments that say RTP is bad, but SDES-SRTP is ok.

Replied above.



> If you need interop, RTP
> is your best option. If you need security DTLS-SRTP is your answer.
> SDES-SRTP does not serve either purpose well.

I'm not a fan of SDES-SRTP, I just say that IMHO it can provide a good
security level.

The only I say is that there is no reason for allowing plain RTP other
than interop with old/legacy SIP devices (those whose vendors have
decided not to implement SRTP, a spec from 2004). Is there any other
case in which using plain RTP within WebRTC context is better than
using SRTP?


Regards.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From lorenzo@meetecho.com  Mon Mar 19 12:55:36 2012
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Date: Mon, 19 Mar 2012 20:52:07 +0100
From: Lorenzo Miniero <lorenzo@meetecho.com>
To: =?UTF-8?B?ScOxYWtp?= Baz Castillo <ibc@aliax.net>
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References: <4F4759DC.7060303@ericsson.com> <CAD5OKxvZbEJ7sV4WPAYoQapzMR_QwAftj-oKg=ioMKHNT792wQ@mail.gmail.com> <6F428EFD2B8C2F49A2FB1317291A76C113563C5A92@USNAVSXCHMBSA1.ndc.alcatel-lucent.com> <CALiegf=jtkDCS_D0ZFe9UpbiadQ0vsJ+4MppQSbLr-wbaXNrfQ@mail.gmail.com> <BLU169-W29E5B86F9E2C6F3126961C93420@phx.gbl> <CALiegfk2aT+6Psr4nT-hG1G7eYRBfFCcT+25On2O4HfUXJ6-ng@mail.gmail.com> <CAD6AjGSmi9j+sdGWPts20-iwGvGij05ek0OKYEPULC6B=aFpQg@mail.gmail.com> <6F428EFD2B8C2F49A2FB1317291A76C113564482A7@USNAVSXCHMBSA1.ndc.alcatel-lucent.com> <CAD5OKxvuEV8Vbq3h7=ZgcKmREjmguvz5n-SpXr2n-EY7a_ddxg@mail.gmail.com> <CALiegfk1ozOKPcDjbd3H_z2Edzh4RcZpYyJSWdw_1DJ04muQXA@mail.gmail.com> <CAD5OKxu8-+0O0=eE7mD1hi=nPUpEXczGj=bRNQCQL1BW8c-c-Q@mail.gmail.com> <52789D17-F7C7-401B-B2E8-6FE3BC5D7CB7@phonefromhere.com> <CAD5OKxtVtzahgk5xniXNvt-WvwNXZwcLau3PuKi1jnHrq4aZAA@mail.gmail.com> <CALiegf=xt1wAx8eid1M=wY0-wetmi9FOX+PoRF3iFd5UXmRgSA@mail.gmail.com>
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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On Mon, 19 Mar 2012 20:47:41 +0100
I=C3=B1aki Baz Castillo <ibc@aliax.net> wrote:

> 2012/3/19 Roman Shpount <roman@telurix.com>:
> > Once again, my point was that application developer would need to prope=
rly
> > develop signaling application (ie deliver it over HTTPS; don't put
> > encryption keys into something that can be easily accessable, etc). Unl=
ess a
> > lot of care is taken, SDES-SRTP is not secure.
>=20
> Again in the airport with open WiFi:
>=20
> - If you use SDES-SRTP and HTTPS (or secure WebSocket), so SDES keys
> are not interceptable in the WiFi network, you can be sure that no one
> in the airport can monitor your media (I assume the server is not
> hacked by a person in the same airport!).
>=20
> - If you use plain RTP you know that every one in the airport can
> monitor your media (regardless there is TLS in the signaling path or
> not).
>=20
>=20
>=20
> > Why do we need to support SDES-SRTP at all?
>=20
> In conjunction with a secure signaling path (HTTPS or secure
> WebSocket) IMHO it provides a reasonable security level for avoiding
> media interception.
>=20
>=20
>=20
> > I do not understand the
> > arguments that say RTP is bad, but SDES-SRTP is ok.
>=20
> Replied above.
>=20
>=20
>=20
> > If you need interop, RTP
> > is your best option. If you need security DTLS-SRTP is your answer.
> > SDES-SRTP does not serve either purpose well.
>=20
> I'm not a fan of SDES-SRTP, I just say that IMHO it can provide a good
> security level.
>=20
> The only I say is that there is no reason for allowing plain RTP other
> than interop with old/legacy SIP devices (those whose vendors have
> decided not to implement SRTP, a spec from 2004). Is there any other
> case in which using plain RTP within WebRTC context is better than
> using SRTP?
>=20


Large scale public events come to mind.

L.

>=20
> Regards.
>=20
>=20
> --=20
> I=C3=B1aki Baz Castillo
> <ibc@aliax.net>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


--=20
Lorenzo Miniero, COB

Meetecho s.r.l.
Web Conferencing and Collaboration Tools
http://www.meetecho.com

From lorenzo@meetecho.com  Mon Mar 19 12:58:48 2012
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From: Lorenzo Miniero <lorenzo@meetecho.com>
To: Lorenzo Miniero <lorenzo@meetecho.com>
Message-ID: <20120319205520.491ed951@rainpc>
In-Reply-To: <20120319205207.1133e5b0@rainpc>
References: <4F4759DC.7060303@ericsson.com> <CAD5OKxvZbEJ7sV4WPAYoQapzMR_QwAftj-oKg=ioMKHNT792wQ@mail.gmail.com> <6F428EFD2B8C2F49A2FB1317291A76C113563C5A92@USNAVSXCHMBSA1.ndc.alcatel-lucent.com> <CALiegf=jtkDCS_D0ZFe9UpbiadQ0vsJ+4MppQSbLr-wbaXNrfQ@mail.gmail.com> <BLU169-W29E5B86F9E2C6F3126961C93420@phx.gbl> <CALiegfk2aT+6Psr4nT-hG1G7eYRBfFCcT+25On2O4HfUXJ6-ng@mail.gmail.com> <CAD6AjGSmi9j+sdGWPts20-iwGvGij05ek0OKYEPULC6B=aFpQg@mail.gmail.com> <6F428EFD2B8C2F49A2FB1317291A76C113564482A7@USNAVSXCHMBSA1.ndc.alcatel-lucent.com> <CAD5OKxvuEV8Vbq3h7=ZgcKmREjmguvz5n-SpXr2n-EY7a_ddxg@mail.gmail.com> <CALiegfk1ozOKPcDjbd3H_z2Edzh4RcZpYyJSWdw_1DJ04muQXA@mail.gmail.com> <CAD5OKxu8-+0O0=eE7mD1hi=nPUpEXczGj=bRNQCQL1BW8c-c-Q@mail.gmail.com> <52789D17-F7C7-401B-B2E8-6FE3BC5D7CB7@phonefromhere.com> <CAD5OKxtVtzahgk5xniXNvt-WvwNXZwcLau3PuKi1jnHrq4aZAA@mail.gmail.com> <CALiegf=xt1wAx8eid1M=wY0-wetmi9FOX+PoRF3iFd5UXmRgSA@mail.gmail.com> <20120319205207.1133e5b0@rainpc>
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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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On Mon, 19 Mar 2012 20:52:07 +0100
Lorenzo Miniero <lorenzo@meetecho.com> wrote:

> On Mon, 19 Mar 2012 20:47:41 +0100
> I=C3=B1aki Baz Castillo <ibc@aliax.net> wrote:
>=20
> > 2012/3/19 Roman Shpount <roman@telurix.com>:
> > > Once again, my point was that application developer would need to pro=
perly
> > > develop signaling application (ie deliver it over HTTPS; don't put
> > > encryption keys into something that can be easily accessable, etc). U=
nless a
> > > lot of care is taken, SDES-SRTP is not secure.
> >=20
> > Again in the airport with open WiFi:
> >=20
> > - If you use SDES-SRTP and HTTPS (or secure WebSocket), so SDES keys
> > are not interceptable in the WiFi network, you can be sure that no one
> > in the airport can monitor your media (I assume the server is not
> > hacked by a person in the same airport!).
> >=20
> > - If you use plain RTP you know that every one in the airport can
> > monitor your media (regardless there is TLS in the signaling path or
> > not).
> >=20
> >=20
> >=20
> > > Why do we need to support SDES-SRTP at all?
> >=20
> > In conjunction with a secure signaling path (HTTPS or secure
> > WebSocket) IMHO it provides a reasonable security level for avoiding
> > media interception.
> >=20
> >=20
> >=20
> > > I do not understand the
> > > arguments that say RTP is bad, but SDES-SRTP is ok.
> >=20
> > Replied above.
> >=20
> >=20
> >=20
> > > If you need interop, RTP
> > > is your best option. If you need security DTLS-SRTP is your answer.
> > > SDES-SRTP does not serve either purpose well.
> >=20
> > I'm not a fan of SDES-SRTP, I just say that IMHO it can provide a good
> > security level.
> >=20
> > The only I say is that there is no reason for allowing plain RTP other
> > than interop with old/legacy SIP devices (those whose vendors have
> > decided not to implement SRTP, a spec from 2004). Is there any other
> > case in which using plain RTP within WebRTC context is better than
> > using SRTP?
> >=20
>=20
>=20
> Large scale public events come to mind.
>=20
> L.
>=20


(even though, thinking about it, they're probably out of scope in here)

L.

> >=20
> > Regards.
> >=20
> >=20
> > --=20
> > I=C3=B1aki Baz Castillo
> > <ibc@aliax.net>
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
>=20
>=20
> --=20
> Lorenzo Miniero, COB
>=20
> Meetecho s.r.l.
> Web Conferencing and Collaboration Tools
> http://www.meetecho.com
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


--=20
Lorenzo Miniero, COB

Meetecho s.r.l.
Web Conferencing and Collaboration Tools
http://www.meetecho.com

From roman@telurix.com  Mon Mar 19 13:08:31 2012
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From: Roman Shpount <roman@telurix.com>
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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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On Mon, Mar 19, 2012 at 3:47 PM, I=F1aki Baz Castillo <ibc@aliax.net> wrote=
:

>
> The only I say is that there is no reason for allowing plain RTP other
> than interop with old/legacy SIP devices (those whose vendors have
> decided not to implement SRTP, a spec from 2004). Is there any other
> case in which using plain RTP within WebRTC context is better than
> using SRTP?
>
>
Is there any case in which using SDES-SRTP is better then using DTLS-SRTP,
except interop with old/legacy devices?
_____________
Roman Shpount

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<br clear=3D"all"><div class=3D"gmail_quote">On Mon, Mar 19, 2012 at 3:47 P=
M, I=F1aki Baz Castillo <span dir=3D"ltr">&lt;<a href=3D"mailto:ibc@aliax.n=
et">ibc@aliax.net</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote=
" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<br>
The only I say is that there is no reason for allowing plain RTP other<br>
than interop with old/legacy SIP devices (those whose vendors have<br>
decided not to implement SRTP, a spec from 2004). Is there any other<br>
case in which using plain RTP within WebRTC context is better than<br>
using SRTP?<br>
<br></blockquote></div><br>Is there any case in which using SDES-SRTP is be=
tter then using DTLS-SRTP, except interop with old/legacy devices?<br>_____=
________<br>Roman Shpount<br>
<br><br>

--047d7b10cde331a9e004bb9e1f01--

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To: Lorenzo Miniero <lorenzo@meetecho.com>
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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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On 19 Mar 2012, at 19:52, Lorenzo Miniero wrote:
>>=20
>> The only I say is that there is no reason for allowing plain RTP =
other
>> than interop with old/legacy SIP devices (those whose vendors have
>> decided not to implement SRTP, a spec from 2004). Is there any other
>> case in which using plain RTP within WebRTC context is better than
>> using SRTP?
>>=20
>=20
>=20
> Large scale public events come to mind.

Assuming that you are thinking of the CPU required to encrypt all the =
streams,=20
I see no reason you couldn't have all the streams use the same keys, the =
you could
write a special SRTP stack that encoded exactly once then echoed the =
same packets to all the recipients.

(Which is close to the case for satellite TV at the moment I believe).

T.=

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<html><head></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
"><br><div><div>On 19 Mar 2012, at 19:52, Lorenzo Miniero =
wrote:</div><blockquote type=3D"cite"><div><blockquote type=3D"cite"><font=
 class=3D"Apple-style-span" =
color=3D"#000000"><br></font></blockquote><blockquote type=3D"cite">The =
only I say is that there is no reason for allowing plain RTP =
other<br></blockquote><blockquote type=3D"cite">than interop with =
old/legacy SIP devices (those whose vendors =
have<br></blockquote><blockquote type=3D"cite">decided not to implement =
SRTP, a spec from 2004). Is there any other<br></blockquote><blockquote =
type=3D"cite">case in which using plain RTP within WebRTC context is =
better than<br></blockquote><blockquote type=3D"cite">using =
SRTP?<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><br><br>Large scale public events come to =
mind.<br></div></blockquote><div><br></div>Assuming that you are =
thinking of the CPU required to encrypt all the =
streams,&nbsp;</div><div>I see no reason you couldn't have all the =
streams use the same keys, the you could</div><div>write a special SRTP =
stack that encoded exactly once then echoed the same packets to all the =
recipients.</div><div><br></div><div>(Which is close to the case for =
satellite TV at the moment I =
believe).</div><div><br></div><div>T.</div></body></html>=

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Mon, 19 Mar 2012 22:04:21 +0100
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2012/3/19 Roman Shpount <roman@telurix.com>:
>> Is there any other
>> case in which using plain RTP within WebRTC context is better than
>> using SRTP?
>>
>
> Is there any case in which using SDES-SRTP is better then using DTLS-SRTP=
,
> except interop with old/legacy devices?

Mandating DTLS-SRTP (this is, not allowing plain RTP neither
SDES-SRTP) is not a problem for me. :)

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Thread-Topic: [rtcweb] Resolving RTP/SDES question in Paris
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Date: Mon, 19 Mar 2012 21:36:47 +0000
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Cc: "<rtcweb@ietf.org>" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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On Mar 19, 2012, at 4:08 PM, Roman Shpount wrote:

> Is there any case in which using SDES-SRTP is better then using DTLS-SRTP=
, except interop with old/legacy devices?


Yup.  The other advantages of SDES are:
1) Shorter call-answer-delay, because there're no additional round-trips be=
fore media other than ICE
2) Cheaper/less-overhead/better-scaling
3) Fewer interop issues - we don't know what DTLS-SRTP interop issues will =
exist, if any, but more code usually means more interop issues, or at least=
 more "bugs".  ICE itself will probably have some issues anyway[*], but mor=
e code and complexity won't help matters.

And if the Identity-model stuff is mandatory to use for DTLS, you can multi=
ply the impact of those factors by some non-zero multiple.

-hadriel
[*] There are already multiple variants of ICE in the wild, with one implem=
entation known not to interop with others.  But I don't know if it's an SDP=
-specific ICE interop issue or a problem with the connectivity checks or st=
ate machine.


From HKaplan@acmepacket.com  Mon Mar 19 15:23:58 2012
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Roman Shpount <roman@telurix.com>
Thread-Topic: [rtcweb] Resolving RTP/SDES question in Paris
Thread-Index: AQHNBh74aMnjJsNUq0eXz+PkH27a4g==
Date: Mon, 19 Mar 2012 22:23:55 +0000
Message-ID: <D75A384B-0F38-4E30-8C03-12E903A69B64@acmepacket.com>
References: <4F4759DC.7060303@ericsson.com> <387F9047F55E8C42850AD6B3A7A03C6C0E1FEB69@inba-mail01.sonusnet.com> <CALiegfnkYVEpmPV-zSL_4wOY-HiFZN-qJCQCiioaS=5NaqhLZw@mail.gmail.com> <CAD5OKxvtOAxMBx6xDnyfTnEq76oDEm6uj1xL6wGjjrtKUAHy3g@mail.gmail.com> <CABcZeBNZiotPmCfT53uEo+O0xw4xv6tXW1M_G-3A5BHuncsduA@mail.gmail.com> <CAD5OKxvYOY5JZ2mYNGiH1poUBQkyOOycePFijH5H+SxtcdqujQ@mail.gmail.com> <CABkgnnVe-b6Sv=R67bMJk_NQqQwdrRUn6rBm7Gu_CMcfPQwtEg@mail.gmail.com> <CAD5OKxvZbEJ7sV4WPAYoQapzMR_QwAftj-oKg=ioMKHNT792wQ@mail.gmail.com> <6F428EFD2B8C2F49A2FB1317291A76C113563C5A92@USNAVSXCHMBSA1.ndc.alcatel-lucent.com> <CALiegf=jtkDCS_D0ZFe9UpbiadQ0vsJ+4MppQSbLr-wbaXNrfQ@mail.gmail.com> <BLU169-W29E5B86F9E2C6F3126961C93420@phx.gbl> <CALiegfk2aT+6Psr4nT-hG1G7eYRBfFCcT+25On2O4HfUXJ6-ng@mail.gmail.com> <CAD6AjGSmi9j+sdGWPts20-iwGvGij05ek0OKYEPULC6B=aFpQg@mail.gmail.com> <6F428EFD2B8C2F49A2FB1317291A76C113564482A7@USNAVSXCHMBSA1.ndc.alcatel-lucent.com> <ADBB75F3-E20C-4EC4-B9C3-EF2E4BFF409C@phonefromhere.com> <CAD5OKxvuEV8Vbq3h7=ZgcKmREjmguvz5n-SpXr2n-EY7a_ddxg@mail.gmail.com> <CALiegfk1ozOKPcDjbd3H_z2Edzh4RcZpYyJSWdw_1DJ04muQXA@mail.gmail.com> <CAD5OKxu8-+0O0=eE7mD1hi=nPUpEXczGj=bRNQCQL1BW8c-c-Q@mail.gmail.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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While I sympathize with the concern that requiring SRTP (regardless of how =
the key is learned) is a burden for those use-cases that literally don't ca=
re about SRTP and for which it does not matter, such as a weather announcem=
ent service or an IETF meeting broadcast, I really think you're 'throwing o=
ut the baby with the bath-water' to propose we either get everything or not=
hing.  People won't have the identity-stuff for some time, and it won't be =
painless to make it work and deployed.  So what you're basically saying is =
"you only get RTP because you can't prove end-to-end security".  But such "=
proof" rarely exists even today.

Today, when you do Email or IM over HTTPS or SMTPS, that email or IM only g=
oes secure to the server you're doing SMTPS or HTTPS with.  You have no ass=
urance whatsoever that it stays secure from that point on, nor that the ser=
ver doesn't log the email/IM in its entirety.  But that doesn't mean there'=
s no value in using HTTPS/SMTPS to reach that first server.  Quite the cont=
rary, there's significant value in it.  Just like there's value in using HT=
TPS to reach a web store-front when purchasing using a credit-card, even th=
ough you have no idea what the security is between the web-store and your a=
ctual credit-card company.

In the WebRTC context, for example, I would expect a quite-popular use-case=
 for WebRTC to be Business websites, such as web-storefronts, or customer-s=
upport sites, or airline/hotel ticket/reservations/upgrades sites, etc.  Wh=
en connecting to the web-page of those sites, I would expect them to be HTT=
PS once I got to some point of the click-flow process, if not from the begi=
nning.  If I click a "talk to us for help" button, I would expect that to b=
e secure as well, as much as the HTTPS portion was anyway.  In this context=
, only allowing them to use RTP is silly.  SDES is reasonably secure in thi=
s context, because there is no other domain the call is going to.  Yes, the=
y have my SRTP key - doesn't matter; yes they can log my call - doesn't mat=
ter.  But at least my media is secure from my laptop to their server's doma=
in - and that does matter, and it's all I ever got from using HTTPS.

-hadriel


On Mar 19, 2012, at 2:15 PM, Roman Shpount wrote:

> I guess my question is, when are we going to tell the user that connectio=
n is "secure"? What I really do not want is making the distinction between =
secured and unsecured connection so muddy, that a normal user will never be=
 able to understand the difference and will agree to anything. Because of t=
his, I would like to see that if connection is over DTLS-SRTP with a remove=
 party verified by an identity provider, user would be asked if they would =
like to communicate with remote party XYZ. If this is anything else, (DTLS-=
SRTP with no identity, SDES-SRTP, or even plain RTP), user should be asked =
either to allow unsecure communication with an unknown party, or, completel=
y prohibited to do so.=20
>=20
> I do not see the difference, as far as security is concerned, between SDE=
S-SRTP and plain RTP. If we allow SDES-SRTP for compatibility reasons, we m=
ight as well allow plain RTP. It would be compatible with more devices and =
as insecure as SDES-SRTP. There is no requirement that WebRTC application m=
ust use HTTPS for signaling. If the application is using HTTP, all the "sit=
ting in the airport" examples are as unsecured with SDES-SRTP as they are w=
ith plain RTP.
>=20
> I do not think we would be able to force WebRTC application developers to=
 create a secure application unless they would decide to create care and ef=
fort to develop it as such. Our goal should be to provide application devel=
opers with tools that would allow them to create secure applications and wi=
th mechanisms that would allow users to identify when their communications =
are actually secure. Everything else is just adding random requirements to =
the standard.
> _____________
> Roman Shpount
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From Jim.Barnett@genesyslab.com  Mon Mar 19 15:50:24 2012
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Thread-Topic: [rtcweb] Resolving RTP/SDES question in Paris
Thread-Index: AQHNBh74aMnjJsNUq0eXz+PkH27a4pZyN+Gg
References: <4F4759DC.7060303@ericsson.com><387F9047F55E8C42850AD6B3A7A03C6C0E1FEB69@inba-mail01.sonusnet.com><CALiegfnkYVEpmPV-zSL_4wOY-HiFZN-qJCQCiioaS=5NaqhLZw@mail.gmail.com><CAD5OKxvtOAxMBx6xDnyfTnEq76oDEm6uj1xL6wGjjrtKUAHy3g@mail.gmail.com><CABcZeBNZiotPmCfT53uEo+O0xw4xv6tXW1M_G-3A5BHuncsduA@mail.gmail.com><CAD5OKxvYOY5JZ2mYNGiH1poUBQkyOOycePFijH5H+SxtcdqujQ@mail.gmail.com><CABkgnnVe-b6Sv=R67bMJk_NQqQwdrRUn6rBm7Gu_CMcfPQwtEg@mail.gmail.com><CAD5OKxvZbEJ7sV4WPAYoQapzMR_QwAftj-oKg=ioMKHNT792wQ@mail.gmail.com><6F428EFD2B8C2F49A2FB1317291A76C113563C5A92@USNAVSXCHMBSA1.ndc.alcatel-lucent.com><CALiegf=jtkDCS_D0ZFe9UpbiadQ0vsJ+4MppQSbLr-wbaXNrfQ@mail.gmail.com><BLU169-W29E5B86F9E2C6F3126961C93420@phx.gbl><CALiegfk2aT+6Psr4nT-hG1G7eYRBfFCcT+25On2O4HfUXJ6-ng@mail.gmail.com><CAD6AjGSmi9j+sdGWPts20-iwGvGij05ek0OKYEPULC6B=aFpQg@mail.gmail.com><6F428EFD2B8C2F49A2FB1317291A76C113564482A7@USNAVSXCHMBSA1.ndc.alcatel-lucent.com><ADBB75F3-E20C-4EC4-B9C3-EF2E4BFF409C@phonefromhere.com><C! AD5OKxvuE V8Vbq3h7=Zgc KmREjmguvz5n-SpXr2n-EY7a_ddxg@mail.gmail.com><CALiegfk1ozOKPcDjbd3H_z2Edzh4RcZpYyJSWdw_1DJ04muQXA@mail.gmail.com><CAD5OKxu8-+0O0=eE7mD1hi=nPUpEXczGj=bRNQCQL1BW8c-c-Q@mail.gmail.com> <D75A384B-0F38-4E30-8C03-12E903A69B64@acmepacket.com>
From: "Jim Barnett" <Jim.Barnett@genesyslab.com>
To: "Hadriel Kaplan" <HKaplan@acmepacket.com>, "Roman Shpount" <roman@telurix.com>
X-OriginalArrivalTime: 19 Mar 2012 22:50:17.0428 (UTC) FILETIME=[A79BE940:01CD0622]
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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On the matter of the call center/travel site recording your call:  can't
the far-end user agent always record your call, no matter what security
you use?  So in the case of SDES, the call center can record at their
SBC/switch/media server (their in-house MITM), which is easier for them,
but even with DTLS/SRTP couldn't they always record it at the agent
desktop, if they wanted to?

- Jim

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
Of Hadriel Kaplan
Sent: Monday, March 19, 2012 6:24 PM
To: Roman Shpount
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris


While I sympathize with the concern that requiring SRTP (regardless of
how the key is learned) is a burden for those use-cases that literally
don't care about SRTP and for which it does not matter, such as a
weather announcement service or an IETF meeting broadcast, I really
think you're 'throwing out the baby with the bath-water' to propose we
either get everything or nothing.  People won't have the identity-stuff
for some time, and it won't be painless to make it work and deployed.
So what you're basically saying is "you only get RTP because you can't
prove end-to-end security".  But such "proof" rarely exists even today.

Today, when you do Email or IM over HTTPS or SMTPS, that email or IM
only goes secure to the server you're doing SMTPS or HTTPS with.  You
have no assurance whatsoever that it stays secure from that point on,
nor that the server doesn't log the email/IM in its entirety.  But that
doesn't mean there's no value in using HTTPS/SMTPS to reach that first
server.  Quite the contrary, there's significant value in it.  Just like
there's value in using HTTPS to reach a web store-front when purchasing
using a credit-card, even though you have no idea what the security is
between the web-store and your actual credit-card company.

In the WebRTC context, for example, I would expect a quite-popular
use-case for WebRTC to be Business websites, such as web-storefronts, or
customer-support sites, or airline/hotel ticket/reservations/upgrades
sites, etc.  When connecting to the web-page of those sites, I would
expect them to be HTTPS once I got to some point of the click-flow
process, if not from the beginning.  If I click a "talk to us for help"
button, I would expect that to be secure as well, as much as the HTTPS
portion was anyway.  In this context, only allowing them to use RTP is
silly.  SDES is reasonably secure in this context, because there is no
other domain the call is going to.  Yes, they have my SRTP key - doesn't
matter; yes they can log my call - doesn't matter.  But at least my
media is secure from my laptop to their server's domain - and that does
matter, and it's all I ever got from using HTTPS.

-hadriel


On Mar 19, 2012, at 2:15 PM, Roman Shpount wrote:

> I guess my question is, when are we going to tell the user that
connection is "secure"? What I really do not want is making the
distinction between secured and unsecured connection so muddy, that a
normal user will never be able to understand the difference and will
agree to anything. Because of this, I would like to see that if
connection is over DTLS-SRTP with a remove party verified by an identity
provider, user would be asked if they would like to communicate with
remote party XYZ. If this is anything else, (DTLS-SRTP with no identity,
SDES-SRTP, or even plain RTP), user should be asked either to allow
unsecure communication with an unknown party, or, completely prohibited
to do so.=20
>=20
> I do not see the difference, as far as security is concerned, between
SDES-SRTP and plain RTP. If we allow SDES-SRTP for compatibility
reasons, we might as well allow plain RTP. It would be compatible with
more devices and as insecure as SDES-SRTP. There is no requirement that
WebRTC application must use HTTPS for signaling. If the application is
using HTTP, all the "sitting in the airport" examples are as unsecured
with SDES-SRTP as they are with plain RTP.
>=20
> I do not think we would be able to force WebRTC application developers
to create a secure application unless they would decide to create care
and effort to develop it as such. Our goal should be to provide
application developers with tools that would allow them to create secure
applications and with mechanisms that would allow users to identify when
their communications are actually secure. Everything else is just adding
random requirements to the standard.
> _____________
> Roman Shpount
>=20
>=20
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Date: Mon, 19 Mar 2012 19:34:16 -0400
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From: Roman Shpount <roman@telurix.com>
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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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--047d7b15aee91f344004bba0ffbb
Content-Type: text/plain; charset=ISO-8859-1

Here are the levels of security that we get right now:

1. DTLS SRTP with Identity provider with signaling over HTTP or HTTPS. As
long as identity provider is not compromised, user knows who they are
talking to and that conversation is reasonably secure.

2. DTLS SRTP or SDES SRTP with signaling over HTTPS. There are no
guarantees on who exactly the user is talking to, but as long we assume
that web server and javascript applications are not compromised,
conversation is secured to some application selected destination. Cannot
tell the user that the call is secure, but probably not going to be
something that can be picked up by a script kiddy with a network sniffer.

3. DTLS SRTP or SDES SRTP with signaling over HTTP. There are no guarantees
on where the application came from or who exactly you are talking to. The
conversation is encrypted, but it can either be listened to by simply
capturing the keys (SDES SRTP) or you can spoof DNS and take over the whole
web application and make DTLS SRTP talk to some sort of monitoring proxy.
No security and probably not much better then plain RTP.

So, my question is, are we going to disable WebRTC from HTTP initiated
sessions completely and show the appropriate UI confirmations (secure for 1
and unsecure for 2) or do we plan to do something different? If we allow
any non identity non DTLS SRTP communications with signaling over HTTP, we
might as well allow all of them. All of those communication modes will have
the same security downsides, and I would argue that RTP have probably the
most upsides.
_____________
Roman Shpount

--047d7b15aee91f344004bba0ffbb
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Here are the levels of security that we get right now:<br><br>1. DTLS SRTP =
with Identity provider with signaling over HTTP or HTTPS. As long as identi=
ty provider is not compromised, user knows who they are talking to and that=
 conversation is reasonably secure.<br>
<br>2. DTLS SRTP or SDES SRTP with signaling over HTTPS. There are no guara=
ntees on who exactly the user is talking to, but as long we assume that web=
 server and javascript applications are not compromised, conversation is se=
cured to some application selected destination. Cannot tell the user that t=
he call is secure, but probably not going to be something that can be picke=
d up by a script kiddy with a network sniffer.<br>
<br>3. DTLS SRTP or SDES SRTP with signaling over HTTP. There are no guaran=
tees on where the application came from or who exactly you are talking to. =
The conversation is encrypted, but it can either be listened to by simply c=
apturing the keys (SDES SRTP) or you can spoof DNS and take over the whole =
web application and make DTLS SRTP talk to some sort of monitoring proxy. N=
o security and probably not much better then plain RTP.<br>
<br>So, my question is, are we going to disable WebRTC from HTTP initiated =
sessions completely and show the appropriate UI confirmations (secure for 1=
 and unsecure for 2) or do we plan to do something different? If we allow a=
ny non identity non DTLS SRTP communications with signaling over HTTP, we m=
ight as well allow all of them. All of those communication modes will have =
the same security downsides, and I would argue that RTP have probably the m=
ost upsides.<br clear=3D"all">
_____________<br>Roman Shpount<br>
<br>

--047d7b15aee91f344004bba0ffbb--

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From: Eric Rescorla <ekr@rtfm.com>
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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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On Mon, Mar 19, 2012 at 4:34 PM, Roman Shpount <roman@telurix.com> wrote:
> Here are the levels of security that we get right now:
>
> 1. DTLS SRTP with Identity provider with signaling over HTTP or HTTPS. As
> long as identity provider is not compromised, user knows who they are
> talking to and that conversation is reasonably secure.
>
> 2. DTLS SRTP or SDES SRTP with signaling over HTTPS. There are no guarantees
> on who exactly the user is talking to, but as long we assume that web server
> and javascript applications are not compromised, conversation is secured to
> some application selected destination. Cannot tell the user that the call is
> secure, but probably not going to be something that can be picked up by a
> script kiddy with a network sniffer.

As discussed in draft-ietf-rtcweb-security, I don't agree that DTLS-SRTP without
identity and SDES-SRTP offer the same level of security. Most of what I have
to say about that is already said there, but briefly:

- DTLS-SRTP is secure against retrospective attack whereas SDES-SRTP is not.
- If there is a mechanism such as fingerprints *is* available for
checking against
MITM, then DTLS-SRTP is secure against active attack as well. Additionally,
the existence of fingerprint type mechanisms acts as a partial deterrent against
MITM attack by the signaling provider even if only a small number of people
check.

-Ekr

From roman@telurix.com  Mon Mar 19 17:59:08 2012
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Date: Mon, 19 Mar 2012 20:59:05 -0400
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From: Roman Shpount <roman@telurix.com>
To: Eric Rescorla <ekr@rtfm.com>
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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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On Mon, Mar 19, 2012 at 8:18 PM, Eric Rescorla <ekr@rtfm.com> wrote:

> As discussed in draft-ietf-rtcweb-security, I don't agree that DTLS-SRTP
> without
> identity and SDES-SRTP offer the same level of security. Most of what I
> have
> to say about that is already said there, but briefly:
>
> - DTLS-SRTP is secure against retrospective attack whereas SDES-SRTP is
> not.
> - If there is a mechanism such as fingerprints *is* available for
> checking against
> MITM, then DTLS-SRTP is secure against active attack as well. Additionally,
> the existence of fingerprint type mechanisms acts as a partial deterrent
> against
> MITM attack by the signaling provider even if only a small number of people
> check.
>
>
I agree with you on all of those points. I would prefer that instead of
SDES SRTP support we spent time optimizing DTLS/ICE handshake (and there
are a few drafts on this list related to that). It should be possible to
conduct a secure key exchange at the same time as ICE candidate selection
with no additional overhead. I also understand the argument that most of
the regular users will never check the fingerprint, and this is why I put
SDES-SRTP and DTLS-SRTP in the same group. DTLS-SRTP is more secure then
SDES-SRTP, but it still not enough to tell the user that communication is
secure.

But regardless of all these points, if we allow signaling over HTTP all the
security considerations go out of the window.

For me the ideal solution is:

1. DTLS-SRTP with Identity and any signaling is always allowed and shows
notification that communication is with a known party and secure.
2. DTLS-SRTP with signaling over HTTPS shows notification that
communication is unsecure and with an unknown party
3. SDES SRTP is not supported at all and DTLS/ICE handshake is optimized
instead to reduce call setup time
4. DTLS-SRTP with no ident and RTP with signaling over HTTP are not
recommended, but allowed after a big red sign (or require some advanced
browser setting to enable).

I would also like to have a simplified DTLS specification for DTLS-SRTP,
with as many as possible unnecessary features (encryption algorithms,
cookies and such) removed and only a few required scenarios supported. For
instance, I do not want DTLS on a WebRTC channel start negotiation and end
up with anything except SRTP as an encryption protocol. Ideally, I want
something were an interop test can be developed with a manageable number of
scenarios.

P.S. Sending RSA public key in the session description instead of
fingerprint and then encrypting the SRTP key using this public key and
sending it in the ICE message might be a much simpler call setup mechanism
to implement then bringing the whole DTLS into the stack. This should also
allow us to setup a secure call with no overhead in addition to ICE.
_____________
Roman Shpount

--e89a8ff1c65a75157104bba22e00
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<br clear=3D"all"><br><div class=3D"gmail_quote">On Mon, Mar 19, 2012 at 8:=
18 PM, Eric Rescorla <span dir=3D"ltr">&lt;<a href=3D"mailto:ekr@rtfm.com">=
ekr@rtfm.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" sty=
le=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
As discussed in draft-ietf-rtcweb-security, I don&#39;t agree that DTLS-SRT=
P without<br>
identity and SDES-SRTP offer the same level of security. Most of what I hav=
e<br>
to say about that is already said there, but briefly:<br>
<br>
- DTLS-SRTP is secure against retrospective attack whereas SDES-SRTP is not=
.<br>
- If there is a mechanism such as fingerprints *is* available for<br>
checking against<br>
MITM, then DTLS-SRTP is secure against active attack as well. Additionally,=
<br>
the existence of fingerprint type mechanisms acts as a partial deterrent ag=
ainst<br>
MITM attack by the signaling provider even if only a small number of people=
<br>
check.<br>
<br></blockquote><div>=A0</div></div>I agree with you on all of those point=
s. I would prefer that instead of SDES SRTP support we spent time optimizin=
g DTLS/ICE handshake (and there are a few drafts on this list related to th=
at). It should be possible to conduct a secure key exchange at the same tim=
e as ICE candidate selection with no additional overhead. I also understand=
 the argument that most of the regular users will never check the fingerpri=
nt, and this is why I put SDES-SRTP and DTLS-SRTP in the same group. DTLS-S=
RTP is more secure then SDES-SRTP, but it still not enough to tell the user=
 that communication is secure.<br>
<br>But regardless of all these points, if we allow signaling over HTTP all=
 the security considerations go out of the window.<br><br>For me the ideal =
solution is:<br><br>1. DTLS-SRTP with Identity and any signaling is always =
allowed and shows notification that communication is with a known party and=
 secure.<br>
2. DTLS-SRTP with signaling over HTTPS shows notification that communicatio=
n is unsecure and with an unknown party<br>3. SDES SRTP is not supported at=
 all and DTLS/ICE handshake is optimized instead to reduce call setup time<=
br>
4. DTLS-SRTP with no ident and RTP with signaling over HTTP are not recomme=
nded, but allowed after a big red sign (or require some advanced browser se=
tting to enable).<br><br>I would also like to have a simplified DTLS specif=
ication for DTLS-SRTP, with as many as possible unnecessary features (encry=
ption algorithms, cookies and such) removed and only a few required scenari=
os supported. For instance, I do not want DTLS on a WebRTC channel start ne=
gotiation and end up with anything except SRTP as an encryption protocol. I=
deally, I want something were an interop test can be developed with a manag=
eable number of scenarios.<br>
<br>P.S. Sending RSA public key in the session description instead of finge=
rprint and then encrypting the SRTP key using this public key and sending i=
t in the ICE message might be a much simpler call setup mechanism to implem=
ent then bringing the whole DTLS into the stack. This should also allow us =
to setup a secure call with no overhead in addition to ICE.<br>
_____________<br>Roman Shpount<br>
<br>

--e89a8ff1c65a75157104bba22e00--

From pravindran@sonusnet.com  Mon Mar 19 18:06:47 2012
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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: Eric Rescorla <ekr@rtfm.com>
Thread-Topic: RTCP consent approach comment
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Eric,

Sec 4.2.3 mentions about RTCP consent approach is not adopted for

1) Small window attack of RTP
2) Tough for legacy non-RTCP endpoint to implement RTCP

I think that small window attack of RTP is possible in ICE as well in case =
attacker has access to offer SDP and the second reason is funny to recommen=
d non-RTCP endpoint to implement ICE instead of implementing RTCP because t=
his reason is equivalent of asking the person to buy cake who is struggling=
 to buy bread.

The missing information is that NAT traversal is not possible using RTCP me=
chanism. Could you please add "NAT traversal" as the one of the reason for =
not adopting RTCP consent approach.

Thanks
Partha

From pravindran@sonusnet.com  Mon Mar 19 18:16:02 2012
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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: Jim Barnett <Jim.Barnett@genesyslab.com>, Hadriel Kaplan <HKaplan@acmepacket.com>, Roman Shpount <roman@telurix.com>
Thread-Topic: [rtcweb] Resolving RTP/SDES question in Paris
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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Jim,

As a customer, you won't really know whether the identity (DTLS-SRTP) of ca=
ll center/travel site is agent or SBC. SBC can perform MITM attack easily a=
s extending SDES-SRTP to DTLS-SRTP for call center site is feasible.

Thanks
Partha

>-----Original Message-----
>From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
>Of Jim Barnett
>Sent: Tuesday, March 20, 2012 4:19 AM
>To: Hadriel Kaplan; Roman Shpount
>Cc: rtcweb@ietf.org
>Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
>
>On the matter of the call center/travel site recording your call:  can't
>the far-end user agent always record your call, no matter what security
>you use?  So in the case of SDES, the call center can record at their
>SBC/switch/media server (their in-house MITM), which is easier for them,
>but even with DTLS/SRTP couldn't they always record it at the agent
>desktop, if they wanted to?
>
>- Jim
>
>-----Original Message-----
>From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
>Of Hadriel Kaplan
>Sent: Monday, March 19, 2012 6:24 PM
>To: Roman Shpount
>Cc: rtcweb@ietf.org
>Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
>
>
>While I sympathize with the concern that requiring SRTP (regardless of
>how the key is learned) is a burden for those use-cases that literally
>don't care about SRTP and for which it does not matter, such as a
>weather announcement service or an IETF meeting broadcast, I really
>think you're 'throwing out the baby with the bath-water' to propose we
>either get everything or nothing.  People won't have the identity-stuff
>for some time, and it won't be painless to make it work and deployed.
>So what you're basically saying is "you only get RTP because you can't
>prove end-to-end security".  But such "proof" rarely exists even today.
>
>Today, when you do Email or IM over HTTPS or SMTPS, that email or IM
>only goes secure to the server you're doing SMTPS or HTTPS with.  You
>have no assurance whatsoever that it stays secure from that point on,
>nor that the server doesn't log the email/IM in its entirety.  But that
>doesn't mean there's no value in using HTTPS/SMTPS to reach that first
>server.  Quite the contrary, there's significant value in it.  Just like
>there's value in using HTTPS to reach a web store-front when purchasing
>using a credit-card, even though you have no idea what the security is
>between the web-store and your actual credit-card company.
>
>In the WebRTC context, for example, I would expect a quite-popular use-
>case for WebRTC to be Business websites, such as web-storefronts, or
>customer-support sites, or airline/hotel ticket/reservations/upgrades
>sites, etc.  When connecting to the web-page of those sites, I would
>expect them to be HTTPS once I got to some point of the click-flow
>process, if not from the beginning.  If I click a "talk to us for help"
>button, I would expect that to be secure as well, as much as the HTTPS
>portion was anyway.  In this context, only allowing them to use RTP is
>silly.  SDES is reasonably secure in this context, because there is no
>other domain the call is going to.  Yes, they have my SRTP key - doesn't
>matter; yes they can log my call - doesn't matter.  But at least my
>media is secure from my laptop to their server's domain - and that does
>matter, and it's all I ever got from using HTTPS.
>
>-hadriel
>
>
>On Mar 19, 2012, at 2:15 PM, Roman Shpount wrote:
>
>> I guess my question is, when are we going to tell the user that
>connection is "secure"? What I really do not want is making the
>distinction between secured and unsecured connection so muddy, that a
>normal user will never be able to understand the difference and will
>agree to anything. Because of this, I would like to see that if
>connection is over DTLS-SRTP with a remove party verified by an identity
>provider, user would be asked if they would like to communicate with
>remote party XYZ. If this is anything else, (DTLS-SRTP with no identity,
>SDES-SRTP, or even plain RTP), user should be asked either to allow
>unsecure communication with an unknown party, or, completely prohibited
>to do so.
>>
>> I do not see the difference, as far as security is concerned, between
>SDES-SRTP and plain RTP. If we allow SDES-SRTP for compatibility
>reasons, we might as well allow plain RTP. It would be compatible with
>more devices and as insecure as SDES-SRTP. There is no requirement that
>WebRTC application must use HTTPS for signaling. If the application is
>using HTTP, all the "sitting in the airport" examples are as unsecured
>with SDES-SRTP as they are with plain RTP.
>>
>> I do not think we would be able to force WebRTC application developers
>to create a secure application unless they would decide to create care
>and effort to develop it as such. Our goal should be to provide
>application developers with tools that would allow them to create secure
>applications and with mechanisms that would allow users to identify when
>their communications are actually secure. Everything else is just adding
>random requirements to the standard.
>> _____________
>> Roman Shpount
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb

From pravindran@sonusnet.com  Mon Mar 19 18:53:15 2012
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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: Hadriel Kaplan <HKaplan@acmepacket.com>, Roman Shpount <roman@telurix.com>, Eric Rescorla <ekr@rtfm.com>, Ted Hardie <ted.ietf@gmail.com>, Randell Jesup <randell-ietf@jesup.org>
Thread-Topic: RTP-IPSec vs SRTP-DTLS [was RE: [rtcweb] Resolving RTP/SDES question in Paris]
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Date: Tue, 20 Mar 2012 01:53:02 +0000
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: [rtcweb] RTP-IPSec vs SRTP-DTLS [was RE: Resolving RTP/SDES question in Paris]
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Hi all,

Could you please let me know the reason why SRTP-DTLS alone MUST be support=
ed by WebRTC web-browser and why not RTP-IPSec. In case my device support I=
PSec for all application, why web-browser has to mandated for double encryp=
tion. Another assumption here is that web-browser is endpoint (commercial b=
rowsers like chrome) which is not required as access entity (like Enterpris=
e access router, SBC) shall acts as web-browser. So, web-browser point-to-p=
oint connection may be between employee browser to Enterprise access entity=
 which is not required to be mandated as SRTP-DTLS. Hadriel mentioned large=
 set of call centre solution wherein web-browser is just the endpoint, one =
way to access call centre and there is no need to change whole call centre =
architecture for the same WebRTC technology in case RTP-IPSec is supported =
by WebRTC browser running in IPSec mobile phone. I disagree with security a=
dvocates that all call centre using RTP will be broken as this security sto=
ry does not sell well in the industry so far.

Also, It is impossible to have the common UI by all web-browser and IETF sp=
ecification should not try to find the way to solve that issue.=20

My problem is not mandatory to implement SRTP-DTLS but the issue is at mand=
atory to use. IMO, RTP-IPSec should be allowed with user-consent in web-bro=
wser and don't argue that all web user in the world are dumb. In case the u=
secase is missing for RTP-IPSec as Randell mentioned, let us discuss in IET=
F-83 RTCWeb meeting.

Thanks
Partha

>-----Original Message-----
>From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
>Of Hadriel Kaplan
>Sent: Tuesday, March 20, 2012 3:54 AM
>To: Roman Shpount
>Cc: rtcweb@ietf.org
>Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
>
>
>While I sympathize with the concern that requiring SRTP (regardless of
>how the key is learned) is a burden for those use-cases that literally
>don't care about SRTP and for which it does not matter, such as a
>weather announcement service or an IETF meeting broadcast, I really
>think you're 'throwing out the baby with the bath-water' to propose we
>either get everything or nothing.  People won't have the identity-stuff
>for some time, and it won't be painless to make it work and deployed.
>So what you're basically saying is "you only get RTP because you can't
>prove end-to-end security".  But such "proof" rarely exists even today.
>
>Today, when you do Email or IM over HTTPS or SMTPS, that email or IM
>only goes secure to the server you're doing SMTPS or HTTPS with.  You
>have no assurance whatsoever that it stays secure from that point on,
>nor that the server doesn't log the email/IM in its entirety.  But that
>doesn't mean there's no value in using HTTPS/SMTPS to reach that first
>server.  Quite the contrary, there's significant value in it.  Just like
>there's value in using HTTPS to reach a web store-front when purchasing
>using a credit-card, even though you have no idea what the security is
>between the web-store and your actual credit-card company.
>
>In the WebRTC context, for example, I would expect a quite-popular use-
>case for WebRTC to be Business websites, such as web-storefronts, or
>customer-support sites, or airline/hotel ticket/reservations/upgrades
>sites, etc.  When connecting to the web-page of those sites, I would
>expect them to be HTTPS once I got to some point of the click-flow
>process, if not from the beginning.  If I click a "talk to us for help"
>button, I would expect that to be secure as well, as much as the HTTPS
>portion was anyway.  In this context, only allowing them to use RTP is
>silly.  SDES is reasonably secure in this context, because there is no
>other domain the call is going to.  Yes, they have my SRTP key - doesn't
>matter; yes they can log my call - doesn't matter.  But at least my
>media is secure from my laptop to their server's domain - and that does
>matter, and it's all I ever got from using HTTPS.
>
>-hadriel
>
>
>On Mar 19, 2012, at 2:15 PM, Roman Shpount wrote:
>
>> I guess my question is, when are we going to tell the user that
>connection is "secure"? What I really do not want is making the
>distinction between secured and unsecured connection so muddy, that a
>normal user will never be able to understand the difference and will
>agree to anything. Because of this, I would like to see that if
>connection is over DTLS-SRTP with a remove party verified by an identity
>provider, user would be asked if they would like to communicate with
>remote party XYZ. If this is anything else, (DTLS-SRTP with no identity,
>SDES-SRTP, or even plain RTP), user should be asked either to allow
>unsecure communication with an unknown party, or, completely prohibited
>to do so.
>>
>> I do not see the difference, as far as security is concerned, between
>SDES-SRTP and plain RTP. If we allow SDES-SRTP for compatibility
>reasons, we might as well allow plain RTP. It would be compatible with
>more devices and as insecure as SDES-SRTP. There is no requirement that
>WebRTC application must use HTTPS for signaling. If the application is
>using HTTP, all the "sitting in the airport" examples are as unsecured
>with SDES-SRTP as they are with plain RTP.
>>
>> I do not think we would be able to force WebRTC application developers
>to create a secure application unless they would decide to create care
>and effort to develop it as such. Our goal should be to provide
>application developers with tools that would allow them to create secure
>applications and with mechanisms that would allow users to identify when
>their communications are actually secure. Everything else is just adding
>random requirements to the standard.
>> _____________
>> Roman Shpount
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb

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Date: Mon, 19 Mar 2012 22:29:15 -0400
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From: Roman Shpount <roman@telurix.com>
To: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
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Cc: Randell Jesup <randell-ietf@jesup.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] RTP-IPSec vs SRTP-DTLS [was RE: Resolving RTP/SDES question in Paris]
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--047d7b10cde3f052bd04bba370f9
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The reason is quite simple actually -- it is quite difficult to confirm
that IPSec is actually present. So browser has no way to determine that it
communicating over IPSec or plain IP.

On the other hand, I actually agree that if we are dealing with an
application on a private IP (like 10.X.X.X) you should be allowed to do
whatever you want. This is why I am proposing to support RTP for
applications delivered over HTTP. If you are in completely secure
environment and you do not want to do encryption, there is no reason why
the application should be delivered over HTTPS. If application is delivered
over HTTP, you either do no care about security (you are calling a public
radio) or security is provided by something else (you are sitting in the
bunker on the network not connected to the internet).

The large contingent of people on this list are convinced that upsides
(security) of requiring encryption outweigh downsides. The mentioned
downsides are:
1. Extra resource requirements. This is not really important as far as I am
concerned. New Xeon server can encrypt 39.7 GB/sec (
http://www.theregister.co.uk/2012/03/06/intel_xeon_e5_launch_event/)
2. Support for legacy devices. Also not as important since most legacy
devices will require a media proxy to support ICE anyway
3. Barrier to entry to develop new services. It is harder to debug and
develop new products if you need to develop larger number of protocols at
once. The understanding is that web browser will include some sort of
hidden setting that will allow plain RTP for debugging and development.
4. Encryption can be illegal or not allowed in some places, such as
military or prisons. Proposal is to enable some sort of configuration in
the web browser that will force it to send all the media through some sort
of recording/monitoring proxy.

While, as far as I am concerned, all of those arguments are addressed, my
main argument against requiring encryption is that there are scenarios when
it is not needed and not actually provided, primarily when application is
delivered over HTTP. All I am saying that if application developer does not
care about security, the application most likely would not be secure,
regardless of what we do. So why force it? Unless developer takes an effort
to buy an SSL certificate and deploy an app on a secure server, or gets the
certificate and sets up an identity provider, communication is not going to
be secure. If I am a high school kid trying to play with real time
communications using a free web hosting provider, I do not care about
security. All I care is that I can write something that makes a call. What
we are effectively doing is making this much more difficult. Innovation
does not have to come from serious projects created for serious purposes.
It often comes from things that are cobbled together and I want us to
enable this, unless it actually breaks something. I want to make WebRTC
easier to play with and hope that new exciting things will come out of this.
_____________
Roman Shpount


On Mon, Mar 19, 2012 at 9:53 PM, Ravindran, Parthasarathi <
pravindran@sonusnet.com> wrote:

> Hi all,
>
> Could you please let me know the reason why SRTP-DTLS alone MUST be
> supported by WebRTC web-browser and why not RTP-IPSec. In case my device
> support IPSec for all application, why web-browser has to mandated for
> double encryption. Another assumption here is that web-browser is endpoint
> (commercial browsers like chrome) which is not required as access entity
> (like Enterprise access router, SBC) shall acts as web-browser. So,
> web-browser point-to-point connection may be between employee browser to
> Enterprise access entity which is not required to be mandated as SRTP-DTLS.
> Hadriel mentioned large set of call centre solution wherein web-browser is
> just the endpoint, one way to access call centre and there is no need to
> change whole call centre architecture for the same WebRTC technology in
> case RTP-IPSec is supported by WebRTC browser running in IPSec mobile
> phone. I disagree with security advocates that all call centre using RTP
> will be broken as this security story does not sell well in the industry so
> far.
>
> Also, It is impossible to have the common UI by all web-browser and IETF
> specification should not try to find the way to solve that issue.
>
> My problem is not mandatory to implement SRTP-DTLS but the issue is at
> mandatory to use. IMO, RTP-IPSec should be allowed with user-consent in
> web-browser and don't argue that all web user in the world are dumb. In
> case the usecase is missing for RTP-IPSec as Randell mentioned, let us
> discuss in IETF-83 RTCWeb meeting.
>
> Thanks
> Partha
>

--047d7b10cde3f052bd04bba370f9
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

The reason is quite simple actually -- it is quite difficult to confirm tha=
t IPSec is actually present. So browser has no way to determine that it com=
municating over IPSec or plain IP.<br><br>On the other hand, I actually agr=
ee that if we are dealing with an application on a private IP (like 10.X.X.=
X) you should be allowed to do whatever you want. This is why I am proposin=
g to support RTP for applications delivered over HTTP. If you are in comple=
tely secure environment and you do not want to do encryption, there is no r=
eason why the application should be delivered over HTTPS. If application is=
 delivered over HTTP, you either do no care about security (you are calling=
 a public radio) or security is provided by something else (you are sitting=
 in the bunker on the network not connected to the internet).<br>
<br>The large contingent of people on this list are convinced that upsides =
(security) of requiring encryption outweigh downsides. The mentioned downsi=
des are:<br>1. Extra resource requirements. This is not really important as=
 far as I am concerned. New Xeon server can encrypt 39.7 GB/sec (<a href=3D=
"http://www.theregister.co.uk/2012/03/06/intel_xeon_e5_launch_event/">http:=
//www.theregister.co.uk/2012/03/06/intel_xeon_e5_launch_event/</a>)<br>
2. Support for legacy devices. Also not as important since most legacy devi=
ces will require a media proxy to support ICE anyway<br>3. Barrier to entry=
 to develop new services. It is harder to debug and develop new products if=
 you need to develop larger number of protocols at once. The understanding =
is that web browser will include some sort of hidden setting that will allo=
w plain RTP for debugging and development.<br>
4. Encryption can be illegal or not allowed in some places, such as militar=
y or prisons. Proposal is to enable some sort of configuration in the web b=
rowser that will force it to send all the media through some sort of record=
ing/monitoring proxy.<br>
<br>While, as far as I am concerned, all of those arguments are addressed, =
my main argument against requiring encryption is that there are scenarios w=
hen it is not needed and not actually provided, primarily when application =
is delivered over HTTP. All I am saying that if application developer does =
not care about security, the application most likely would not be secure, r=
egardless of what we do. So why force it? Unless developer takes an effort =
to buy an SSL certificate and deploy an app on a secure server, or gets the=
 certificate and sets up an identity provider, communication is not going t=
o be secure. If I am a high school kid trying to play with real time commun=
ications using a free web hosting provider, I do not care about security. A=
ll I care is that I can write something that makes a call. What we are effe=
ctively doing is making this much more difficult. Innovation does not have =
to come from serious projects created for serious purposes. It often comes =
from things that are cobbled together and I want us to enable this, unless =
it actually breaks something. I want to make WebRTC easier to play with and=
 hope that new exciting things will come out of this.<br clear=3D"all">
_____________<br>Roman Shpount<br>
<br><br><div class=3D"gmail_quote">On Mon, Mar 19, 2012 at 9:53 PM, Ravindr=
an, Parthasarathi <span dir=3D"ltr">&lt;<a href=3D"mailto:pravindran@sonusn=
et.com">pravindran@sonusnet.com</a>&gt;</span> wrote:<br><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex">
Hi all,<br>
<br>
Could you please let me know the reason why SRTP-DTLS alone MUST be support=
ed by WebRTC web-browser and why not RTP-IPSec. In case my device support I=
PSec for all application, why web-browser has to mandated for double encryp=
tion. Another assumption here is that web-browser is endpoint (commercial b=
rowsers like chrome) which is not required as access entity (like Enterpris=
e access router, SBC) shall acts as web-browser. So, web-browser point-to-p=
oint connection may be between employee browser to Enterprise access entity=
 which is not required to be mandated as SRTP-DTLS. Hadriel mentioned large=
 set of call centre solution wherein web-browser is just the endpoint, one =
way to access call centre and there is no need to change whole call centre =
architecture for the same WebRTC technology in case RTP-IPSec is supported =
by WebRTC browser running in IPSec mobile phone. I disagree with security a=
dvocates that all call centre using RTP will be broken as this security sto=
ry does not sell well in the industry so far.<br>

<br>
Also, It is impossible to have the common UI by all web-browser and IETF sp=
ecification should not try to find the way to solve that issue.<br>
<br>
My problem is not mandatory to implement SRTP-DTLS but the issue is at mand=
atory to use. IMO, RTP-IPSec should be allowed with user-consent in web-bro=
wser and don&#39;t argue that all web user in the world are dumb. In case t=
he usecase is missing for RTP-IPSec as Randell mentioned, let us discuss in=
 IETF-83 RTCWeb meeting.<br>

<br>
Thanks<br>
Partha<br></blockquote></div><br>

--047d7b10cde3f052bd04bba370f9--

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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
Thread-Topic: [rtcweb] Resolving RTP/SDES question in Paris
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Date: Tue, 20 Mar 2012 03:20:46 +0000
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References: <4F4759DC.7060303@ericsson.com><387F9047F55E8C42850AD6B3A7A03C6C0E1FEB69@inba-mail01.sonusnet.com><CALiegfnkYVEpmPV-zSL_4wOY-HiFZN-qJCQCiioaS=5NaqhLZw@mail.gmail.com><CAD5OKxvtOAxMBx6xDnyfTnEq76oDEm6uj1xL6wGjjrtKUAHy3g@mail.gmail.com><CABcZeBNZiotPmCfT53uEo+O0xw4xv6tXW1M_G-3A5BHuncsduA@mail.gmail.com><CAD5OKxvYOY5JZ2mYNGiH1poUBQkyOOycePFijH5H+SxtcdqujQ@mail.gmail.com><CABkgnnVe-b6Sv=R67bMJk_NQqQwdrRUn6rBm7Gu_CMcfPQwtEg@mail.gmail.com><CAD5OKxvZbEJ7sV4WPAYoQapzMR_QwAftj-oKg=ioMKHNT792wQ@mail.gmail.com><6F428EFD2B8C2F49A2FB1317291A76C113563C5A92@USNAVSXCHMBSA1.ndc.alcatel-lucent.com><CALiegf=jtkDCS_D0ZFe9UpbiadQ0vsJ+4MppQSbLr-wbaXNrfQ@mail.gmail.com><BLU169-W29E5B86F9E2C6F3126961C93420@phx.gbl><CALiegfk2aT+6Psr4nT-hG1G7eYRBfFCcT+25On2O4HfUXJ6-ng@mail.gmail.com><CAD6AjGSmi9j+sdGWPts20-iwGvGij05ek0OKYEPULC6B=aFpQg@mail.gmail.com><6F428EFD2B8C2F49A2FB1317291A76C113564482A7@USNAVSXCHMBSA1.ndc.alcatel-lucent.com><ADBB75F3-E20C-4EC4-B9C3-EF2E4BFF409C@phonefromhere.com><C! AD5OKxvuE V8Vbq3h7=Zgc KmREjmguvz5n-SpXr2n-EY7a_ddxg@mail.gmail.com><CALiegfk1ozOKPcDjbd3H_z2Edzh4RcZpYyJSWdw_1DJ04muQXA@mail.gmail.com><CAD5OKxu8-+0O0=eE7mD1hi=nPUpEXczGj=bRNQCQL1BW8c-c-Q@mail.gmail.com> <D75A384B-0F38-4E30-8C03-12E903A69B64@acmepacket.com> <E17CAD772E76C742B645BD4DC602CD8105EBE8CF@NAHALD.us.int.genesyslab.com> <387F9047F55E8C42850AD6B3A7A03C6C0E1FFE23@inba-mail01.sonusnet.com>
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And, it should be noted, they *want* it that way.  They don't generally wan=
t you to know which agent picked up the call, or that the call got transfer=
red from an IVR to a specific agent, or got put in a queue with elevator-mu=
sic announcement server, or any of that.  You "called" Travel Co., and they=
 answered.  What goes on behind that SBC is equivalent to the back-end data=
base stuff that goes on between the web server and back-end systems, while =
all you see is the web-server your browser happens to be connected to. =20

The "identity" you'll get is www.travel.co, which you already had when your=
 browser did HTTPS cert verification of their web-server.  HTTPS became the=
 key-exchange transport for the SRTP key.  Since they already proved they o=
wn the cert for www.travel.co, having them claim to be www.travel.co should=
n't require yet more verification.

-hadriel


On Mar 19, 2012, at 9:15 PM, Ravindran, Parthasarathi wrote:

> Jim,
>=20
> As a customer, you won't really know whether the identity (DTLS-SRTP) of =
call center/travel site is agent or SBC. SBC can perform MITM attack easily=
 as extending SDES-SRTP to DTLS-SRTP for call center site is feasible.
>=20
> Thanks
> Partha


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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: Bernard Aboba <bernard_aboba@hotmail.com>, "ibc@aliax.net" <ibc@aliax.net>
Thread-Topic: [rtcweb] Resolving RTP/SDES question in Paris
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Bernard,

In case you expect SRTP-DTLS implementation will be prevalent during RTCWeb=
 specification publication, I seriously doubt it.

Thanks
Partha

From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Bernard Aboba
Sent: Monday, March 19, 2012 5:44 PM
To: ibc@aliax.net
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris

> Which ones? I just see one: interoperability with legacy SIP devices.

[BA] Agreed.  And as we've already seen, SDES/SRTP support is prevalent eno=
ugh among legacy devices that the legacy use case does not require RTP.

> Even if you use an VPN for your enterprise WebRTC application there is NO=
 problem at all in using SRTP
> over the VPN.

[BA] And in fact, this is exactly the NSA "Secure VoIP" architecture (with =
SDES/SRTP, by the way).

> I can make business with my legacy and not secure SIP devices,
> those that don't implement SRTP regardless SRTP was designed for SIP".

[BA] At this point, support for SRTP is an expected feature on legacy equip=
ment.  For
example, all the leading PSTN gateway vendors support SRTP already.  By the=
 time
RTCWEB specs are final, SRTP support will be very prevalent.









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<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Bernard,<o:p></o:p></span=
></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">In case you expect SRTP-D=
TLS implementation will be prevalent during RTCWeb specification publicatio=
n, I seriously doubt it.
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Thanks<o:p></o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Partha<o:p></o:p></span><=
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<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
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<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> rtcweb-b=
ounces@ietf.org [mailto:rtcweb-bounces@ietf.org]
<b>On Behalf Of </b>Bernard Aboba<br>
<b>Sent:</b> Monday, March 19, 2012 5:44 PM<br>
<b>To:</b> ibc@aliax.net<br>
<b>Cc:</b> rtcweb@ietf.org<br>
<b>Subject:</b> Re: [rtcweb] Resolving RTP/SDES question in Paris<o:p></o:p=
></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ta=
homa&quot;,&quot;sans-serif&quot;">&gt; Which ones? I just see one: interop=
erability with legacy SIP devices.<br>
<br>
[BA] Agreed.&nbsp; And as we've already seen, SDES/SRTP support is prevalen=
t enough among legacy devices that the legacy use case does not require RTP=
.
<br>
<br>
&gt; Even if you use an VPN for your enterprise WebRTC application there is=
 NO problem at all in using SRTP<o:p></o:p></span></p>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">&gt; over=
 the VPN.<br>
<br>
[BA] And in fact, this is exactly the NSA &quot;Secure VoIP&quot; architect=
ure (with SDES/SRTP, by the way).
<br>
<br>
&gt; I can make business with my legacy and not secure SIP devices,<br>
&gt; those that don't implement SRTP regardless SRTP was designed for SIP&q=
uot;.<br>
<br>
[BA] At this point, support for SRTP is an expected feature on legacy equip=
ment.&nbsp; For<br>
example, all the leading PSTN gateway vendors support SRTP already.&nbsp; B=
y the time<br>
RTCWEB specs are final, SRTP support will be very prevalent. <br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<o:p></o:p></span></p>
</div>
</div>
</div>
</div>
</body>
</html>

--_000_387F9047F55E8C42850AD6B3A7A03C6C0E2000F9inbamail01sonus_--

From salvatore.loreto@ericsson.com  Tue Mar 20 01:55:42 2012
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Subject: Re: [rtcweb] ICE and draft-ietf-mmusic-sctp-sdp
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Hi Cullen,

On 3/19/12 3:00 PM, Cullen Jennings wrote:
> I like draft-ietf-mmusic-sctp-sdp but I'd like to see it have more stuff in it. Specifically I would like it to
>
> 1) clearly define how to do the SDP for use of SCTP over DTLS over UPD with ICE. This is the use case we need to solve
right,
in Quebec we decided not to specify the ICE usage for *plain* SCTP in 
this draft...
but for sure we have to define the usage of SCTP over DTLS with ICE in 
the draft

>
> 2) make sure it works with bundle
+1

>
> 3) SCTP has sub addresses (SCTP Channels), I think these need to be negotiated as well in some cases so I would like to see an optional way of negotiating and signaling the channels including correlation with some sort of application defined label

yep,  we can define a way to signal/negotiate the number of channels we 
want initially in the association vs having this number configured;
and eventually even go further e.g. negotiate the addition of channels, 
and perhaps some sort of correlation of a channel
with an application defined label


Salvatore

-- 
Salvatore Loreto, PhD
www.sloreto.com


From andrew.hutton@siemens-enterprise.com  Tue Mar 20 03:06:55 2012
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From: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>
To: Hadriel Kaplan <HKaplan@acmepacket.com>, "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
Date: Tue, 20 Mar 2012 11:06:51 +0100
Thread-Topic: [rtcweb] Resolving RTP/SDES question in Paris
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References: <4F4759DC.7060303@ericsson.com><387F9047F55E8C42850AD6B3A7A03C6C0E1FEB69@inba-mail01.sonusnet.com><CALiegfnkYVEpmPV-zSL_4wOY-HiFZN-qJCQCiioaS=5NaqhLZw@mail.gmail.com><CAD5OKxvtOAxMBx6xDnyfTnEq76oDEm6uj1xL6wGjjrtKUAHy3g@mail.gmail.com><CABcZeBNZiotPmCfT53uEo+O0xw4xv6tXW1M_G-3A5BHuncsduA@mail.gmail.com><CAD5OKxvYOY5JZ2mYNGiH1poUBQkyOOycePFijH5H+SxtcdqujQ@mail.gmail.com><CABkgnnVe-b6Sv=R67bMJk_NQqQwdrRUn6rBm7Gu_CMcfPQwtEg@mail.gmail.com><CAD5OKxvZbEJ7sV4WPAYoQapzMR_QwAftj-oKg=ioMKHNT792wQ@mail.gmail.com><6F428EFD2B8C2F49A2FB1317291A76C113563C5A92@USNAVSXCHMBSA1.ndc.alcatel-lucent.com><CALiegf=jtkDCS_D0ZFe9UpbiadQ0vsJ+4MppQSbLr-wbaXNrfQ@mail.gmail.com><BLU169-W29E5B86F9E2C6F3126961C93420@phx.gbl><CALiegfk2aT+6Psr4nT-hG1G7eYRBfFCcT+25On2O4HfUXJ6-ng@mail.gmail.com><CAD6AjGSmi9j+sdGWPts20-iwGvGij05ek0OKYEPULC6B=aFpQg@mail.gmail.com><6F428EFD2B8C2F49A2FB1317291A76C113564482A7@USNAVSXCHMBSA1.ndc.alcatel-lucent.com><ADBB75F3-E20C-4EC4-B9C3-EF2E4BFF409C@phonefromhere.com><C! AD5OKxvuE V8Vbq3h7=Zgc KmREjmguvz5n-SpXr2n-EY7a_ddxg@mail.gmail.com><CALiegfk1ozOKPcDjbd3H_z2Edzh4RcZpYyJSWdw_1DJ04muQXA@mail.gmail.com><CAD5OKxu8-+0O0=eE7mD1hi=nPUpEXczGj=bRNQCQL1BW8c-c-Q@mail.gmail.com> <D75A384B-0F38-4E30-8C03-12E903A69B64@acmepacket.com> <E17CAD772E76C742B645BD4DC602CD8105EBE8CF@NAHALD.us.int.genesyslab.com> <387F9047F55E8C42850AD6B3A7A03C6C0E1FFE23@inba-mail01.sonusnet.com> <2E10EB15-7E2E-47B9-80D1-5244DDE5FDF7@acmepacket.com>
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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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Hi,

Completely agree with Hadriel's statement below.

In addition to wanting to know that you are connected to "www.travel.co" or=
 "www.mybank.co" you as a consumer and the company really want/need the med=
ia to be recorded by the companies system for your own protection. This is =
going to involve some kind of MITM or as we call it in SIPREC a Session Rec=
ording Client (SRC) to be able to intercept the media and route it to a Ses=
sion Recording Server (SRS).=20

The agent's desktop device could act as the SRC but more likely it will be =
some kind of B2BUA that does this. =20

In these scenario's you as a consumer using this service really want the me=
dia to be secure at least as far as reaching the bank but after that like a=
ny existing web based system you have to trust the bank with your data this=
 is no different whether that data is speech or text that you entered on a =
web page.

The recording system should be able to work with DTLS-SRTP but it will most=
 likely act as a MITM so will change the DTLS fingerprint. I don't see that=
 as a problem.

Regards
Andy



> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Hadriel Kaplan
> Sent: 20 March 2012 03:21
> To: Ravindran, Parthasarathi
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
>=20
>=20
> And, it should be noted, they *want* it that way.  They don't generally
> want you to know which agent picked up the call, or that the call got
> transferred from an IVR to a specific agent, or got put in a queue with
> elevator-music announcement server, or any of that.  You "called"
> Travel Co., and they answered.  What goes on behind that SBC is
> equivalent to the back-end database stuff that goes on between the web
> server and back-end systems, while all you see is the web-server your
> browser happens to be connected to.
>=20
> The "identity" you'll get is www.travel.co, which you already had when
> your browser did HTTPS cert verification of their web-server.  HTTPS
> became the key-exchange transport for the SRTP key.  Since they already
> proved they own the cert for www.travel.co, having them claim to be
> www.travel.co shouldn't require yet more verification.
>=20
> -hadriel
>=20
>=20
> On Mar 19, 2012, at 9:15 PM, Ravindran, Parthasarathi wrote:
>=20
> > Jim,
> >
> > As a customer, you won't really know whether the identity (DTLS-SRTP)
> of call center/travel site is agent or SBC. SBC can perform MITM attack
> easily as extending SDES-SRTP to DTLS-SRTP for call center site is
> feasible.
> >
> > Thanks
> > Partha
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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On 03/20/2012 02:06 AM, Ravindran, Parthasarathi wrote:
> Eric,
>
> Sec 4.2.3 mentions about RTCP consent approach is not adopted for
>
> 1) Small window attack of RTP
> 2) Tough for legacy non-RTCP endpoint to implement RTCP
>
> I think that small window attack of RTP is possible in ICE as well in case attacker has access to offer SDP and the second reason is funny to recommend non-RTCP endpoint to implement ICE instead of implementing RTCP because this reason is equivalent of asking the person to buy cake who is struggling to buy bread.
>
> The missing information is that NAT traversal is not possible using RTCP mechanism. Could you please add "NAT traversal" as the one of the reason for not adopting RTCP consent approach.
Ravindran,

the section already limits itself to "... cases where the legacy 
endpoint has a public address...." - in those cases, NAT traversal is 
not necessary, so I don't believe your comment applies.

Can you explain why you see a window of attack in the ICE case? I don't 
understand that comment.


> Thanks
> Partha
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] SDES vs DTLS-SRTP revisited
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On 03/17/2012 04:35 AM, Hadriel Kaplan wrote:
> On Mar 16, 2012, at 6:32 PM, Eric Rescorla wrote:
>> Yes, I think the fingerprint mismatch in case 1 serves as a stronger indication
>> that something was wrong, because there is no technical reason for the
>> fingerprint not to match if both sides believe they are doing DTLS. I.e.,
>> it always indicates either a software defect or a man in the middle.
>> And the suspicious user of course should hang up, but they should also
>> post pictures to /., which, I would think, serves
>> as a pretty significant deterrent to providers tampering with people's calls!
>> By contrast, with SDES, the attack scenario and the legitimate scenario
>> look the same.
> I don't think you can jump to the conclusion that there's either a software defect or a MitM.  Imagine if WebRTC user Alice calls Bob through provider-A, using Bob's telephone number.  Provider-A supports DTLS-SRTP on its PSTN gateway, so it handles that and generates a call into the PSTN for Bob, who happens to use Provider-B.  Bob is currently on a WebRTC client connected to Provider-B, and Provider-B happens to also support DTLS-SRTP on its PSTN gateway.  So both Alice and Bob see "DTLS-SRTP", but different fingerprints.  It's not a software defect.  You could argue it's a MitM (the PSTN gateways doing what they do), and certainly there may *be* someone recording the call in the middle for all you know - but you really don't know either way.  Alice dialed Bob, and got the best case scenario given the situation.
I don't get this scenario. If Alice calls Bob using two different 
gateways, won't she go through a credentials and fingerprint exchange 
with the gateways?
In that case, wouldn't the fingerprint belong to the gateway?
> The good news is the media is secure from the browsers to the PSTN, which is not an insignificant accomplishment.  If we're worried about WiFi - and I am, for at least marketing/perception reasons if nothing else - then we've alleviated that concern.
>
> So then there's slashdot.  If Alice or Bob posts to slashdot, it's bad news.  But we can hardly blame Provider-A or Provider-B, since they had no alternative.
It wouldn't be the first time someone posted a story on slashdot that 
got the facts wrong...
>    Ironically, if WebRTC supported SDES, then Provider-A and B could do that instead, and make it clear/explicit no end-to-end guarantee can be made for the call - while *still* avoiding the main concern, which is the WiFi problem.  In some sense *we*, the IETF/W3C, would be more honest if we allowed SDES (over HTTPS) for gateway cases, since that's actually the level of security being provided.  Mandating DTLS-SRTP provides a false sense of security, and false sense of insecurity if it fails.
>
> -hadriel
>
> _______________________________________________
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> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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On 03/17/2012 10:14 PM, Igor Faynberg wrote:
> ..
>
> On 3/17/2012 4:45 PM, Martin Thomson wrote:
>> ... Then I explicitly place a trust anchor for that
>> certificate in your browser.
>
> How?  I thought the browser would never allow that to happen. (I 
> assumed it would come provisioned with anchors,  or would allow anchor 
> provision through some different channel.)
>
Igor,

in Chrome, go to chrome://settings/certificates, choose the 
"authorities" tab, and note the presence of the "import" button.

All browsers (as far as I know) have similar mechanisms.
"Owning" your computer will achieve this goal both in its corporate 
meaning and in its hacker meaning.



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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
  <head>
    <meta content="text/html; charset=ISO-8859-1"
      http-equiv="Content-Type">
  </head>
  <body bgcolor="#ffffff" text="#000000">
    On 03/17/2012 10:14 PM, Igor Faynberg wrote:
    <blockquote cite="mid:4F64FE98.3070605@alcatel-lucent.com"
      type="cite">..
      <br>
      <br>
      On 3/17/2012 4:45 PM, Martin Thomson wrote:
      <br>
      <blockquote type="cite">... Then I explicitly place a trust anchor
        for that
        <br>
        certificate in your browser.
        <br>
      </blockquote>
      <br>
      How?&nbsp; I thought the browser would never allow that to happen. (I
      assumed it would come provisioned with anchors,&nbsp; or would allow
      anchor provision through some different channel.)
      <br>
      <br>
    </blockquote>
    Igor,<br>
    <br>
    in Chrome, go to <a>chrome://settings/certificates, choose the
      "authorities" tab, and note the presence of the "import" button.<br>
      <br>
      All browsers (as far as I know) have similar mechanisms.<br>
      "Owning" your computer will achieve this goal both in its
      corporate meaning and in its hacker meaning.<br>
      <br>
    </a><br>
  </body>
</html>

--------------020905000402050109040802--

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Subject: [rtcweb] Telling the user the connection is secure (Re: Resolving RTP/SDES question in Paris)
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I believe I have said this before, but...

We should never tell the user the connection is secure.
We should tell the user when we know he's exposed to risks that he 
usually isn't.

Thus - we should not give any indication that we're using DTLS-SRTP with 
verified identities (if that's what we normally support). We SHOULD give 
a warning saying "hey, since the gateway you've connected to isn't doing 
normal authentication procedures, but instead insists on exchanging keys 
on the signalling channel, you are less sure who you're talking to than 
usual, and there are more boxes that might record your call in the way, 
but the script kiddie on your hotel WLAN still can't see your packets 
(translation: legacy SDES key exchange is in use, but SRTP is still on).

All this will of course be iconified into a single cryptic graphic 
probably involving a padlock :-)

On 03/19/2012 07:47 PM, Igor Faynberg wrote:
> This is the question that I have been asking for a while...  I don't 
> expect a complete fireproof answer, of course, and I also understand 
> that the browser today is telling me a few things about the security 
> of a site and warns me when "the site is trying to access the data it 
> should not be accessing."
>
> But I  also imagine that a rogue site could display a message 
> mimicking the security assurance as though it comes from the browser.
>
> So it would be good to have a very clear idea when  the determination 
> about the security of the connection and such is made and how the end 
> user can verify that it actually comes from the browser.
>
> (To this end, the user MUST trust the browser, of course.)
>
> Igor
>
> On 3/19/2012 2:15 PM, Roman Shpount wrote:
>> I guess my question is, when are we going to tell the user that 
>> connection is "secure"?...
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


From stefan.lk.hakansson@ericsson.com  Tue Mar 20 03:58:31 2012
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Cc: "Banerjee, Atin" <Atin.Banerjee@Polycom.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Use-case document  - additional recording use case
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Hi Ranjit,

recording was discussed at length last fall (and is to some extent 
captured in the document) without really concluding.

This fact, and the fact that there was really no support for your 
proposal makes me think we should not add this use-case (at least not now).

Br,
Stefan

On 03/07/2012 07:07 AM, Avasarala, Ranjit wrote:
> Hi Stefan
>
> Could we add a use case related to session recording? The use case would be
>
> 4.2. x Simple Video Recording Service
>
> 4.2. x.1 Description
>
> The user who has joined a video communication service wants to record the communication data from the web browser.
>
> 4.2.x.1.1  Derived requirements
>
> TBD
>
> I also feel there is a need for a API requirement - the Web API should provide a means to record the incoming media - both audio&  video or audio only or video only
>
> Regards
> Ranjit
>
> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of Stefan Hakansson LK
> Sent: Tuesday, March 06, 2012 2:59 PM
> To: rtcweb@ietf.org
> Subject: [rtcweb] Use-case document
>
> Hi!
>
> The Use-case document
> (<http://datatracker.ietf.org/doc/draft-ietf-rtcweb-use-cases-and-requirements/?include_text=1>)
> will expire in early April.
>
> The editors propose to add a new use-case when updating to -07. The new use-case is about large sessions and is inspired by input made earlier by Arno Bakker
> (http://www.ietf.org/mail-archive/web/rtcweb/current/msg00530.html) and recently by Harald Alvestrand (http://lists.w3.org/Archives/Public/public-webrtc/2012Feb/0066.html).
>
> Proposed text for the new use-case:
>
> On-line lecture
> ===============
>
> A lecture is given on-line by a very popular lecturer. The attendance is very large. Because of this it is not really feasible to have direct audiovisual feedback (for e.g. questions) for every participant, instead participants can request the floor by clicking a button, and gets a signal when audio and video has been set up (and the lecturer is ready to take the question). The question is heard by all participants (not only the lecturer).
>
> Participants can join and leave the lecture at any time (i.e. they do not need to be there at the start to be able to connect).
>
> The service is operated by the university of the lecturer. It is set up such that all participants connect to a central node. The university can not invest much in HW, so the processing required (per participant) must be minimized.
>
>
> New requirement:
>
> - It must be possible to set up media streams and encryption in such a way that processing in a central node is minimized (no transcoding required, no decryption/re-encryption for every outgoing flow - just simple forwarding).
>
>
> Let us know what you think. If there is support we will add this use-case, if not we will just make a new revision of the document (with no major changes to the content).
>
> Stefan for the editors
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From stefan.lk.hakansson@ericsson.com  Tue Mar 20 03:59:52 2012
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Subject: Re: [rtcweb] Use-case document
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On 03/08/2012 04:12 PM, Stefan Hakansson LK wrote:
> On 03/06/2012 02:01 PM, Eric Rescorla wrote:
>> Obviously, this is an interesting use case, but it's not clear to me that
>> it's something you would want to address entirely via WebRTC
>>
>> 1. It seems likely that the lecturer will be using something
>> rather more sophisticated than a Web browser in many cases
>> (often these lecturers are actually joint on-line and in-person
>> so you need microphone control in the room as well; see for
>> instance the popular Sandel Justice lecturers out of Harvard).
>>
>> 2. You also probably want to actually support not just real-time
>> viewing but also podcasting, time-shifting, etc., as well as
>> support for people who don't have WebRTC.
>>
>> When you put these together, it might well be natural to have
>> a one-way stream via Flash, HTML5 video, etc., and then individual
>> WebRTC uplinks which get mixed into that single stream as necessary.
>> That makes questions about transcoding, re-encryption, etc. a lot
>> simpler.
>
> I agree to that the use-case proposed was perhaps not that well selected
> for the reasons you give above.
>
> What we were really after was to enable multi-party with really low cost
> for the central node (given that you're not doing a mesh). If you have
> to decrypt, alter RTP parameters, and then re-encrypt you quickly get
> into a situation where the central node needs a lot of processing power.
>
> If the packets simply can be forwarded you're much better off - and this
> is what we wanted to capture. It is also a way to in the use-case and
> requirement document reflect a bit of what JSEP enables; if the
> application can *set* certain parts of the media settings used by the
> browser, the app can set it up such that central node would not have to
> alter anything - just forward. We thought it would be nice to have a
> JSEP related req in the document!
>
> Stefan

I hear no support for this, and will not add it to the use-case document 
at this stage.

Stefan

>
>>
>> -Ekr
>>
>>
>> On Tue, Mar 6, 2012 at 1:28 AM, Stefan Hakansson LK
>> <stefan.lk.hakansson@ericsson.com>   wrote:
>>> Hi!
>>>
>>> The Use-case document
>>> (<http://datatracker.ietf.org/doc/draft-ietf-rtcweb-use-cases-and-requirements/?include_text=1>)
>>> will expire in early April.
>>>
>>> The editors propose to add a new use-case when updating to -07. The new
>>> use-case is about large sessions and is inspired by input made earlier
>>> by Arno Bakker
>>> (http://www.ietf.org/mail-archive/web/rtcweb/current/msg00530.html) and
>>> recently by Harald Alvestrand
>>> (http://lists.w3.org/Archives/Public/public-webrtc/2012Feb/0066.html).
>>>
>>> Proposed text for the new use-case:
>>>
>>> On-line lecture
>>> ===============
>>>
>>> A lecture is given on-line by a very popular lecturer. The attendance is
>>> very large. Because of this it is not really feasible to have
>>> direct audiovisual feedback (for e.g. questions) for every participant,
>>> instead participants can request the floor by clicking a button, and
>>> gets a signal when audio and video has been set up (and the lecturer is
>>> ready to take the question). The question is heard by all participants
>>> (not only the lecturer).
>>>
>>> Participants can join and leave the lecture at any time (i.e. they do
>>> not need to be there at the start to be able to connect).
>>>
>>> The service is operated by the university of the lecturer. It is set up such
>>> that all participants connect to a central node. The university can not
>>> invest much in HW, so the processing required (per participant) must be
>>> minimized.
>>>
>>>
>>> New requirement:
>>>
>>> - It must be possible to set up media streams and encryption in such a
>>> way that processing in a central node is minimized (no transcoding
>>> required, no decryption/re-encryption for every outgoing flow - just
>>> simple forwarding).
>>>
>>>
>>> Let us know what you think. If there is support we will add this use-case,
>>> if not we will just make a new revision of the document (with no major
>>> changes to the content).
>>>
>>> Stefan for the editors
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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Any said:=20

> The recording system should be able to work with DTLS-SRTP but it will mo=
st likely act as a MITM so will change the DTLS fingerprint. I don't see th=
at as a problem.

[BA] One of the questions I have had about the usage of DTLS-SRTP contempla=
ted in RTCWEB is whether it will be interoperable with the SIP usage of DTL=
S-SRTP. =20

In the SIP framework it is possible to do RFC 4474 re-signing in the B2BUA =
if the destination is an E.164 number.=20

However=2C the DTLS-SRTP/IdP framework seems to assume that the identities =
are email-style addresses. =20


 		 	   		  =

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<body class=3D'hmmessage'><div dir=3D'ltr'>
Any said: <br><br><div>&gt=3B The recording system should be able to work w=
ith DTLS-SRTP but it will most likely act as a MITM so will change the DTLS=
 fingerprint. I don't see that as a problem.<br><br>[BA] One of the questio=
ns I have had about the usage of DTLS-SRTP contemplated in RTCWEB is whethe=
r it will be interoperable with the SIP usage of DTLS-SRTP.&nbsp=3B <br><br=
>In the SIP framework it is possible to do RFC 4474 re-signing in the B2BUA=
 if the destination is an E.164 number. <br><br>However=2C the DTLS-SRTP/Id=
P framework seems to assume that the identities are email-style addresses.&=
nbsp=3B <br><br><br></div> 		 	   		  </div></body>
</html>=

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Tue, 20 Mar 2012 12:20:10 +0100
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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2012/3/20 Roman Shpount <roman@telurix.com>:
> I would also like to have a simplified DTLS specification for DTLS-SRTP,
> with as many as possible unnecessary features (encryption algorithms,
> cookies and such) removed and only a few required scenarios supported. Fo=
r
> instance, I do not want DTLS on a WebRTC channel start negotiation and en=
d
> up with anything except SRTP as an encryption protocol. Ideally, I want
> something were an interop test can be developed with a manageable number =
of
> scenarios.
>
> P.S. Sending RSA public key in the session description instead of
> fingerprint and then encrypting the SRTP key using this public key and
> sending it in the ICE message might be a much simpler call setup mechanis=
m
> to implement then bringing the whole DTLS into the stack. This should als=
o
> allow us to setup a secure call with no overhead in addition to ICE.

Would this DTLS-SRTP simplification interoperate with other
technologies also implementing DTLS-SRTP+SDP?

AFAIK there are (not yet) SIP devices supporting DTLS-SRTP (maybe I'm
wrong), however my question is: once there are SIP implementations (or
Jingle ones implementing DTLS-SRTP) would those implementations
interoperate with your simplified version of DTLS for WebRTC?

Regards.

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Date: Tue, 20 Mar 2012 09:40:37 -0400
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From: Roman Shpount <roman@telurix.com>
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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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--047d7b10d17bea946f04bbacd1bd
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Once again, I understand how this helps in case of HTTPS, but how would
this help in case of WebRTC? Media description is carried in some sort of
application defined protocol (can even be transmitted over an encrypted
SCTP data channel or encrypted in javaScript), so monitoring proxy cannot
reliably modify it. I understand how it can be allowed to fake the
signature on modified SDP by inserting a certificate in the browser, but
how would the proxy get to the SDP in the first place?
_____________
Roman Shpount


On Tue, Mar 20, 2012 at 6:41 AM, Harald Alvestrand <harald@alvestrand.no>wrote:

> **
> On 03/17/2012 10:14 PM, Igor Faynberg wrote:
>
> ..
>
> On 3/17/2012 4:45 PM, Martin Thomson wrote:
>
> ... Then I explicitly place a trust anchor for that
> certificate in your browser.
>
>
> How?  I thought the browser would never allow that to happen. (I assumed
> it would come provisioned with anchors,  or would allow anchor provision
> through some different channel.)
>
>  Igor,
>
> in Chrome, go to chrome://settings/certificates, choose the "authorities"
> tab, and note the presence of the "import" button.
>
> All browsers (as far as I know) have similar mechanisms.
> "Owning" your computer will achieve this goal both in its corporate
> meaning and in its hacker meaning.
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--047d7b10d17bea946f04bbacd1bd
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Once again, I understand how this helps in case of HTTPS, but how would thi=
s help in case of WebRTC? Media description is carried in some sort of appl=
ication defined protocol (can even be transmitted over an encrypted SCTP da=
ta channel or encrypted in javaScript), so monitoring proxy cannot reliably=
 modify it. I understand how it can be allowed to fake the signature on mod=
ified SDP by inserting a certificate in the browser, but how would the prox=
y get to the SDP in the first place?<br clear=3D"all">
_____________<br>Roman Shpount<br>
<br><br><div class=3D"gmail_quote">On Tue, Mar 20, 2012 at 6:41 AM, Harald =
Alvestrand <span dir=3D"ltr">&lt;<a href=3D"mailto:harald@alvestrand.no">ha=
rald@alvestrand.no</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quot=
e" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<u></u>

 =20
   =20
 =20
  <div bgcolor=3D"#ffffff" text=3D"#000000"><div class=3D"im">
    On 03/17/2012 10:14 PM, Igor Faynberg wrote:
    <blockquote type=3D"cite">..
      <br>
      <br>
      On 3/17/2012 4:45 PM, Martin Thomson wrote:
      <br>
      <blockquote type=3D"cite">... Then I explicitly place a trust anchor
        for that
        <br>
        certificate in your browser.
        <br>
      </blockquote>
      <br>
      How?=A0 I thought the browser would never allow that to happen. (I
      assumed it would come provisioned with anchors,=A0 or would allow
      anchor provision through some different channel.)
      <br>
      <br>
    </blockquote></div>
    Igor,<br>
    <br>
    in Chrome, go to <a>chrome://settings/certificates, choose the
      &quot;authorities&quot; tab, and note the presence of the &quot;impor=
t&quot; button.<br>
      <br>
      All browsers (as far as I know) have similar mechanisms.<br>
      &quot;Owning&quot; your computer will achieve this goal both in its
      corporate meaning and in its hacker meaning.<br>
      <br>
    </a><br>
  </div>

<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br>

--047d7b10d17bea946f04bbacd1bd--

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Tue, 20 Mar 2012 14:48:33 +0100
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2012/3/20 Roman Shpount <roman@telurix.com>:
> Once again, I understand how this helps in case of HTTPS, but how would t=
his
> help in case of WebRTC? Media description is carried in some sort of
> application defined protocol (can even be transmitted over an encrypted S=
CTP
> data channel or encrypted in javaScript), so monitoring proxy cannot
> reliably modify it. I understand how it can be allowed to fake the signat=
ure
> on modified SDP by inserting a certificate in the browser, but how would =
the
> proxy get to the SDP in the first place?

That's up to the implementor, is not? This is: if a WebRTC provider
wants to monitor the media by modifying the SDP it should design some
system in which the SDP is carried by its systems (i.e. via HTTP(s) or
WebSocket).

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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On Tue, Mar 20, 2012 at 7:20 AM, I=F1aki Baz Castillo <ibc@aliax.net> wrote=
:

> 2012/3/20 Roman Shpount <roman@telurix.com>:
> > I would also like to have a simplified DTLS specification for DTLS-SRTP=
,
> > with as many as possible unnecessary features (encryption algorithms,
> > cookies and such) removed and only a few required scenarios supported.
> For
> > instance, I do not want DTLS on a WebRTC channel start negotiation and
> end
> > up with anything except SRTP as an encryption protocol. Ideally, I want
> > something were an interop test can be developed with a manageable numbe=
r
> of
> > scenarios.
> >
> > P.S. Sending RSA public key in the session description instead of
> > fingerprint and then encrypting the SRTP key using this public key and
> > sending it in the ICE message might be a much simpler call setup
> mechanism
> > to implement then bringing the whole DTLS into the stack. This should
> also
> > allow us to setup a secure call with no overhead in addition to ICE.
>
> Would this DTLS-SRTP simplification interoperate with other
> technologies also implementing DTLS-SRTP+SDP?
>
> AFAIK there are (not yet) SIP devices supporting DTLS-SRTP (maybe I'm
> wrong), however my question is: once there are SIP implementations (or
> Jingle ones implementing DTLS-SRTP) would those implementations
> interoperate with your simplified version of DTLS for WebRTC?
>
>
There are actually two different options that I am proposing here.
Simplified DTLS-SRTP is just a clarification on what encryption protocols
and features are supported. Interop would be no different then interop with
DTLS-SRTP implementation which would require certificate validation. If one
of the parties request the feature not supported by the other, DTLS
handshake fails and connection would not be established.

The other proposal was not to use DTLS at all, since it is an overkill both
complexity and feature wise. DTLS is designed the way it is primarily to
extend TLS to UDP. DTLS-SRTP is a bolt on of SRTP into DTLS. So we actually
got a protocol twice reused with design goals changing dramatically in the
process. The desired attribute of DTLS-SRTP is that encryption keys are not
transmitted in clear text. The rest (security handshake separated from
signaling and ICE handshake, support for data encryption schemes other then
SRTP, support for certificate validation, etc) is actually just a
liability. All of those things imply slower setup and interop problems. So,
my second proposal was not to use DTLS at all, just include the public keys
for supported key exchange protocols in the SDP and send the encrypted SRTP
private key in the ICE message. Or something similar to that. This will
definitely not interop with DTLS, but will make implementation of
encryption very simple.
_____________
Roman Shpount

--047d7b2edf03d09baf04bbacf58b
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<br clear=3D"all"><br><br><div class=3D"gmail_quote">On Tue, Mar 20, 2012 a=
t 7:20 AM, I=F1aki Baz Castillo <span dir=3D"ltr">&lt;<a href=3D"mailto:ibc=
@aliax.net">ibc@aliax.net</a>&gt;</span> wrote:<br><blockquote class=3D"gma=
il_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-lef=
t:1ex">
2012/3/20 Roman Shpount &lt;<a href=3D"mailto:roman@telurix.com">roman@telu=
rix.com</a>&gt;:<br>
<div class=3D"im">&gt; I would also like to have a simplified DTLS specific=
ation for DTLS-SRTP,<br>
&gt; with as many as possible unnecessary features (encryption algorithms,<=
br>
&gt; cookies and such) removed and only a few required scenarios supported.=
 For<br>
&gt; instance, I do not want DTLS on a WebRTC channel start negotiation and=
 end<br>
&gt; up with anything except SRTP as an encryption protocol. Ideally, I wan=
t<br>
&gt; something were an interop test can be developed with a manageable numb=
er of<br>
&gt; scenarios.<br>
&gt;<br>
&gt; P.S. Sending RSA public key in the session description instead of<br>
&gt; fingerprint and then encrypting the SRTP key using this public key and=
<br>
&gt; sending it in the ICE message might be a much simpler call setup mecha=
nism<br>
&gt; to implement then bringing the whole DTLS into the stack. This should =
also<br>
&gt; allow us to setup a secure call with no overhead in addition to ICE.<b=
r>
<br>
</div>Would this DTLS-SRTP simplification interoperate with other<br>
technologies also implementing DTLS-SRTP+SDP?<br>
<br>
AFAIK there are (not yet) SIP devices supporting DTLS-SRTP (maybe I&#39;m<b=
r>
wrong), however my question is: once there are SIP implementations (or<br>
Jingle ones implementing DTLS-SRTP) would those implementations<br>
interoperate with your simplified version of DTLS for WebRTC?<br>
<br></blockquote></div><br>There are actually two different options that I =
am proposing here. Simplified DTLS-SRTP is just a clarification on what enc=
ryption protocols and features are supported. Interop would be no different=
 then interop with DTLS-SRTP implementation which would require certificate=
 validation. If one of the parties request the feature not supported by the=
 other, DTLS handshake fails and connection would not be established.<br>
<br>The other proposal was not to use DTLS at all, since it is an overkill =
both complexity and feature wise. DTLS is designed the way it is primarily =
to extend TLS to UDP. DTLS-SRTP is a bolt on of SRTP into DTLS. So we actua=
lly got a protocol twice reused with design goals changing dramatically in =
the process. The desired attribute of DTLS-SRTP is that encryption keys are=
 not transmitted in clear text. The rest (security handshake separated from=
 signaling and ICE handshake, support for data encryption schemes other the=
n SRTP, support for certificate validation, etc) is actually just a liabili=
ty. All of those things imply slower setup and interop problems. So, my sec=
ond proposal was not to use DTLS at all, just include the public keys for s=
upported key exchange protocols in the SDP and send the encrypted SRTP priv=
ate key in the ICE message. Or something similar to that. This will definit=
ely not interop with DTLS, but will make implementation of encryption very =
simple.<br>
_____________<br>Roman Shpount<br>
<br>

--047d7b2edf03d09baf04bbacf58b--

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References: <4F4759DC.7060303@ericsson.com> <387F9047F55E8C42850AD6B3A7A03C6C0E1FEB69@inba-mail01.sonusnet.com> <CALiegfnkYVEpmPV-zSL_4wOY-HiFZN-qJCQCiioaS=5NaqhLZw@mail.gmail.com> <CAD5OKxvtOAxMBx6xDnyfTnEq76oDEm6uj1xL6wGjjrtKUAHy3g@mail.gmail.com> <CABcZeBNZiotPmCfT53uEo+O0xw4xv6tXW1M_G-3A5BHuncsduA@mail.gmail.com> <CAD5OKxvYOY5JZ2mYNGiH1poUBQkyOOycePFijH5H+SxtcdqujQ@mail.gmail.com> <CABkgnnVe-b6Sv=R67bMJk_NQqQwdrRUn6rBm7Gu_CMcfPQwtEg@mail.gmail.com> <4F64FE98.3070605@alcatel-lucent.com> <4F685ED9.2050109@alvestrand.no> <CAD5OKxsVp7px9bHAgxgdqPMxRgppcVUDKt8JHBhyq9qqW3pAMg@mail.gmail.com> <CALiegfnfoMvJ2EbAUoF5o_R_GNYSxMWAE1h+8Js1T8u5eKbc=Q@mail.gmail.com>
Date: Tue, 20 Mar 2012 09:56:51 -0400
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From: Roman Shpount <roman@telurix.com>
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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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On Tue, Mar 20, 2012 at 9:48 AM, I=F1aki Baz Castillo <ibc@aliax.net> wrote=
:

> 2012/3/20 Roman Shpount <roman@telurix.com>:
> > Once again, I understand how this helps in case of HTTPS, but how would
> this
> > help in case of WebRTC? Media description is carried in some sort of
> > application defined protocol (can even be transmitted over an encrypted
> SCTP
> > data channel or encrypted in javaScript), so monitoring proxy cannot
> > reliably modify it. I understand how it can be allowed to fake the
> signature
> > on modified SDP by inserting a certificate in the browser, but how woul=
d
> the
> > proxy get to the SDP in the first place?
>
> That's up to the implementor, is not? This is: if a WebRTC provider
> wants to monitor the media by modifying the SDP it should design some
> system in which the SDP is carried by its systems (i.e. via HTTP(s) or
> WebSocket).
>
>
We got one group of people designing browsers, another group of people
designing monitoring proxies and the third group designing WebRTC
application. For the monitoring proxy to work it would need some sort of
network protocol to communicate with the browser (similar how HTTP proxy
needs a network protocol to work with the browser). I would expect what's
needed is some sort of network based protocol to send the SDP to a proxy,
get modified SDP back and pass it to JavaScript (or take the SDP from
JavaScript, through the proxy and to the media stack). I do not think this
is up to implementer, this is typically what the standards are for.
_____________
Roman Shpount

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<br clear=3D"all"><br><div class=3D"gmail_quote">On Tue, Mar 20, 2012 at 9:=
48 AM, I=F1aki Baz Castillo <span dir=3D"ltr">&lt;<a href=3D"mailto:ibc@ali=
ax.net">ibc@aliax.net</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_q=
uote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1e=
x">
2012/3/20 Roman Shpount &lt;<a href=3D"mailto:roman@telurix.com">roman@telu=
rix.com</a>&gt;:<br>
<div class=3D"im">&gt; Once again, I understand how this helps in case of H=
TTPS, but how would this<br>
&gt; help in case of WebRTC? Media description is carried in some sort of<b=
r>
&gt; application defined protocol (can even be transmitted over an encrypte=
d SCTP<br>
&gt; data channel or encrypted in javaScript), so monitoring proxy cannot<b=
r>
&gt; reliably modify it. I understand how it can be allowed to fake the sig=
nature<br>
&gt; on modified SDP by inserting a certificate in the browser, but how wou=
ld the<br>
&gt; proxy get to the SDP in the first place?<br>
<br>
</div>That&#39;s up to the implementor, is not? This is: if a WebRTC provid=
er<br>
wants to monitor the media by modifying the SDP it should design some<br>
system in which the SDP is carried by its systems (i.e. via HTTP(s) or<br>
WebSocket).<br>
<div><div></div><br></div></blockquote></div><br>We got one group of people=
 designing browsers, another group of people designing monitoring proxies a=
nd the third group designing WebRTC application. For the monitoring proxy t=
o work it would need some sort of network protocol to communicate with the =
browser (similar how HTTP proxy needs a network protocol to work with the b=
rowser). I would expect what&#39;s needed is some sort of network based pro=
tocol to send the SDP to a proxy, get modified SDP back and pass it to Java=
Script (or take the SDP from JavaScript, through the proxy and to the media=
 stack). I do not think this is up to implementer, this is typically what t=
he standards are for.<br>
_____________<br>Roman Shpount<br>
<br>

--047d7b2edf03f8a91204bbad0b7d--

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References: <4F4759DC.7060303@ericsson.com> <CAD5OKxvYOY5JZ2mYNGiH1poUBQkyOOycePFijH5H+SxtcdqujQ@mail.gmail.com> <CABkgnnVe-b6Sv=R67bMJk_NQqQwdrRUn6rBm7Gu_CMcfPQwtEg@mail.gmail.com> <CAD5OKxvZbEJ7sV4WPAYoQapzMR_QwAftj-oKg=ioMKHNT792wQ@mail.gmail.com> <6F428EFD2B8C2F49A2FB1317291A76C113563C5A92@USNAVSXCHMBSA1.ndc.alcatel-lucent.com> <CALiegf=jtkDCS_D0ZFe9UpbiadQ0vsJ+4MppQSbLr-wbaXNrfQ@mail.gmail.com> <BLU169-W29E5B86F9E2C6F3126961C93420@phx.gbl> <CALiegfk2aT+6Psr4nT-hG1G7eYRBfFCcT+25On2O4HfUXJ6-ng@mail.gmail.com> <CAD6AjGSmi9j+sdGWPts20-iwGvGij05ek0OKYEPULC6B=aFpQg@mail.gmail.com> <6F428EFD2B8C2F49A2FB1317291A76C113564482A7@USNAVSXCHMBSA1.ndc.alcatel-lucent.com> <ADBB75F3-E20C-4EC4-B9C3-EF2E4BFF409C@phonefromhere.com> <CAD5OKxvuEV8Vbq3h7=ZgcKmREjmguvz5n-SpXr2n-EY7a_ddxg@mail.gmail.com> <CALiegfk1ozOKPcDjbd3H_z2Edzh4RcZpYyJSWdw_1DJ04muQXA@mail.gmail.com> <CAD5OKxu8-+0O0=eE7mD1hi=nPUpEXczGj=bRNQCQL1BW8c-c-Q@mail.gmail.com> <4F677F3B.3040407@alcatel-lucent.com> <4F686183.6040201@alvestrand.no>
Date: Tue, 20 Mar 2012 10:03:46 -0400
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From: Roman Shpount <roman@telurix.com>
To: Harald Alvestrand <harald@alvestrand.no>
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Telling the user the connection is secure (Re: Resolving RTP/SDES question in Paris)
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On Tue, Mar 20, 2012 at 6:52 AM, Harald Alvestrand <harald@alvestrand.no>wrote:

> I believe I have said this before, but...
>
> We should never tell the user the connection is secure.
> We should tell the user when we know he's exposed to risks that he usually
> isn't.
>
> Thus - we should not give any indication that we're using DTLS-SRTP with
> verified identities (if that's what we normally support). We SHOULD give a
> warning saying "hey, since the gateway you've connected to isn't doing
> normal authentication procedures, but instead insists on exchanging keys on
> the signalling channel, you are less sure who you're talking to than usual,
> and there are more boxes that might record your call in the way, but the
> script kiddie on your hotel WLAN still can't see your packets (translation:
> legacy SDES key exchange is in use, but SRTP is still on).
>
> All this will of course be iconified into a single cryptic graphic
> probably involving a padlock :-)
>
>
What we actually got here has to be iconified into padlocks of different
size and degree of openness.

In case of DTLS-SRTP with verified identities we still need to inform the
user of the indentity of the person they are communicating with. It has to
come from the browser, so that it can be presented with some degree of
trust. Since we do not trust web server or the javascript app, we should be
asking for user consent to get access to microphone and camera for each
call anyway. When asking for this consent, we should be either telling the
identity of the remote party or give a warning that remote party is unknown.
_____________
Roman Shpount

--e89a8ff1c35eb45aa904bbad24f3
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<br clear=3D"all"><div class=3D"gmail_quote">On Tue, Mar 20, 2012 at 6:52 A=
M, Harald Alvestrand <span dir=3D"ltr">&lt;<a href=3D"mailto:harald@alvestr=
and.no">harald@alvestrand.no</a>&gt;</span> wrote:<br><blockquote class=3D"=
gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-=
left:1ex">
I believe I have said this before, but...<br>
<br>
We should never tell the user the connection is secure.<br>
We should tell the user when we know he&#39;s exposed to risks that he usua=
lly isn&#39;t.<br>
<br>
Thus - we should not give any indication that we&#39;re using DTLS-SRTP wit=
h verified identities (if that&#39;s what we normally support). We SHOULD g=
ive a warning saying &quot;hey, since the gateway you&#39;ve connected to i=
sn&#39;t doing normal authentication procedures, but instead insists on exc=
hanging keys on the signalling channel, you are less sure who you&#39;re ta=
lking to than usual, and there are more boxes that might record your call i=
n the way, but the script kiddie on your hotel WLAN still can&#39;t see you=
r packets (translation: legacy SDES key exchange is in use, but SRTP is sti=
ll on).<br>

<br>
All this will of course be iconified into a single cryptic graphic probably=
 involving a padlock :-)<br>
<br></blockquote></div><br>What we actually got here has to be iconified in=
to padlocks of different size and degree of openness.<br><br>In case of DTL=
S-SRTP with verified identities we still need to inform the user of the ind=
entity of the person they are communicating with. It has to come from the b=
rowser, so that it can be presented with some degree of trust. Since we do =
not trust web server or the javascript app, we should be asking for user co=
nsent to get access to microphone and camera for each call anyway. When ask=
ing for this consent, we should be either telling the identity of the remot=
e party or give a warning that remote party is unknown.<br>
_____________<br>Roman Shpount<br>
<br><br>

--e89a8ff1c35eb45aa904bbad24f3--

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Tue, 20 Mar 2012 15:16:11 +0100
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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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2012/3/20 Roman Shpount <roman@telurix.com>:
> The other proposal was not to use DTLS at all, since it is an overkill bo=
th
> complexity and feature wise. DTLS is designed the way it is primarily to
> extend TLS to UDP. DTLS-SRTP is a bolt on of SRTP into DTLS. So we actual=
ly
> got a protocol twice reused with design goals changing dramatically in th=
e
> process. The desired attribute of DTLS-SRTP is that encryption keys are n=
ot
> transmitted in clear text. The rest (security handshake separated from
> signaling and ICE handshake, support for data encryption schemes other th=
en
> SRTP, support for certificate validation, etc) is actually just a liabili=
ty.
> All of those things imply slower setup and interop problems. So, my secon=
d
> proposal was not to use DTLS at all, just include the public keys for
> supported key exchange protocols in the SDP and send the encrypted SRTP
> private key in the ICE message. Or something similar to that. This will
> definitely not interop with DTLS, but will make implementation of encrypt=
ion
> very simple.

So that proposal would modify even ICE protocol, am I right? Too much
dramatic IMHO :)

Ok, given the fact that DTLS is still an utopia (all the people know
about it but it seems that nobody has seen it in action), why not to
leave it for a future version of WebRTC and go now with SDES +
encrypted signaling path? In this way:

- We get interoperability with other technologies *already*
implementing SDES+SRTP (so ***tested***).

- DTLS can be added once it's mature/tested/widely-implemented and we
can learn from other implementations (what do they implement, interop
problems, bugs...?).

- Adding DTLS later to WebRTC does not break nothing since its usage
would be still negotiable during the session initiation.


Yes, I do know that SDES+SRTP is not the best solution, but I've never
seen a scenario in which all the cool and fantastic features of DTLS
are required. Given the comments about DTLS in this long thread it
seems that choosing DTLS would be a risk (WebRTC would become a DTLS
beta-tester...).


Regards.

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Cc: Leon Portman <Leon.Portman@nice.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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From: Eric Rescorla <ekr@rtfm.com>
Date: Tue, 20 Mar 2012 08:01:43 -0700
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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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On Tue, Mar 20, 2012 at 7:16 AM, I=F1aki Baz Castillo <ibc@aliax.net> wrote=
:
> 2012/3/20 Roman Shpount <roman@telurix.com>:
>> The other proposal was not to use DTLS at all, since it is an overkill b=
oth
>> complexity and feature wise. DTLS is designed the way it is primarily to
>> extend TLS to UDP. DTLS-SRTP is a bolt on of SRTP into DTLS. So we actua=
lly
>> got a protocol twice reused with design goals changing dramatically in t=
he
>> process. The desired attribute of DTLS-SRTP is that encryption keys are =
not
>> transmitted in clear text. The rest (security handshake separated from
>> signaling and ICE handshake, support for data encryption schemes other t=
hen
>> SRTP, support for certificate validation, etc) is actually just a liabil=
ity.
>> All of those things imply slower setup and interop problems. So, my seco=
nd
>> proposal was not to use DTLS at all, just include the public keys for
>> supported key exchange protocols in the SDP and send the encrypted SRTP
>> private key in the ICE message. Or something similar to that. This will
>> definitely not interop with DTLS, but will make implementation of encryp=
tion
>> very simple.
>
> So that proposal would modify even ICE protocol, am I right? Too much
> dramatic IMHO :)
>
> Ok, given the fact that DTLS is still an utopia (all the people know
> about it but it seems that nobody has seen it in action),

This is not true for DTLS in general. It's been in stacks for
a while now and is widely used in VPNs.

There certainly is rather less use of DTLS-SRTP, but it isn't unknown
either, as I believe Cullen has mentioned on the list before.

-Ekr

From harald@alvestrand.no  Tue Mar 20 08:40:05 2012
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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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On 03/20/2012 02:40 PM, Roman Shpount wrote:
> Once again, I understand how this helps in case of HTTPS, but how 
> would this help in case of WebRTC? Media description is carried in 
> some sort of application defined protocol (can even be transmitted 
> over an encrypted SCTP data channel or encrypted in javaScript), so 
> monitoring proxy cannot reliably modify it. I understand how it can be 
> allowed to fake the signature on modified SDP by inserting a 
> certificate in the browser, but how would the proxy get to the SDP in 
> the first place?

"this" = inserting a trust root in the browser?

Once you trust the root I inserted, I can intercept all your HTTPS 
connections, and thus I can replace all the Javascript you think you are 
loading from your trusted provider with Javascript of my choice.

Once your browser is running my Javascript, I can do anything the 
original application could - with the privilleges you granted to that 
application.

I think that's more or less "game over" when it comes to security.


> _____________
> Roman Shpount
>
>
> On Tue, Mar 20, 2012 at 6:41 AM, Harald Alvestrand 
> <harald@alvestrand.no <mailto:harald@alvestrand.no>> wrote:
>
>     On 03/17/2012 10:14 PM, Igor Faynberg wrote:
>>     ..
>>
>>     On 3/17/2012 4:45 PM, Martin Thomson wrote:
>>>     ... Then I explicitly place a trust anchor for that
>>>     certificate in your browser.
>>
>>     How?  I thought the browser would never allow that to happen. (I
>>     assumed it would come provisioned with anchors,  or would allow
>>     anchor provision through some different channel.)
>>
>     Igor,
>
>     in Chrome, go to chrome://settings/certificates, choose the
>     "authorities" tab, and note the presence of the "import" button.
>
>     All browsers (as far as I know) have similar mechanisms.
>     "Owning" your computer will achieve this goal both in its
>     corporate meaning and in its hacker meaning.
>
>
>
>     _______________________________________________
>     rtcweb mailing list
>     rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>     https://www.ietf.org/mailman/listinfo/rtcweb
>
>


--------------010107050405090001030409
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
  <head>
    <meta content="text/html; charset=ISO-8859-1"
      http-equiv="Content-Type">
  </head>
  <body bgcolor="#ffffff" text="#000000">
    On 03/20/2012 02:40 PM, Roman Shpount wrote:
    <blockquote
cite="mid:CAD5OKxsVp7px9bHAgxgdqPMxRgppcVUDKt8JHBhyq9qqW3pAMg@mail.gmail.com"
      type="cite">Once again, I understand how this helps in case of
      HTTPS, but how would this help in case of WebRTC? Media
      description is carried in some sort of application defined
      protocol (can even be transmitted over an encrypted SCTP data
      channel or encrypted in javaScript), so monitoring proxy cannot
      reliably modify it. I understand how it can be allowed to fake the
      signature on modified SDP by inserting a certificate in the
      browser, but how would the proxy get to the SDP in the first
      place?<br clear="all">
    </blockquote>
    <br>
    "this" = inserting a trust root in the browser?<br>
    <br>
    Once you trust the root I inserted, I can intercept all your HTTPS
    connections, and thus I can replace all the Javascript you think you
    are loading from your trusted provider with Javascript of my choice.<br>
    <br>
    Once your browser is running my Javascript, I can do anything the
    original application could - with the privilleges you granted to
    that application.<br>
    <br>
    I think that's more or less "game over" when it comes to security.<br>
    <br>
    <br>
    <blockquote
cite="mid:CAD5OKxsVp7px9bHAgxgdqPMxRgppcVUDKt8JHBhyq9qqW3pAMg@mail.gmail.com"
      type="cite">
      _____________<br>
      Roman Shpount<br>
      <br>
      <br>
      <div class="gmail_quote">On Tue, Mar 20, 2012 at 6:41 AM, Harald
        Alvestrand <span dir="ltr">&lt;<a moz-do-not-send="true"
            href="mailto:harald@alvestrand.no">harald@alvestrand.no</a>&gt;</span>
        wrote:<br>
        <blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt
          0.8ex; border-left: 1px solid rgb(204, 204, 204);
          padding-left: 1ex;">
          <div bgcolor="#ffffff" text="#000000">
            <div class="im"> On 03/17/2012 10:14 PM, Igor Faynberg
              wrote:
              <blockquote type="cite">.. <br>
                <br>
                On 3/17/2012 4:45 PM, Martin Thomson wrote: <br>
                <blockquote type="cite">... Then I explicitly place a
                  trust anchor for that <br>
                  certificate in your browser. <br>
                </blockquote>
                <br>
                How?&nbsp; I thought the browser would never allow that to
                happen. (I assumed it would come provisioned with
                anchors,&nbsp; or would allow anchor provision through some
                different channel.) <br>
                <br>
              </blockquote>
            </div>
            Igor,<br>
            <br>
            in Chrome, go to <a moz-do-not-send="true">chrome://settings/certificates,
              choose the "authorities" tab, and note the presence of the
              "import" button.<br>
              <br>
              All browsers (as far as I know) have similar mechanisms.<br>
              "Owning" your computer will achieve this goal both in its
              corporate meaning and in its hacker meaning.<br>
              <br>
            </a><br>
          </div>
          <br>
          _______________________________________________<br>
          rtcweb mailing list<br>
          <a moz-do-not-send="true" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
          <a moz-do-not-send="true"
            href="https://www.ietf.org/mailman/listinfo/rtcweb"
            target="_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
          <br>
        </blockquote>
      </div>
      <br>
    </blockquote>
    <br>
  </body>
</html>

--------------010107050405090001030409--

From igor.faynberg@alcatel-lucent.com  Tue Mar 20 08:41:50 2012
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Telling the user the connection is secure (Re: Resolving RTP/SDES question in Paris)
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+1

Igor

On 3/20/2012 6:52 AM, Harald Alvestrand wrote:
> I believe I have said this before, but...
>
> We should never tell the user the connection is secure.
> We should tell the user when we know he's exposed to risks that he 
> usually isn't.
>
> Thus - we should not give any indication that we're using DTLS-SRTP 
> with verified identities (if that's what we normally support). We 
> SHOULD give a warning saying "hey, since the gateway you've connected 
> to isn't doing normal authentication procedures, but instead insists 
> on exchanging keys on the signalling channel, you are less sure who 
> you're talking to than usual, and there are more boxes that might 
> record your call in the way, but the script kiddie on your hotel WLAN 
> still can't see your packets (translation: legacy SDES key exchange is 
> in use, but SRTP is still on).
>
> All this will of course be iconified into a single cryptic graphic 
> probably involving a padlock :-)
>
> On 03/19/2012 07:47 PM, Igor Faynberg wrote:
>> This is the question that I have been asking for a while...  I don't 
>> expect a complete fireproof answer, of course, and I also understand 
>> that the browser today is telling me a few things about the security 
>> of a site and warns me when "the site is trying to access the data it 
>> should not be accessing."
>>
>> But I  also imagine that a rogue site could display a message 
>> mimicking the security assurance as though it comes from the browser.
>>
>> So it would be good to have a very clear idea when  the determination 
>> about the security of the connection and such is made and how the end 
>> user can verify that it actually comes from the browser.
>>
>> (To this end, the user MUST trust the browser, of course.)
>>
>> Igor
>>
>> On 3/19/2012 2:15 PM, Roman Shpount wrote:
>>> I guess my question is, when are we going to tell the user that 
>>> connection is "secure"?...
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>

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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Harald Alvestrand <harald@alvestrand.no>
Thread-Topic: [rtcweb] SDES vs DTLS-SRTP revisited
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Subject: Re: [rtcweb] SDES vs DTLS-SRTP revisited
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On Mar 20, 2012, at 6:30 AM, Harald Alvestrand wrote:

> I don't get this scenario. If Alice calls Bob using two different gateway=
s, won't she go through a credentials and fingerprint exchange with the gat=
eways?
> In that case, wouldn't the fingerprint belong to the gateway?

Yes, but the statement being made was in the context of Alice calling Bob, =
them seeing their browsers claim DTLS-SRTP "secured" or whatever, and them =
being super-geeks and checking the detailed info of what the actual DTLS fi=
ngerprints were, and finding they don't both see the same fingerprints... a=
nd that they would thus believe there was either a software bug or a malici=
ous middleman.

So my point was that's not a good conclusion to jump to, since both caller =
and called parties can see DTLS-SRTP "secured" mode lock-icon, but with dif=
ferent fingerprints, and yet it being neither a software bug nor a maliciou=
s middleman.  I realize no distinction can be made between a PSTN-gateway a=
nd a malicious MitM, by design, but that's not a good thing - because it me=
ans people will simply learn to ignore the warnings generated by the browse=
r, because from a user perspective they'll all be false positives.

-hadriel


From harald@alvestrand.no  Tue Mar 20 09:01:02 2012
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Subject: Re: [rtcweb] SDES vs DTLS-SRTP revisited
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On 03/20/2012 04:44 PM, Hadriel Kaplan wrote:
> On Mar 20, 2012, at 6:30 AM, Harald Alvestrand wrote:
>
>> I don't get this scenario. If Alice calls Bob using two different gateways, won't she go through a credentials and fingerprint exchange with the gateways?
>> In that case, wouldn't the fingerprint belong to the gateway?
> Yes, but the statement being made was in the context of Alice calling Bob, them seeing their browsers claim DTLS-SRTP "secured" or whatever, and them being super-geeks and checking the detailed info of what the actual DTLS fingerprints were, and finding they don't both see the same fingerprints... and that they would thus believe there was either a software bug or a malicious middleman.
So we've uncovered a requirement here: When a fingerprint is shown, it 
should clearly identify what it is the fingerprint of - in this case, 
the gateway that relays the call to Bob, not Bob.
> So my point was that's not a good conclusion to jump to, since both caller and called parties can see DTLS-SRTP "secured" mode lock-icon, but with different fingerprints, and yet it being neither a software bug nor a malicious middleman.  I realize no distinction can be made between a PSTN-gateway and a malicious MitM, by design, but that's not a good thing - because it means people will simply learn to ignore the warnings generated by the browser, because from a user perspective they'll all be false positives.
So in this case, the browser needs to not warn; it needs to say 
"connected to gateway in order to reach Bob".

From roman@telurix.com  Tue Mar 20 09:10:21 2012
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On Tue, Mar 20, 2012 at 11:39 AM, Harald Alvestrand <harald@alvestrand.no>wrote:

> **
> On 03/20/2012 02:40 PM, Roman Shpount wrote:
>
> Once again, I understand how this helps in case of HTTPS, but how would
> this help in case of WebRTC? Media description is carried in some sort of
> application defined protocol (can even be transmitted over an encrypted
> SCTP data channel or encrypted in javaScript), so monitoring proxy cannot
> reliably modify it. I understand how it can be allowed to fake the
> signature on modified SDP by inserting a certificate in the browser, but
> how would the proxy get to the SDP in the first place?
>
>
> "this" = inserting a trust root in the browser?
>
> Once you trust the root I inserted, I can intercept all your HTTPS
> connections, and thus I can replace all the Javascript you think you are
> loading from your trusted provider with Javascript of my choice.
>
> Once your browser is running my Javascript, I can do anything the original
> application could - with the privilleges you granted to that application.
>
> I think that's more or less "game over" when it comes to security.
>
>
"this" is inserting the root certificate. It is a game over as far as
security is concerned, but the rest of your email completely missing my
point.

If, as an owner of a computer, you want to enable media monitoring with an
existing standard you need to know exactly how each application is
implemented. This is a solution for hacker but not the normal enterprise.
In order to have monitoring support in the enterprise environment you need
an ability to force the browser to proxy all the media through a monitoring
application without any application modification, via browser settings
alone.
_____________
Roman Shpount

--047d7b15a68b390dda04bbaee97b
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<br clear=3D"all"><div class=3D"gmail_quote">On Tue, Mar 20, 2012 at 11:39 =
AM, Harald Alvestrand <span dir=3D"ltr">&lt;<a href=3D"mailto:harald@alvest=
rand.no">harald@alvestrand.no</a>&gt;</span> wrote:<br><blockquote class=3D=
"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding=
-left:1ex">
<u></u>

 =20
   =20
 =20
  <div bgcolor=3D"#ffffff" text=3D"#000000"><div class=3D"im">
    On 03/20/2012 02:40 PM, Roman Shpount wrote:
    <blockquote type=3D"cite">Once again, I understand how this helps in ca=
se of
      HTTPS, but how would this help in case of WebRTC? Media
      description is carried in some sort of application defined
      protocol (can even be transmitted over an encrypted SCTP data
      channel or encrypted in javaScript), so monitoring proxy cannot
      reliably modify it. I understand how it can be allowed to fake the
      signature on modified SDP by inserting a certificate in the
      browser, but how would the proxy get to the SDP in the first
      place?<br clear=3D"all">
    </blockquote>
    <br></div>
    &quot;this&quot; =3D inserting a trust root in the browser?<br>
    <br>
    Once you trust the root I inserted, I can intercept all your HTTPS
    connections, and thus I can replace all the Javascript you think you
    are loading from your trusted provider with Javascript of my choice.<br=
>
    <br>
    Once your browser is running my Javascript, I can do anything the
    original application could - with the privilleges you granted to
    that application.<br>
    <br>
    I think that&#39;s more or less &quot;game over&quot; when it comes to =
security.<div class=3D"im"><br></div></div></blockquote></div><br>&quot;thi=
s&quot; is inserting the root certificate. It is a game over as far as secu=
rity is concerned, but the rest of your email completely missing my point.<=
br>
<br>If, as an owner of a computer, you want to enable media monitoring with=
 an existing standard you need to know exactly how each application is impl=
emented. This is a solution for hacker but not the normal enterprise. In or=
der to have monitoring support in the enterprise environment you need an ab=
ility to force the browser to proxy all the media through a monitoring appl=
ication without any application modification, via browser settings alone.<b=
r>
_____________<br>Roman Shpount<br>
<br><br>

--047d7b15a68b390dda04bbaee97b--

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To: Harald Alvestrand <harald@alvestrand.no>
References: <4F4759DC.7060303@ericsson.com>	<387F9047F55E8C42850AD6B3A7A03C6C0E1FEB69@inba-mail01.sonusnet.com>	<CALiegfnkYVEpmPV-zSL_4wOY-HiFZN-qJCQCiioaS=5NaqhLZw@mail.gmail.com>	<CAD5OKxvtOAxMBx6xDnyfTnEq76oDEm6uj1xL6wGjjrtKUAHy3g@mail.gmail.com>	<CABcZeBNZiotPmCfT53uEo+O0xw4xv6tXW1M_G-3A5BHuncsduA@mail.gmail.com>	<CAD5OKxvYOY5JZ2mYNGiH1poUBQkyOOycePFijH5H+SxtcdqujQ@mail.gmail.com>	<CABkgnnVe-b6Sv=R67bMJk_NQqQwdrRUn6rBm7Gu_CMcfPQwtEg@mail.gmail.com> <4F64FE98.3070605@alcatel-lucent.com> <4F685ED9.2050109@alvestrand.no>
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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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Harald,

Many thanks!  I did just what you suggested, and spent happy three hours 
figuring out what is going on.  I found something that claims to be my 
own certificate, allegedly signed by me (which I swear I have never 
done) as well as many other certificates from the authorities I never 
knew (nor could find any information about).  On top of all, I found two 
certificates that allege being issued by Microsoft, while they are 
marked as fraudulent. (So somenone is checking this stuff and takes action?)

The above report speaks for itself.

The key finding though is that the browser interface in NOT the only way 
to get the certificates in.  Anyone can put them in a directory, and 
this a big problem for the "hacker meaning."   Incidentally, this is not 
only the browser matter--same holds for the OS.

If root certificates were to be installed only through the browser 
administrative interface, I imagine the browser could send me a warning 
or even deny the installation.  I know there is a lot to think through 
on how to bootstrap this process, but I think it should  be possible to 
achieve.  But in this case it appears to me that the browser would need 
to have its own independent file system controlled only through the 
administrative interface.

It also appears to me that the interface should be password-protected so 
that only the legitimate owner (including corporate) can control the 
installation of root certificates. If the hacker who does not know the 
administrative password just got  access to my computer (say, via JS), 
there is no way for this hacker to  install the certificate.

I know that this issue takes off on a tangent from the RTCWeb 
discussion, so I don't want to start that. But I wanted to acknowledge 
your message and thank you for responding.  Maybe we could talk about 
this with you, Hui-Lan, and Eric off-line.

Igor

On 3/20/2012 6:41 AM, Harald Alvestrand wrote:
> Igor,
>
>
> in Chrome, go to chrome://settings/certificates, choose the 
> "authorities" tab, and note the presence of the "import" button.
>
> All browsers (as far as I know) have similar mechanisms.
> "Owning" your computer will achieve this goal both in its corporate 
> meaning and in its hacker meaning.
>
>

--------------040506050507000604020604
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
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    <meta content="text/html; charset=ISO-8859-1"
      http-equiv="Content-Type">
  </head>
  <body text="#000000" bgcolor="#ffffff">
    Harald,<br>
    <br>
    Many thanks! &nbsp;I did just what you suggested, and spent happy three
    hours figuring out what is going on. &nbsp;I found something that claims
    to be my own certificate, allegedly signed by me (which I swear I
    have never done) as well as many other certificates from the
    authorities I never knew (nor could find any information about). &nbsp;On
    top of all, I found two certificates that allege being issued by
    Microsoft, while they are marked as fraudulent. (So somenone is
    checking this stuff and takes action?)<br>
    <br>
    The above report speaks for itself.<br>
    <br>
    The key finding though is that the browser interface in NOT the only
    way to get the certificates in. &nbsp;Anyone can put them in a directory,
    and this a big problem for the "hacker meaning." &nbsp; Incidentally,
    this is not only the browser matter--same holds for the OS.<br>
    <br>
    If root certificates were to be installed only through the browser
    administrative interface, I imagine the browser could send me a
    warning or even deny the installation. &nbsp;I know there is a lot to
    think through on how to bootstrap this process, but I think it
    should &nbsp;be possible to achieve. &nbsp;But in this case it appears to me
    that the browser would need to have its own independent file system
    controlled only through the administrative interface. <br>
    <br>
    It also appears to me that the interface should be
    password-protected so that only the legitimate owner (including
    corporate) can control the installation of root certificates. If the
    hacker who does not know the administrative password just got
    &nbsp;access to my computer (say, via JS), there is no way for this
    hacker to&nbsp; install the certificate.<br>
    <br>
    I know that this issue takes off on a tangent from the RTCWeb
    discussion, so I don't want to start that. But I wanted to
    acknowledge your message and thank you for responding. &nbsp;Maybe we
    could talk about this with you, Hui-Lan, and Eric off-line.<br>
    <br>
    Igor<br>
    <br>
    On 3/20/2012 6:41 AM, Harald Alvestrand wrote:
    <blockquote cite="mid:4F685ED9.2050109@alvestrand.no" type="cite">
      Igor,<br>
      <br>
      <br>
      in Chrome, go to <a moz-do-not-send="true">chrome://settings/certificates,
        choose the "authorities" tab, and note the presence of the
        "import" button.<br>
        <br>
        All browsers (as far as I know) have similar mechanisms.<br>
        "Owning" your computer will achieve this goal both in its
        corporate meaning and in its hacker meaning.<br>
        <br>
      </a><br>
    </blockquote>
  </body>
</html>

--------------040506050507000604020604--

From Jim.Barnett@genesyslab.com  Wed Mar 21 06:47:47 2012
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Thread-Topic: [rtcweb] Resolving RTP/SDES question in Paris
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References: <4F4759DC.7060303@ericsson.com><387F9047F55E8C42850AD6B3A7A03C6C0E1FEB69@inba-mail01.sonusnet.com><CALiegfnkYVEpmPV-zSL_4wOY-HiFZN-qJCQCiioaS=5NaqhLZw@mail.gmail.com><CAD5OKxvtOAxMBx6xDnyfTnEq76oDEm6uj1xL6wGjjrtKUAHy3g@mail.gmail.com><CABcZeBNZiotPmCfT53uEo+O0xw4xv6tXW1M_G-3A5BHuncsduA@mail.gmail.com><CAD5OKxvYOY5JZ2mYNGiH1poUBQkyOOycePFijH5H+SxtcdqujQ@mail.gmail.com><CABkgnnVe-b6Sv=R67bMJk_NQqQwdrRUn6rBm7Gu_CMcfPQwtEg@mail.gmail.com><CAD5OKxvZbEJ7sV4WPAYoQapzMR_QwAftj-oKg=ioMKHNT792wQ@mail.gmail.com><6F428EFD2B8C2F49A2FB1317291A76C113563C5A92@USNAVSXCHMBSA1.ndc.alcatel-lucent.com><CALiegf=jtkDCS_D0ZFe9UpbiadQ0vsJ+4MppQSbLr-wbaXNrfQ@mail.gmail.com><BLU169-W29E5B86F9E2C6F3126961C93420@phx.gbl><CALiegfk2aT+6Psr4nT-hG1G7eYRBfFCcT+25On2O4HfUXJ6-ng@mail.gmail.com><CAD6AjGSmi9j+sdGWPts20-iwGvGij05ek0OKYEPULC6B=aFpQg@mail.gmail.com><6F428EFD2B8C2F49A2FB1317291A76C113564482A7@USNAVSXCHMBSA1.ndc.alcatel-lucent.com><ADBB75F3-E20C-4EC4-B9C3-EF2E4BFF409C@phonefromhere.com><C! !AD5OKxvu E V8Vbq3h7=Z gcKmREjmguvz5n-SpXr2n-EY7a_ddxg@mail.gmail.com><CALiegfk1ozOKPcDjbd3H_z2Edzh4RcZpYyJSWdw_1DJ04muQXA@mail.gmail.com><CAD5OKxu8-+0O0=eE7mD1hi=nPUpEXczGj=bRNQCQL1BW8c-c-Q@mail.gmail.com><D75A384B-0F38-4E30-8C03-12E903A69B64@acmepacket.com><E17CAD772E76C742B645BD4DC602CD8105EBE8CF@NAHALD.us.int.genesyslab.com><387F9047F55E8C42850AD6B3A7A03C6C0E1FFE23@inba-mail01.sonusnet.com><2E10EB15-7E2E-47B9-80D1-5244DDE5FDF7@acmepacket.com> <101C6067BEC68246B0C3F6843BCCC1E31296AE222A@MCHP058A.global-ad.net>
From: "Jim Barnett" <Jim.Barnett@genesyslab.com>
To: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>, "Hadriel Kaplan" <HKaplan@acmepacket.com>, "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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One way to look at this case would be to say that it's the server at the
boundary of www.travel.co that is the RTCWeb endpoint,  _not_ the agent
desktop. The boundary server is acting as an RTCWeb-to-corporate network
gateway, and systems inside the corporate network are not affected.
Callers see themselves as connected to www.travel.co, rather than having
a direct RTCWeb-all-the-way connection to the agent.  I'm pretty sure
that call centers would like to do things this way since it doesn't
require any changes to their corporate infrastructure. (They can
add/upgrade the gateway without changing their recording systems or
agent desktops, etc.)

I don't see anything like this in the use case document.  Is this a use
case we want to support? =20

- Jim

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
Of Hutton, Andrew
Sent: Tuesday, March 20, 2012 6:07 AM
To: Hadriel Kaplan; Ravindran, Parthasarathi
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris

Hi,

Completely agree with Hadriel's statement below.

In addition to wanting to know that you are connected to "www.travel.co"
or "www.mybank.co" you as a consumer and the company really want/need
the media to be recorded by the companies system for your own
protection. This is going to involve some kind of MITM or as we call it
in SIPREC a Session Recording Client (SRC) to be able to intercept the
media and route it to a Session Recording Server (SRS).=20

The agent's desktop device could act as the SRC but more likely it will
be some kind of B2BUA that does this. =20

In these scenario's you as a consumer using this service really want the
media to be secure at least as far as reaching the bank but after that
like any existing web based system you have to trust the bank with your
data this is no different whether that data is speech or text that you
entered on a web page.

The recording system should be able to work with DTLS-SRTP but it will
most likely act as a MITM so will change the DTLS fingerprint. I don't
see that as a problem.

Regards
Andy



> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On=20
> Behalf Of Hadriel Kaplan
> Sent: 20 March 2012 03:21
> To: Ravindran, Parthasarathi
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
>=20
>=20
> And, it should be noted, they *want* it that way.  They don't=20
> generally want you to know which agent picked up the call, or that the

> call got transferred from an IVR to a specific agent, or got put in a=20
> queue with elevator-music announcement server, or any of that.  You
"called"
> Travel Co., and they answered.  What goes on behind that SBC is=20
> equivalent to the back-end database stuff that goes on between the web

> server and back-end systems, while all you see is the web-server your=20
> browser happens to be connected to.
>=20
> The "identity" you'll get is www.travel.co, which you already had when

> your browser did HTTPS cert verification of their web-server.  HTTPS=20
> became the key-exchange transport for the SRTP key.  Since they=20
> already proved they own the cert for www.travel.co, having them claim=20
> to be www.travel.co shouldn't require yet more verification.
>=20
> -hadriel
>=20
>=20
> On Mar 19, 2012, at 9:15 PM, Ravindran, Parthasarathi wrote:
>=20
> > Jim,
> >
> > As a customer, you won't really know whether the identity=20
> > (DTLS-SRTP)
> of call center/travel site is agent or SBC. SBC can perform MITM=20
> attack easily as extending SDES-SRTP to DTLS-SRTP for call center site

> is feasible.
> >
> > Thanks
> > Partha
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

From pravindran@sonusnet.com  Wed Mar 21 06:51:15 2012
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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: Jim Barnett <Jim.Barnett@genesyslab.com>, "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>, Hadriel Kaplan <HKaplan@acmepacket.com>
Thread-Topic: [rtcweb] Resolving RTP/SDES question in Paris
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Date: Wed, 21 Mar 2012 13:51:07 +0000
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References: <4F4759DC.7060303@ericsson.com><387F9047F55E8C42850AD6B3A7A03C6C0E1FEB69@inba-mail01.sonusnet.com><CALiegfnkYVEpmPV-zSL_4wOY-HiFZN-qJCQCiioaS=5NaqhLZw@mail.gmail.com><CAD5OKxvtOAxMBx6xDnyfTnEq76oDEm6uj1xL6wGjjrtKUAHy3g@mail.gmail.com><CABcZeBNZiotPmCfT53uEo+O0xw4xv6tXW1M_G-3A5BHuncsduA@mail.gmail.com><CAD5OKxvYOY5JZ2mYNGiH1poUBQkyOOycePFijH5H+SxtcdqujQ@mail.gmail.com><CABkgnnVe-b6Sv=R67bMJk_NQqQwdrRUn6rBm7Gu_CMcfPQwtEg@mail.gmail.com><CAD5OKxvZbEJ7sV4WPAYoQapzMR_QwAftj-oKg=ioMKHNT792wQ@mail.gmail.com><6F428EFD2B8C2F49A2FB1317291A76C113563C5A92@USNAVSXCHMBSA1.ndc.alcatel-lucent.com><CALiegf=jtkDCS_D0ZFe9UpbiadQ0vsJ+4MppQSbLr-wbaXNrfQ@mail.gmail.com><BLU169-W29E5B86F9E2C6F3126961C93420@phx.gbl><CALiegfk2aT+6Psr4nT-hG1G7eYRBfFCcT+25On2O4HfUXJ6-ng@mail.gmail.com><CAD6AjGSmi9j+sdGWPts20-iwGvGij05ek0OKYEPULC6B=aFpQg@mail.gmail.com><6F428EFD2B8C2F49A2FB1317291A76C113564482A7@USNAVSXCHMBSA1.ndc.alcatel-lucent.com><ADBB75F3-E20C-4EC4-B9C3-EF2E4BFF409C@phonefromhere.com><C! !AD5OKxvu	E V8Vbq3h7=Z gcKmREjmguvz5n-SpXr2n-EY7a_ddxg@mail.gmail.com><CALiegfk1ozOKPcDjbd3H_z2Edzh4RcZpYyJSWdw_1DJ04muQXA@mail.gmail.com><CAD5OKxu8-+0O0=eE7mD1hi=nPUpEXczGj=bRNQCQL1BW8c-c-Q@mail.gmail.com><D75A384B-0F38-4E30-8C03-12E903A69B64@acmepacket.com><E17CAD772E76C742B645BD4DC602CD8105EBE8CF@NAHALD.us.int.genesyslab.com><387F9047F55E8C42850AD6B3A7A03C6C0E1FFE23@inba-mail01.sonusnet.com><2E10EB15-7E2E-47B9-80D1-5244DDE5FDF7@acmepacket.com> <101C6067BEC68246B0C3F6843BCCC1E31296AE222A@MCHP058A.global-ad.net> <E17CAD772E76C742B645BD4DC602CD8105EBEB17@NAHALD.us.int.genesyslab.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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It will be good to add this CallCenter WebRTC usecase.

Thanks
Partha

>-----Original Message-----
>From: Jim Barnett [mailto:Jim.Barnett@genesyslab.com]
>Sent: Wednesday, March 21, 2012 7:18 PM
>To: Hutton, Andrew; Hadriel Kaplan; Ravindran, Parthasarathi
>Cc: rtcweb@ietf.org
>Subject: RE: [rtcweb] Resolving RTP/SDES question in Paris
>
>One way to look at this case would be to say that it's the server at the
>boundary of www.travel.co that is the RTCWeb endpoint,  _not_ the agent
>desktop. The boundary server is acting as an RTCWeb-to-corporate network
>gateway, and systems inside the corporate network are not affected.
>Callers see themselves as connected to www.travel.co, rather than having
>a direct RTCWeb-all-the-way connection to the agent.  I'm pretty sure
>that call centers would like to do things this way since it doesn't
>require any changes to their corporate infrastructure. (They can
>add/upgrade the gateway without changing their recording systems or
>agent desktops, etc.)
>
>I don't see anything like this in the use case document.  Is this a use
>case we want to support?
>
>- Jim
>
>-----Original Message-----
>From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
>Of Hutton, Andrew
>Sent: Tuesday, March 20, 2012 6:07 AM
>To: Hadriel Kaplan; Ravindran, Parthasarathi
>Cc: rtcweb@ietf.org
>Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
>
>Hi,
>
>Completely agree with Hadriel's statement below.
>
>In addition to wanting to know that you are connected to "www.travel.co"
>or "www.mybank.co" you as a consumer and the company really want/need
>the media to be recorded by the companies system for your own
>protection. This is going to involve some kind of MITM or as we call it
>in SIPREC a Session Recording Client (SRC) to be able to intercept the
>media and route it to a Session Recording Server (SRS).
>
>The agent's desktop device could act as the SRC but more likely it will
>be some kind of B2BUA that does this.
>
>In these scenario's you as a consumer using this service really want the
>media to be secure at least as far as reaching the bank but after that
>like any existing web based system you have to trust the bank with your
>data this is no different whether that data is speech or text that you
>entered on a web page.
>
>The recording system should be able to work with DTLS-SRTP but it will
>most likely act as a MITM so will change the DTLS fingerprint. I don't
>see that as a problem.
>
>Regards
>Andy
>
>
>
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
>> Behalf Of Hadriel Kaplan
>> Sent: 20 March 2012 03:21
>> To: Ravindran, Parthasarathi
>> Cc: rtcweb@ietf.org
>> Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
>>
>>
>> And, it should be noted, they *want* it that way.  They don't
>> generally want you to know which agent picked up the call, or that the
>
>> call got transferred from an IVR to a specific agent, or got put in a
>> queue with elevator-music announcement server, or any of that.  You
>"called"
>> Travel Co., and they answered.  What goes on behind that SBC is
>> equivalent to the back-end database stuff that goes on between the web
>
>> server and back-end systems, while all you see is the web-server your
>> browser happens to be connected to.
>>
>> The "identity" you'll get is www.travel.co, which you already had when
>
>> your browser did HTTPS cert verification of their web-server.  HTTPS
>> became the key-exchange transport for the SRTP key.  Since they
>> already proved they own the cert for www.travel.co, having them claim
>> to be www.travel.co shouldn't require yet more verification.
>>
>> -hadriel
>>
>>
>> On Mar 19, 2012, at 9:15 PM, Ravindran, Parthasarathi wrote:
>>
>> > Jim,
>> >
>> > As a customer, you won't really know whether the identity
>> > (DTLS-SRTP)
>> of call center/travel site is agent or SBC. SBC can perform MITM
>> attack easily as extending SDES-SRTP to DTLS-SRTP for call center site
>
>> is feasible.
>> >
>> > Thanks
>> > Partha
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb

From igor.faynberg@alcatel-lucent.com  Wed Mar 21 07:58:21 2012
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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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+1

Indeed, this is the only way call centers could work.

Igor

On 3/21/2012 9:51 AM, Ravindran, Parthasarathi wrote:
> It will be good to add this CallCenter WebRTC usecase.
>
> Thanks
> Partha
>
>> -----Original Message-----
>> From: Jim Barnett [mailto:Jim.Barnett@genesyslab.com]
>> Sent: Wednesday, March 21, 2012 7:18 PM
>> To: Hutton, Andrew; Hadriel Kaplan; Ravindran, Parthasarathi
>> Cc: rtcweb@ietf.org
>> Subject: RE: [rtcweb] Resolving RTP/SDES question in Paris
>>
>> One way to look at this case would be to say that it's the server at the
>> boundary of www.travel.co that is the RTCWeb endpoint,  _not_ the agent
>> desktop. The boundary server is acting as an RTCWeb-to-corporate network
>> gateway, and systems inside the corporate network are not affected.
>> Callers see themselves as connected to www.travel.co, rather than having
>> a direct RTCWeb-all-the-way connection to the agent.  I'm pretty sure
>> that call centers would like to do things this way since it doesn't
>> require any changes to their corporate infrastructure. (They can
>> add/upgrade the gateway without changing their recording systems or
>> agent desktops, etc.)
>>
>> I don't see anything like this in the use case document.  Is this a use
>> case we want to support?
>>
>> - Jim
>>
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
>> Of Hutton, Andrew
>> Sent: Tuesday, March 20, 2012 6:07 AM
>> To: Hadriel Kaplan; Ravindran, Parthasarathi
>> Cc: rtcweb@ietf.org
>> Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
>>
>> Hi,
>>
>> Completely agree with Hadriel's statement below.
>>
>> In addition to wanting to know that you are connected to "www.travel.co"
>> or "www.mybank.co" you as a consumer and the company really want/need
>> the media to be recorded by the companies system for your own
>> protection. This is going to involve some kind of MITM or as we call it
>> in SIPREC a Session Recording Client (SRC) to be able to intercept the
>> media and route it to a Session Recording Server (SRS).
>>
>> The agent's desktop device could act as the SRC but more likely it will
>> be some kind of B2BUA that does this.
>>
>> In these scenario's you as a consumer using this service really want the
>> media to be secure at least as far as reaching the bank but after that
>> like any existing web based system you have to trust the bank with your
>> data this is no different whether that data is speech or text that you
>> entered on a web page.
>>
>> The recording system should be able to work with DTLS-SRTP but it will
>> most likely act as a MITM so will change the DTLS fingerprint. I don't
>> see that as a problem.
>>
>> Regards
>> Andy
>>
>>
>>
>>> -----Original Message-----
>>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
>>> Behalf Of Hadriel Kaplan
>>> Sent: 20 March 2012 03:21
>>> To: Ravindran, Parthasarathi
>>> Cc: rtcweb@ietf.org
>>> Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
>>>
>>>
>>> And, it should be noted, they *want* it that way.  They don't
>>> generally want you to know which agent picked up the call, or that the
>>> call got transferred from an IVR to a specific agent, or got put in a
>>> queue with elevator-music announcement server, or any of that.  You
>> "called"
>>> Travel Co., and they answered.  What goes on behind that SBC is
>>> equivalent to the back-end database stuff that goes on between the web
>>> server and back-end systems, while all you see is the web-server your
>>> browser happens to be connected to.
>>>
>>> The "identity" you'll get is www.travel.co, which you already had when
>>> your browser did HTTPS cert verification of their web-server.  HTTPS
>>> became the key-exchange transport for the SRTP key.  Since they
>>> already proved they own the cert for www.travel.co, having them claim
>>> to be www.travel.co shouldn't require yet more verification.
>>>
>>> -hadriel
>>>
>>>
>>> On Mar 19, 2012, at 9:15 PM, Ravindran, Parthasarathi wrote:
>>>
>>>> Jim,
>>>>
>>>> As a customer, you won't really know whether the identity
>>>> (DTLS-SRTP)
>>> of call center/travel site is agent or SBC. SBC can perform MITM
>>> attack easily as extending SDES-SRTP to DTLS-SRTP for call center site
>>> is feasible.
>>>> Thanks
>>>> Partha
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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Subject: [rtcweb] Modular Congestion Control in rtcweb
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Dear all,

is there any plan to make the congestion control algorithm
modular for multimedia transmission via PeerConnections?

A particular congestion control module could be selected
by the connecting peers at the establishment of the
PeerConnection similarly to DCCP.

Best regards,
--
Luca De Cicco, PhD, Eng.
Politecnico di Bari
Dipartimento di Elettrotecnica ed Elettronica
Via Re David, 200 - Bari - ITALY
Office: +39 080 596 3851

From harald@alvestrand.no  Thu Mar 22 00:00:47 2012
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Subject: Re: [rtcweb] Modular Congestion Control in rtcweb
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On 03/21/2012 08:53 PM, Luca De Cicco wrote:
> Dear all,
>
> is there any plan to make the congestion control algorithm
> modular for multimedia transmission via PeerConnections?
>
> A particular congestion control module could be selected
> by the connecting peers at the establishment of the
> PeerConnection similarly to DCCP.
So far, the discussions (which have mainly been done on the 
rtp-congestion@alvestrand.no list) have focused on making a mechanism 
for reporting bandwidth estimates available, and on discussing one 
particular new congestion control mechanism based on delay estimation.

It's implicit that we'd like to try out more than one estimation 
mechanism. But it's not clear to me that this translates to "modular 
algorithms".

Please join the discussion!

                 Harald


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From: Harald Alvestrand <harald@alvestrand.no>
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To: Roman Shpount <roman@telurix.com>
References: <4F4759DC.7060303@ericsson.com>	<387F9047F55E8C42850AD6B3A7A03C6C0E1FEB69@inba-mail01.sonusnet.com>	<CALiegfnkYVEpmPV-zSL_4wOY-HiFZN-qJCQCiioaS=5NaqhLZw@mail.gmail.com>	<CAD5OKxvtOAxMBx6xDnyfTnEq76oDEm6uj1xL6wGjjrtKUAHy3g@mail.gmail.com>	<CABcZeBNZiotPmCfT53uEo+O0xw4xv6tXW1M_G-3A5BHuncsduA@mail.gmail.com>	<CAD5OKxvYOY5JZ2mYNGiH1poUBQkyOOycePFijH5H+SxtcdqujQ@mail.gmail.com>	<CABkgnnVe-b6Sv=R67bMJk_NQqQwdrRUn6rBm7Gu_CMcfPQwtEg@mail.gmail.com>	<4F64FE98.3070605@alcatel-lucent.com>	<4F685ED9.2050109@alvestrand.no>	<CAD5OKxsVp7px9bHAgxgdqPMxRgppcVUDKt8JHBhyq9qqW3pAMg@mail.gmail.com>	<4F68A4CC.9090306@alvestrand.no> <CAD5OKxuiApLKRASc2YuBfkM_8h8wGDPPQ3TdOYGum2yauidA5A@mail.gmail.com>
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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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On 03/20/2012 05:10 PM, Roman Shpount wrote:
>
> On Tue, Mar 20, 2012 at 11:39 AM, Harald Alvestrand 
> <harald@alvestrand.no <mailto:harald@alvestrand.no>> wrote:
>
>     On 03/20/2012 02:40 PM, Roman Shpount wrote:
>>     Once again, I understand how this helps in case of HTTPS, but how
>>     would this help in case of WebRTC? Media description is carried
>>     in some sort of application defined protocol (can even be
>>     transmitted over an encrypted SCTP data channel or encrypted in
>>     javaScript), so monitoring proxy cannot reliably modify it. I
>>     understand how it can be allowed to fake the signature on
>>     modified SDP by inserting a certificate in the browser, but how
>>     would the proxy get to the SDP in the first place?
>
>     "this" = inserting a trust root in the browser?
>
>     Once you trust the root I inserted, I can intercept all your HTTPS
>     connections, and thus I can replace all the Javascript you think
>     you are loading from your trusted provider with Javascript of my
>     choice.
>
>     Once your browser is running my Javascript, I can do anything the
>     original application could - with the privilleges you granted to
>     that application.
>
>     I think that's more or less "game over" when it comes to security.
>
>
> "this" is inserting the root certificate. It is a game over as far as 
> security is concerned, but the rest of your email completely missing 
> my point.
>
> If, as an owner of a computer, you want to enable media monitoring 
> with an existing standard you need to know exactly how each 
> application is implemented. This is a solution for hacker but not the 
> normal enterprise. In order to have monitoring support in the 
> enterprise environment you need an ability to force the browser to 
> proxy all the media through a monitoring application without any 
> application modification, via browser settings alone.

OK, you're worried about supporting the interceptor, not about 
preventing the interception... right?

My gut feeling is that the legitimate interceptor has to be able to say 
"only these apps are used for business purposes" - for those, he should 
be able to know how to do it.

I don't think it's possible even in theory to intercept all possible 
applications, even with your own root-cert in the user's browser. It's 
probably even possible to prove it by showing that it's equivalent to 
solving the halting problem.

So I guess administrators who have the need just have to say "these 
applications and no other".

                   Harald



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      http-equiv="Content-Type">
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    On 03/20/2012 05:10 PM, Roman Shpount wrote:
    <blockquote
cite="mid:CAD5OKxuiApLKRASc2YuBfkM_8h8wGDPPQ3TdOYGum2yauidA5A@mail.gmail.com"
      type="cite"><br clear="all">
      <div class="gmail_quote">On Tue, Mar 20, 2012 at 11:39 AM, Harald
        Alvestrand <span dir="ltr">&lt;<a moz-do-not-send="true"
            href="mailto:harald@alvestrand.no">harald@alvestrand.no</a>&gt;</span>
        wrote:<br>
        <blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt
          0.8ex; border-left: 1px solid rgb(204, 204, 204);
          padding-left: 1ex;">
          <div bgcolor="#ffffff" text="#000000">
            <div class="im"> On 03/20/2012 02:40 PM, Roman Shpount
              wrote:
              <blockquote type="cite">Once again, I understand how this
                helps in case of HTTPS, but how would this help in case
                of WebRTC? Media description is carried in some sort of
                application defined protocol (can even be transmitted
                over an encrypted SCTP data channel or encrypted in
                javaScript), so monitoring proxy cannot reliably modify
                it. I understand how it can be allowed to fake the
                signature on modified SDP by inserting a certificate in
                the browser, but how would the proxy get to the SDP in
                the first place?<br clear="all">
              </blockquote>
              <br>
            </div>
            "this" = inserting a trust root in the browser?<br>
            <br>
            Once you trust the root I inserted, I can intercept all your
            HTTPS connections, and thus I can replace all the Javascript
            you think you are loading from your trusted provider with
            Javascript of my choice.<br>
            <br>
            Once your browser is running my Javascript, I can do
            anything the original application could - with the
            privilleges you granted to that application.<br>
            <br>
            I think that's more or less "game over" when it comes to
            security.
            <div class="im"><br>
            </div>
          </div>
        </blockquote>
      </div>
      <br>
      "this" is inserting the root certificate. It is a game over as far
      as security is concerned, but the rest of your email completely
      missing my point.<br>
      <br>
      If, as an owner of a computer, you want to enable media monitoring
      with an existing standard you need to know exactly how each
      application is implemented. This is a solution for hacker but not
      the normal enterprise. In order to have monitoring support in the
      enterprise environment you need an ability to force the browser to
      proxy all the media through a monitoring application without any
      application modification, via browser settings alone.<br>
    </blockquote>
    <br>
    OK, you're worried about supporting the interceptor, not about
    preventing the interception... right?<br>
    <br>
    My gut feeling is that the legitimate interceptor has to be able to
    say "only these apps are used for business purposes" - for those, he
    should be able to know how to do it.<br>
    <br>
    I don't think it's possible even in theory to intercept all possible
    applications, even with your own root-cert in the user's browser.
    It's probably even possible to prove it by showing that it's
    equivalent to solving the halting problem.<br>
    <br>
    So I guess administrators who have the need just have to say "these
    applications and no other".<br>
    <br>
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Harald<br>
    <br>
    <br>
  </body>
</html>

--------------090404050608040300080802--

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In-Reply-To: <4F6AECC6.8020004@alvestrand.no>
References: <4F4759DC.7060303@ericsson.com> <387F9047F55E8C42850AD6B3A7A03C6C0E1FEB69@inba-mail01.sonusnet.com> <CALiegfnkYVEpmPV-zSL_4wOY-HiFZN-qJCQCiioaS=5NaqhLZw@mail.gmail.com> <CAD5OKxvtOAxMBx6xDnyfTnEq76oDEm6uj1xL6wGjjrtKUAHy3g@mail.gmail.com> <CABcZeBNZiotPmCfT53uEo+O0xw4xv6tXW1M_G-3A5BHuncsduA@mail.gmail.com> <CAD5OKxvYOY5JZ2mYNGiH1poUBQkyOOycePFijH5H+SxtcdqujQ@mail.gmail.com> <CABkgnnVe-b6Sv=R67bMJk_NQqQwdrRUn6rBm7Gu_CMcfPQwtEg@mail.gmail.com> <4F64FE98.3070605@alcatel-lucent.com> <4F685ED9.2050109@alvestrand.no> <CAD5OKxsVp7px9bHAgxgdqPMxRgppcVUDKt8JHBhyq9qqW3pAMg@mail.gmail.com> <4F68A4CC.9090306@alvestrand.no> <CAD5OKxuiApLKRASc2YuBfkM_8h8wGDPPQ3TdOYGum2yauidA5A@mail.gmail.com> <4F6AECC6.8020004@alvestrand.no>
Date: Thu, 22 Mar 2012 10:18:56 -0400
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From: Roman Shpount <roman@telurix.com>
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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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On Thu, Mar 22, 2012 at 5:11 AM, Harald Alvestrand <harald@alvestrand.no>wrote:

> **
> OK, you're worried about supporting the interceptor, not about preventing
> the interception... right?
>
> My gut feeling is that the legitimate interceptor has to be able to say
> "only these apps are used for business purposes" - for those, he should be
> able to know how to do it.
>
> I don't think it's possible even in theory to intercept all possible
> applications, even with your own root-cert in the user's browser. It's
> probably even possible to prove it by showing that it's equivalent to
> solving the halting problem.
>
> So I guess administrators who have the need just have to say "these
> applications and no other".
>
>
Yes, I was worried about supporting the interceptor. This is all related to
my question of support of WebRTC applications in military, prisons, or
financial organizations. I think WebRTC would be completely disabled in
such locations unless web browser can be configured to confirm to some sort
of communications and monitoring policy. I do not think enabling only
certain applications is a sustainable solution, since application can only
be enabled based on its URL, but this in no way implies that the actual
protocol used for signaling exchange by this application will stay the
same. In the best case this will result in tens of specialized monitoring
rules that will need to be maintained. In the worst case, WebRTC would be
simply disabled. At this point the only workable solution that I see is
some sort of media proxy protocol.
_____________
Roman Shpount

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<br clear=3D"all"><div class=3D"gmail_quote">On Thu, Mar 22, 2012 at 5:11 A=
M, Harald Alvestrand <span dir=3D"ltr">&lt;<a href=3D"mailto:harald@alvestr=
and.no">harald@alvestrand.no</a>&gt;</span> wrote:<br><blockquote class=3D"=
gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-=
left:1ex">
<u></u>

 =20
   =20
   =20
 =20
  <div bgcolor=3D"#ffffff" text=3D"#000000"><div><div></div>OK, you&#39;re =
worried about supporting the interceptor, not about
    preventing the interception... right?<br></div>
    <br>
    My gut feeling is that the legitimate interceptor has to be able to
    say &quot;only these apps are used for business purposes&quot; - for th=
ose, he
    should be able to know how to do it.<br>
    <br>
    I don&#39;t think it&#39;s possible even in theory to intercept all pos=
sible
    applications, even with your own root-cert in the user&#39;s browser.
    It&#39;s probably even possible to prove it by showing that it&#39;s
    equivalent to solving the halting problem.<br>
    <br>
    So I guess administrators who have the need just have to say &quot;thes=
e
    applications and no other&quot;.<br><font color=3D"#888888">
    </font><br></div></blockquote></div><br>Yes, I was worried about suppor=
ting the interceptor. This is all related to my question of support of WebR=
TC applications in military, prisons, or financial organizations. I think W=
ebRTC would be completely disabled in such locations unless web browser can=
 be configured to confirm to some sort of communications and monitoring pol=
icy. I do not think enabling only certain applications is a sustainable sol=
ution, since application can only be enabled based on its URL, but this in =
no way implies that the actual protocol used for signaling exchange by this=
 application will stay the same. In the best case this will result in tens =
of specialized monitoring rules that will need to be maintained. In the wor=
st case, WebRTC would be simply disabled. At this point the only workable s=
olution that I see is some sort of media proxy protocol.<br>
_____________<br>Roman Shpount<br>
<br><br>

--e89a8ff1c35eb2941a04bbd5968e--

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From: Martin Thomson <martin.thomson@gmail.com>
To: Roman Shpount <roman@telurix.com>
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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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On 22 March 2012 07:18, Roman Shpount <roman@telurix.com> wrote:
> Yes, I was worried about supporting the interceptor.

Please see: http://tools.ietf.org/html/rfc2804

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--047d7b10d17b7a286804bbd6b7c1
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On Thu, Mar 22, 2012 at 11:33 AM, Martin Thomson
<martin.thomson@gmail.com>wrote:

> On 22 March 2012 07:18, Roman Shpount <roman@telurix.com> wrote:
> > Yes, I was worried about supporting the interceptor.
>
> Please see: http://tools.ietf.org/html/rfc2804
>

And I did see it. I suggest you do the same. I am not talking about
providing a back door into the system to allow wiretapping. I am talking
about a user voluntarily enabling network based recording of its own WebRTC
traffic. So, if I am an enterprise, and I own the computer, I should be
able to enforce a recording and communications policy on it. Enterprise can
do it now with HTTP/HTTPS, why not WebRTC?
_____________
Roman Shpount

--047d7b10d17b7a286804bbd6b7c1
Content-Type: text/html; charset=ISO-8859-1
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<br clear=3D"all"><br><div class=3D"gmail_quote">On Thu, Mar 22, 2012 at 11=
:33 AM, Martin Thomson <span dir=3D"ltr">&lt;<a href=3D"mailto:martin.thoms=
on@gmail.com">martin.thomson@gmail.com</a>&gt;</span> wrote:<br><blockquote=
 class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc soli=
d;padding-left:1ex">
<div class=3D"im">On 22 March 2012 07:18, Roman Shpount &lt;<a href=3D"mail=
to:roman@telurix.com">roman@telurix.com</a>&gt; wrote:<br>
&gt; Yes, I was worried about supporting the interceptor.<br>
<br>
</div>Please see: <a href=3D"http://tools.ietf.org/html/rfc2804" target=3D"=
_blank">http://tools.ietf.org/html/rfc2804</a><br>
</blockquote></div><br>And I did see it. I suggest you do the same. I am no=
t talking about providing a back door into the system to allow wiretapping.=
 I am talking about a user voluntarily enabling network based recording of =
its own WebRTC traffic. So, if I am an enterprise, and I own the computer, =
I should be able to enforce a recording and communications policy on it. En=
terprise can do it now with HTTP/HTTPS, why not WebRTC? <br>
_____________<br>Roman Shpount<br>
<br>

--047d7b10d17b7a286804bbd6b7c1--

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On 22 March 2012 08:39, Roman Shpount <roman@telurix.com> wrote:
> So, if I am an enterprise, and I own the computer, I should be able
> to enforce a recording and communications policy on it. Enterprise can do it
> now with HTTP/HTTPS, why not WebRTC?

If you can install trust anchors, what more do you need?  A set of instructions?

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Date: Thu, 22 Mar 2012 12:13:33 -0400
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From: Roman Shpount <roman@telurix.com>
To: Martin Thomson <martin.thomson@gmail.com>
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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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On Thu, Mar 22, 2012 at 11:55 AM, Martin Thomson
<martin.thomson@gmail.com>wrote:

> On 22 March 2012 08:39, Roman Shpount <roman@telurix.com> wrote:
> > So, if I am an enterprise, and I own the computer, I should be able
> > to enforce a recording and communications policy on it. Enterprise can
> do it
> > now with HTTP/HTTPS, why not WebRTC?
>
> If you can install trust anchors, what more do you need?  A set of
> instructions?
>

Ok, repeating my previous email:

If, as an owner of a computer, you want to enable media monitoring with an
existing standard you need to know exactly how each application is
implemented. This is a solution for hacker but not the normal enterprise.
In order to have monitoring support in the enterprise environment you need
an ability to force the browser to proxy all the media through a monitoring
application without any application modification, via browser settings
alone. For the monitoring proxy to work it would need some sort of network
protocol to communicate with the browser (similar how HTTP proxy needs a
network protocol to work with the browser). I would expect what's needed is
some sort of network based protocol to send the SDP to a proxy, get
modified SDP back and pass it to JavaScript (or take the SDP from
JavaScript, through the proxy and to the media stack).
_____________
Roman Shpount

--047d7b10d17b8cb87504bbd7305c
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<br clear=3D"all"><div class=3D"gmail_quote">On Thu, Mar 22, 2012 at 11:55 =
AM, Martin Thomson <span dir=3D"ltr">&lt;<a href=3D"mailto:martin.thomson@g=
mail.com">martin.thomson@gmail.com</a>&gt;</span> wrote:<br><blockquote cla=
ss=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;pa=
dding-left:1ex">
<div class=3D"im">On 22 March 2012 08:39, Roman Shpount &lt;<a href=3D"mail=
to:roman@telurix.com">roman@telurix.com</a>&gt; wrote:<br>
&gt; So, if I am an enterprise, and I own the computer, I should be able<br=
>
&gt; to enforce a recording and communications policy on it. Enterprise can=
 do it<br>
&gt; now with HTTP/HTTPS, why not WebRTC?<br>
<br>
</div>If you can install trust anchors, what more do you need? =A0A set of =
instructions?<br>
</blockquote></div><br>Ok, repeating my previous email:<br><br>If, as an ow=
ner of a computer, you want to enable media monitoring with=20
an existing standard you need to know exactly how each application is=20
implemented. This is a solution for hacker but not the normal=20
enterprise. In order to have monitoring support in the enterprise=20
environment you need an ability to force the browser to proxy all the=20
media through a monitoring application without any application=20
modification, via browser settings alone. For the monitoring proxy to work =
it would need some sort of network=20
protocol to communicate with the browser (similar how HTTP proxy needs a
 network protocol to work with the browser). I would expect what&#39;s=20
needed is some sort of network based protocol to send the SDP to a=20
proxy, get modified SDP back and pass it to JavaScript (or take the SDP=20
from JavaScript, through the proxy and to the media stack).<br>____________=
_<br>Roman Shpount<br>
<br><br>

--047d7b10d17b8cb87504bbd7305c--

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To: Roman Shpount <roman@telurix.com>
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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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On Thu, Mar 22, 2012 at 9:13 AM, Roman Shpount <roman@telurix.com> wrote:
>
>
>> If you can install trust anchors, what more do you need? =A0A set of
>> instructions?
>
>
> Ok, repeating my previous email:
>
> If, as an owner of a computer, you want to enable media monitoring with a=
n
> existing standard you need to know exactly how each application is
> implemented.

Given that we are talking about downloadable applications, this is
going to require some doing.  You're not going to know how each
application is implemented without vetting them in some way, which
seems to result in the whitelisting exercise that Harald pointed at
earlier.

>This is a solution for hacker but not the normal enterprise. In
> order to have monitoring support in the enterprise environment you need a=
n
> ability to force the browser to proxy all the media through a monitoring
> application without any application modification, via browser settings
> alone.

But you are not suggesting, I hope, that all browsers must have such a
setting, and clearly we would not suggest that it must always be on.
In the environment you describe you're basically going to have to have
both a trust anchor that allows you to man-in-the-middle all traffic
and a network configuration that makes the path through the
man-in-the-middle the most effective one.  Neither of those is within
the charter of this working group, in my personal opinion.  If you
would like me to caucus with the other chairs to see whether it is our
opinion with our hats on, please let me know.

regards,

Ted Hardie

From roman@telurix.com  Thu Mar 22 09:37:02 2012
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On Thu, Mar 22, 2012 at 12:24 PM, Ted Hardie <ted.ietf@gmail.com> wrote:

>
> But you are not suggesting, I hope, that all browsers must have such a
> setting, and clearly we would not suggest that it must always be on.
> In the environment you describe you're basically going to have to have
> both a trust anchor that allows you to man-in-the-middle all traffic
> and a network configuration that makes the path through the
> man-in-the-middle the most effective one.  Neither of those is within
> the charter of this working group, in my personal opinion.  If you
> would like me to caucus with the other chairs to see whether it is our
> opinion with our hats on, please let me know.
>
>
What I am suggesting is that there is a need for some sort of network
protocol to enable an external network element to monitor/modify SDP media
descriptions being passed to and from WebRTC. I would assume that
implementation and use of this protocol would be optional to a web browser
(similar how implementation and use of a SOCKS HTTP proxy is optional to a
web browser). In a sense, this should not be any different then any of the
HTTP proxy protocols. I do agree that the whole trust anchor issue is
outside of the group charter, especially since it is already supported and
implemented by the web browsers, but I would like to ask the chairs to see
if media proxy protocol actually falls under the WebRTC charter.

Thank You,
_____________
Roman Shpount

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<br clear=3D"all"><div class=3D"gmail_quote">On Thu, Mar 22, 2012 at 12:24 =
PM, Ted Hardie <span dir=3D"ltr">&lt;<a href=3D"mailto:ted.ietf@gmail.com">=
ted.ietf@gmail.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quot=
e" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<br>
But you are not suggesting, I hope, that all browsers must have such a<br>
setting, and clearly we would not suggest that it must always be on.<br>
In the environment you describe you&#39;re basically going to have to have<=
br>
both a trust anchor that allows you to man-in-the-middle all traffic<br>
and a network configuration that makes the path through the<br>
man-in-the-middle the most effective one. =A0Neither of those is within<br>
the charter of this working group, in my personal opinion. =A0If you<br>
would like me to caucus with the other chairs to see whether it is our<br>
opinion with our hats on, please let me know.<br>
<br></blockquote></div><br>What I am suggesting is that there is a need for=
 some sort of network protocol to enable an external network element to mon=
itor/modify SDP media descriptions being passed to and from WebRTC. I would=
 assume that implementation and use of this protocol would be optional to a=
 web browser (similar how implementation and use of a SOCKS HTTP proxy is o=
ptional to a web browser). In a sense, this should not be any different the=
n any of the HTTP proxy protocols. I do agree that the whole trust anchor i=
ssue is outside of the group charter, especially since it is already suppor=
ted and implemented by the web browsers, but I would like to ask the chairs=
 to see if media proxy protocol actually falls under the WebRTC charter. <b=
r>
<br>Thank You,<br>_____________<br>Roman Shpount<br>
<br><br>

--047d7b15a68b4aff0a04bbd78473--

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Eric,

A question:  For the case of BrowserID, I understand that a client gets 
a certificate from an IdP.   If so, why you and I would not use the 
certificates from our respective IdPs in order to authenticate each 
other  in DTLS?

Igor




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From: Ted Hardie <ted.ietf@gmail.com>
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On Thu, Mar 22, 2012 at 9:36 AM, Roman Shpount <roman@telurix.com> wrote:

> I do agree that the whole trust anchor issue is
> outside of the group charter, especially since it is already supported and
> implemented by the web browsers, but I would like to ask the chairs to see
> if media proxy protocol actually falls under the WebRTC charter.
>

Okay, what language in the current charter do you see as  covering
this activity?

regards,

Ted Hardie

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From: Eric Rescorla <ekr@rtfm.com>
Date: Thu, 22 Mar 2012 18:53:42 +0100
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Good question.

1. BrowserID certificates aren't X.509, and that's all the TLS/DTLS will do=
.
2. This lets us be agnostic about IdP mechanisms without having to change
TLS every time we add a new one.

-Ekr


On Thu, Mar 22, 2012 at 6:22 PM, Igor Faynberg
<igor.faynberg@alcatel-lucent.com> wrote:
> Eric,
>
> A question: =A0For the case of BrowserID, I understand that a client gets=
 a
> certificate from an IdP. =A0 If so, why you and I would not use the
> certificates from our respective IdPs in order to authenticate each other
> =A0in DTLS?
>
> Igor
>
>
>

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Thanks! Now I understand.

My only follow-up question (and that could probably wait until the next 
week) is why not make BrowserID certificate adhere to X.509.  If we 
issue self-signed X.509 certificates ourselves, what is in the way of 
any IdP doing just that?

Igor

On 3/22/2012 1:53 PM, Eric Rescorla wrote:
> Good question.
>
> 1. BrowserID certificates aren't X.509, and that's all the TLS/DTLS will do.
> 2. This lets us be agnostic about IdP mechanisms without having to change
> TLS every time we add a new one.
>
> -Ekr
>
>
> On Thu, Mar 22, 2012 at 6:22 PM, Igor Faynberg
> <igor.faynberg@alcatel-lucent.com>  wrote:
>> Eric,
>>
>> A question:  For the case of BrowserID, I understand that a client gets a
>> certificate from an IdP.   If so, why you and I would not use the
>> certificates from our respective IdPs in order to authenticate each other
>>   in DTLS?
>>
>> Igor
>>
>>
>>

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From: "Dan Wing" <dwing@cisco.com>
To: "'Roman Shpount'" <roman@telurix.com>, "'Harald Alvestrand'" <harald@alvestrand.no>, "'Ted Hardie'" <ted.ietf@gmail.com>
References: <4F4759DC.7060303@ericsson.com>	<387F9047F55E8C42850AD6B3A7A03C6C0E1FEB69@inba-mail01.sonusnet.com>	<CALiegfnkYVEpmPV-zSL_4wOY-HiFZN-qJCQCiioaS=5NaqhLZw@mail.gmail.com>	<CAD5OKxvtOAxMBx6xDnyfTnEq76oDEm6uj1xL6wGjjrtKUAHy3g@mail.gmail.com>	<CABcZeBNZiotPmCfT53uEo+O0xw4xv6tXW1M_G-3A5BHuncsduA@mail.gmail.com>	<CAD5OKxvYOY5JZ2mYNGiH1poUBQkyOOycePFijH5H+SxtcdqujQ@mail.gmail.com>	<CABkgnnVe-b6Sv=R67bMJk_NQqQwdrRUn6rBm7Gu_CMcfPQwtEg@mail.gmail.com>	<4F64FE98.3070605@alcatel-lucent.com>	<4F685ED9.2050109@alvestrand.no>	<CAD5OKxsVp7px9bHAgxgdqPMxRgppcVUDKt8JHBhyq9qqW3pAMg@mail.gmail.com>	<4F68A4CC.9090306@alvestrand.no>	<CAD5OKxuiApLKRASc2YuBfkM_8h8wGDPPQ3TdOYGum2yauidA5A@mail.gmail.com>	<4F6AECC6.8020004@alvestrand.no> <CAD5OKxsSUeMFYXZMZVqQFWdeEB=30HJuJ=mP9GaYkksBmp1mOA@mail.gmail.com>
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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Roman Shpount
> Sent: Thursday, March 22, 2012 7:19 AM
> To: Harald Alvestrand
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
...
> Yes, I was worried about supporting the interceptor. This is all
> related to my question of support of WebRTC applications in military,
> prisons, or financial organizations. I think WebRTC would be completely
> disabled in such locations unless web browser can be configured to
> confirm to some sort of communications and monitoring policy. I do not
> think enabling only certain applications is a sustainable solution,
> since application can only be enabled based on its URL, but this in no
> way implies that the actual protocol used for signaling exchange by
> this application will stay the same. In the best case this will result
> in tens of specialized monitoring rules that will need to be
> maintained. In the worst case, WebRTC would be simply disabled. At this
> point the only workable solution that I see is some sort of media proxy
> protocol.

The SIPREC working group, formed in March 2010, has solutions for
this problem.  The problem is not created by RTCWEB.  Disabling 
encryption is not necessary -- nor desirable.  "Jails" get brought up
in this context often, so using that same example, consider a prisoner 
at a jail communicating with their lawyer.  The prisoner needs secure 
communications with their lawyer without the risk of the 
communications being leaked to the press by anyone along the media 
path.  For details see http://tools.ietf.org/wg/siprec/

We will have a brietf presentation on SIPREC during my session
at RTCWEB, explaining how it can work with RTCWEB.

-d




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--047d7b15aee96726bd04bbdefbd1
Content-Type: text/plain; charset=ISO-8859-1

On Thu, Mar 22, 2012 at 1:30 PM, Ted Hardie <ted.ietf@gmail.com> wrote:

> Okay, what language in the current charter do you see as  covering
> this activity?
>
>
I would say that this activity would be covered by:

2. Define a security model that describes the security and privacy
goals and specifies the security protocol mechanisms necessary
to achieve those goals.

3. Define the solution - protocols and API requirements - for
firewall and NAT traversal.

I do consider ability to enforce the enterprise or location communications
policy on WebRTC clients to be one of the requirements for security and
privacy. I expect that unless ability to enforce such communication policy
within enterprises or other organizations is provided, WebRTC would be
simply disabled.

In practical terms, I expect that an organization that enforces a
communication policy disables routing to public internet and only allows
communications via a company controlled HTTP proxy. Such setup would
effectively prevent WebRTC from operations. What I am suggesting is to
provide a complimentary protocol for inspection and modification of WebRTC
SDP messages which would allow WebRTC enabled browser to control an
enterprise edge media proxy and would allow WebRTC media traffic to
traverse the enterprise firewall and to operate in such environment.

I do consider such functionality as installing trust anchors, SDP
resigning, media re-encoding and media recording by the edge proxy to be
implementation details and to be outside of the scope of this chapter.
_____________
Roman Shpount

--047d7b15aee96726bd04bbdefbd1
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<br clear=3D"all"><div class=3D"gmail_quote">On Thu, Mar 22, 2012 at 1:30 P=
M, Ted Hardie <span dir=3D"ltr">&lt;<a href=3D"mailto:ted.ietf@gmail.com">t=
ed.ietf@gmail.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote=
" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Okay, what language in the current charter do you see as =A0covering<br>
this activity?<br>
<br></blockquote></div><br>I would say that this activity would be covered =
by:<br><br>  2.  Define a security model that describes the security and pr=
ivacy<br>      goals and specifies the security protocol mechanisms necessa=
ry<br>
      to achieve those goals.
<br><br>  3.  Define the solution - protocols and API requirements - for<br=
>      firewall and NAT traversal.<br><br>I do consider ability to enforce =
the enterprise or location communications policy on WebRTC clients to be on=
e of the requirements for security and privacy. I expect that unless abilit=
y to enforce such communication policy within enterprises or other organiza=
tions is provided, WebRTC would be simply disabled. <br>
<br>In practical terms, I expect that an organization that enforces a commu=
nication policy disables routing to public internet and only allows communi=
cations via a company controlled HTTP proxy. Such setup would effectively p=
revent WebRTC from operations. What I am suggesting is to provide a complim=
entary protocol for inspection and modification of WebRTC SDP messages whic=
h would allow WebRTC enabled browser to control an enterprise edge media pr=
oxy and would allow WebRTC media traffic to traverse the enterprise firewal=
l and to operate in such environment.<br>
<br>I do consider such functionality as installing trust anchors, SDP resig=
ning, media re-encoding and media recording by the edge proxy to be impleme=
ntation details and to be outside of the scope of this chapter.<br>________=
_____<br>
Roman Shpount<br>
<br>

--047d7b15aee96726bd04bbdefbd1--

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comments inline=85

On Mar 22, 2012, at 9:31 PM, Roman Shpount wrote:

> I would say that this activity would be covered by:

Did you mean to have a #1 before #2 below?  Your email was missing a #1.

> 2. Define a security model that describes the security and privacy
> goals and specifies the security protocol mechanisms necessary
> to achieve those goals.=20

We've got that:
http://tools.ietf.org/html/draft-ietf-rtcweb-security
http://tools.ietf.org/html/draft-ietf-rtcweb-security-arch


> 3. Define the solution - protocols and API requirements - for
> firewall and NAT traversal.

We've got that, though not in one doc:
http://tools.ietf.org/html/draft-ietf-rtcweb-overview
http://tools.ietf.org/html/draft-ietf-rtcweb-use-cases-and-requirements
http://tools.ietf.org/html/draft-ietf-rtcweb-jsep


> I do consider ability to enforce the enterprise or location communication=
s policy on WebRTC clients to be one of the requirements for security and p=
rivacy.

OK, so what you're really saying is we have #2 and #3, but you don't agree =
they cover the needs, and you want more added to #2 (and ultimately #3), ri=
ght?


> I expect that unless ability to enforce such communication policy within =
enterprises or other organizations is provided, WebRTC would be simply disa=
bled.=20
> In practical terms, I expect that an organization that enforces a communi=
cation policy disables routing to public internet and only allows communica=
tions via a company controlled HTTP proxy. Such setup would effectively pre=
vent WebRTC from operations. What I am suggesting is to provide a complimen=
tary protocol for inspection and modification of WebRTC SDP messages which =
would allow WebRTC enabled browser to control an enterprise edge media prox=
y and would allow WebRTC media traffic to traverse the enterprise firewall =
and to operate in such environment.

Well... if they really can't enforce their policies, then by definition the=
y *should* disable it.  That's a feature, not a bug.  And luckily it *can* =
be disabled, which is also a feature of WebRTC. =20

Having said that, I don't understand why you think an organization that has=
 strict monitoring policies can't monitor and enforce WebRTC traffic, both =
the HTTP *and* the media/data channels.  They *can* do it.  It's just not "=
automagic", and not cheap.  But that's ok - I don't think we need to optimi=
ze the deployment complexity or cost structure for that model.

-hadriel


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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Harald Alvestrand <harald@alvestrand.no>
Thread-Topic: [rtcweb] SDES vs DTLS-SRTP revisited
Thread-Index: AQHNCK+9SrtHmQaTeUW0LPb8el1DRw==
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] SDES vs DTLS-SRTP revisited
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On Mar 20, 2012, at 12:00 PM, Harald Alvestrand wrote:

> On 03/20/2012 04:44 PM, Hadriel Kaplan wrote:
>> On Mar 20, 2012, at 6:30 AM, Harald Alvestrand wrote:
>>=20
>>> I don't get this scenario. If Alice calls Bob using two different gatew=
ays, won't she go through a credentials and fingerprint exchange with the g=
ateways?
>>> In that case, wouldn't the fingerprint belong to the gateway?
>> Yes, but the statement being made was in the context of Alice calling Bo=
b, them seeing their browsers claim DTLS-SRTP "secured" or whatever, and th=
em being super-geeks and checking the detailed info of what the actual DTLS=
 fingerprints were, and finding they don't both see the same fingerprints..=
. and that they would thus believe there was either a software bug or a mal=
icious middleman.
> So we've uncovered a requirement here: When a fingerprint is shown, it sh=
ould clearly identify what it is the fingerprint of - in this case, the gat=
eway that relays the call to Bob, not Bob.

Yup.  So long as no gateways are named "Bob", of course, or no humans are n=
amed "Gateway". ;)

-hadriel


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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Roman Shpount <roman@telurix.com>
Thread-Topic: [rtcweb] Resolving RTP/SDES question in Paris
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On Mar 20, 2012, at 9:40 AM, Roman Shpount wrote:

> Once again, I understand how this helps in case of HTTPS, but how would t=
his help in case of WebRTC? Media description is carried in some sort of ap=
plication defined protocol (can even be transmitted over an encrypted SCTP =
data channel or encrypted in javaScript), so monitoring proxy cannot reliab=
ly modify it. I understand how it can be allowed to fake the signature on m=
odified SDP by inserting a certificate in the browser, but how would the pr=
oxy get to the SDP in the first place?

I was trying to follow this thread backwards from today, to try to grok how=
 you got to the conclusion that an organization with strict media monitorin=
g policies wouldn't be able to do so for WebRTC.  I think the above email i=
s where I get disconnected form the train of thought.

If, as in another email you sent, an organization such as the military or j=
ail or whatever controls its network such that they use the whitelist model=
 of only authorized applications allowed our of their network, and everythi=
ng goes through the HTTP proxy, then the above statement of:

> Media description is carried in some sort of application defined protocol=
 (can even be transmitted over an encrypted SCTP data channel or encrypted =
in javaScript), so monitoring proxy cannot reliably modify it.

=85 makes no sense to me.

If you were preventing unknown applications, you would not allow unknown UD=
P or TCP destination ports to be used to get out of the network.  The HTTP =
Proxy has to (1) know your web application wants to use another set of port=
s, and (2) has to allow it.  Ergo, there can be no "encrypted SCTP data cha=
nnel" that was not previously authorized; nor would encrypting SDP-type-inf=
o in Javascript help you one iota, because you won't be able to actually us=
e it to send media, since the HTTP proxy is not opening new ports for you t=
o reach the Internet.

So I don't understand what the problem is.  If you run a super-tight networ=
k like that and you *do not* want WebRTC to work because you can't monitor =
it, then by design it won't work.  If you run a super-tight network like th=
at and you *do* want WebRTC to work, then you either have to be willing to =
install custom browsers/apps, or your HTTP Proxy has to be able to see the =
HTTP traffic and correctly parse the Javascript-passed data and be a media =
b2bua.  That's a really nasty set of constraints, especially trying to grok=
 the javascript enough to be a media b2bua, and undoubtedly means it won't =
work for many WebRTC websites.  But that's arguably the right thing to happ=
en - downloading javascript is akin to downloading a new application for ea=
ch website, and again by definition this is a network that only wants to al=
low applications it wants to allow, so that's what it gets: per-application=
 control, and thus requires per-application knowledge/support.  Anything yo=
ur proxy doesn't understand doesn't work, but again that's kinda the point =
of a super-tight policy.

Personally, if I were running such an org, and I don't know if it's possibl=
e, but I'd go get some form of rootkit/kernel-mod to intercept audio/video/=
keystrokes well below the user layer, so that you wouldn't have to worry wh=
at application was running - just intercept the audio/screen/keyboard somew=
here near the drivers.  Presumably some companies make such things, for jus=
t such organizations? (sort of like the equivalent of CarrierIQ)  That way =
it doesn't matter whether it's WebRTC, Flash, or native apps.

-hadriel


From pravindran@sonusnet.com  Thu Mar 22 23:40:12 2012
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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: Dan Wing <dwing@cisco.com>, 'Roman Shpount' <roman@telurix.com>, "'Harald Alvestrand'" <harald@alvestrand.no>, 'Ted Hardie' <ted.ietf@gmail.com>
Thread-Topic: [rtcweb] Resolving RTP/SDES question in Paris
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Even in case disabling encryption is not the desired behavior, I don't see =
the problem in using RTP-IPSec instead of SRTP-DTLS. I understand that web-=
browser has no standard way to identify whether plain IP or IPsec in Layer =
3. So, it shall be configured by browser-user and it shall be considered as=
 user consent. This approach helps in case in avoiding double encryption te=
chnically wherein IPSec is mandated. I know the argument that double encryp=
tion is not problem in the endpoint as it has huge CPU and memory resources=
 but in case web-browser as part of access devices, it will have impact on =
performance. My issue is with mandating to use SRTP-DTLS in WebRTC and not =
with mandate to implement SRTP-DTLS.

Also, AFAIK HTTPS certificate is not free of cost. Whereever the cost matte=
rs like social welfare site, educational site and some free blogs, the tend=
ency will be towards HTTP and user of those sites will have the consent to =
use RTP.=20

Thanks
Partha

>-----Original Message-----
>From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
>Of Dan Wing
>Sent: Friday, March 23, 2012 4:14 AM
>To: 'Roman Shpount'; 'Harald Alvestrand'; 'Ted Hardie'
>Cc: rtcweb@ietf.org
>Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
>
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
>> Behalf Of Roman Shpount
>> Sent: Thursday, March 22, 2012 7:19 AM
>> To: Harald Alvestrand
>> Cc: rtcweb@ietf.org
>> Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
>...
>> Yes, I was worried about supporting the interceptor. This is all
>> related to my question of support of WebRTC applications in military,
>> prisons, or financial organizations. I think WebRTC would be
>> completely disabled in such locations unless web browser can be
>> configured to confirm to some sort of communications and monitoring
>> policy. I do not think enabling only certain applications is a
>> sustainable solution, since application can only be enabled based on
>> its URL, but this in no way implies that the actual protocol used for
>> signaling exchange by this application will stay the same. In the best
>> case this will result in tens of specialized monitoring rules that
>> will need to be maintained. In the worst case, WebRTC would be simply
>> disabled. At this point the only workable solution that I see is some
>> sort of media proxy protocol.
>
>The SIPREC working group, formed in March 2010, has solutions for this
>problem.  The problem is not created by RTCWEB.  Disabling encryption is
>not necessary -- nor desirable.  "Jails" get brought up in this context
>often, so using that same example, consider a prisoner at a jail
>communicating with their lawyer.  The prisoner needs secure
>communications with their lawyer without the risk of the communications
>being leaked to the press by anyone along the media path.  For details
>see http://tools.ietf.org/wg/siprec/
>
>We will have a brietf presentation on SIPREC during my session at
>RTCWEB, explaining how it can work with RTCWEB.
>
>-d
>
>
>
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb

From stefan.lk.hakansson@ericsson.com  Fri Mar 23 04:11:36 2012
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From: Stefan Hakansson LK <stefan.lk.hakansson@ericsson.com>
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Subject: [rtcweb] On video codec for rtcweb
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I think most agrees to that it would be advantageous if there could be 
an agreement on mandatory to implement codecs (at least one for audio 
and at least one for video), for interoperability between 
implementations (among other things).

I would like to discuss a bit on mandatory to implement video codec 
support. There are many factors to consider, three of them are:

1. Coding efficiency: As more and more of the available capacity on the 
Internet is consumed by video, I think that codecs that are clearly 
inferior to the state of art regarding compression efficiency should not 
be considered.

2. Implementation status: Another factor to consider is the 
implementation status in end user equipment, and, especially for battery 
powered devices, to what extend platforms have (optimized/HW 
accelerated) implementations.

3. Interoperability with existing services (it can be argued that webrtc 
is a green field, but on the other hand there seems to be a lot of 
people wanting interoperability).

To me, h.264 is a good candidate considering the factors above. It is 
competitive regarding compression efficiency, it is widely implemented 
(in all kinds of devices and platforms) and is the by far mostly used 
video codec currently. Notably, a recent study ([1]) says that h.264 is 
the leading format also for the html video element.

In addition, licensing for h.264 seems straightforward, and the large 
number of implementations in commercial use indicates that vendors do 
not perceive the risk for licensing/patent issues down the road as being 
unacceptable big (this is an observation from the outside; I'm not a 
lawyer and I don't deal with licensing).

So, I propose that h.264 should be mandatory to implement.

Stefan

[1] http://blog.mefeedia.com/html5-dec-2011


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On 03/22/2012 07:17 PM, Igor Faynberg wrote:
> Thanks! Now I understand.
>
> My only follow-up question (and that could probably wait until the 
> next week) is why not make BrowserID certificate adhere to X.509.  If 
> we issue self-signed X.509 certificates ourselves, what is in the way 
> of any IdP doing just that?
The BrowserID specification is the work item of a group that is entirely 
outside of this group.
You can certainly go there (browserid.org) to argue that they should 
change to X.509 certificates, but we can't decide in the RTCWEB WG that 
they should change.

>
> Igor
>
> On 3/22/2012 1:53 PM, Eric Rescorla wrote:
>> Good question.
>>
>> 1. BrowserID certificates aren't X.509, and that's all the TLS/DTLS 
>> will do.
>> 2. This lets us be agnostic about IdP mechanisms without having to 
>> change
>> TLS every time we add a new one.
>>
>> -Ekr
>>
>>
>> On Thu, Mar 22, 2012 at 6:22 PM, Igor Faynberg
>> <igor.faynberg@alcatel-lucent.com>  wrote:
>>> Eric,
>>>
>>> A question:  For the case of BrowserID, I understand that a client 
>>> gets a
>>> certificate from an IdP.   If so, why you and I would not use the
>>> certificates from our respective IdPs in order to authenticate each 
>>> other
>>>   in DTLS?
>>>
>>> Igor
>>>
>>>
>>>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


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Stefan Hakansson LK wrote:
> So, I propose that h.264 should be mandatory to implement.

Mozilla strongly opposes such a proposal. Many will be familiar with our 
announcements regarding our use of platform codecs on B2G and possibly 
Android last week. The primary reasoning behind that decision, i.e. the 
one factor that was not in our power to change, was the availability of 
content on websites. This is not a problem for WebRTC, as browsers 
control both ends of the communication.

What _is_ a problem is licensing and distributing encoders in an 
open-source project. This is quite a bit different than using system 
decoders, as outside of mobile, encoders are usually not available, and 
even if they were, they might not have particularly good quality, nor 
the features to control latency, loss robustness, and other aspects 
necessary for real-time communication.

If you thought that a concession in one place would make Mozilla more 
likely to compromise in others, I assure you the opposite is true. 
Maintaining the use of RF codecs in WebRTC is now even more important 
than it was before. See the words directly from Mozilla's CTO: 
http://groups.google.com/group/mozilla.dev.platform/msg/53892ad3ea9eeb98

From lorenzo@meetecho.com  Fri Mar 23 04:48:12 2012
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+1,

L.


Il giorno Fri, 23 Mar 2012 04:40:40 -0700
"Timothy B. Terriberry" <tterriberry@mozilla.com> ha scritto:

> Stefan Hakansson LK wrote:
> > So, I propose that h.264 should be mandatory to implement.
> 
> Mozilla strongly opposes such a proposal. Many will be familiar with
> our announcements regarding our use of platform codecs on B2G and
> possibly Android last week. The primary reasoning behind that
> decision, i.e. the one factor that was not in our power to change,
> was the availability of content on websites. This is not a problem
> for WebRTC, as browsers control both ends of the communication.
> 
> What _is_ a problem is licensing and distributing encoders in an 
> open-source project. This is quite a bit different than using system 
> decoders, as outside of mobile, encoders are usually not available,
> and even if they were, they might not have particularly good quality,
> nor the features to control latency, loss robustness, and other
> aspects necessary for real-time communication.
> 
> If you thought that a concession in one place would make Mozilla more 
> likely to compromise in others, I assure you the opposite is true. 
> Maintaining the use of RF codecs in WebRTC is now even more important 
> than it was before. See the words directly from Mozilla's CTO: 
> http://groups.google.com/group/mozilla.dev.platform/msg/53892ad3ea9eeb98
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb



-- 
Lorenzo Miniero, COB

Meetecho s.r.l.
Web Conferencing and Collaboration Tools
http://www.meetecho.com

From Markus.Isomaki@nokia.com  Fri Mar 23 05:15:15 2012
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Hi,

Timothy B. Terriberry wrote:
>
>The primary reasoning behind that decision, i.e. the one
>factor that was not in our power to change, was the availability of conten=
t on
>websites. This is not a problem for WebRTC, as browsers control both ends =
of
>the communication.
>

In WebRTC context what is somewhat analogous is the interconnect to existin=
g non-web video systems. What codecs are used in them is not under browser =
control. I believe those mostly use H.264. It's probably possible to use tr=
anscoders but so it is between browsers too. I'd say that even if couldn't =
mandate a single video codec (as it has seemed to me since the beginning), =
in practice H.264 support would be highly RECOMMENDED (a SHOULD). There mig=
ht be good reasons not to follow that SHOULD, as it seems to be the case fo=
r Mozilla. =20

Markus


>-----Original Message-----
>From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
>Of ext Timothy B. Terriberry
>Sent: 23 March, 2012 13:41
>To: Stefan Hakansson LK
>Cc: rtcweb@ietf.org
>Subject: Re: [rtcweb] On video codec for rtcweb
>
>Stefan Hakansson LK wrote:
>> So, I propose that h.264 should be mandatory to implement.
>
>Mozilla strongly opposes such a proposal. Many will be familiar with our
>announcements regarding our use of platform codecs on B2G and possibly
>Android last week. The primary reasoning behind that decision, i.e. the on=
e
>factor that was not in our power to change, was the availability of conten=
t on
>websites. This is not a problem for WebRTC, as browsers control both ends =
of
>the communication.
>
>What _is_ a problem is licensing and distributing encoders in an open-sour=
ce
>project. This is quite a bit different than using system decoders, as outs=
ide of
>mobile, encoders are usually not available, and even if they were, they mi=
ght
>not have particularly good quality, nor the features to control latency, l=
oss
>robustness, and other aspects necessary for real-time communication.
>
>If you thought that a concession in one place would make Mozilla more like=
ly
>to compromise in others, I assure you the opposite is true.
>Maintaining the use of RF codecs in WebRTC is now even more important than
>it was before. See the words directly from Mozilla's CTO:
>http://groups.google.com/group/mozilla.dev.platform/msg/53892ad3ea9eeb
>98
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb

From tterriberry@mozilla.com  Fri Mar 23 05:34:17 2012
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Subject: Re: [rtcweb] On video codec for rtcweb
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Markus.Isomaki@nokia.com wrote:
> control. I believe those mostly use H.264. It's probably possible to use
> transcoders but so it is between browsers too. I'd say that even if couldn't

With a direct browser-to-browser connection, there is nothing sitting 
between them to do the transcoding. In the DTLS-SRTP case with identity 
verification, which I think a number of people here view as highly 
important, you are in fact _guaranteeing_ that nothing but the browser 
can encode or decode the video.

As for existing non-web video systems, I am happy to send them G.711, 
and no more, if all they wish to support is H.264. To me, interoperation 
beyond that is no more than a "nice to have". WebRTC was meant to be 
much more expansive than traditional telephony. If all WebRTC turns out 
to be is another vehicle for old-world video conferencing style systems, 
then we will have failed.

From chris.cavigioli@intel.com  Fri Mar 23 05:35:04 2012
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From: "Cavigioli, Chris" <chris.cavigioli@intel.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] On video codec for rtcweb
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Subject: Re: [rtcweb] On video codec for rtcweb
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Let's be clear on profiles.  If we seek widest mobile hardware footprint:
- ITU-T H.264 Constrained Baseline Profile is probably the most ubiquitous =
hardware video codec today with reasonable power/performance ratio
- ITU-T G.711, both u_law (North America) and A_law (Europe) formats, can b=
e supported by everyone with very trivial software burden
-chris

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Markus.Isomaki@nokia.com
Sent: Friday, March 23, 2012 5:15 AM
To: tterriberry@mozilla.com; stefan.lk.hakansson@ericsson.com
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] On video codec for rtcweb

Hi,

Timothy B. Terriberry wrote:
>
>The primary reasoning behind that decision, i.e. the one
>factor that was not in our power to change, was the availability of conten=
t on
>websites. This is not a problem for WebRTC, as browsers control both ends =
of
>the communication.
>

In WebRTC context what is somewhat analogous is the interconnect to existin=
g non-web video systems. What codecs are used in them is not under browser =
control. I believe those mostly use H.264. It's probably possible to use tr=
anscoders but so it is between browsers too. I'd say that even if couldn't =
mandate a single video codec (as it has seemed to me since the beginning), =
in practice H.264 support would be highly RECOMMENDED (a SHOULD). There mig=
ht be good reasons not to follow that SHOULD, as it seems to be the case fo=
r Mozilla. =20

Markus


>-----Original Message-----
>From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
>Of ext Timothy B. Terriberry
>Sent: 23 March, 2012 13:41
>To: Stefan Hakansson LK
>Cc: rtcweb@ietf.org
>Subject: Re: [rtcweb] On video codec for rtcweb
>
>Stefan Hakansson LK wrote:
>> So, I propose that h.264 should be mandatory to implement.
>
>Mozilla strongly opposes such a proposal. Many will be familiar with our
>announcements regarding our use of platform codecs on B2G and possibly
>Android last week. The primary reasoning behind that decision, i.e. the on=
e
>factor that was not in our power to change, was the availability of conten=
t on
>websites. This is not a problem for WebRTC, as browsers control both ends =
of
>the communication.
>
>What _is_ a problem is licensing and distributing encoders in an open-sour=
ce
>project. This is quite a bit different than using system decoders, as outs=
ide of
>mobile, encoders are usually not available, and even if they were, they mi=
ght
>not have particularly good quality, nor the features to control latency, l=
oss
>robustness, and other aspects necessary for real-time communication.
>
>If you thought that a concession in one place would make Mozilla more like=
ly
>to compromise in others, I assure you the opposite is true.
>Maintaining the use of RF codecs in WebRTC is now even more important than
>it was before. See the words directly from Mozilla's CTO:
>http://groups.google.com/group/mozilla.dev.platform/msg/53892ad3ea9eeb
>98
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb
_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

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Hi

Timothy B. Terriberry wrote:
>
>Markus.Isomaki@nokia.com wrote:
>> control. I believe those mostly use H.264. It's probably possible to
>> use transcoders but so it is between browsers too. I'd say that even
>> if couldn't
>
>With a direct browser-to-browser connection, there is nothing sitting
>between them to do the transcoding. In the DTLS-SRTP case with identity
>verification, which I think a number of people here view as highly importa=
nt,
>you are in fact _guaranteeing_ that nothing but the browser can encode or
>decode the video.
>

Understood. So it would be important to agree on a common codec. However, I=
'm just wondering if forcing the decision would be wise for the IETF. I hav=
e heard many people say that if X was chosen, they wouldn't still suport it=
. My opinion is that a codec should only be mandated if there is clear cons=
ensus about it. Exaggerating a bit but not much that would probably leave u=
s with G.711.

>As for existing non-web video systems, I am happy to send them G.711, and
>no more, if all they wish to support is H.264.=20

Indeed :-) That's fine, but for some other people video support in that sce=
nario could be more important.=20

>To me, interoperation beyond
>that is no more than a "nice to have". WebRTC was meant to be much more
>expansive than traditional telephony. If all WebRTC turns out to be is ano=
ther
>vehicle for old-world video conferencing style systems, then we will have
>failed.
>

Here we are in full agreement. I don't think either that the "legacy" inter=
connect is the highest priority. I don't find the current video call market=
 that impressive that we should somehow limit ourselves to serve it.

Markus

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On 03/23/2012 08:34 AM, Cavigioli, Chris wrote:
> Let's be clear on profiles.  If we seek widest mobile hardware footprint:
> - ITU-T H.264 Constrained Baseline Profile is probably the most ubiquitous hardware video codec today with reasonable power/performance ratio
> - ITU-T G.711, both u_law (North America) and A_law (Europe) formats, can be supported by everyone with very trivial software burden
> -chris
Seeking the widest mobile hardware footprint is a very narrow view
considering the scope of WebRTC.  WebRTC is a technology with browsers
as the main endpoints.  While mobile devices make up a significant
portion of that population, they are by no means the only one, and there
are no guarantees that this will be the major use case of WebRTC in the
future.

Time and again, success and adoption of standards across all markets,
devices, and platforms has been rooted in the usage of a open,
royalty-free formats.  Mandating H.264 for this or any other purpose
while it remains unlicensable for usage in free/open source software
projects invalidates the universality of WebRTC.  This would setup the
protocol to be yet another proprietary system that claims to be open but
in practice will be limited to usage solely by existing MPEG-LA
licensors/licensees and other existing players in the telecommunications
market.  This makes the project, and the standard, dead in the water as
far as a lot of the players interested in this project are concerned.

Having said that, WebRTC at this point remains open and free, and to
ensure that, the only technologies that should be included are those
that are royalty free and licensed for use within free/open-source
software project.  This is the *only* way for WebRTC to be successful
and to achieve the goals of its (intended) open nature.

From tterriberry@mozilla.com  Fri Mar 23 06:28:21 2012
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Subject: Re: [rtcweb] On video codec for rtcweb
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Markus.Isomaki@nokia.com wrote:
 > My opinion is that a codec should only be mandated if there is clear
 > consensus about it. Exaggerating a bit but not much that would 
probably leave
 > us with G.711.

We've had this conversation on this list before. See
http://www.ietf.org/mail-archive/web/rtcweb/current/msg03047.html
wherein you find that even Stephan _Wenger_ and I managed to agree that 
your standard is too strong. Such consensus is a rare thing indeed.

From bernard_aboba@hotmail.com  Fri Mar 23 07:24:54 2012
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--_6fa0bf81-9f22-465b-8a03-e993800d6472_
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Tim said:=20

> As for existing non-web video systems=2C I am happy to send them G.711=2C=
=20
> and no more=2C if all they wish to support is H.264. To me=2C interoperat=
ion=20
> beyond that is no more than a "nice to have". WebRTC was meant to be=20
> much more expansive than traditional telephony. If all WebRTC turns out=20
> to be is another vehicle for old-world video conferencing style systems=
=2C=20
> then we will have failed.

[BA]  While "old-world video conferencing" systems (H.323) are still widely=
 used=2C
I think it is fair to say that there is a strong trend towards adoption of =
SIP in a=20
wide range of markets=2C including government=2C educational=2C medical=2C =
financials=2C etc.=20

I have talked to customers in many of those markets=2C and one of the consi=
stently
repeated requests is for interoperability between mobile video and video
conferencing platforms=2C *without the need for transcoding*.   The reason =
why
transcoding is unacceptable is that the cost of transcoding gateways is hig=
h
on a per user basis=2C and for organizations where high use of video is ant=
icipated=2C
the cost of transcoding (as opposed to a signaling gateway) is a major prob=
lem.  =20
Another frequently cited issue is the ability to use hardware encode/decode=
 on mobile devices=2C so
as to save power.  In scenarios such as emergency services=2C this can lite=
rally be a
matter of life and death.=20

Given that mobile devices and video conferencing systems overwhelmingly sup=
port=20
H.264 encode/decode in hardware=2C and that we do not yet have general hard=
ware support
for encode/decode operations=2C the answer to customer requests can only be=
 provided by
H.264 at this time=2C or perhaps by H.265 (which is likely to be rapidly ad=
opted once it is
ratified in early 2013). =20

While I understand the reasons why browser vendors might not want to
include software implementations of encumbered codecs=2C so they can remain=
 free to users=2C=20
that logic does not extend to a refusal to rely on operating system or hard=
ware
support=2C which would presumably come without cost to the browser vendors.=
=20

My discussions with customers indicate that support for underlying encode/d=
ecode
capabilities is considered an important enough feature that it will cause t=
hose customers to
switch to browsers that support it.=20
 		 	   		  =

--_6fa0bf81-9f22-465b-8a03-e993800d6472_
Content-Type: text/html; charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<html>
<head>
<style><!--
.hmmessage P
{
margin:0px=3B
padding:0px
}
body.hmmessage
{
font-size: 10pt=3B
font-family:Tahoma
}
--></style></head>
<body class=3D'hmmessage'><div dir=3D'ltr'>
Tim said: <br><br><div>&gt=3B As for existing non-web video systems=2C I am=
 happy to send them G.711=2C <br>&gt=3B and no more=2C if all they wish to =
support is H.264. To me=2C interoperation <br>&gt=3B beyond that is no more=
 than a "nice to have". WebRTC was meant to be <br>&gt=3B much more expansi=
ve than traditional telephony. If all WebRTC turns out <br>&gt=3B to be is =
another vehicle for old-world video conferencing style systems=2C <br>&gt=
=3B then we will have failed.<br><br>[BA]&nbsp=3B While "old-world video co=
nferencing" systems (H.323) are still widely used=2C<br>I think it is fair =
to say that there is a strong trend towards adoption of SIP in a <br>wide r=
ange of markets=2C including government=2C educational=2C medical=2C financ=
ials=2C etc. <br><br>I have talked to customers in many of those markets=2C=
 and one of the consistently<br>repeated requests is for interoperability b=
etween mobile video and video<br>conferencing platforms=2C *without the nee=
d for transcoding*. &nbsp=3B The reason why<br>transcoding is unacceptable =
is that the cost of transcoding gateways is high<br>on a per user basis=2C =
and for organizations where high use of video is anticipated=2C<br>the cost=
 of transcoding (as opposed to a signaling gateway) is a major problem.&nbs=
p=3B&nbsp=3B <br>Another frequently cited issue is the ability to use hardw=
are encode/decode on mobile devices=2C so<br>as to save power.&nbsp=3B In s=
cenarios such as emergency services=2C this can literally be a<br>matter of=
 life and death. <br><br>Given that mobile devices and video conferencing s=
ystems overwhelmingly support <br>H.264 encode/decode in hardware=2C and th=
at we do not yet have general hardware support<br>for encode/decode operati=
ons=2C the answer to customer requests can only be provided by<br>H.264 at =
this time=2C or perhaps by H.265 (which is likely to be rapidly adopted onc=
e it is<br>ratified in early 2013).&nbsp=3B <br><br>While I understand the =
reasons why browser vendors might not want to<br>include software implement=
ations of encumbered codecs=2C so they can remain free to users=2C <br>that=
 logic does not extend to a refusal to rely on operating system or hardware=
<br>support=2C which would presumably come without cost to the browser vend=
ors. <br><br>My discussions with customers indicate that support for underl=
ying encode/decode<br>capabilities is considered an important enough featur=
e that it will cause those customers to<br>switch to browsers that support =
it. <br></div> 		 	   		  </div></body>
</html>=

--_6fa0bf81-9f22-465b-8a03-e993800d6472_--

From matthew.kaufman@skype.net  Fri Mar 23 07:35:41 2012
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From: Matthew Kaufman <matthew.kaufman@skype.net>
To: "Timothy B. Terriberry" <tterriberry@mozilla.com>, Stefan Hakansson LK <stefan.lk.hakansson@ericsson.com>
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Timothy B. Terriberry:

"Mozilla strongly opposes such a proposal. Many will be familiar with our a=
nnouncements regarding our use of platform codecs on B2G and possibly Andro=
id last week. The primary reasoning behind that decision, i.e. the one fact=
or that was not in our power to change, was the availability of content on =
websites. This is not a problem for WebRTC, as browsers control both ends o=
f the communication."


While I understand the reasons for disliking a codec that isn't available r=
oyalty-free, I have to say that the above argument is exactly backwards.

While a website that wished to offer content to two different browsers with=
 different codecs could simply transcode and store the video in two formats=
 (albeit at some cost in storage, which is getting cheaper all the time), a=
 website that wishes to offer RTC capabilities to users with differing brow=
sers would be forced to stand up servers, in-path, transcoding in real time=
... a much more onerous requirement in order to achieve interoperability.

While browsers "control both ends of the communication" in RTCWEB, the site=
s offering RTC services don't necessarily "control both ends of the communi=
cation" insofar as choosing what browsers their users use to visit... or ra=
ther, in order to achieve interoperable communication without the cost (and=
 security concerns) of transcoding sites will in fact start to deny service=
 to particular users based on their choice of browser.

Matthew Kaufman


From tterriberry@mozilla.com  Fri Mar 23 07:47:13 2012
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Bernard Aboba wrote:
> While I understand the reasons why browser vendors might not want to
> include software implementations of encumbered codecs, so they can
> remain free to users,
> that logic does not extend to a refusal to rely on operating system or
> hardware
> support, which would presumably come without cost to the browser vendors.

Oh, but it does. My understanding is that even Windows 7 does not ship 
with a licensed encoder. Even if it did, nearly a plurality of Mozilla's 
users still run Windows XP. I would like those users to be able to talk 
to each other, and with mobile devices, and I view not supporting an 
encumbered codec as the best route to ensuring that support for a free 
one is deployed and working on all of these systems.

From tterriberry@mozilla.com  Fri Mar 23 07:51:31 2012
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Matthew Kaufman wrote:
 > or rather, in order to achieve interoperable communication without 
the cost
 > (and security concerns) of transcoding sites will in fact start to deny
 > service to particular users based on their choice of browser.

I agree this is an excellent argument for having _a_ mandatory to 
implement codec. But it's an argument against having that codec be an 
encumbered codec. Much better to have it be a free one that everyone is 
able to deploy. Even Microsoft (via Skype) now deploys VP8.

From tterriberry@mozilla.com  Fri Mar 23 07:51:39 2012
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Subject: Re: [rtcweb] On video codec for rtcweb
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Matthew Kaufman wrote:
 > or rather, in order to achieve interoperable communication without 
the cost
 > (and security concerns) of transcoding sites will in fact start to deny
 > service to particular users based on their choice of browser.

I agree this is an excellent argument for having _a_ mandatory to 
implement codec. But it's an argument against having that codec be an 
encumbered codec. Much better to have it be a free one that everyone is 
able to deploy. Even Microsoft (via Skype) now deploys VP8.

From andrew.hutton@siemens-enterprise.com  Fri Mar 23 07:54:38 2012
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From: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>
To: Dan Wing <dwing@cisco.com>, 'Roman Shpount' <roman@telurix.com>, 'Harald Alvestrand' <harald@alvestrand.no>, 'Ted Hardie' <ted.ietf@gmail.com>
Date: Fri, 23 Mar 2012 15:54:31 +0100
Thread-Topic: [rtcweb] Resolving RTP/SDES question in Paris
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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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Hi Dan,

I don't think SIPREC has a solution to the particular problem that Roman is=
 talking about and it would probably not be within scope of SIPREC to solve=
 this even assuming that there is some SIP involved.

If the prison deployed its own RTCWeb application and restricted the prison=
er to using only this application then of course there would be no problem =
with intercepting the media and a SIPREC type solution could be deployed wi=
th some kind of B2BUA which has access to both signaling and media initiati=
ng the media recording and of course informing the prisoner that they are b=
eing recorded.

However if the prisoner is allowed to access their lawyers RTCWeb applicati=
on and download the JavaScript from there then SIPREC does not help the pri=
son record the resulting media exchanged.

Regards
Andy=20


> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Dan Wing
> Sent: 22 March 2012 22:44
> To: 'Roman Shpount'; 'Harald Alvestrand'; 'Ted Hardie'
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
>=20
> > -----Original Message-----
> > From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> > Behalf Of Roman Shpount
> > Sent: Thursday, March 22, 2012 7:19 AM
> > To: Harald Alvestrand
> > Cc: rtcweb@ietf.org
> > Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
> ...
> > Yes, I was worried about supporting the interceptor. This is all
> > related to my question of support of WebRTC applications in military,
> > prisons, or financial organizations. I think WebRTC would be
> completely
> > disabled in such locations unless web browser can be configured to
> > confirm to some sort of communications and monitoring policy. I do
> not
> > think enabling only certain applications is a sustainable solution,
> > since application can only be enabled based on its URL, but this in
> no
> > way implies that the actual protocol used for signaling exchange by
> > this application will stay the same. In the best case this will
> result
> > in tens of specialized monitoring rules that will need to be
> > maintained. In the worst case, WebRTC would be simply disabled. At
> this
> > point the only workable solution that I see is some sort of media
> proxy
> > protocol.
>=20
> The SIPREC working group, formed in March 2010, has solutions for
> this problem.  The problem is not created by RTCWEB.  Disabling
> encryption is not necessary -- nor desirable.  "Jails" get brought up
> in this context often, so using that same example, consider a prisoner
> at a jail communicating with their lawyer.  The prisoner needs secure
> communications with their lawyer without the risk of the
> communications being leaked to the press by anyone along the media
> path.  For details see http://tools.ietf.org/wg/siprec/
>=20
> We will have a brietf presentation on SIPREC during my session
> at RTCWEB, explaining how it can work with RTCWEB.
>=20
> -d
>=20
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From dwing@cisco.com  Fri Mar 23 08:02:04 2012
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From: "Dan Wing" <dwing@cisco.com>
To: "'Ravindran, Parthasarathi'" <pravindran@sonusnet.com>, "'Roman Shpount'" <roman@telurix.com>, "'Harald Alvestrand'" <harald@alvestrand.no>, "'Ted Hardie'" <ted.ietf@gmail.com>
References: <4F4759DC.7060303@ericsson.com>	<387F9047F55E8C42850AD6B3A7A03C6C0E1FEB69@inba-mail01.sonusnet.com>	<CALiegfnkYVEpmPV-zSL_4wOY-HiFZN-qJCQCiioaS=5NaqhLZw@mail.gmail.com>	<CAD5OKxvtOAxMBx6xDnyfTnEq76oDEm6uj1xL6wGjjrtKUAHy3g@mail.gmail.com>	<CABcZeBNZiotPmCfT53uEo+O0xw4xv6tXW1M_G-3A5BHuncsduA@mail.gmail.com>	<CAD5OKxvYOY5JZ2mYNGiH1poUBQkyOOycePFijH5H+SxtcdqujQ@mail.gmail.com>	<CABkgnnVe-b6Sv=R67bMJk_NQqQwdrRUn6rBm7Gu_CMcfPQwtEg@mail.gmail.com>	<4F64FE98.3070605@alcatel-lucent.com>	<4F685ED9.2050109@alvestrand.no>	<CAD5OKxsVp7px9bHAgxgdqPMxRgppcVUDKt8JHBhyq9qqW3pAMg@mail.gmail.com>	<4F68A4CC.9090306@alvestrand.no>	<CAD5OKxuiApLKRASc2YuBfkM_8h8wGDPPQ3TdOYGum2yauidA5A@mail.gmail.com>	<4F6AECC6.8020004@alvestrand.no>	<CAD5OKxsSUeMFYXZMZVqQFWdeEB=30HJuJ=mP9GaYkksBmp1mOA@mail.gmail.com> <03fa01cd087d$57899120$069cb360$@com> <387F9047F55E8C42850AD6B3A7A03C6C0E21E8F5@inba-mail01.sonusnet.com>
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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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> -----Original Message-----
> From: Ravindran, Parthasarathi [mailto:pravindran@sonusnet.com]
> Sent: Thursday, March 22, 2012 11:40 PM
> To: Dan Wing; 'Roman Shpount'; 'Harald Alvestrand'; 'Ted Hardie'
> Cc: rtcweb@ietf.org
> Subject: RE: [rtcweb] Resolving RTP/SDES question in Paris
> 
> Even in case disabling encryption is not the desired behavior, I don't
> see the problem in using RTP-IPSec instead of SRTP-DTLS.

Several.

IPsec ESP tunnel or transport mode?  With NAT-T support?  Is there
an API for applications to determine if the underlying OS provides
IPsec ESP for that flow (I know for Cisco IOS, the answer to that
API question is 'no'.  And that was my understanding in the version 
of Windows Server that we used to use for our call control system,
although of course that could have changed with Windows 7 or 8).
RTP over IPsec breaks link header compression (cRTP and ROHC),
which are used by some networks.

> I understand
> that web-browser has no standard way to identify whether plain IP or
> IPsec in Layer 3. So, it shall be configured by browser-user and it
> shall be considered as user consent. This approach helps in case in
> avoiding double encryption technically wherein IPSec is mandated. I
> know the argument that double encryption is not problem in the endpoint
> as it has huge CPU and memory resources but in case web-browser as part
> of access devices, it will have impact on performance. My issue is with
> mandating to use SRTP-DTLS in WebRTC and not with mandate to implement
> SRTP-DTLS.

'Double encryption' is done because it's the only way to achieve
security at various layers.

Let's take a simple example:  someone is using their PC at a coffee
shop or a conference, and (because the person is using a possibly-hostile
network, and the person is ostensibly working) they bring up a VPN.  All 
of the person's traffic goes into their VPN.  The person then accesses
paypal (or gmail, or hundreds of other example sites), which forces them 
to use HTTPS.  They are now "double encrypted".  

Can you see the difficulty in removing the HTTPS connection to Paypal
(or gmail, etc.).  Can you see the difficulty in the PC determining
the HTTPS connection is "good enough" and routing that outside the 
VPN -- it can't be done simply by port number or else someone could
accidentally run un-encrypted traffic on port 443.

And this person may well even be triple encryption, if the WiFi access 
point uses wireless encryption.  Taking your argument further, if we 
are running IPsec we should have a way to disable the WiFi encryption?
But the WiFi encryption is being used to protect that layer -- just
like the IPsec is used in the VPN to protect that layer and the HTTPS
to PayPal is used to protect that layer.

> Also, AFAIK HTTPS certificate is not free of cost. Whereever the cost
> matters like social welfare site, educational site and some free blogs,
> the tendency will be towards HTTP

Ok.  (Separate problem.)

> and user of those sites will have the
> consent to use RTP.

I don't see why those sites cannot use SRTP.

-d


> Thanks
> Partha
> 
> >-----Original Message-----
> >From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf
> >Of Dan Wing
> >Sent: Friday, March 23, 2012 4:14 AM
> >To: 'Roman Shpount'; 'Harald Alvestrand'; 'Ted Hardie'
> >Cc: rtcweb@ietf.org
> >Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
> >
> >> -----Original Message-----
> >> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> >> Behalf Of Roman Shpount
> >> Sent: Thursday, March 22, 2012 7:19 AM
> >> To: Harald Alvestrand
> >> Cc: rtcweb@ietf.org
> >> Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
> >...
> >> Yes, I was worried about supporting the interceptor. This is all
> >> related to my question of support of WebRTC applications in
> military,
> >> prisons, or financial organizations. I think WebRTC would be
> >> completely disabled in such locations unless web browser can be
> >> configured to confirm to some sort of communications and monitoring
> >> policy. I do not think enabling only certain applications is a
> >> sustainable solution, since application can only be enabled based on
> >> its URL, but this in no way implies that the actual protocol used
> for
> >> signaling exchange by this application will stay the same. In the
> best
> >> case this will result in tens of specialized monitoring rules that
> >> will need to be maintained. In the worst case, WebRTC would be
> simply
> >> disabled. At this point the only workable solution that I see is
> some
> >> sort of media proxy protocol.
> >
> >The SIPREC working group, formed in March 2010, has solutions for this
> >problem.  The problem is not created by RTCWEB.  Disabling encryption
> is
> >not necessary -- nor desirable.  "Jails" get brought up in this
> context
> >often, so using that same example, consider a prisoner at a jail
> >communicating with their lawyer.  The prisoner needs secure
> >communications with their lawyer without the risk of the
> communications
> >being leaked to the press by anyone along the media path.  For details
> >see http://tools.ietf.org/wg/siprec/
> >
> >We will have a brietf presentation on SIPREC during my session at
> >RTCWEB, explaining how it can work with RTCWEB.
> >
> >-d
> >
> >
> >
> >_______________________________________________
> >rtcweb mailing list
> >rtcweb@ietf.org
> >https://www.ietf.org/mailman/listinfo/rtcweb


From martin.thomson@gmail.com  Fri Mar 23 08:22:00 2012
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Subject: Re: [rtcweb] On video codec for rtcweb
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On 23 March 2012 05:34, Timothy B. Terriberry <tterriberry@mozilla.com> wrote:
> With a direct browser-to-browser connection, there is nothing sitting
> between them to do the transcoding. In the DTLS-SRTP case with identity
> verification, which I think a number of people here view as highly
> important, you are in fact _guaranteeing_ that nothing but the browser can
> encode or decode the video.

This is incorrect.  The browser can implement the DTLS handshake and
crypto renegotiation, plus the SRTP shim.  It is possible for the
browser to perform the crypto without performing encode or decode if
the source is capable of providing RTP.  From RFC 3711:

   [...]  Conceptually, we consider SRTP to be a "bump in the stack"
   implementation which resides between the RTP application and the
   transport layer.  SRTP intercepts RTP packets and then forwards an
   equivalent SRTP packet on the sending side, and intercepts SRTP
   packets and passes an equivalent RTP packet up the stack on the
   receiving side.

From matthew.kaufman@skype.net  Fri Mar 23 08:27:08 2012
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From: Matthew Kaufman <matthew.kaufman@skype.net>
To: "Timothy B. Terriberry" <tterriberry@mozilla.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] On video codec for rtcweb
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Timothy B. Terriberry:


> Matthew Kaufman wrote:
 > > or rather, in order to achieve interoperable communication without the=
 cost  > (and security concerns) of transcoding sites will in fact start to=
 deny  > service to particular users based on their choice of browser.

> I agree this is an excellent argument for having _a_ mandatory to impleme=
nt codec. But it's an argument against having that codec be an encumbered c=
odec. Much better to have it be a free one that everyone is able to deploy.=
 Even Microsoft (via Skype) now deploys VP8.

I wasn't commenting on that part, just that if Mozilla is going to soften u=
p on H.264 you'd think it'd happen to RTCWEB first, VOD second... not the r=
everse.

As for VP8, we don't actually know if it falls into the set "a free one tha=
t everyone is able to deploy", do we?

Matthew Kaufman


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On 03/23/2012 11:24 AM, Matthew Kaufman wrote:
> Timothy B. Terriberry:
>
>
>> Matthew Kaufman wrote:
>  > > or rather, in order to achieve interoperable communication without the cost  > (and security concerns) of transcoding sites will in fact start to deny  > service to particular users based on their choice of browser.
>
>> I agree this is an excellent argument for having _a_ mandatory to implement codec. But it's an argument against having that codec be an encumbered codec. Much better to have it be a free one that everyone is able to deploy. Even Microsoft (via Skype) now deploys VP8.
> I wasn't commenting on that part, just that if Mozilla is going to soften up on H.264 you'd think it'd happen to RTCWEB first, VOD second... not the reverse.
>
> As for VP8, we don't actually know if it falls into the set "a free one that everyone is able to deploy", do we?
>
> Matthew Kaufman
With all due respect, this is FUD that is baseless.  A blog post[1], an
abandoned patent pool[2], and vague comments by Steve Jobs[3] are all
that really stand as doubt as to the royalty-free basis on VP8.  What we
do know is that On2 had some patents on VP8, and these have been
licensed for royalty-free usage after their acquisition by Google. 
Google themselves have deployed VP8 widely on YouTube via the WebM
format and implemented and distributed it on Android, which is quite a
large surface area to expose if they had doubts about its safety. 
Patent licensing targets don't get much more juicy than Google.

As for implementation, any choice of format will come with challenges,
and the licensing on H.264 vs. VP8 make it clear that VP8 can receive a
much wider adoption in the long run, especially one that does not
unnecessarily restrict this forthcoming standard only to existing
players in the telecommunications market.  There are already plenty of
hardware chips that natively support VP8 decoding and encoding[4], and
that will grow over time, as well, because many parties are strongly
interested in a royalty-free alternative to H.264.  Let alone there are
ARM implementations of VP8 as well[5].

Where is the legitimate doubt, then, that VP8 is not an acceptable,
royalty-free format that can be used for the purposes of WebRTC?  We're
talking initial deployment.  Technology is always changed, and WebRTC 2
is free to adopt Daala[6] when it is released.  Believe me, the
developers would love to ensure it is suitable for WebRTC, amongst other
things.

[1] http://x264dev.multimedia.cx/archives/377#more-377
[2] http://www.mpegla.org/main/pages/CurrentPools_LightBox.aspx (note,
it's not listed there!)
[3] http://www.theregister.co.uk/2010/05/20/jobs_on_vp8/
[4] http://wiki.webmproject.org/hardware/arm-socs
[5] http://www.webmproject.org/tools/vp8-sdk/changelog.html
[6] https://www.xiph.org/daala/

From roman@telurix.com  Fri Mar 23 09:48:34 2012
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--047d7b15a68b6510fe04bbebcb04
Content-Type: text/plain; charset=ISO-8859-1

On Fri, Mar 23, 2012 at 8:34 AM, Timothy B. Terriberry <
tterriberry@mozilla.com> wrote:

> With a direct browser-to-browser connection, there is nothing sitting
> between them to do the transcoding. In the DTLS-SRTP case with identity
> verification, which I think a number of people here view as highly
> important, you are in fact _guaranteeing_ that nothing but the browser can
> encode or decode the video.
>

This is not technically correct. You can still build a system that
implements a media proxy that does nothing but handling ICE, encryption,
and identity verification without ever transcoding the content. The cost of
re-encrypting the media is low (ie single E5 Xeon CPU can probably do about
10GB/s), but the cost of transcoding video, or even audio is huge in
comparison. Because of this, supporting a comprehensive set of codecs is
highly desirable.
_____________
Roman Shpount

--047d7b15a68b6510fe04bbebcb04
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<br clear=3D"all"><br><div class=3D"gmail_quote">On Fri, Mar 23, 2012 at 8:=
34 AM, Timothy B. Terriberry <span dir=3D"ltr">&lt;<a href=3D"mailto:tterri=
berry@mozilla.com">tterriberry@mozilla.com</a>&gt;</span> wrote:<br><blockq=
uote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc =
solid;padding-left:1ex">
<div class=3D"im"></div>
With a direct browser-to-browser connection, there is nothing sitting betwe=
en them to do the transcoding. In the DTLS-SRTP case with identity verifica=
tion, which I think a number of people here view as highly important, you a=
re in fact _guaranteeing_ that nothing but the browser can encode or decode=
 the video.<br>
</blockquote><div><br>This is not technically correct. You can still build =
a system that implements a media proxy that does nothing but handling ICE, =
encryption, and identity verification without ever transcoding the content.=
 The cost of re-encrypting the media is low (ie single E5 Xeon CPU can prob=
ably do about 10GB/s), but the cost of transcoding video, or even audio is =
huge in comparison. Because of this, supporting a comprehensive set of code=
cs is highly desirable.<br>
</div></div>_____________<br>Roman Shpount<br>

--047d7b15a68b6510fe04bbebcb04--

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Date: Fri, 23 Mar 2012 13:14:22 -0400
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From: Roman Shpount <roman@telurix.com>
To: Hadriel Kaplan <HKaplan@acmepacket.com>
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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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--047d7b10d17be3aa9b04bbec2761
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inline

On Fri, Mar 23, 2012 at 12:29 AM, Hadriel Kaplan <HKaplan@acmepacket.com>wr=
ote:

> comments inline=85
>
> On Mar 22, 2012, at 9:31 PM, Roman Shpount wrote:
>
> > I would say that this activity would be covered by:
>
> Did you mean to have a #1 before #2 below?  Your email was missing a #1.
>
>
No, I did not mean to have #1 before #2 and #3. I was quoting the exact
language in the chapter definition that should cover the activity related
to WebRTC specific proxy.


> > 2. Define a security model that describes the security and privacy
> > goals and specifies the security protocol mechanisms necessary
> > to achieve those goals.
>
> We've got that:
> http://tools.ietf.org/html/draft-ietf-rtcweb-security
> http://tools.ietf.org/html/draft-ietf-rtcweb-security-arch
>
>
These documents do not cover scenario when organization forces all the
traffic to go through HTTP proxy to enforce its communication policy.

> 3. Define the solution - protocols and API requirements - for
> > firewall and NAT traversal.
>
> We've got that, though not in one doc:
> http://tools.ietf.org/html/draft-ietf-rtcweb-overview
> http://tools.ietf.org/html/draft-ietf-rtcweb-use-cases-and-requirements
> http://tools.ietf.org/html/draft-ietf-rtcweb-jsep
>
>
Once again, these documents do not cover scenario when organization forces
all the traffic to go through HTTP proxy. In such cases WebRTC simply will
not work. At best, we can advise such organizations to deploy their own
TURN server, but this allow all traffic to traverse into public internet,
not just webrtc related one, which might be undesirable.



> > I do consider ability to enforce the enterprise or location
> communications policy on WebRTC clients to be one of the requirements for
> security and privacy.
>
> OK, so what you're really saying is we have #2 and #3, but you don't agre=
e
> they cover the needs, and you want more added to #2 (and ultimately #3),
> right?
>

I think we are currently missing one of the common network deployment
scenarios for web browsers, where web browsers are deployed on a network
non-routable to public internet with an HTTP proxy used to allow access to
the web. I believe that we should add capabilities necessary to support
such deployments.


> > I expect that unless ability to enforce such communication policy withi=
n
> enterprises or other organizations is provided, WebRTC would be simply
> disabled.
> > In practical terms, I expect that an organization that enforces a
> communication policy disables routing to public internet and only allows
> communications via a company controlled HTTP proxy. Such setup would
> effectively prevent WebRTC from operations. What I am suggesting is to
> provide a complimentary protocol for inspection and modification of WebRT=
C
> SDP messages which would allow WebRTC enabled browser to control an
> enterprise edge media proxy and would allow WebRTC media traffic to
> traverse the enterprise firewall and to operate in such environment.
>
> Well... if they really can't enforce their policies, then by definition
> they *should* disable it.  That's a feature, not a bug.  And luckily it
> *can* be disabled, which is also a feature of WebRTC.
>
> Having said that, I don't understand why you think an organization that
> has strict monitoring policies can't monitor and enforce WebRTC traffic,
> both the HTTP *and* the media/data channels.  They *can* do it.  It's jus=
t
> not "automagic", and not cheap.  But that's ok - I don't think we need to
> optimize the deployment complexity or cost structure for that model.
>

The reason is that with custom signaling protocols implementing monitoring
or enforcing policies would be very hard. BTW, this is not only and not
primarily about recording traffic. This can be used to force all the
traffic outside of the enterprise to be DTLS-SRTP only, but still allow
DES-SRTP (or RTP if WebRTC will support it) within the enterprise network.
_____________
Roman Shpount

--047d7b10d17be3aa9b04bbec2761
Content-Type: text/html; charset=windows-1252
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<br clear=3D"all">inline<br><br><div class=3D"gmail_quote">On Fri, Mar 23, =
2012 at 12:29 AM, Hadriel Kaplan <span dir=3D"ltr">&lt;<a href=3D"mailto:HK=
aplan@acmepacket.com">HKaplan@acmepacket.com</a>&gt;</span> wrote:<br><bloc=
kquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #cc=
c solid;padding-left:1ex">
comments inline=85<br>
<div class=3D"im"><br>
On Mar 22, 2012, at 9:31 PM, Roman Shpount wrote:<br>
<br>
&gt; I would say that this activity would be covered by:<br>
<br>
</div>Did you mean to have a #1 before #2 below? =A0Your email was missing =
a #1.<br>
<div class=3D"im"><br></div></blockquote><div><br>No, I did not mean to hav=
e #1 before #2 and #3. I was quoting the exact language in the chapter defi=
nition that should cover the activity related to WebRTC specific proxy.<br>
=A0<br></div><blockquote class=3D"gmail_quote" style=3D"margin:0pt 0pt 0pt =
0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div class=
=3D"im">
&gt; 2. Define a security model that describes the security and privacy<br>
&gt; goals and specifies the security protocol mechanisms necessary<br>
&gt; to achieve those goals.<br>
<br>
</div>We&#39;ve got that:<br>
<a href=3D"http://tools.ietf.org/html/draft-ietf-rtcweb-security" target=3D=
"_blank">http://tools.ietf.org/html/draft-ietf-rtcweb-security</a><br>
<a href=3D"http://tools.ietf.org/html/draft-ietf-rtcweb-security-arch" targ=
et=3D"_blank">http://tools.ietf.org/html/draft-ietf-rtcweb-security-arch</a=
><br>
<div class=3D"im"><br></div></blockquote><div>=A0<br>These documents do not=
 cover scenario when organization forces all the traffic to go through HTTP=
 proxy to enforce its communication policy.<br><br></div><blockquote class=
=3D"gmail_quote" style=3D"margin:0pt 0pt 0pt 0.8ex;border-left:1px solid rg=
b(204,204,204);padding-left:1ex">
<div class=3D"im">
&gt; 3. Define the solution - protocols and API requirements - for<br>
&gt; firewall and NAT traversal.<br>
<br>
</div>We&#39;ve got that, though not in one doc:<br>
<a href=3D"http://tools.ietf.org/html/draft-ietf-rtcweb-overview" target=3D=
"_blank">http://tools.ietf.org/html/draft-ietf-rtcweb-overview</a><br>
<a href=3D"http://tools.ietf.org/html/draft-ietf-rtcweb-use-cases-and-requi=
rements" target=3D"_blank">http://tools.ietf.org/html/draft-ietf-rtcweb-use=
-cases-and-requirements</a><br>
<a href=3D"http://tools.ietf.org/html/draft-ietf-rtcweb-jsep" target=3D"_bl=
ank">http://tools.ietf.org/html/draft-ietf-rtcweb-jsep</a><br>
<div class=3D"im"><br></div></blockquote><div><br>Once again, these documen=
ts do not cover scenario when organization forces all the traffic to go thr=
ough HTTP proxy. In such cases WebRTC simply will not work. At best, we can=
 advise such organizations to deploy their own TURN server, but this allow =
all traffic to traverse into public internet, not just webrtc related one, =
which might be undesirable.<br>
=A0
<br></div><div>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0=
pt 0pt 0pt 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><=
div class=3D"im">
&gt; I do consider ability to enforce the enterprise or location communicat=
ions policy on WebRTC clients to be one of the requirements for security an=
d privacy.<br>
<br>
</div>OK, so what you&#39;re really saying is we have #2 and #3, but you do=
n&#39;t agree they cover the needs, and you want more added to #2 (and ulti=
mately #3), right?<br><div class=3D"im"></div></blockquote><div><br>I think=
 we are currently missing one of the common network deployment scenarios fo=
r web browsers, where web browsers are deployed on a network non-routable t=
o public internet with an HTTP proxy used to allow access to the web. I bel=
ieve that we should add capabilities necessary to support such deployments.=
 <br>
=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0pt 0pt 0pt 0.8e=
x;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div class=3D"im=
">

&gt; I expect that unless ability to enforce such communication policy with=
in enterprises or other organizations is provided, WebRTC would be simply d=
isabled.<br>
&gt; In practical terms, I expect that an organization that enforces a comm=
unication policy disables routing to public internet and only allows commun=
ications via a company controlled HTTP proxy. Such setup would effectively =
prevent WebRTC from operations. What I am suggesting is to provide a compli=
mentary protocol for inspection and modification of WebRTC SDP messages whi=
ch would allow WebRTC enabled browser to control an enterprise edge media p=
roxy and would allow WebRTC media traffic to traverse the enterprise firewa=
ll and to operate in such environment.<br>

<br>
</div>Well... if they really can&#39;t enforce their policies, then by defi=
nition they *should* disable it. =A0That&#39;s a feature, not a bug. =A0And=
 luckily it *can* be disabled, which is also a feature of WebRTC.<br>
<br>
Having said that, I don&#39;t understand why you think an organization that=
 has strict monitoring policies can&#39;t monitor and enforce WebRTC traffi=
c, both the HTTP *and* the media/data channels. =A0They *can* do it. =A0It&=
#39;s just not &quot;automagic&quot;, and not cheap. =A0But that&#39;s ok -=
 I don&#39;t think we need to optimize the deployment complexity or cost st=
ructure for that model.<font color=3D"#888888"><br>

</font></blockquote></div><br>The reason is that with custom signaling prot=
ocols implementing monitoring or enforcing policies would be very hard. BTW=
, this is not only and not primarily about recording traffic. This can be u=
sed to force all the traffic outside of the enterprise to be DTLS-SRTP only=
, but still allow DES-SRTP (or RTP if WebRTC will support it) within the en=
terprise network. <br>
_____________<br>Roman Shpount<br>

--047d7b10d17be3aa9b04bbec2761--

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References: <4F4759DC.7060303@ericsson.com> <387F9047F55E8C42850AD6B3A7A03C6C0E1FEB69@inba-mail01.sonusnet.com> <CALiegfnkYVEpmPV-zSL_4wOY-HiFZN-qJCQCiioaS=5NaqhLZw@mail.gmail.com> <CAD5OKxvtOAxMBx6xDnyfTnEq76oDEm6uj1xL6wGjjrtKUAHy3g@mail.gmail.com> <CABcZeBNZiotPmCfT53uEo+O0xw4xv6tXW1M_G-3A5BHuncsduA@mail.gmail.com> <CAD5OKxvYOY5JZ2mYNGiH1poUBQkyOOycePFijH5H+SxtcdqujQ@mail.gmail.com> <CABkgnnVe-b6Sv=R67bMJk_NQqQwdrRUn6rBm7Gu_CMcfPQwtEg@mail.gmail.com> <4F64FE98.3070605@alcatel-lucent.com> <4F685ED9.2050109@alvestrand.no> <CAD5OKxsVp7px9bHAgxgdqPMxRgppcVUDKt8JHBhyq9qqW3pAMg@mail.gmail.com> <4F68A4CC.9090306@alvestrand.no> <CAD5OKxuiApLKRASc2YuBfkM_8h8wGDPPQ3TdOYGum2yauidA5A@mail.gmail.com> <4F6AECC6.8020004@alvestrand.no> <CAD5OKxsSUeMFYXZMZVqQFWdeEB=30HJuJ=mP9GaYkksBmp1mOA@mail.gmail.com> <03fa01cd087d$57899120$069cb360$@com> <387F9047F55E8C42850AD6B3A7A03C6C0E21E8F5@inba-mail01.sonusnet.com> <062901cd0905$e48bb520$ada31f60$@com>
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From: Roman Shpount <roman@telurix.com>
To: Dan Wing <dwing@cisco.com>
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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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On Fri, Mar 23, 2012 at 11:01 AM, Dan Wing <dwing@cisco.com> wrote:

>
> 'Double encryption' is done because it's the only way to achieve
> security at various layers.
>
>
Probably a side note, but double encryption is often done since each
encryption layer serves completely different purposes.

In case of organizations that care about security of their communications
(such as NSA), it is not only the content of the communication, but the
fact that communication took place between certain parties is a secret. For
example, the fact that president is calling Israeli prime minister in the
middle of the night usually means that something significant is about to
happen and normally is a national secret, even if the content of such
conversation is unknown. To prevent eavesdropping on the communication
parties, some sort of VPN protocol such IPsec is used.

On the other hand, parties that are involved in the secure communication
need to confirm each others identities and ensure that only identified
parties will receive contents of the communication. This is insured by
HTTPS with SRTP in case of WebRTC, or by SIPS with SRTP in case of NSA
deployment.

Bottom line, in most cases double encryption is a required feature and not
a bug.
_____________
Roman Shpount

--047d7b15aee9534f2304bbec5f21
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<br clear=3D"all"><div class=3D"gmail_quote">On Fri, Mar 23, 2012 at 11:01 =
AM, Dan Wing <span dir=3D"ltr">&lt;<a href=3D"mailto:dwing@cisco.com">dwing=
@cisco.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=
=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<br>
&#39;Double encryption&#39; is done because it&#39;s the only way to achiev=
e<br>
security at various layers.<br>
<br></blockquote></div><br>Probably a side note, but double encryption is o=
ften done since each encryption layer serves completely different purposes.=
 <br><br>In case of organizations that care about security of their communi=
cations (such as NSA), it is not only the content of the communication, but=
 the fact that communication took place between certain parties is a secret=
. For example, the fact that president is calling Israeli prime minister in=
 the middle of the night usually means that something significant is about =
to happen and normally is a national secret, even if the content of such co=
nversation is unknown. To prevent eavesdropping on the communication partie=
s, some sort of VPN protocol such IPsec is used.<br>
<br>On the other hand, parties that are involved in the secure communicatio=
n need to confirm each others identities and ensure that only identified pa=
rties will receive contents of the communication. This is insured by HTTPS =
with SRTP in case of WebRTC, or by SIPS with SRTP in case of NSA deployment=
.<br>
<br>Bottom line, in most cases double encryption is a required feature and =
not a bug. <br>_____________<br>Roman Shpount<br>
<br><br>

--047d7b15aee9534f2304bbec5f21--

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I will be on official travel from 3/23 - 4/02 and will be slow in responding
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For Kalliste questions please contact Susie Wee
For RTCWeb questions please contact Ethan Hugg.

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From dwing@cisco.com  Fri Mar 23 10:35:32 2012
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From: "Dan Wing" <dwing@cisco.com>
To: "'Hutton, Andrew'" <andrew.hutton@siemens-enterprise.com>, "'Roman Shpount'" <roman@telurix.com>, "'Harald Alvestrand'" <harald@alvestrand.no>, "'Ted Hardie'" <ted.ietf@gmail.com>
References: <4F4759DC.7060303@ericsson.com>	<387F9047F55E8C42850AD6B3A7A03C6C0E1FEB69@inba-mail01.sonusnet.com>	<CALiegfnkYVEpmPV-zSL_4wOY-HiFZN-qJCQCiioaS=5NaqhLZw@mail.gmail.com>	<CAD5OKxvtOAxMBx6xDnyfTnEq76oDEm6uj1xL6wGjjrtKUAHy3g@mail.gmail.com>	<CABcZeBNZiotPmCfT53uEo+O0xw4xv6tXW1M_G-3A5BHuncsduA@mail.gmail.com>	<CAD5OKxvYOY5JZ2mYNGiH1poUBQkyOOycePFijH5H+SxtcdqujQ@mail.gmail.com>	<CABkgnnVe-b6Sv=R67bMJk_NQqQwdrRUn6rBm7Gu_CMcfPQwtEg@mail.gmail.com>	<4F64FE98.3070605@alcatel-lucent.com>	<4F685ED9.2050109@alvestrand.no>	<CAD5OKxsVp7px9bHAgxgdqPMxRgppcVUDKt8JHBhyq9qqW3pAMg@mail.gmail.com>	<4F68A4CC.9090306@alvestrand.no>	<CAD5OKxuiApLKRASc2YuBfkM_8h8wGDPPQ3TdOYGum2yauidA5A@mail.gmail.com>	<4F6AECC6.8020004@alvestrand.no>	<CAD5OKxsSUeMFYXZMZVqQFWdeEB=30HJuJ=mP9GaYkksBmp1mOA@mail.gmail.com> <03fa01cd087d$57899120$069cb360$@com> <101C6067BEC68246B0C3F6843BCCC1E31296B58E75@MCHP058A.global-ad.net>
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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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> -----Original Message-----
> From: Hutton, Andrew [mailto:andrew.hutton@siemens-enterprise.com]
> Sent: Friday, March 23, 2012 7:55 AM
> To: Dan Wing; 'Roman Shpount'; 'Harald Alvestrand'; 'Ted Hardie'
> Cc: rtcweb@ietf.org
> Subject: RE: [rtcweb] Resolving RTP/SDES question in Paris
> 
> Hi Dan,
> 
> I don't think SIPREC has a solution to the particular problem that
> Roman is talking about and it would probably not be within scope of
> SIPREC to solve this even assuming that there is some SIP involved.
> 
> If the prison deployed its own RTCWeb application and restricted the
> prisoner to using only this application then of course there would be
> no problem with intercepting the media and a SIPREC type solution could
> be deployed with some kind of B2BUA which has access to both signaling
> and media initiating the media recording and of course informing the
> prisoner that they are being recorded.


If recording by the jail is necessary, the identity provider would need
to the jail when you're in jail.  Or, if working at a company that
needs to record, that company would be your identity provider.  I 
work for Cisco, and my email address is dwing@cisco.com -- Cisco clearly
owns my @cisco.com identity.

The identity provider can do many things.  For WEBRTC, the identity
provider can force themselves into being able to decrypt the SRTP session,
and they can prevent a session from being established if the end user 
(dwing@cisco.com) refuses to allow that decryption.

> However if the prisoner is allowed to access their lawyers RTCWeb
> application and download the JavaScript from there then SIPREC does not
> help the prison record the resulting media exchanged.

Putting RTCWEB aside for a moment, Javascript can establish a 
Diffie-Hellman exchange between a client and server.  After the DH exchange,
the prisoner can now securely communicate with their lawyer -- but has
to type and read (tomorrow, with RTCWEB, the prisoner could speak and
listen).  If the lawyer or prisoner were worried the DH exchange or the
Javascript itself might be compromised, they can use out of band 
techniques (exchanging pieces of paper) to verify the DH exchange 
and checksum the Javascript.  Javascript could also be used to operate 
a one-time-pad encryption system.  I think the problem with 'jails'
is not new to RTCWEB, and the problem exists if the problem space 
includes a prisoner with has access to a computer that can run code
obtained from outside the jail, such as from the Internet, such as
any Javascript.

If the concern is Javascript providing a stenography channel,
that problem is unsolvable.  That Javascript could, for all we know,
send SRTP with RTCWEB, or encrypted text messages without RTCWEB,
over any channel it wants -- including over DNS, for example.  
There is code available that sends IP over DNS, such as 
http://code.kryo.se/iodine)

-d


> Regards
> Andy
> 
> 
> > -----Original Message-----
> > From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> > Behalf Of Dan Wing
> > Sent: 22 March 2012 22:44
> > To: 'Roman Shpount'; 'Harald Alvestrand'; 'Ted Hardie'
> > Cc: rtcweb@ietf.org
> > Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
> >
> > > -----Original Message-----
> > > From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> > > Behalf Of Roman Shpount
> > > Sent: Thursday, March 22, 2012 7:19 AM
> > > To: Harald Alvestrand
> > > Cc: rtcweb@ietf.org
> > > Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
> > ...
> > > Yes, I was worried about supporting the interceptor. This is all
> > > related to my question of support of WebRTC applications in
> military,
> > > prisons, or financial organizations. I think WebRTC would be
> > completely
> > > disabled in such locations unless web browser can be configured to
> > > confirm to some sort of communications and monitoring policy. I do
> > not
> > > think enabling only certain applications is a sustainable solution,
> > > since application can only be enabled based on its URL, but this in
> > no
> > > way implies that the actual protocol used for signaling exchange by
> > > this application will stay the same. In the best case this will
> > result
> > > in tens of specialized monitoring rules that will need to be
> > > maintained. In the worst case, WebRTC would be simply disabled. At
> > this
> > > point the only workable solution that I see is some sort of media
> > proxy
> > > protocol.
> >
> > The SIPREC working group, formed in March 2010, has solutions for
> > this problem.  The problem is not created by RTCWEB.  Disabling
> > encryption is not necessary -- nor desirable.  "Jails" get brought up
> > in this context often, so using that same example, consider a
> prisoner
> > at a jail communicating with their lawyer.  The prisoner needs secure
> > communications with their lawyer without the risk of the
> > communications being leaked to the press by anyone along the media
> > path.  For details see http://tools.ietf.org/wg/siprec/
> >
> > We will have a brietf presentation on SIPREC during my session
> > at RTCWEB, explaining how it can work with RTCWEB.
> >
> > -d
> >
> >
> >
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb


From roman@telurix.com  Fri Mar 23 10:45:13 2012
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References: <4F4759DC.7060303@ericsson.com> <387F9047F55E8C42850AD6B3A7A03C6C0E1FEB69@inba-mail01.sonusnet.com> <CALiegfnkYVEpmPV-zSL_4wOY-HiFZN-qJCQCiioaS=5NaqhLZw@mail.gmail.com> <CAD5OKxvtOAxMBx6xDnyfTnEq76oDEm6uj1xL6wGjjrtKUAHy3g@mail.gmail.com> <CABcZeBNZiotPmCfT53uEo+O0xw4xv6tXW1M_G-3A5BHuncsduA@mail.gmail.com> <CAD5OKxvYOY5JZ2mYNGiH1poUBQkyOOycePFijH5H+SxtcdqujQ@mail.gmail.com> <CABkgnnVe-b6Sv=R67bMJk_NQqQwdrRUn6rBm7Gu_CMcfPQwtEg@mail.gmail.com> <4F64FE98.3070605@alcatel-lucent.com> <4F685ED9.2050109@alvestrand.no> <CAD5OKxsVp7px9bHAgxgdqPMxRgppcVUDKt8JHBhyq9qqW3pAMg@mail.gmail.com> <B60F5A76-8ADB-4A55-8F1C-0438F7D7EA80@acmepacket.com>
Date: Fri, 23 Mar 2012 13:45:10 -0400
Message-ID: <CAD5OKxv4t-5LS3u=u7SnaLwnmFnE2uCSQ6D06tYqgp8g-oBPYQ@mail.gmail.com>
From: Roman Shpount <roman@telurix.com>
To: Hadriel Kaplan <HKaplan@acmepacket.com>
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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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On Fri, Mar 23, 2012 at 1:39 AM, Hadriel Kaplan <HKaplan@acmepacket.com>wro=
te:

>
> I was trying to follow this thread backwards from today, to try to grok
> how you got to the conclusion that an organization with strict media
> monitoring policies wouldn't be able to do so for WebRTC.  I think the
> above email is where I get disconnected form the train of thought.
>
> If, as in another email you sent, an organization such as the military or
> jail or whatever controls its network such that they use the whitelist
> model of only authorized applications allowed our of their network, and
> everything goes through the HTTP proxy, then the above statement of:
>
> > Media description is carried in some sort of application defined
> protocol (can even be transmitted over an encrypted SCTP data channel or
> encrypted in javaScript), so monitoring proxy cannot reliably modify it.
>
> =85 makes no sense to me.
>
> If you were preventing unknown applications, you would not allow unknown
> UDP or TCP destination ports to be used to get out of the network.  The
> HTTP Proxy has to (1) know your web application wants to use another set =
of
> ports, and (2) has to allow it.  Ergo, there can be no "encrypted SCTP da=
ta
> channel" that was not previously authorized; nor would encrypting
> SDP-type-info in Javascript help you one iota, because you won't be able =
to
> actually use it to send media, since the HTTP proxy is not opening new
> ports for you to reach the Internet.
>
So I don't understand what the problem is.  If you run a super-tight
> network like that and you *do not* want WebRTC to work because you can't
> monitor it, then by design it won't work.  If you run a super-tight netwo=
rk
> like that and you *do* want WebRTC to work, then you either have to be
> willing to install custom browsers/apps, or your HTTP Proxy has to be abl=
e
> to see the HTTP traffic and correctly parse the Javascript-passed data an=
d
> be a media b2bua.  That's a really nasty set of constraints, especially
> trying to grok the javascript enough to be a media b2bua, and undoubtedly
> means it won't work for many WebRTC websites.  But that's arguably the
> right thing to happen - downloading javascript is akin to downloading a n=
ew
> application for each website, and again by definition this is a network
> that only wants to allow applications it wants to allow, so that's what i=
t
> gets: per-application control, and thus requires per-application
> knowledge/support.  Anything your proxy doesn't understand doesn't work,
> but again that's kinda the point of a super-tight policy.
>
>
My problem with all of this that as far as communication policies are
concerned, for all the practical purposes managing WebRTC becomes just a
big on/off switch. It is not practical to reverse engineer JavaScript
application to enable media proxy/ B2BUA to work transparently just by
monitoring  HTTP communications. Even if such HTTP proxy based B2BUA can be
developed, it will break after the first javascript application update. The
solution you describe is possible but is extremely brittle.


> Personally, if I were running such an org, and I don't know if it's
> possible, but I'd go get some form of rootkit/kernel-mod to intercept
> audio/video/keystrokes well below the user layer, so that you wouldn't ha=
ve
> to worry what application was running - just intercept the
> audio/screen/keyboard somewhere near the drivers.  Presumably some
> companies make such things, for just such organizations? (sort of like th=
e
> equivalent of CarrierIQ)  That way it doesn't matter whether it's WebRTC,
> Flash, or native apps.
>
>
Unfortunately this does not always work. What you are looking for is a way
to enforceme that all the devices on the organizational network either
confirms to organization communication policy or not allowed to
communicate. This also applies to other devices that are being brought to
your organization (ie lawyer laptop in prison). Organization cannot force
everybody to install a key logger to their computers that they bring in,
but they can force all the communications coming from within the
organization to go through some sort of proxy or not to be allowed to
communicate.
_____________
Roman Shpount

--e89a8ff2480106e68104bbec961f
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<br><div class=3D"gmail_quote">On Fri, Mar 23, 2012 at 1:39 AM, Hadriel Kap=
lan <span dir=3D"ltr">&lt;<a href=3D"mailto:HKaplan@acmepacket.com">HKaplan=
@acmepacket.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" =
style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div class=3D"im"><br>
I was trying to follow this thread backwards from today, to try to grok how=
 you got to the conclusion that an organization with strict media monitorin=
g policies wouldn&#39;t be able to do so for WebRTC. =A0I think the above e=
mail is where I get disconnected form the train of thought.<br>
</div>
<br>
If, as in another email you sent, an organization such as the military or j=
ail or whatever controls its network such that they use the whitelist model=
 of only authorized applications allowed our of their network, and everythi=
ng goes through the HTTP proxy, then the above statement of:<br>

<div class=3D"im"><br>
&gt; Media description is carried in some sort of application defined proto=
col (can even be transmitted over an encrypted SCTP data channel or encrypt=
ed in javaScript), so monitoring proxy cannot reliably modify it.<br>
<br>
</div>=85 makes no sense to me.<br>
<br>
If you were preventing unknown applications, you would not allow unknown UD=
P or TCP destination ports to be used to get out of the network. =A0The HTT=
P Proxy has to (1) know your web application wants to use another set of po=
rts, and (2) has to allow it. =A0Ergo, there can be no &quot;encrypted SCTP=
 data channel&quot; that was not previously authorized; nor would encryptin=
g SDP-type-info in Javascript help you one iota, because you won&#39;t be a=
ble to actually use it to send media, since the HTTP proxy is not opening n=
ew ports for you to reach the Internet.<br>
</blockquote><div></div><div></div><blockquote class=3D"gmail_quote" style=
=3D"margin:0pt 0pt 0pt 0.8ex;border-left:1px solid rgb(204,204,204);padding=
-left:1ex">
So I don&#39;t understand what the problem is. =A0If you run a super-tight =
network like that and you *do not* want WebRTC to work because you can&#39;=
t monitor it, then by design it won&#39;t work. =A0If you run a super-tight=
 network like that and you *do* want WebRTC to work, then you either have t=
o be willing to install custom browsers/apps, or your HTTP Proxy has to be =
able to see the HTTP traffic and correctly parse the Javascript-passed data=
 and be a media b2bua. =A0That&#39;s a really nasty set of constraints, esp=
ecially trying to grok the javascript enough to be a media b2bua, and undou=
btedly means it won&#39;t work for many WebRTC websites. =A0But that&#39;s =
arguably the right thing to happen - downloading javascript is akin to down=
loading a new application for each website, and again by definition this is=
 a network that only wants to allow applications it wants to allow, so that=
&#39;s what it gets: per-application control, and thus requires per-applica=
tion knowledge/support. =A0Anything your proxy doesn&#39;t understand doesn=
&#39;t work, but again that&#39;s kinda the point of a super-tight policy.<=
br>

<br></blockquote><div><div><br>
My problem with all of this that as far as communication policies are=20
concerned, for all the practical purposes managing WebRTC becomes just a
 big on/off switch. It is not practical to reverse engineer JavaScript=20
application to enable media proxy/ B2BUA to work transparently just by=20
monitoring=A0 HTTP communications. Even if such HTTP proxy based B2BUA can
 be developed, it will break after the first javascript application=20
update. The solution you describe is possible but is extremely brittle. <br=
>
</div>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0pt 0pt 0p=
t 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
Personally, if I were running such an org, and I don&#39;t know if it&#39;s=
 possible, but I&#39;d go get some form of rootkit/kernel-mod to intercept =
audio/video/keystrokes well below the user layer, so that you wouldn&#39;t =
have to worry what application was running - just intercept the audio/scree=
n/keyboard somewhere near the drivers. =A0Presumably some companies make su=
ch things, for just such organizations? (sort of like the equivalent of Car=
rierIQ) =A0That way it doesn&#39;t matter whether it&#39;s WebRTC, Flash, o=
r native apps.<br>

<font color=3D"#888888"><br></font></blockquote><div>=A0</div><div>Unfortun=
ately this does not always work. What you are looking for is a way to enfor=
ceme that all the devices on the organizational network either confirms to =
organization communication policy or not allowed to communicate. This also =
applies to other devices that are being brought to your organization (ie la=
wyer laptop in prison). Organization cannot force everybody to install a ke=
y logger to their computers that they bring in, but they can force all the =
communications coming from within the organization to go through some sort =
of proxy or not to be allowed to communicate.<br>
</div>_____________<br></div>Roman Shpount<br>

--e89a8ff2480106e68104bbec961f--

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Subject: Re: [rtcweb] On video codec for rtcweb
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--e89a8fb1fffaa5aaa904bbed1e13
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Chrome's WebRTC stack will only include the VP8 codec.

Best Regards,

/Serge

On Fri, Mar 23, 2012 at 12:40, Timothy B. Terriberry <
tterriberry@mozilla.com> wrote:

> Stefan Hakansson LK wrote:
>
>> So, I propose that h.264 should be mandatory to implement.
>>
>
> Mozilla strongly opposes such a proposal. Many will be familiar with our
> announcements regarding our use of platform codecs on B2G and possibly
> Android last week. The primary reasoning behind that decision, i.e. the one
> factor that was not in our power to change, was the availability of content
> on websites. This is not a problem for WebRTC, as browsers control both
> ends of the communication.
>
> What _is_ a problem is licensing and distributing encoders in an
> open-source project. This is quite a bit different than using system
> decoders, as outside of mobile, encoders are usually not available, and
> even if they were, they might not have particularly good quality, nor the
> features to control latency, loss robustness, and other aspects necessary
> for real-time communication.
>
> If you thought that a concession in one place would make Mozilla more
> likely to compromise in others, I assure you the opposite is true.
> Maintaining the use of RF codecs in WebRTC is now even more important than
> it was before. See the words directly from Mozilla's CTO:
> http://groups.google.com/**group/mozilla.dev.platform/**
> msg/53892ad3ea9eeb98<http://groups.google.com/group/mozilla.dev.platform/msg/53892ad3ea9eeb98>
>
> ______________________________**_________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/**listinfo/rtcweb<https://www.ietf.org/mailman/listinfo/rtcweb>
> Google Sweden AB | Kungsbron 2, SE-111 22 Stockholm | Org. nr. 556656-6880
>
> Apparently, this footer is required in Europe. Apologies. This email may
> be confidential or privileged.  If you received this communication by
> mistake, please don't forward it to anyone else,please erase all copies and
> attachments, and please let me know that it went to the wrong person.
>  Thanks.
>  <https://www.ietf.org/mailman/listinfo/rtcweb>

--e89a8fb1fffaa5aaa904bbed1e13
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<div style>Chrome&#39;s WebRTC stack will only include the VP8 codec.</div>=
<div style><br></div><div style>Best Regards,=A0</div><div style><br></div>=
<div style>/Serge</div><br><div class=3D"gmail_quote">On Fri, Mar 23, 2012 =
at 12:40, Timothy B. Terriberry <span dir=3D"ltr">&lt;<a href=3D"mailto:tte=
rriberry@mozilla.com">tterriberry@mozilla.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"im">Stefan Hakansson LK wrote:=
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
So, I propose that h.264 should be mandatory to implement.<br>
</blockquote>
<br></div>
Mozilla strongly opposes such a proposal. Many will be familiar with our an=
nouncements regarding our use of platform codecs on B2G and possibly Androi=
d last week. The primary reasoning behind that decision, i.e. the one facto=
r that was not in our power to change, was the availability of content on w=
ebsites. This is not a problem for WebRTC, as browsers control both ends of=
 the communication.<br>


<br>
What _is_ a problem is licensing and distributing encoders in an open-sourc=
e project. This is quite a bit different than using system decoders, as out=
side of mobile, encoders are usually not available, and even if they were, =
they might not have particularly good quality, nor the features to control =
latency, loss robustness, and other aspects necessary for real-time communi=
cation.<br>


<br>
If you thought that a concession in one place would make Mozilla more likel=
y to compromise in others, I assure you the opposite is true. Maintaining t=
he use of RF codecs in WebRTC is now even more important than it was before=
. See the words directly from Mozilla&#39;s CTO: <a href=3D"http://groups.g=
oogle.com/group/mozilla.dev.platform/msg/53892ad3ea9eeb98" target=3D"_blank=
">http://groups.google.com/<u></u>group/mozilla.dev.platform/<u></u>msg/538=
92ad3ea9eeb98</a><div class=3D"HOEnZb">

<div class=3D"h5"><br>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a></div></div><a href=
=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank"><font c=
lass=3D"Apple-style-span" color=3D"#999999" size=3D"1">Google Sweden AB | K=
ungsbron 2, SE-111 22 Stockholm | Org. nr. 556656-6880=A0</font><br>

<br><font class=3D"Apple-style-span" color=3D"#999999" size=3D"1">Apparentl=
y, this footer is required in Europe. Apologies. This email may be confiden=
tial or privileged. =A0If you received this communication by mistake, pleas=
e don&#39;t forward it to anyone else,please erase all copies and attachmen=
ts, and please let me know that it went to the wrong person. =A0Thanks.</fo=
nt><br>


</a></blockquote></div>

--e89a8fb1fffaa5aaa904bbed1e13--

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To: Roman Shpount <roman@telurix.com>
References: <4F4759DC.7060303@ericsson.com>	<CAD5OKxsVp7px9bHAgxgdqPMxRgppcVUDKt8JHBhyq9qqW3pAMg@mail.gmail.com>	<4F68A4CC.9090306@alvestrand.no>	<CAD5OKxuiApLKRASc2YuBfkM_8h8wGDPPQ3TdOYGum2yauidA5A@mail.gmail.com>	<4F6AECC6.8020004@alvestrand.no>	<CAD5OKxsSUeMFYXZMZVqQFWdeEB=30HJuJ=mP9GaYkksBmp1mOA@mail.gmail.com>	<CABkgnnW3AyREj9T2zDzJf64Bhjdfc1K8-ebZe-V0a-tbiTYEpw@mail.gmail.com>	<CAD5OKxswXun5nVhKcN2oXkTdr-wKuOB7PhbXgdGSE4RexhTd3A@mail.gmail.com>	<CABkgnnUZgtgEWZvYiyxDVNsDRDNsdr30Pd38cZeR18UB1Z04mw@mail.gmail.com>	<CAD5OKxuLY2hZVqmna_GmpcBdFbZiP6ybKTJ19DN4f25mabLTrg@mail.gmail.com>	<CA+9kkMDFEmJVuZu_BCBvU34YbWB-0CCR7SbKt-3PF=XV0vs3JQ@mail.gmail.com>	<CAD5OKxu=JPJyWWPii=t8JGKAKTqVvB5JFQJ-jBBnXmiK4xpPEQ@mail.gmail.com>	<CA+9kkMALJW2H5bM9yiykfBUqWoZifkZWpve=ih+ors1u9mz=7A@mail.gmail.com>	<CAD5OKxva=zm0NUwUE0mhaEOmrDYCvtn3kP701NeRbiBRKhPFJQ@mail.gmail.com>	<13C620B3-1297-4212-B61C-5643E08E9748@acmepacket.com> <CAD5OKxt_i2eJ1hK6tR7aKUt4T+6sitfxdMaBfN-ASR6-UPRE2A@mail.gmail.com >
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Cc: "<rtcweb@ietf.org>" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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On 03/23/2012 06:14 PM, Roman Shpount wrote:
>
>     > 3. Define the solution - protocols and API requirements - for
>     > firewall and NAT traversal.
>
>     We've got that, though not in one doc:
>     http://tools.ietf.org/html/draft-ietf-rtcweb-overview
>     http://tools.ietf.org/html/draft-ietf-rtcweb-use-cases-and-requirements
>     http://tools.ietf.org/html/draft-ietf-rtcweb-jsep
>
>
> Once again, these documents do not cover scenario when organization 
> forces all the traffic to go through HTTP proxy.
Roman,
It seems to me that you are arguing that the scenarios in section 4.1 of 
the use cases document do not cover that specific case, and I think you 
are right in that; the list is:

    The following considerations are applicable to all use cases:
    o  Clients can be on IPv4-only
    o  Clients can be on IPv6-only
    o  Clients can be on dual-stack
    o  Clients can be on wideband (10s of Mbits/sec)
    o  Clients can be on narrowband (10s to 100s of Kbits/sec)
    o  Clients can be on variable-media-quality networks (wireless)
    o  Clients can be on congested networks
    o  Clients can be on firewalled networks with no UDP allowed
    o  Clients can be on networks with cone NAT
    o  Clients can be on networks with symmetric NAT

Now, there are two ways to interpret this omission:

- The WG did not think of that use case when the list was created
- The WG does not want that use case on the list because it constrains 
the solution space too much

If (re)opening this issue, I think I'd find myself in the "do not want 
that use case" camp.

             Harald



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    On 03/23/2012 06:14 PM, Roman Shpount wrote:
    <blockquote
cite="mid:CAD5OKxt_i2eJ1hK6tR7aKUt4T+6sitfxdMaBfN-ASR6-UPRE2A@mail.gmail.com"
      type="cite">
      <div><br>
      </div>
      <blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex;
        border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
        <div class="im">
          &gt; 3. Define the solution - protocols and API requirements -
          for<br>
          &gt; firewall and NAT traversal.<br>
          <br>
        </div>
        We've got that, though not in one doc:<br>
        <a moz-do-not-send="true"
          href="http://tools.ietf.org/html/draft-ietf-rtcweb-overview"
          target="_blank">http://tools.ietf.org/html/draft-ietf-rtcweb-overview</a><br>
        <a moz-do-not-send="true"
href="http://tools.ietf.org/html/draft-ietf-rtcweb-use-cases-and-requirements"
          target="_blank">http://tools.ietf.org/html/draft-ietf-rtcweb-use-cases-and-requirements</a><br>
        <a moz-do-not-send="true"
          href="http://tools.ietf.org/html/draft-ietf-rtcweb-jsep"
          target="_blank">http://tools.ietf.org/html/draft-ietf-rtcweb-jsep</a><br>
        <div class="im"><br>
        </div>
      </blockquote>
      <div><br>
        Once again, these documents do not cover scenario when
        organization forces all the traffic to go through HTTP proxy.</div>
    </blockquote>
    Roman,<br>
    It seems to me that you are arguing that the scenarios in section
    4.1 of the use cases document do not cover that specific case, and I
    think you are right in that; the list is:<br>
    <br>
       The following considerations are applicable to all use cases:<br>
       o  Clients can be on IPv4-only<br>
       o  Clients can be on IPv6-only<br>
       o  Clients can be on dual-stack<br>
       o  Clients can be on wideband (10s of Mbits/sec)<br>
       o  Clients can be on narrowband (10s to 100s of Kbits/sec)<br>
       o  Clients can be on variable-media-quality networks (wireless)<br>
       o  Clients can be on congested networks<br>
       o  Clients can be on firewalled networks with no UDP allowed<br>
       o  Clients can be on networks with cone NAT<br>
       o  Clients can be on networks with symmetric NAT<br>
    <br>
    Now, there are two ways to interpret this omission:<br>
    <br>
    - The WG did not think of that use case when the list was created<br>
    - The WG does not want that use case on the list because it
    constrains the solution space too much<br>
    <br>
    If (re)opening this issue, I think I'd find myself in the "do not
    want that use case" camp.<br>
    <br>
                Harald<br>
    <br>
    <br>
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To: Roman Shpount <roman@telurix.com>
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Cc: "<rtcweb@ietf.org>" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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On 03/23/2012 06:45 PM, Roman Shpount wrote:
>
> Unfortunately this does not always work. What you are looking for is a 
> way to enforceme that all the devices on the organizational network 
> either confirms to organization communication policy or not allowed to 
> communicate. This also applies to other devices that are being brought 
> to your organization (ie lawyer laptop in prison). Organization cannot 
> force everybody to install a key logger to their computers that they 
> bring in, but they can force all the communications coming from within 
> the organization to go through some sort of proxy or not to be allowed 
> to communicate.
>

I think you are looking for http://tools.ietf.org/wg/nea/charters, not 
RTCWEB.



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    On 03/23/2012 06:45 PM, Roman Shpount wrote:
    <blockquote
cite="mid:CAD5OKxv4t-5LS3u=u7SnaLwnmFnE2uCSQ6D06tYqgp8g-oBPYQ@mail.gmail.com"
      type="cite"><font color="#888888"><br>
      </font>
      <div class="gmail_quote">
        <div> </div>
        <div>Unfortunately this does not always work. What you are
          looking for is a way to enforceme that all the devices on the
          organizational network either confirms to organization
          communication policy or not allowed to communicate. This also
          applies to other devices that are being brought to your
          organization (ie lawyer laptop in prison). Organization cannot
          force everybody to install a key logger to their computers
          that they bring in, but they can force all the communications
          coming from within the organization to go through some sort of
          proxy or not to be allowed to communicate.<br>
        </div>
        <br>
      </div>
    </blockquote>
    <br>
    I think you are looking for <a
      href="http://tools.ietf.org/wg/nea/charters">http://tools.ietf.org/wg/nea/charters</a>,
    not RTCWEB.<br>
    <br>
    <br>
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From igor.faynberg@alcatel-lucent.com  Fri Mar 23 11:53:41 2012
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Now I understand  and agree that we cannot change their specification.  
But... do we need to use it?

Igor

On 3/23/2012 7:22 AM, Harald Alvestrand wrote:
> On 03/22/2012 07:17 PM, Igor Faynberg wrote:
>> Thanks! Now I understand.
>>
>> My only follow-up question (and that could probably wait until the 
>> next week) is why not make BrowserID certificate adhere to X.509.  If 
>> we issue self-signed X.509 certificates ourselves, what is in the way 
>> of any IdP doing just that?
> The BrowserID specification is the work item of a group that is 
> entirely outside of this group.
> You can certainly go there (browserid.org) to argue that they should 
> change to X.509 certificates, but we can't decide in the RTCWEB WG 
> that they should change.
>
>>
>> Igor
>>
>> On 3/22/2012 1:53 PM, Eric Rescorla wrote:
>>> Good question.
>>>
>>> 1. BrowserID certificates aren't X.509, and that's all the TLS/DTLS 
>>> will do.
>>> 2. This lets us be agnostic about IdP mechanisms without having to 
>>> change
>>> TLS every time we add a new one.
>>>
>>> -Ekr
>>>
>>>
>>> On Thu, Mar 22, 2012 at 6:22 PM, Igor Faynberg
>>> <igor.faynberg@alcatel-lucent.com>  wrote:
>>>> Eric,
>>>>
>>>> A question:  For the case of BrowserID, I understand that a client 
>>>> gets a
>>>> certificate from an IdP.   If so, why you and I would not use the
>>>> certificates from our respective IdPs in order to authenticate each 
>>>> other
>>>>   in DTLS?
>>>>
>>>> Igor
>>>>
>>>>
>>>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>

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From: snandaku <snandaku@cisco.com>
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From abu_hurayrah@hidayahonline.org  Fri Mar 23 12:01:05 2012
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Date: Fri, 23 Mar 2012 15:00:58 -0400
From: Basil Mohamed Gohar <abu_hurayrah@hidayahonline.org>
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Subject: Re: [rtcweb] On video codec for rtcweb
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On 03/23/2012 12:48 PM, Roman Shpount wrote:
> On Fri, Mar 23, 2012 at 8:34 AM, Timothy B. Terriberry
> <tterriberry@mozilla.com <mailto:tterriberry@mozilla.com>> wrote:
>
>     With a direct browser-to-browser connection, there is nothing
>     sitting between them to do the transcoding. In the DTLS-SRTP case
>     with identity verification, which I think a number of people here
>     view as highly important, you are in fact _guaranteeing_ that
>     nothing but the browser can encode or decode the video.
>
>
> This is not technically correct. You can still build a system that
> implements a media proxy that does nothing but handling ICE,
> encryption, and identity verification without ever transcoding the
> content. The cost of re-encrypting the media is low (ie single E5 Xeon
> CPU can probably do about 10GB/s), but the cost of transcoding video,
> or even audio is huge in comparison. Because of this, supporting a
> comprehensive set of codecs is highly desirable.
While perhaps technically desirable for those already knee-deep in the
existing telecommunications, it will serve to lock-out any WebRTC
implementations that either choose to or cannot implement an encumbered
format, such as free/open source software developers and companies.  As
of right now, there *are* no official complete WebRTC implementations to
inconvenience by choosing a mandatory-to-implement codec.  It would be
only an inconvenience to existing infrastructure, but because of the
free and open nature of the proposed format, adding support for it in
software would be a minimal effort, especially as it would standardized.

The reference implementation of VP8 (libvpx) is also licensed in such as
a way as to be equally available to all types of software.

From snandaku@cisco.com  Fri Mar 23 12:02:21 2012
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From: snandaku <snandaku@cisco.com>
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From ds@ti.com  Fri Mar 23 15:38:50 2012
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From: "Schleef, David" <ds@ti.com>
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From: Roman Shpount <roman@telurix.com>
To: Basil Mohamed Gohar <abu_hurayrah@hidayahonline.org>
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Subject: Re: [rtcweb] On video codec for rtcweb
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--e89a8ff24801be75a604bbf14bfe
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On Fri, Mar 23, 2012 at 3:00 PM, Basil Mohamed Gohar <
abu_hurayrah@hidayahonline.org> wrote:

> On 03/23/2012 12:48 PM, Roman Shpount wrote:
> > On Fri, Mar 23, 2012 at 8:34 AM, Timothy B. Terriberry
> > <tterriberry@mozilla.com <mailto:tterriberry@mozilla.com>> wrote:
> >
> >     With a direct browser-to-browser connection, there is nothing
> >     sitting between them to do the transcoding. In the DTLS-SRTP case
> >     with identity verification, which I think a number of people here
> >     view as highly important, you are in fact _guaranteeing_ that
> >     nothing but the browser can encode or decode the video.
> >
> >
> > This is not technically correct. You can still build a system that
> > implements a media proxy that does nothing but handling ICE,
> > encryption, and identity verification without ever transcoding the
> > content. The cost of re-encrypting the media is low (ie single E5 Xeon
> > CPU can probably do about 10GB/s), but the cost of transcoding video,
> > or even audio is huge in comparison. Because of this, supporting a
> > comprehensive set of codecs is highly desirable.
> While perhaps technically desirable for those already knee-deep in the
> existing telecommunications, it will serve to lock-out any WebRTC
> implementations that either choose to or cannot implement an encumbered
> format, such as free/open source software developers and companies.  As
> of right now, there *are* no official complete WebRTC implementations to
> inconvenience by choosing a mandatory-to-implement codec.  It would be
> only an inconvenience to existing infrastructure, but because of the
> free and open nature of the proposed format, adding support for it in
> software would be a minimal effort, especially as it would standardized.
>
> The reference implementation of VP8 (libvpx) is also licensed in such as
> a way as to be equally available to all types of software.
>
>
I am not arguing for making H.264 a base codec. I would actually prefer
VP8. All I was saying that despite lack of interoperability on encryption,
ICE, and identity, there are still significant benefits of supporting the
same codecs when communicating with legacy equipment, since re-encryption
and media proxy can be handled separately and using much less resources
then transcoding.
_____________
Roman Shpount

--e89a8ff24801be75a604bbf14bfe
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<br clear=3D"all"><div class=3D"gmail_quote">On Fri, Mar 23, 2012 at 3:00 P=
M, Basil Mohamed Gohar <span dir=3D"ltr">&lt;<a href=3D"mailto:abu_hurayrah=
@hidayahonline.org">abu_hurayrah@hidayahonline.org</a>&gt;</span> wrote:<br=
><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1=
px #ccc solid;padding-left:1ex">
<div class=3D"im">On 03/23/2012 12:48 PM, Roman Shpount wrote:<br>
&gt; On Fri, Mar 23, 2012 at 8:34 AM, Timothy B. Terriberry<br>
</div><div><div></div><div class=3D"h5">&gt; &lt;<a href=3D"mailto:tterribe=
rry@mozilla.com">tterriberry@mozilla.com</a> &lt;mailto:<a href=3D"mailto:t=
terriberry@mozilla.com">tterriberry@mozilla.com</a>&gt;&gt; wrote:<br>
&gt;<br>
&gt; =A0 =A0 With a direct browser-to-browser connection, there is nothing<=
br>
&gt; =A0 =A0 sitting between them to do the transcoding. In the DTLS-SRTP c=
ase<br>
&gt; =A0 =A0 with identity verification, which I think a number of people h=
ere<br>
&gt; =A0 =A0 view as highly important, you are in fact _guaranteeing_ that<=
br>
&gt; =A0 =A0 nothing but the browser can encode or decode the video.<br>
&gt;<br>
&gt;<br>
&gt; This is not technically correct. You can still build a system that<br>
&gt; implements a media proxy that does nothing but handling ICE,<br>
&gt; encryption, and identity verification without ever transcoding the<br>
&gt; content. The cost of re-encrypting the media is low (ie single E5 Xeon=
<br>
&gt; CPU can probably do about 10GB/s), but the cost of transcoding video,<=
br>
&gt; or even audio is huge in comparison. Because of this, supporting a<br>
&gt; comprehensive set of codecs is highly desirable.<br>
</div></div>While perhaps technically desirable for those already knee-deep=
 in the<br>
existing telecommunications, it will serve to lock-out any WebRTC<br>
implementations that either choose to or cannot implement an encumbered<br>
format, such as free/open source software developers and companies. =A0As<b=
r>
of right now, there *are* no official complete WebRTC implementations to<br=
>
inconvenience by choosing a mandatory-to-implement codec. =A0It would be<br=
>
only an inconvenience to existing infrastructure, but because of the<br>
free and open nature of the proposed format, adding support for it in<br>
software would be a minimal effort, especially as it would standardized.<br=
>
<br>
The reference implementation of VP8 (libvpx) is also licensed in such as<br=
>
a way as to be equally available to all types of software.<br>
<div><div></div><br></div></blockquote></div><br>I am not arguing for makin=
g H.264 a base codec. I would actually prefer VP8. All I was saying that de=
spite lack of interoperability on encryption, ICE, and identity, there are =
still significant benefits of supporting the same codecs when communicating=
 with legacy equipment, since re-encryption and media proxy can be handled =
separately and using much less resources then transcoding.<br>
_____________<br>Roman Shpount<br>
<br><br>

--e89a8ff24801be75a604bbf14bfe--

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From: Gregory Maxwell <gmaxwell@juniper.net>
To: Roman Shpount <roman@telurix.com>, Basil Mohamed Gohar <abu_hurayrah@hidayahonline.org>
Date: Fri, 23 Mar 2012 17:48:38 -0700
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Roman Shpount wrote:
> I am not arguing for making H.264 a base codec. I would actually prefer=20
> VP8. All I was saying that despite lack of interoperability on=20
> encryption, ICE, and identity, there are still significant benefits of=20
> supporting the same codecs when communicating with legacy equipment,=20

This is all true, but there won't always be a gateway or SBC in the media
path and users may not want one even if all it did was transcode. The case
with no media gateway can _only_ work reliably by virtue of compatibility
being baked into the specification=97 in the other cases interoperability
is 'simply' a matter of different tradeoffs.

An interesting observation on the cost of transcoding: a number of
mobile providers are now transparently transcoding their users' http
video downloads on the fly, not for the purposes of changing the codec,
but solely to make them smaller. This can do terrible things to the video
quality (beyond just the reduction in bitrate: MP4 wasn't designed to
stream from a live source). Perhaps this is an argument for more pervasive
end-to-end encryption=97 but the point remains that CPU power, especially
CPU power in high density data-centers, is still relatively cheap.


From cb.list6@gmail.com  Fri Mar 23 18:10:25 2012
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Subject: Re: [rtcweb] On video codec for rtcweb
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On Mar 23, 2012 5:49 PM, "Gregory Maxwell" <gmaxwell@juniper.net> wrote:
>
> Roman Shpount wrote:
> > I am not arguing for making H.264 a base codec. I would actually prefer
> > VP8. All I was saying that despite lack of interoperability on
> > encryption, ICE, and identity, there are still significant benefits of
> > supporting the same codecs when communicating with legacy equipment,
>
> This is all true, but there won't always be a gateway or SBC in the media
> path and users may not want one even if all it did was transcode. The cas=
e
> with no media gateway can _only_ work reliably by virtue of compatibility
> being baked into the specification=97 in the other cases interoperability
> is 'simply' a matter of different tradeoffs.
>
> An interesting observation on the cost of transcoding: a number of
> mobile providers are now transparently transcoding their users' http
> video downloads on the fly, not for the purposes of changing the codec,
> but solely to make them smaller. This can do terrible things to the video
> quality (beyond just the reduction in bitrate: MP4 wasn't designed to
> stream from a live source). Perhaps this is an argument for more pervasiv=
e
> end-to-end encryption=97 but the point remains that CPU power, especially

Excellent points both about encryption and codec. This carrier transcoding
mess is a real challenge and should be architecturally thwarted.

Cb

> CPU power in high density data-centers, is still relatively cheap.
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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<p><br>
On Mar 23, 2012 5:49 PM, &quot;Gregory Maxwell&quot; &lt;<a href=3D"mailto:=
gmaxwell@juniper.net">gmaxwell@juniper.net</a>&gt; wrote:<br>
&gt;<br>
&gt; Roman Shpount wrote:<br>
&gt; &gt; I am not arguing for making H.264 a base codec. I would actually =
prefer<br>
&gt; &gt; VP8. All I was saying that despite lack of interoperability on<br=
>
&gt; &gt; encryption, ICE, and identity, there are still significant benefi=
ts of<br>
&gt; &gt; supporting the same codecs when communicating with legacy equipme=
nt,<br>
&gt;<br>
&gt; This is all true, but there won&#39;t always be a gateway or SBC in th=
e media<br>
&gt; path and users may not want one even if all it did was transcode. The =
case<br>
&gt; with no media gateway can _only_ work reliably by virtue of compatibil=
ity<br>
&gt; being baked into the specification=97 in the other cases interoperabil=
ity<br>
&gt; is &#39;simply&#39; a matter of different tradeoffs.<br>
&gt;<br>
&gt; An interesting observation on the cost of transcoding: a number of<br>
&gt; mobile providers are now transparently transcoding their users&#39; ht=
tp<br>
&gt; video downloads on the fly, not for the purposes of changing the codec=
,<br>
&gt; but solely to make them smaller. This can do terrible things to the vi=
deo<br>
&gt; quality (beyond just the reduction in bitrate: MP4 wasn&#39;t designed=
 to<br>
&gt; stream from a live source). Perhaps this is an argument for more perva=
sive<br>
&gt; end-to-end encryption=97 but the point remains that CPU power, especia=
lly<br></p>
<p>Excellent points both about encryption and codec. This carrier transcodi=
ng mess is a real challenge and should be architecturally thwarted. </p>
<p>Cb</p>
<p>&gt; CPU power in high density data-centers, is still relatively cheap.<=
br>
&gt;<br>
&gt; _______________________________________________<br>
&gt; rtcweb mailing list<br>
&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.i=
etf.org/mailman/listinfo/rtcweb</a><br>
</p>

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On 23 March 2012 11:53, Igor Faynberg <igor.faynberg@alcatel-lucent.com> wr=
ote:

> Now I understand =C2=A0and agree that we cannot change their specificatio=
n.
> =C2=A0But... do we need to use it?

We intend to use it in Firefox.

 -r

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From: Hadriel Kaplan <HKaplan@acmepacket.com>
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Thread-Topic: [rtcweb] Resolving RTP/SDES question in Paris
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On Mar 23, 2012, at 1:45 PM, Roman Shpount wrote:

Personally, if I were running such an org, and I don't know if it's possibl=
e, but I'd go get some form of rootkit/kernel-mod to intercept audio/video/=
keystrokes well below the user layer, so that you wouldn't have to worry wh=
at application was running - just intercept the audio/screen/keyboard somew=
here near the drivers.  Presumably some companies make such things, for jus=
t such organizations? (sort of like the equivalent of CarrierIQ)  That way =
it doesn't matter whether it's WebRTC, Flash, or native apps.


Unfortunately this does not always work. What you are looking for is a way =
to enforceme that all the devices on the organizational network either conf=
irms to organization communication policy or not allowed to communicate. Th=
is also applies to other devices that are being brought to your organizatio=
n (ie lawyer laptop in prison). Organization cannot force everybody to inst=
all a key logger to their computers that they bring in, but they can force =
all the communications coming from within the organization to go through so=
me sort of proxy or not to be allowed to communicate.

But that *is* their policy, and it *is* being followed: as you said, their =
policy is "all the devices on the organizational network either confirms to=
 organization communication policy or not allowed to communicate".  That is=
 being followed, because all web communications go through the Organization=
's proxy, and if a lawyer comes in his/her web traffic goes through the pro=
xy=85  but the lawyer can't use it for WebRTC-created media - which is exac=
tly matching the Organization's policy, because the lawyer's laptop must no=
t and will not work for WebRTC communications.

I'm not trying to be pedantic or difficult here, or trying to dismiss the u=
se-case.  I know there are many such "super-tight" networks, but I just don=
't see how we can reasonably accommodate their use-case in this WG.  And I'=
m pointing out that we really are following their policies, inasmuch as Web=
RTC media will fail to escape their network - I would be far more worried i=
f we somehow bypassed their policies instead.  At least this way it's simil=
ar behavior they get for unknown/unauthorized applications, which is arguab=
ly what going to random websites and running downloaded javascript is equiv=
alent to.

-hadriel




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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3DWindows-1=
252">
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space; ">
<br>
<div>
<div>On Mar 23, 2012, at 1:45 PM, Roman Shpount wrote:</div>
<br>
<blockquote type=3D"cite">
<div class=3D"gmail_quote">
<blockquote class=3D"gmail_quote" style=3D"margin-top: 0pt; margin-right: 0=
pt; margin-bottom: 0pt; margin-left: 0.8ex; border-left-width: 1px; border-=
left-style: solid; border-left-color: rgb(204, 204, 204); padding-left: 1ex=
; position: static; z-index: auto; ">
Personally, if I were running such an org, and I don't know if it's possibl=
e, but I'd go get some form of rootkit/kernel-mod to intercept audio/video/=
keystrokes well below the user layer, so that you wouldn't have to worry wh=
at application was running - just
 intercept the audio/screen/keyboard somewhere near the drivers. &nbsp;Pres=
umably some companies make such things, for just such organizations? (sort =
of like the equivalent of CarrierIQ) &nbsp;That way it doesn't matter wheth=
er it's WebRTC, Flash, or native apps.<br>
<font color=3D"#888888"><br>
</font></blockquote>
<div>&nbsp;</div>
<div>Unfortunately this does not always work. What you are looking for is a=
 way to enforceme that all the devices on the organizational network either=
 confirms to organization communication policy or not allowed to communicat=
e. This also applies to other devices
 that are being brought to your organization (ie lawyer laptop in prison). =
Organization cannot force everybody to install a key logger to their comput=
ers that they bring in, but they can force all the communications coming fr=
om within the organization to go
 through some sort of proxy or not to be allowed to communicate.<br>
</div>
</div>
</blockquote>
<br>
</div>
<div>But that *is* their policy, and it *is* being followed: as you said, t=
heir policy is &quot;all the devices on the organizational network either c=
onfirms to organization communication policy or not allowed to communicate&=
quot;. &nbsp;That is being followed, because all
 web communications go through the Organization's proxy, and if a lawyer co=
mes in his/her
<i>web</i> traffic goes through the proxy=85 &nbsp;but the lawyer can't use=
 it for WebRTC-created media - which is exactly matching the Organization's=
 policy, because the lawyer's laptop must not and will not work for WebRTC =
communications.</div>
<div><br>
</div>
<div>I'm not trying to be pedantic or difficult here, or trying to dismiss =
the use-case. &nbsp;I know there are many such &quot;super-tight&quot; netw=
orks, but I just don't see how we can reasonably accommodate their use-case=
 in this WG. &nbsp;And I'm pointing out that we really
 are following their policies, inasmuch as WebRTC media will fail to escape=
 their network - I would be far more worried if we somehow bypassed their p=
olicies instead. &nbsp;At least this way it's similar behavior they get for=
 unknown/unauthorized applications, which
 is arguably what going to random websites and running downloaded javascrip=
t is equivalent to.</div>
<div><br>
</div>
<div>-hadriel</div>
<div><br>
</div>
<div><br>
</div>
<br>
</body>
</html>

--_000_86CD900905574F949E9F89DAD2D80FF1acmepacketcom_--

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From: Tim Panton <tim@phonefromhere.com>
Date: Sat, 24 Mar 2012 09:15:09 +0000
To: "Cavigioli, Chris" <chris.cavigioli@intel.com>
Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] On video codec for rtcweb
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Sent from my iPhone

On 23 Mar 2012, at 12:34, "Cavigioli, Chris" <chris.cavigioli@intel.com> wro=
te:

> ITU-T G.711, both u_law (North America) and A_law (Europe) formats, can be=
 supported by everyone with very trivial software burden

I hope that doesn't mean we forget the plan to support Opus.=20

We need an adaptive codec with fec in order to get decent audio out of real w=
orld networks, like 3G, cafe wifi, vsat etc.=20

G711 is ok for perfect networks with legacy interop needs, but pretty much u=
seless anywhere else.=20

Tim.=

From fluffy@cisco.com  Sat Mar 24 04:38:24 2012
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Subject: [rtcweb] RTCWeb DEMO's Thursday at Lunch
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We will be doing RTCWeb related demos in room 242AB on Thursday =
1130-1300. Please let the chairs know if there is a demo you would like =
to show.=20

There will be no food provided - I suspect what we will do is just run =
demos from 11:30 to 12:00 and then give people time to go for lunch. The =
chairs will see what people want to demo and then plan a schedule.=20

If you have some code that might run, lets us know and lets see it. =20

Thanks, Cullen







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Subject: [rtcweb] Encryption consideration: SDES and TLS-SRP
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Hi all,

i wanted to ask the WG opinion on the encryption framework for the RTCWEB.

In particular as we know today there is a strong need to have an
encryption-by-default on most new protocols.

Today:
- Email provider encryption by default traffic
- Mobile operators encrypt by default traffic
- Almost all web transaction use encryption

So it's realistic and practical to think that VoIP future is
encrypted-by-default to at least basically protect against the casual
wiretapper. (Can we evaluate this assumption?)

In that case we may have different threat scenario:
- Eavesdropping by third party with access to user's communication line
- Eavesdropping by who is providing the telephony service

A simpler protection scenario would at least provide protection against
third party with access to user's communication line.
A stronger protection scenario would also provide protection against who
is providing the service.

Here we should consider the security of:
- Signaling
- Media

Signaling has to be protected with TLS.
Media has to be protected with SRTP that rely on DTLS or on the new
proposal to use SDES [1] (idea that i like).

So basically in both cases there is the need to rely on a TLS-secured
channel and here we should evaluate the possible weak points.

As you know the TLS based trust model based on "Browser/OS Pre-loaded
Trusted CA" revealed serious risks posed to internet trust model [2] .

So, because we are defining a stack that should look forward the future,
we cannot not-consider seriously the serious issues related with TLS and
way to manage it.

An idea that came to my mind is that the specification, to face a
future-proof approach to the internet trust issues, could mandate to
support the implementation of TLS-SRP.

TLS-SRP is a new standard that let the TLS key exchange be authenticated
by a login/password with no need to trust a specific CA.
Status of implementation of RFC5054 http://tools.ietf.org/html/rfc5054
is available on http://trustedhttp.org/wiki/Main_Page .

If we embrace SDES key exchange and TLS-SRP authentication the overall
protocol stack would be simpler than having to deal with DTLS and
relying only on Trusted-CA architectures.

[1]
http://www.ietf.org/proceedings/interim/2012/01/31/rtcweb/slides/rtcweb-7.pdf
[2] http://ritter.vg/p/2012-TLS-Survey.pdf

p.s. I'm quite new to the list, so i may repeat some discussion
elements. In that case i kindly ask to leave to archive pointer to
already made discussion.

From roman@telurix.com  Sat Mar 24 16:56:44 2012
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References: <4F6C5A5E.6050100@ericsson.com> <4F6C6138.6010908@mozilla.com> <E44893DD4E290745BB608EB23FDDB76219E813@008-AM1MPN1-041.mgdnok.nokia.com> <E36D1A4AE0B6AA4091F1728D584A6AD215EC3936@ORSMSX104.amr.corp.intel.com> <7D8EC27B-2446-4992-B5F5-5CC7A17D2BCE@phonefromhere.com>
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On Sat, Mar 24, 2012 at 5:15 AM, Tim Panton <tim@phonefromhere.com> wrote:

> G711 is ok for perfect networks with legacy interop needs, but pretty much
> useless anywhere else.
>
>
Not that I have anything against OPUS (except that it is not a standard
yet), but G711 works just fine for calls within local LAN or over a
wireline network. If you have the bandwidth, you can run it with 16 or 32
KHz sampling rate and it will sound considerably better then any compressed
codec. Since we are getting more and more bandwidth between end points, in
the future we might not need to compress audio at all. It is not just for
interop.
_____________
Roman Shpount

--047d7b15aee97214b504bc05e437
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<br clear=3D"all"><div class=3D"gmail_quote">On Sat, Mar 24, 2012 at 5:15 A=
M, Tim Panton <span dir=3D"ltr">&lt;<a href=3D"mailto:tim@phonefromhere.com=
">tim@phonefromhere.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail=
_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:=
1ex">
G711 is ok for perfect networks with legacy interop needs, but pretty much =
useless anywhere else.<br>
<font color=3D"#888888"></font><br></blockquote></div><br>Not that I have a=
nything against OPUS (except that it is not a standard yet), but G711 works=
 just fine for calls within local LAN or over a wireline network. If you ha=
ve the bandwidth, you can run it with 16 or 32 KHz sampling rate and it wil=
l sound considerably better then any compressed codec. Since we are getting=
 more and more bandwidth between end points, in the future we might not nee=
d to compress audio at all. It is not just for interop.<br>
_____________<br>Roman Shpount<br>
<br><br>

--047d7b15aee97214b504bc05e437--

From jmvalin@mozilla.com  Sat Mar 24 17:34:37 2012
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] On video codec for rtcweb
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On 24/03/12 07:56 PM, Roman Shpount wrote:
> Not that I have anything against OPUS (except that it is not a standard
> yet), but G711 works just fine for calls within local LAN or over a
> wireline network. If you have the bandwidth, you can run it with 16 or
> 32 KHz sampling rate and it will sound considerably better then any
> compressed codec. 

Have you ever listened to G.711 at 16 kHz? Certainly not "better then
any compressed codec". In fact, the white quantization noise will
usually be louder than the higher frequencies of the signal. And 32 kHz
will just be even worse.

	Jean-Marc

From harald@alvestrand.no  Sun Mar 25 13:12:09 2012
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Subject: Re: [rtcweb] Encryption consideration: SDES and TLS-SRP
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Please read draft-ietf-rtcweb-security and draft-ietf-rtcweb-security-arch.

I'm sorry, but not even checking the names of drafts the working group 
is working on before asking questions is below the minimum bar we expect 
of participants posting long emails.

                 Harald

On 03/24/2012 11:41 PM, Fabio Pietrosanti (naif) wrote:
> Hi all,
>
> i wanted to ask the WG opinion on the encryption framework for the RTCWEB.
>
> In particular as we know today there is a strong need to have an
> encryption-by-default on most new protocols.
>
> Today:
> - Email provider encryption by default traffic
> - Mobile operators encrypt by default traffic
> - Almost all web transaction use encryption
>
> So it's realistic and practical to think that VoIP future is
> encrypted-by-default to at least basically protect against the casual
> wiretapper. (Can we evaluate this assumption?)
>
> In that case we may have different threat scenario:
> - Eavesdropping by third party with access to user's communication line
> - Eavesdropping by who is providing the telephony service
>
> A simpler protection scenario would at least provide protection against
> third party with access to user's communication line.
> A stronger protection scenario would also provide protection against who
> is providing the service.
>
> Here we should consider the security of:
> - Signaling
> - Media
>
> Signaling has to be protected with TLS.
> Media has to be protected with SRTP that rely on DTLS or on the new
> proposal to use SDES [1] (idea that i like).
>
> So basically in both cases there is the need to rely on a TLS-secured
> channel and here we should evaluate the possible weak points.
>
> As you know the TLS based trust model based on "Browser/OS Pre-loaded
> Trusted CA" revealed serious risks posed to internet trust model [2] .
>
> So, because we are defining a stack that should look forward the future,
> we cannot not-consider seriously the serious issues related with TLS and
> way to manage it.
>
> An idea that came to my mind is that the specification, to face a
> future-proof approach to the internet trust issues, could mandate to
> support the implementation of TLS-SRP.
>
> TLS-SRP is a new standard that let the TLS key exchange be authenticated
> by a login/password with no need to trust a specific CA.
> Status of implementation of RFC5054 http://tools.ietf.org/html/rfc5054
> is available on http://trustedhttp.org/wiki/Main_Page .
>
> If we embrace SDES key exchange and TLS-SRP authentication the overall
> protocol stack would be simpler than having to deal with DTLS and
> relying only on Trusted-CA architectures.
>
> [1]
> http://www.ietf.org/proceedings/interim/2012/01/31/rtcweb/slides/rtcweb-7.pdf
> [2] http://ritter.vg/p/2012-TLS-Survey.pdf
>
> p.s. I'm quite new to the list, so i may repeat some discussion
> elements. In that case i kindly ask to leave to archive pointer to
> already made discussion.
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


From harald@alvestrand.no  Mon Mar 26 02:12:17 2012
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Subject: [rtcweb] Dependency graphs for RTCWEB
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Just FYI....

the automatically generated dependency graphs for RTCWEB are visible here:

http://fenron.net/~fenner/ietf/deps/viz/rtcweb-norm.pdf 
<http://fenron.net/%7Efenner/ietf/deps/viz/rtcweb-norm.pdf> (normative 
references only)

http://fenron.net/~fenner/ietf/deps/viz/rtcweb.pdf 
<http://fenron.net/%7Efenner/ietf/deps/viz/rtcweb.pdf> (all references)

Worthy of note.

                     Harald




From fluffy@cisco.com  Mon Mar 26 05:17:20 2012
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Subject: [rtcweb] Moving to WEDNESDAY Re: RTCWeb DEMO's Thursday at Lunch -
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This meeting is going to move to Wednesday 1130-1300 in room 242AB.=20

This was moved to avoid the conflict with the lunch talk titled "Status =
of Standards for Cloud Storage" being given by Scality as they are a =
paying sponsor of this IETF. Instead we will move to conflict with the =
Cisco lunch talk - I really doubt that Cisco will have a problem with =
this.

I'd got some side questions about what this was and wanted it to be =
clear - this is to look at the running code of folks building the RTCWeb =
specs and gets some feedback from implementors about the specs as well =
as help folks understand how the implementations matched (or fail to =
match) the specs.=20

Cullen

On Mar 24, 2012, at 12:38 , Cullen Jennings wrote:

>=20
> We will be doing RTCWeb related demos in room 242AB on Thursday =
1130-1300. Please let the chairs know if there is a demo you would like =
to show.=20
>=20
> There will be no food provided - I suspect what we will do is just run =
demos from 11:30 to 12:00 and then give people time to go for lunch. The =
chairs will see what people want to demo and then plan a schedule.=20
>=20
> If you have some code that might run, lets us know and lets see it. =20=

>=20
> Thanks, Cullen
>=20
>=20
>=20
>=20
>=20
>=20


From ajrmeyn@gmail.com  Tue Mar 27 00:52:04 2012
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Subject: [rtcweb] Query regarding rtcweb use cases and requirements.
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Hi, This is with regards to the following document:Web Real-Time =
Communication Use-cases and Requirements
draft-ietf-rtcweb-use-cases-and-requirements-06.txt
http://datatracker.ietf.org/doc/draft-ietf-rtcweb-use-cases-and-requireme=
nts/?include_text=3D1=20
I have noticed that the use cases with regards to Data API of web real =
time communication, have not been addressed. For example: use cases that =
are related to file sharing between browsers.=20

is there any particular reason for the same?

Regards,
Antony Meyn
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<HTML><HEAD>
<META content=3D"text/html; charset=3Diso-8859-1" =
http-equiv=3DContent-Type></HEAD>
<BODY dir=3Dltr>
<DIV dir=3Dltr>
<DIV style=3D"FONT-FAMILY: 'Arial'; COLOR: #000000; FONT-SIZE: 10pt">
<DIV=20
style=3D"FONT-STYLE: normal; DISPLAY: inline; FONT-FAMILY: 'Calibri'; =
COLOR: #000000; FONT-SIZE: small; FONT-WEIGHT: normal; TEXT-DECORATION: =
none">
<DIV dir=3Dltr>
<DIV style=3D"FONT-FAMILY: 'Arial'; COLOR: #000000; FONT-SIZE: =
10pt"><PRE style=3D"LINE-HEIGHT: 12pt; BACKGROUND-COLOR: =
rgb(255,255,255); MARGIN-TOP: 0px; MARGIN-BOTTOM: 0px">Hi,</PRE><PRE =
style=3D"LINE-HEIGHT: 12pt; BACKGROUND-COLOR: rgb(255,255,255); =
MARGIN-TOP: 0px; MARGIN-BOTTOM: 0px">&nbsp;</PRE><PRE =
style=3D"LINE-HEIGHT: 12pt; BACKGROUND-COLOR: rgb(255,255,255); =
MARGIN-TOP: 0px; MARGIN-BOTTOM: 0px">This is with regards to the =
following document:</PRE><PRE style=3D"LINE-HEIGHT: 12pt; =
BACKGROUND-COLOR: rgb(255,255,255); MARGIN-TOP: 0px; MARGIN-BOTTOM: =
0px">Web Real-Time Communication Use-cases and Requirements
draft-ietf-rtcweb-use-cases-and-requirements-06.txt
</PRE><A=20
href=3D"http://datatracker.ietf.org/doc/draft-ietf-rtcweb-use-cases-and-r=
equirements/?include_text=3D1"=20
target=3D_blank><FONT=20
face=3DTahoma>http://datatracker.ietf.org/doc/draft-ietf-rtcweb-use-cases=
-and-requirements/?include_text=3D1</FONT></A><FONT=20
face=3DTahoma> </FONT>
<DIV><FONT face=3DTahoma>I have noticed that the use cases with regards =
to Data=20
API of web real time communication, have not been addressed. For =
example: use=20
cases that are related to file sharing between browsers. </FONT></DIV>
<DIV><FONT face=3DTahoma></FONT>&nbsp;</DIV>
<DIV><FONT face=3DTahoma>is there any particular reason for the =
same?</FONT></DIV>
<DIV><FONT face=3DTahoma></FONT>&nbsp;</DIV>
<DIV><FONT face=3DTahoma>Regards,</FONT></DIV>
<DIV><FONT face=3DTahoma>Antony=20
Meyn</FONT></DIV></DIV></DIV></DIV></DIV></DIV></BODY></HTML>

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From harald@alvestrand.no  Tue Mar 27 03:36:28 2012
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Subject: Re: [rtcweb] Query regarding rtcweb use cases and requirements.
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On 03/27/2012 09:51 AM, ajrmeyn@gmail.com wrote:
> Hi,
>   
> This is with regards to the following document:
> Web Real-Time Communication Use-cases and Requirements
> draft-ietf-rtcweb-use-cases-and-requirements-06.txt
> http://datatracker.ietf.org/doc/draft-ietf-rtcweb-use-cases-and-requirements/?include_text=1
> I have noticed that the use cases with regards to Data API of web real 
> time communication, have not been addressed. For example: use cases 
> that are related to file sharing between browsers.
> is there any particular reason for the same?
Requirement F23 is a data-passing requirement.

For communications that did not need low latency, the WG felt that the 
RTCWEB requirements were not different from the requirements of any 
other application, so those could be fulfilled with other types of 
interfaces, including server-mediated file transfers.

The WG is suggesting data API/protocol solutions that can be used for 
other things than fulfiling the most narrow interpretation of the 
requirement listed, but did not feel a need to add a requirement 
covering those.

The purpose of the "use cases and requirements" is to find a minimum set 
of things that the protocol suite has to be able to do in order to be 
"good enough", not to delineate everything the protocol can be used for.

My interpretation....

                       Harald




> Regards,
> Antony Meyn
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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    On 03/27/2012 09:51 AM, <a class="moz-txt-link-abbreviated" href="mailto:ajrmeyn@gmail.com">ajrmeyn@gmail.com</a> wrote:
    <blockquote cite="mid:FA86515700364A16B4EDF69E88E3FA1A@devbox"
      type="cite">
      <meta content="text/html; charset=ISO-8859-1"
        http-equiv="Content-Type">
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        <div style="font-family: 'Arial'; color: rgb(0, 0, 0);
          font-size: 10pt;">
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            font-weight: normal; text-decoration: none;">
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                font-size: 10pt;">
                <pre style="line-height: 12pt; background-color: rgb(255, 255, 255); margin-top: 0px; margin-bottom: 0px;">Hi,</pre>
                <pre style="line-height: 12pt; background-color: rgb(255, 255, 255); margin-top: 0px; margin-bottom: 0px;">&nbsp;</pre>
                <pre style="line-height: 12pt; background-color: rgb(255, 255, 255); margin-top: 0px; margin-bottom: 0px;">This is with regards to the following document:</pre>
                <pre style="line-height: 12pt; background-color: rgb(255, 255, 255); margin-top: 0px; margin-bottom: 0px;">Web Real-Time Communication Use-cases and Requirements
draft-ietf-rtcweb-use-cases-and-requirements-06.txt
</pre>
                <a moz-do-not-send="true"
href="http://datatracker.ietf.org/doc/draft-ietf-rtcweb-use-cases-and-requirements/?include_text=1"
                  target="_blank"><font face="Tahoma">http://datatracker.ietf.org/doc/draft-ietf-rtcweb-use-cases-and-requirements/?include_text=1</font></a><font
                  face="Tahoma"> </font>
                <div><font face="Tahoma">I have noticed that the use
                    cases with regards to Data API of web real time
                    communication, have not been addressed. For example:
                    use cases that are related to file sharing between
                    browsers. </font></div>
                <div>&nbsp;</div>
                <div><font face="Tahoma">is there any particular reason
                    for the same?</font></div>
              </div>
            </div>
          </div>
        </div>
      </div>
    </blockquote>
    Requirement F23 is a data-passing requirement.<br>
    <br>
    For communications that did not need low latency, the WG felt that
    the RTCWEB requirements were not different from the requirements of
    any other application, so those could be fulfilled with other types
    of interfaces, including server-mediated file transfers.<br>
    <br>
    The WG is suggesting data API/protocol solutions that can be used
    for other things than fulfiling the most narrow interpretation of
    the requirement listed, but did not feel a need to add a requirement
    covering those.<br>
    <br>
    The purpose of the "use cases and requirements" is to find a minimum
    set of things that the protocol suite has to be able to do in order
    to be "good enough", not to delineate everything the protocol can be
    used for.<br>
    <br>
    My interpretation....<br>
    <br>
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Harald<br>
    <br>
    <br>
    <br>
    <br>
    <blockquote cite="mid:FA86515700364A16B4EDF69E88E3FA1A@devbox"
      type="cite">
      <div dir="ltr">
        <div style="font-family: 'Arial'; color: rgb(0, 0, 0);
          font-size: 10pt;">
          <div style="font-style: normal; display: inline; font-family:
            'Calibri'; color: rgb(0, 0, 0); font-size: small;
            font-weight: normal; text-decoration: none;">
            <div dir="ltr">
              <div style="font-family: 'Arial'; color: rgb(0, 0, 0);
                font-size: 10pt;">
                <div>&nbsp;</div>
                <div><font face="Tahoma">Regards,</font></div>
                <div><font face="Tahoma">Antony Meyn</font></div>
              </div>
            </div>
          </div>
        </div>
      </div>
      <pre wrap="">
<fieldset class="mimeAttachmentHeader"></fieldset>
_______________________________________________
rtcweb mailing list
<a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
    </blockquote>
    <br>
  </body>
</html>

--------------040102050509010701030701--

From ietf@meetecho.com  Tue Mar 27 07:43:58 2012
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Subject: [rtcweb] Meetecho support for RTCWEB WG meeting session
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Hi all,

a virtual room has been reserved on the Meetecho system for Wednesday's 
RTCWEB WG meeting session.

Access to the on-line session (including audio and video streams) will 
be available at:
http://www.meetecho.com/ietf83/rtcweb

The Meetecho session automatically logs you into the standard IETF 
jabber room. So, from there, you can have an integrated experience 
involving all media and allowing you to interact with the room.
Remote participants might also send their own voice to the room, if they 
want to, by either calling a landline phone number, or using our 
embedded VoIP applet (in this last case they are *strongly* advised to 
use a headset).

A tutorial of interactivity features of the tool can be found at:
http://www.meetecho.com/ietf83/tutorials

Cheers,
the Meetecho team

-- 
Meetecho s.r.l.
Web Conferencing and Collaboration Tools
www.meetecho.com

From martin.thomson@gmail.com  Tue Mar 27 20:28:27 2012
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Date: Wed, 28 Mar 2012 05:28:24 +0200
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I finally got around to posting this.

Since the version that I circulated, I've fleshed out the security
considerations a little.  I didn't remove the extra round trip,
because I forgot.

From:  <internet-drafts@ietf.org>
A new version of I-D, draft-thomson-rtcweb-ice-dtls-00.txt has been
successfully submitted by Martin Thomson and posted to the IETF
repository.

From rbarnes@bbn.com  Wed Mar 28 02:15:34 2012
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Subject: [rtcweb] SRTP and "marketing"
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I didn't make it to the mic at the meeting today, but I wanted to 
express one concern about the possibility of making RTCWEB SRTP-only.

Hadriel mentioned the "marketing value" of having always-on encryption, 
this idea that only supporting SRTP will make RTCWEB look like something 
secure and trustworthy.  I'm concerned that this might not be the case, 
and in fact that being SRTP-only might effectively be an over-promise, 
in light of the fact the absence of universal authentication.

Hadriel noted that the competitors to this technology are Skype and 
Flash, and it's worth considering the security situation with these 
technologies, because they kind of bracket RTCWEB.  With Skype (assuming 
they've designed it properly), there is actually a universal 
authentication, under a single authority.  So you really do know that 
you're talking to whatever Skype ID you intend to, and nobody else. 
With Flash, well, does anyone expect it to be secure anyway?

What I'm concerned about in the RTCWEB context is that without a 
universal authentication/identity infrastructure, we will end up 
*promising* a secure call, but not *delivering* it.  I haven't done the 
analysis, but it does not seem implausible to me that FireSheep-like 
vulnerabilities are lurking here.

So ISTM the "marketing" argument carries with it some serious risks as 
well as some small possible benefit.

--Richard

From harald@alvestrand.no  Wed Mar 28 03:13:41 2012
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On 03/28/2012 11:15 AM, Richard L. Barnes wrote:
> I didn't make it to the mic at the meeting today, but I wanted to 
> express one concern about the possibility of making RTCWEB SRTP-only.
>
> Hadriel mentioned the "marketing value" of having always-on 
> encryption, this idea that only supporting SRTP will make RTCWEB look 
> like something secure and trustworthy.  I'm concerned that this might 
> not be the case, and in fact that being SRTP-only might effectively be 
> an over-promise, in light of the fact the absence of universal 
> authentication.
>
> Hadriel noted that the competitors to this technology are Skype and 
> Flash, and it's worth considering the security situation with these 
> technologies, because they kind of bracket RTCWEB.  With Skype 
> (assuming they've designed it properly), there is actually a universal 
> authentication, under a single authority.  So you really do know that 
> you're talking to whatever Skype ID you intend to, and nobody else. 
> With Flash, well, does anyone expect it to be secure anyway?
>
> What I'm concerned about in the RTCWEB context is that without a 
> universal authentication/identity infrastructure, we will end up 
> *promising* a secure call, but not *delivering* it.  I haven't done 
> the analysis, but it does not seem implausible to me that 
> FireSheep-like vulnerabilities are lurking here.
If there are, we need to close them before we ship the specs.
Given reasonable practices (such as using only HTTPS for loading the 
pages and JS libraries), if we can't deliver security (against known 
attacks), we shouldn't ship the spec.
>
> So ISTM the "marketing" argument carries with it some serious risks as 
> well as some small possible benefit.
Only if we don't deliver.
>
> --Richard
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


From rbarnes@bbn.com  Wed Mar 28 03:43:36 2012
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>> I didn't make it to the mic at the meeting today, but I wanted to
>> express one concern about the possibility of making RTCWEB SRTP-only.
>>
>> Hadriel mentioned the "marketing value" of having always-on
>> encryption, this idea that only supporting SRTP will make RTCWEB look
>> like something secure and trustworthy. I'm concerned that this might
>> not be the case, and in fact that being SRTP-only might effectively be
>> an over-promise, in light of the fact the absence of universal
>> authentication.
>>
>> Hadriel noted that the competitors to this technology are Skype and
>> Flash, and it's worth considering the security situation with these
>> technologies, because they kind of bracket RTCWEB. With Skype
>> (assuming they've designed it properly), there is actually a universal
>> authentication, under a single authority. So you really do know that
>> you're talking to whatever Skype ID you intend to, and nobody else.
>> With Flash, well, does anyone expect it to be secure anyway?
>>
>> What I'm concerned about in the RTCWEB context is that without a
>> universal authentication/identity infrastructure, we will end up
>> *promising* a secure call, but not *delivering* it. I haven't done the
>> analysis, but it does not seem implausible to me that FireSheep-like
>> vulnerabilities are lurking here.
> If there are, we need to close them before we ship the specs.
> Given reasonable practices (such as using only HTTPS for loading the
> pages and JS libraries), if we can't deliver security (against known
> attacks), we shouldn't ship the spec.

I agree that we should lock things down to the extent we can, and I 
think we will be able to do pretty well, using things such as you note.

But at base, without authentication, you cannot prevent MitM if someone 
is in the right place in the network.  You can just constrain the set of 
places from which an MitM can be launched.


>> So ISTM the "marketing" argument carries with it some serious risks as
>> well as some small possible benefit.
> Only if we don't deliver.

Except that "deliver" in this case entails a global 
authentication/identity infrastructure.  And it seems unlikely that 
we'll deliver that in the short run.

--Richard

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--e89a8fb1ede0561c3004bc4b71f7
Content-Type: text/plain; charset=ISO-8859-1

On Wed, Mar 28, 2012 at 2:15 AM, Richard L. Barnes <rbarnes@bbn.com> wrote:

> [...]
> What I'm concerned about in the RTCWEB context is that without a universal
> authentication/identity infrastructure, we will end up *promising* a secure
> call, but not *delivering* it.  I haven't done the analysis, but it does
> not seem implausible to me that FireSheep-like vulnerabilities are lurking
> here.
>
>
>

The choices of framework proposed in today's meeting still carry an overall
undercurrent of the same generic mechanism as a SAML-based authentication
and authorization.
Even if a universal authentication infrastructure should exist - it becomes
a potential single point of failure (imagine that being the defunct
diginotar) or non-success (MS Passport).
Too many trust-anchors (IdPs) is a problem as well for the single end-user
(non-enterprise). But in the end - would users prefer to go with the
trust-anchors they have come to associate with and have gained a reputation
for; or something completely new?

Even with identity - the authoritative case proposes a <name>:<domain>
paradigm and in the 3rd party case too - assertions are based on
association of a user to domain - by an outside idP. Thus, there's
significant closeness in the identity form - regardless of whether it is
the most common RFC822 (email address), SIP URI (with a slight exception of
OpenID) or other URI forms.

-mani

> So ISTM the "marketing" argument carries with it some serious risks as
> well as some small possible benefit.
>
> --Richard
> ______________________________**_________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/**listinfo/rtcweb<https://www.ietf.org/mailman/listinfo/rtcweb>
>

--e89a8fb1ede0561c3004bc4b71f7
Content-Type: text/html; charset=ISO-8859-1
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<br><br><div class=3D"gmail_quote">On Wed, Mar 28, 2012 at 2:15 AM, Richard=
 L. Barnes <span dir=3D"ltr">&lt;<a href=3D"mailto:rbarnes@bbn.com">rbarnes=
@bbn.com</a>&gt;</span> wrote:<br><blockquote style=3D"margin:0px 0px 0px 0=
.8ex;padding-left:1ex;border-left-color:rgb(204,204,204);border-left-width:=
1px;border-left-style:solid" class=3D"gmail_quote">
<p>[...]<br>
What I&#39;m concerned about in the RTCWEB context is that without a univer=
sal authentication/identity infrastructure, we will end up *promising* a se=
cure call, but not *delivering* it. =A0I haven&#39;t done the analysis, but=
 it does not seem implausible to me that FireSheep-like vulnerabilities are=
 lurking here.<br>

</p><p>=A0</p></blockquote><div>=A0</div><div>The choices of framework prop=
osed in today&#39;s meeting still carry an overall undercurrent of the same=
 generic mechanism as a SAML-based authentication and authorization.</div>
<div>Even if a universal authentication infrastructure should exist - it be=
comes a potential single point of failure (imagine that being the defunct d=
iginotar) or non-success (MS Passport).</div><div>Too many trust-anchors (I=
dPs) is a problem as well for the single end-user (non-enterprise). But in =
the end - would users prefer to go with the trust-anchors they have come to=
 associate with and have gained a reputation for; or something completely n=
ew?</div>
<div>=A0</div><div>Even with identity - the authoritative case proposes a &=
lt;name&gt;:&lt;domain&gt; paradigm and in the 3rd party case too - asserti=
ons are based on association of a user to domain - by an outside idP. Thus,=
 there&#39;s significant closeness in the identity form - regardless of whe=
ther it is the most common RFC822 (email address), SIP URI (with a slight e=
xception of OpenID) or other URI forms.</div>
<div>=A0</div><div>-mani</div><blockquote style=3D"margin:0px 0px 0px 0.8ex=
;padding-left:1ex;border-left-color:rgb(204,204,204);border-left-width:1px;=
border-left-style:solid" class=3D"gmail_quote">
So ISTM the &quot;marketing&quot; argument carries with it some serious ris=
ks as well as some small possible benefit.<br>
<br>
--Richard<br>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</blockquote></div><br>

--e89a8fb1ede0561c3004bc4b71f7--

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Changing the subject line...

On 03/28/2012 12:55 PM, Mahalingam Mani wrote:
>
>
> On Wed, Mar 28, 2012 at 2:15 AM, Richard L. Barnes <rbarnes@bbn.com 
> <mailto:rbarnes@bbn.com>> wrote:
>
>     [...]
>     What I'm concerned about in the RTCWEB context is that without a
>     universal authentication/identity infrastructure, we will end up
>     *promising* a secure call, but not *delivering* it.  I haven't
>     done the analysis, but it does not seem implausible to me that
>     FireSheep-like vulnerabilities are lurking here.
>
> The choices of framework proposed in today's meeting still carry an 
> overall undercurrent of the same generic mechanism as a SAML-based 
> authentication and authorization.
> Even if a universal authentication infrastructure should exist - it 
> becomes a potential single point of failure (imagine that being the 
> defunct diginotar) or non-success (MS Passport).
> Too many trust-anchors (IdPs) is a problem as well for the single 
> end-user (non-enterprise). But in the end - would users prefer to go 
> with the trust-anchors they have come to associate with and have 
> gained a reputation for; or something completely new?
> Even with identity - the authoritative case proposes a <name>:<domain> 
> paradigm and in the 3rd party case too - assertions are based on 
> association of a user to domain - by an outside idP. Thus, there's 
> significant closeness in the identity form - regardless of whether it 
> is the most common RFC822 (email address), SIP URI (with a slight 
> exception of OpenID) or other URI forms.

This actually shows that the IdP proposal brings a dependency with it on 
the domain hierarchy.
This has real issues, the most important one perhaps being that domain 
identities can't be guaranteed over time; even when real-life 
organizational identities exist, it's not unusual to see the domain name 
change hands multiple times, either voluntarily (selling) or 
involuntarily (UDRP, court cases, forgetting to renew and being 
drop-kicked).

Not sure where discussion of this problem belongs; it might not be a 
problem if one never has to verify a stored identity.

I think the proposals bring significant value even in the absence of 
standardized identity verification - FireSheep is a good example, for 
instance; with HTTPS, FireSheep is powerless as long as it's not in 
cahoots with a corrupted CA - which defeats a huge subset of the set of 
potential attackers.




--------------020302080109010309070300
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: 7bit

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
  <head>
    <meta content="text/html; charset=ISO-8859-1"
      http-equiv="Content-Type">
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  <body bgcolor="#ffffff" text="#000000">
    Changing the subject line...<br>
    <br>
    On 03/28/2012 12:55 PM, Mahalingam Mani wrote:
    <blockquote
cite="mid:CAN8ZsXCtRcFG4a9MOFa-pgBBZG-yCXAJ47K4wh31JtprArgNjA@mail.gmail.com"
      type="cite"><br>
      <br>
      <div class="gmail_quote">On Wed, Mar 28, 2012 at 2:15 AM, Richard
        L. Barnes <span dir="ltr">&lt;<a moz-do-not-send="true"
            href="mailto:rbarnes@bbn.com">rbarnes@bbn.com</a>&gt;</span>
        wrote:<br>
        <blockquote style="margin: 0px 0px 0px 0.8ex; padding-left: 1ex;
          border-left: 1px solid rgb(204, 204, 204);"
          class="gmail_quote">
          <p>[...]<br>
            What I'm concerned about in the RTCWEB context is that
            without a universal authentication/identity infrastructure,
            we will end up *promising* a secure call, but not
            *delivering* it. &nbsp;I haven't done the analysis, but it does
            not seem implausible to me that FireSheep-like
            vulnerabilities are lurking here.<br>
          </p>
          <p>&nbsp;</p>
        </blockquote>
        <div>&nbsp;</div>
        <div>The choices of framework proposed in today's meeting still
          carry an overall undercurrent of the same generic mechanism as
          a SAML-based authentication and authorization.</div>
        <div>Even if a universal authentication infrastructure should
          exist - it becomes a potential single point of failure
          (imagine that being the defunct diginotar) or non-success (MS
          Passport).</div>
        <div>Too many trust-anchors (IdPs) is a problem as well for the
          single end-user (non-enterprise). But in the end - would users
          prefer to go with the trust-anchors they have come to
          associate with and have gained a reputation for; or something
          completely new?</div>
        <div>&nbsp;</div>
        <div>Even with identity - the authoritative case proposes a
          &lt;name&gt;:&lt;domain&gt; paradigm and in the 3rd party case
          too - assertions are based on association of a user to domain
          - by an outside idP. Thus, there's significant closeness in
          the identity form - regardless of whether it is the most
          common RFC822 (email address), SIP URI (with a slight
          exception of OpenID) or other URI forms.</div>
      </div>
    </blockquote>
    <br>
    This actually shows that the IdP proposal brings a dependency with
    it on the domain hierarchy.<br>
    This has real issues, the most important one perhaps being that
    domain identities can't be guaranteed over time; even when real-life
    organizational identities exist, it's not unusual to see the domain
    name change hands multiple times, either voluntarily (selling) or
    involuntarily (UDRP, court cases, forgetting to renew and being
    drop-kicked).<br>
    <br>
    Not sure where discussion of this problem belongs; it might not be a
    problem if one never has to verify a stored identity.<br>
    <br>
    I think the proposals bring significant value even in the absence of
    standardized identity verification - FireSheep is a good example,
    for instance; with HTTPS, FireSheep is powerless as long as it's not
    in cahoots with a corrupted CA - which defeats a huge subset of the
    set of potential attackers.<br>
    <br>
    <br>
    <br>
  </body>
</html>

--------------020302080109010309070300--

From magnus.westerlund@ericsson.com  Wed Mar 28 07:50:29 2012
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WG,

In todays RTCWEB WG meeting there was discussion around media security
mechanism. In this meeting there was some clear consensus in the
meeting which we would like to confirm on the list.

The first was that there was overwhelming consensus that all RTP
packets SHALL be protected by SRTP.

Secondly that no one objected against making DTLS-SRTP a mandatory to
implement and the default keying mechanism. Additional mechanisms are
not precluded.

WG participants may state their position regarding these consensus calls
until 12th of April when the chairs will declare the final consensus. If
you where present in the meeting room and comment on this, please
indicate that.

Best Regards

Magnus Westerlund
For the WG chairs


From abu_hurayrah@hidayahonline.org  Wed Mar 28 07:51:35 2012
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On 03/28/2012 05:15 AM, Richard L. Barnes wrote:
> I didn't make it to the mic at the meeting today, but I wanted to
> express one concern about the possibility of making RTCWEB SRTP-only.
>
> Hadriel mentioned the "marketing value" of having always-on
> encryption, this idea that only supporting SRTP will make RTCWEB look
> like something secure and trustworthy.  I'm concerned that this might
> not be the case, and in fact that being SRTP-only might effectively be
> an over-promise, in light of the fact the absence of universal
> authentication.
>
> Hadriel noted that the competitors to this technology are Skype and
> Flash, and it's worth considering the security situation with these
> technologies, because they kind of bracket RTCWEB.  With Skype
> (assuming they've designed it properly), there is actually a universal
> authentication, under a single authority.  So you really do know that
> you're talking to whatever Skype ID you intend to, and nobody else.
> With Flash, well, does anyone expect it to be secure anyway?
>
> What I'm concerned about in the RTCWEB context is that without a
> universal authentication/identity infrastructure, we will end up
> *promising* a secure call, but not *delivering* it.  I haven't done
> the analysis, but it does not seem implausible to me that
> FireSheep-like vulnerabilities are lurking here.
>
> So ISTM the "marketing" argument carries with it some serious risks as
> well as some small possible benefit.
>
> --Richard
If I'm totally missing the point (not having attended the event
mentioned), please disregard what I have to say.

I think that there is a value to having an encrypted connection, even if
it does not combine with it authentication from both ends, as this
would, at the very least, prevent snooping of the communications between
the two end points.  This would be the equivalent of using a self-signed
certificate for HTTPS, without the drama of the certificate warnings we
get in browsers.

What I mean is, we can make the higher degree of security be one that
combines authentication with encryption, but always-on encryption can be
the default, and this is what I meant by saying it still has value.  At
least a 3rd-party would not be able to spy on communications between two
people and the connection has a minimum degree of privacy to it. 
Combined with a trusted method of establishing authentication, whether
via a central authority (e.g., current CA's) or allowing users to
exchanged keys via other channels (e.g., PGP chain of trust style), then
a fully trustworthy and private method of communications can be established.

I hope the two degrees I am pointing out are clear.

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On 03/28/2012 10:50 AM, Magnus Westerlund wrote:
> WG,
>
> In todays RTCWEB WG meeting there was discussion around media security
> mechanism. In this meeting there was some clear consensus in the
> meeting which we would like to confirm on the list.
>
> The first was that there was overwhelming consensus that all RTP
> packets SHALL be protected by SRTP.
>
> Secondly that no one objected against making DTLS-SRTP a mandatory to
> implement and the default keying mechanism. Additional mechanisms are
> not precluded.
>
> WG participants may state their position regarding these consensus calls
> until 12th of April when the chairs will declare the final consensus. If
> you where present in the meeting room and comment on this, please
> indicate that.
>
> Best Regards
>
> Magnus Westerlund
> For the WG chairs
I already brought-up my concerns in the other thread, so I'll summarize
the core point I was making here.  Would using SRTP *require* a central
authority for establishing authenticity, or can authenticity be
established via a point-to-point means (e.g., how it's traditionally
done via SSH [i.e., upon first connection or via previous key exchange])?

This is about degrees of trust that the user is will to place upon
various methods, of course.  I am stating that the option should exist
for authenticity of an end point to be established outside of a central
authority (e.g., key exchange via other means).

From ekr@rtfm.com  Wed Mar 28 08:14:35 2012
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From: Eric Rescorla <ekr@rtfm.com>
Date: Wed, 28 Mar 2012 17:13:53 +0200
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On Wed, Mar 28, 2012 at 4:55 PM, Basil Mohamed Gohar
<abu_hurayrah@hidayahonline.org> wrote:
> On 03/28/2012 10:50 AM, Magnus Westerlund wrote:
>> WG,
>>
>> In todays RTCWEB WG meeting there was discussion around media security
>> mechanism. In this meeting there was some clear consensus in the
>> meeting which we would like to confirm on the list.
>>
>> The first was that there was overwhelming consensus that all RTP
>> packets SHALL be protected by SRTP.
>>
>> Secondly that no one objected against making DTLS-SRTP a mandatory to
>> implement and the default keying mechanism. Additional mechanisms are
>> not precluded.
>>
>> WG participants may state their position regarding these consensus calls
>> until 12th of April when the chairs will declare the final consensus. If
>> you where present in the meeting room and comment on this, please
>> indicate that.
>>
>> Best Regards
>>
>> Magnus Westerlund
>> For the WG chairs
> I already brought-up my concerns in the other thread, so I'll summarize
> the core point I was making here. =A0Would using SRTP *require* a central
> authority for establishing authenticity, or can authenticity be
> established via a point-to-point means (e.g., how it's traditionally
> done via SSH [i.e., upon first connection or via previous key exchange])?
>
> This is about degrees of trust that the user is will to place upon
> various methods, of course. =A0I am stating that the option should exist
> for authenticity of an end point to be established outside of a central
> authority (e.g., key exchange via other means).

This will still be possible. I agree that it's very important in some
applications.

-Ekr

From roman@telurix.com  Wed Mar 28 08:49:26 2012
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As I have mentioned before on this list I am strongly against making SRTP
protection for RTP a requirement. I think this is an unnecessary
requirement that serves little real purpose except feeding into some
marketing message that most of the WebRTC users would not care about.
Unless use of identity is also a requirement, requiring SRTP will provide
security only in a very narrow sense of the word. At the same time I do
believe that extra standard requirements will stifle innovation and  will
complicate new service or application creation.

I have no objection to making DTLS-SRTP a required to implement protocol.
_____________
Roman Shpount


On Wed, Mar 28, 2012 at 10:50 AM, Magnus Westerlund <
magnus.westerlund@ericsson.com> wrote:

> WG,
>
> In todays RTCWEB WG meeting there was discussion around media security
> mechanism. In this meeting there was some clear consensus in the
> meeting which we would like to confirm on the list.
>
> The first was that there was overwhelming consensus that all RTP
> packets SHALL be protected by SRTP.
>
> Secondly that no one objected against making DTLS-SRTP a mandatory to
> implement and the default keying mechanism. Additional mechanisms are
> not precluded.
>
> WG participants may state their position regarding these consensus calls
> until 12th of April when the chairs will declare the final consensus. If
> you where present in the meeting room and comment on this, please
> indicate that.
>
> Best Regards
>
> Magnus Westerlund
> For the WG chairs
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--047d7b3395a11f86ed04bc4f8d3c
Content-Type: text/html; charset=ISO-8859-1
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As I have mentioned before on this list I am strongly against making SRTP p=
rotection for RTP a requirement. I think this is an unnecessary requirement=
 that serves little real purpose except feeding into some marketing message=
 that most of the WebRTC users would not care about. Unless use of identity=
 is also a requirement, requiring SRTP will provide security only in a very=
 narrow sense of the word. At the same time I do believe that extra standar=
d requirements will stifle innovation and=A0 will complicate new service or=
 application creation.<br>
<br>I have no objection to making DTLS-SRTP a required to implement protoco=
l.<br clear=3D"all">_____________<br>Roman Shpount<br>
<br><br><div class=3D"gmail_quote">On Wed, Mar 28, 2012 at 10:50 AM, Magnus=
 Westerlund <span dir=3D"ltr">&lt;<a href=3D"mailto:magnus.westerlund@erics=
son.com">magnus.westerlund@ericsson.com</a>&gt;</span> wrote:<br><blockquot=
e class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc sol=
id;padding-left:1ex">
WG,<br>
<br>
In todays RTCWEB WG meeting there was discussion around media security<br>
mechanism. In this meeting there was some clear consensus in the<br>
meeting which we would like to confirm on the list.<br>
<br>
The first was that there was overwhelming consensus that all RTP<br>
packets SHALL be protected by SRTP.<br>
<br>
Secondly that no one objected against making DTLS-SRTP a mandatory to<br>
implement and the default keying mechanism. Additional mechanisms are<br>
not precluded.<br>
<br>
WG participants may state their position regarding these consensus calls<br=
>
until 12th of April when the chairs will declare the final consensus. If<br=
>
you where present in the meeting room and comment on this, please<br>
indicate that.<br>
<br>
Best Regards<br>
<br>
Magnus Westerlund<br>
For the WG chairs<br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div><br>

--047d7b3395a11f86ed04bc4f8d3c--

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Roman,

I think there is a misunderstanding (I assume you did not attend the 
meeting today).  It has been clarified that SRTP allows the NULL 
encryption algorithm, and that this option will be available.

Igor

On 3/28/2012 11:49 AM, Roman Shpount wrote:
> As I have mentioned before on this list I am strongly against making 
> SRTP protection for RTP a requirement. I think this is an unnecessary 
> requirement that serves little real purpose except feeding into some 
> marketing message that most of the WebRTC users would not care about. 
> Unless use of identity is also a requirement, requiring SRTP will 
> provide security only in a very narrow sense of the word. At the same 
> time I do believe that extra standard requirements will stifle 
> innovation and  will complicate new service or application creation.
>
> I have no objection to making DTLS-SRTP a required to implement protocol.
> _____________
> Roman Shpount
>
>
> On Wed, Mar 28, 2012 at 10:50 AM, Magnus Westerlund 
> <magnus.westerlund@ericsson.com 
> <mailto:magnus.westerlund@ericsson.com>> wrote:
>
>     WG,
>
>     In todays RTCWEB WG meeting there was discussion around media security
>     mechanism. In this meeting there was some clear consensus in the
>     meeting which we would like to confirm on the list.
>
>     The first was that there was overwhelming consensus that all RTP
>     packets SHALL be protected by SRTP.
>
>     Secondly that no one objected against making DTLS-SRTP a mandatory to
>     implement and the default keying mechanism. Additional mechanisms are
>     not precluded.
>
>     WG participants may state their position regarding these consensus
>     calls
>     until 12th of April when the chairs will declare the final
>     consensus. If
>     you where present in the meeting room and comment on this, please
>     indicate that.
>
>     Best Regards
>
>     Magnus Westerlund
>     For the WG chairs
>
>     _______________________________________________
>     rtcweb mailing list
>     rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>     https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

--------------000406000806040000090208
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<html>
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    <meta content="text/html; charset=ISO-8859-1"
      http-equiv="Content-Type">
  </head>
  <body text="#000000" bgcolor="#ffffff">
    Roman,<br>
    <br>
    I think there is a misunderstanding (I assume you did not attend the
    meeting today).&nbsp; It has been clarified that SRTP allows the NULL
    encryption algorithm, and that this option will be available.<br>
    <br>
    Igor<br>
    <br>
    On 3/28/2012 11:49 AM, Roman Shpount wrote:
    <blockquote
cite="mid:CAD5OKxs6NHha2egNSTumEaHYJ0bB6qu_nfshmBM6dntx2n49HQ@mail.gmail.com"
      type="cite">As I have mentioned before on this list I am strongly
      against making SRTP protection for RTP a requirement. I think this
      is an unnecessary requirement that serves little real purpose
      except feeding into some marketing message that most of the WebRTC
      users would not care about. Unless use of identity is also a
      requirement, requiring SRTP will provide security only in a very
      narrow sense of the word. At the same time I do believe that extra
      standard requirements will stifle innovation and&nbsp; will complicate
      new service or application creation.<br>
      <br>
      I have no objection to making DTLS-SRTP a required to implement
      protocol.<br clear="all">
      _____________<br>
      Roman Shpount<br>
      <br>
      <br>
      <div class="gmail_quote">On Wed, Mar 28, 2012 at 10:50 AM, Magnus
        Westerlund <span dir="ltr">&lt;<a moz-do-not-send="true"
            href="mailto:magnus.westerlund@ericsson.com">magnus.westerlund@ericsson.com</a>&gt;</span>
        wrote:<br>
        <blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt
          0.8ex; border-left: 1px solid rgb(204, 204, 204);
          padding-left: 1ex;">
          WG,<br>
          <br>
          In todays RTCWEB WG meeting there was discussion around media
          security<br>
          mechanism. In this meeting there was some clear consensus in
          the<br>
          meeting which we would like to confirm on the list.<br>
          <br>
          The first was that there was overwhelming consensus that all
          RTP<br>
          packets SHALL be protected by SRTP.<br>
          <br>
          Secondly that no one objected against making DTLS-SRTP a
          mandatory to<br>
          implement and the default keying mechanism. Additional
          mechanisms are<br>
          not precluded.<br>
          <br>
          WG participants may state their position regarding these
          consensus calls<br>
          until 12th of April when the chairs will declare the final
          consensus. If<br>
          you where present in the meeting room and comment on this,
          please<br>
          indicate that.<br>
          <br>
          Best Regards<br>
          <br>
          Magnus Westerlund<br>
          For the WG chairs<br>
          <br>
          _______________________________________________<br>
          rtcweb mailing list<br>
          <a moz-do-not-send="true" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
          <a moz-do-not-send="true"
            href="https://www.ietf.org/mailman/listinfo/rtcweb"
            target="_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
        </blockquote>
      </div>
      <br>
      <pre wrap="">
<fieldset class="mimeAttachmentHeader"></fieldset>
_______________________________________________
rtcweb mailing list
<a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
    </blockquote>
  </body>
</html>

--------------000406000806040000090208--

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Actually, I was confused about that - it seems to me it's still an open ite=
m whether to allow null cipher or not.

-hadriel


On Mar 28, 2012, at 5:56 PM, Igor Faynberg wrote:

Roman,

I think there is a misunderstanding (I assume you did not attend the meetin=
g today).  It has been clarified that SRTP allows the NULL encryption algor=
ithm, and that this option will be available.

Igor

On 3/28/2012 11:49 AM, Roman Shpount wrote:
As I have mentioned before on this list I am strongly against making SRTP p=
rotection for RTP a requirement. I think this is an unnecessary requirement=
 that serves little real purpose except feeding into some marketing message=
 that most of the WebRTC users would not care about. Unless use of identity=
 is also a requirement, requiring SRTP will provide security only in a very=
 narrow sense of the word. At the same time I do believe that extra standar=
d requirements will stifle innovation and  will complicate new service or a=
pplication creation.

I have no objection to making DTLS-SRTP a required to implement protocol.
_____________
Roman Shpount


On Wed, Mar 28, 2012 at 10:50 AM, Magnus Westerlund <magnus.westerlund@eric=
sson.com<mailto:magnus.westerlund@ericsson.com>> wrote:
WG,

In todays RTCWEB WG meeting there was discussion around media security
mechanism. In this meeting there was some clear consensus in the
meeting which we would like to confirm on the list.

The first was that there was overwhelming consensus that all RTP
packets SHALL be protected by SRTP.

Secondly that no one objected against making DTLS-SRTP a mandatory to
implement and the default keying mechanism. Additional mechanisms are
not precluded.

WG participants may state their position regarding these consensus calls
until 12th of April when the chairs will declare the final consensus. If
you where present in the meeting room and comment on this, please
indicate that.

Best Regards

Magnus Westerlund
For the WG chairs

_______________________________________________
rtcweb mailing list
rtcweb@ietf.org<mailto:rtcweb@ietf.org>
https://www.ietf.org/mailman/listinfo/rtcweb



_______________________________________________
rtcweb mailing list
rtcweb@ietf.org<mailto:rtcweb@ietf.org>
https://www.ietf.org/mailman/listinfo/rtcweb


_______________________________________________
rtcweb mailing list
rtcweb@ietf.org<mailto:rtcweb@ietf.org>
https://www.ietf.org/mailman/listinfo/rtcweb


--_000_91F0E66EA75E4369A0109712FF90258Dacmepacketcom_
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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Diso-8859-=
1">
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space; ">
<div><br>
</div>
<div>Actually, I was confused about that - it seems to me it's still an ope=
n item whether to allow null cipher or not.</div>
<div><br>
</div>
<div>-hadriel</div>
<div><br>
</div>
<br>
<div>
<div>On Mar 28, 2012, at 5:56 PM, Igor Faynberg wrote:</div>
<br class=3D"Apple-interchange-newline">
<blockquote type=3D"cite">
<div text=3D"#000000" bgcolor=3D"#ffffff">Roman,<br>
<br>
I think there is a misunderstanding (I assume you did not attend the meetin=
g today).&nbsp; It has been clarified that SRTP allows the NULL encryption =
algorithm, and that this option will be available.<br>
<br>
Igor<br>
<br>
On 3/28/2012 11:49 AM, Roman Shpount wrote:
<blockquote cite=3D"mid:CAD5OKxs6NHha2egNSTumEaHYJ0bB6qu_nfshmBM6dntx2n49HQ=
@mail.gmail.com" type=3D"cite">
As I have mentioned before on this list I am strongly against making SRTP p=
rotection for RTP a requirement. I think this is an unnecessary requirement=
 that serves little real purpose except feeding into some marketing message=
 that most of the WebRTC users would
 not care about. Unless use of identity is also a requirement, requiring SR=
TP will provide security only in a very narrow sense of the word. At the sa=
me time I do believe that extra standard requirements will stifle innovatio=
n and&nbsp; will complicate new service
 or application creation.<br>
<br>
I have no objection to making DTLS-SRTP a required to implement protocol.<b=
r clear=3D"all">
_____________<br>
Roman Shpount<br>
<br>
<br>
<div class=3D"gmail_quote">On Wed, Mar 28, 2012 at 10:50 AM, Magnus Westerl=
und <span dir=3D"ltr">
&lt;<a moz-do-not-send=3D"true" href=3D"mailto:magnus.westerlund@ericsson.c=
om">magnus.westerlund@ericsson.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin: 0pt 0pt 0pt
          0.8ex; border-left: 1px solid rgb(204, 204, 204);
          padding-left: 1ex;">
WG,<br>
<br>
In todays RTCWEB WG meeting there was discussion around media security<br>
mechanism. In this meeting there was some clear consensus in the<br>
meeting which we would like to confirm on the list.<br>
<br>
The first was that there was overwhelming consensus that all RTP<br>
packets SHALL be protected by SRTP.<br>
<br>
Secondly that no one objected against making DTLS-SRTP a mandatory to<br>
implement and the default keying mechanism. Additional mechanisms are<br>
not precluded.<br>
<br>
WG participants may state their position regarding these consensus calls<br=
>
until 12th of April when the chairs will declare the final consensus. If<br=
>
you where present in the meeting room and comment on this, please<br>
indicate that.<br>
<br>
Best Regards<br>
<br>
Magnus Westerlund<br>
For the WG chairs<br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a moz-do-not-send=3D"true" href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org=
</a><br>
<a moz-do-not-send=3D"true" href=3D"https://www.ietf.org/mailman/listinfo/r=
tcweb" target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><b=
r>
</blockquote>
</div>
<br>
<pre wrap=3D""><fieldset class=3D"mimeAttachmentHeader"></fieldset>
_______________________________________________
rtcweb mailing list
<a class=3D"moz-txt-link-abbreviated" href=3D"mailto:rtcweb@ietf.org">rtcwe=
b@ietf.org</a>
<a class=3D"moz-txt-link-freetext" href=3D"https://www.ietf.org/mailman/lis=
tinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
</blockquote>
</div>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
https://www.ietf.org/mailman/listinfo/rtcweb<br>
</blockquote>
</div>
<br>
</body>
</html>

--_000_91F0E66EA75E4369A0109712FF90258Dacmepacketcom_--

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On 03/28/2012 10:56 AM, Igor Faynberg wrote:

> I think there is a misunderstanding (I assume you did not attend the
> meeting today). It has been clarified that SRTP allows the NULL
> encryption algorithm, and that this option will be available.

The cipher algorithms are not the issue, and in many cases, neither is 
the CPU consumption required to do the encryption/decryption. I believe 
the larger concern that has been expressed is making WebRTC media 
dependent on DTLS-SRTP (the protocol) itself, since it is still 
relatively immature deployment-wise, and there are not many 
implementations available for developers to use to build products.

As an example, if the WG had chosen SDES-SRTP instead (even though it 
has its own set of faults), this concern would not be present, because 
that is already widely deployed, well understood and there are proven 
implementations available to choose from.

I'm not in any way lobbying against DTLS-SRTP being a MUST, though: I 
support that position.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kfleming@digium.com | SIP: kpfleming@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

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I agree with Igor and Magnus, it's better to have "Security by Default"
in place, and then leave to the open options for:
- Key Agreement
- Encryption algorithms

Interesting consideration on RTP encryption by default:

          Why RTP Does Not Mandate a Single Security Mechanism
	http://tools.ietf.org/html/draft-ietf-avt-srtp-not-mandatory-08

Also a must read, while designing IETF widely spread protocols:

                   Strong Security Requirements for
           Internet Engineering Task Force Standard Protocols
		http://tools.ietf.org/html/rfc3365


It's up the Telecommunication Regulators, Telecommunication Providers,
Telecommunication Equipment Manufacturer and Security Monitoring
equipment to handle the rules for Lawful interception, where an agency
have the rights to listen to a conversation.

But this is "a particular case" for which, even when there is
encryption-by-default, it's possible to listen/record the conversation.

Please remind that today there are multiple asymmetric cyberwar in
action and is an irresponsible action to even think not to include
security-by-default in any new protocol.

It's ok not to over-complicate the first version of the standard by
introducing dozen of different key management systems (SDES/ZRTP/MIKEY),
but imho it's not ok to leave in 2012 the standardizaton of a
non-encrypted-by-default protocol.

Again, imho it's an act of irresponsibility, what's defined in IETF
environment, have chance to get adopted by *billions* of people.

We must act responsibly.

-- 
Fabio Pietrosanti
Founder, CTO

Tel: +39 02 911930893 + ext: 907
Mobile: +39 340 1801049
E-mail: fabio.pietrosanti@privatewave.com
Skype: fpietrosanti
Linkedin: http://linkedin.com/in/secret

PrivateWave Italia S.p.A.
Via Gaetano Giardino 1 - 20123 Milano - Italy
www.privatewave.com

On 3/28/12 5:56 PM, Igor Faynberg wrote:
> Roman,
> 
> I think there is a misunderstanding (I assume you did not attend the
> meeting today).  It has been clarified that SRTP allows the NULL
> encryption algorithm, and that this option will be available.
> 
> Igor

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Content-Type: text/plain; charset=ISO-8859-1

Igor,

I could not attend the meeting today, but I am responding to the call of
consent on the list, that did not mention that or any other conditions on
SRTP encryption support. Does it also mean that the call can be established
using plain RTP, or is DTLS handshake, even in order to setup NULL
encruption, is still a requirement? If DTLS handshake is a requirement, I
still would object to it, since this will require an extra implementation
and interop step in anything communicating with WebRTC.

On the related note, where is SRTP profile with NULL encoding defined? I
have seen quite a few SRTP implementations but not a single one of them
supported NULL codec in any of its profiles.  Also, we should be very
careful not to support NULL encoding unless it is specifically enabled in
API. Even though I do want to support unencrypted media, I do not want
anything that allows to bid down from encrypted to unencrypted connection,
for instance during DTLS negotiation, unless specifically enabled.
_____________
Roman Shpount


On Wed, Mar 28, 2012 at 11:56 AM, Igor Faynberg <
igor.faynberg@alcatel-lucent.com> wrote:

> **
> Roman,
>
> I think there is a misunderstanding (I assume you did not attend the
> meeting today).  It has been clarified that SRTP allows the NULL encryption
> algorithm, and that this option will be available.
>
> Igor
>
>
> On 3/28/2012 11:49 AM, Roman Shpount wrote:
>
> As I have mentioned before on this list I am strongly against making SRTP
> protection for RTP a requirement. I think this is an unnecessary
> requirement that serves little real purpose except feeding into some
> marketing message that most of the WebRTC users would not care about.
> Unless use of identity is also a requirement, requiring SRTP will provide
> security only in a very narrow sense of the word. At the same time I do
> believe that extra standard requirements will stifle innovation and  will
> complicate new service or application creation.
>
> I have no objection to making DTLS-SRTP a required to implement protocol.
> _____________
> Roman Shpount
>
>
> On Wed, Mar 28, 2012 at 10:50 AM, Magnus Westerlund <
> magnus.westerlund@ericsson.com> wrote:
>
>> WG,
>>
>> In todays RTCWEB WG meeting there was discussion around media security
>> mechanism. In this meeting there was some clear consensus in the
>> meeting which we would like to confirm on the list.
>>
>> The first was that there was overwhelming consensus that all RTP
>> packets SHALL be protected by SRTP.
>>
>> Secondly that no one objected against making DTLS-SRTP a mandatory to
>> implement and the default keying mechanism. Additional mechanisms are
>> not precluded.
>>
>> WG participants may state their position regarding these consensus calls
>> until 12th of April when the chairs will declare the final consensus. If
>> you where present in the meeting room and comment on this, please
>> indicate that.
>>
>> Best Regards
>>
>> Magnus Westerlund
>> For the WG chairs
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>
>
> _______________________________________________
> rtcweb mailing listrtcweb@ietf.orghttps://www.ietf.org/mailman/listinfo/rtcweb
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--0023544477cd5c31a904bc4fe9c0
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Igor,<br><br>I could not attend the meeting today, but I am responding to t=
he call of consent on the list, that did not mention that or any other cond=
itions on SRTP encryption support. Does it also mean that the call can be e=
stablished using plain RTP, or is DTLS handshake, even in order to setup NU=
LL encruption, is still a requirement? If DTLS handshake is a requirement, =
I still would object to it, since this will require an extra implementation=
 and interop step in anything communicating with WebRTC.<br>
<br>On the related note, where is SRTP profile with NULL encoding defined? =
I have seen quite a few SRTP implementations but not a single one of them s=
upported NULL codec in any of its profiles.=A0 Also, we should be very care=
ful not to support NULL encoding unless it is specifically enabled in API. =
Even though I do want to support unencrypted media, I do not want anything =
that allows to bid down from encrypted to unencrypted connection, for insta=
nce during DTLS negotiation, unless specifically enabled.<br clear=3D"all">
_____________<br>Roman Shpount<br>
<br><br><div class=3D"gmail_quote">On Wed, Mar 28, 2012 at 11:56 AM, Igor F=
aynberg <span dir=3D"ltr">&lt;<a href=3D"mailto:igor.faynberg@alcatel-lucen=
t.com">igor.faynberg@alcatel-lucent.com</a>&gt;</span> wrote:<br><blockquot=
e class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc sol=
id;padding-left:1ex">
<u></u>

 =20
   =20
 =20
  <div text=3D"#000000" bgcolor=3D"#ffffff">
    Roman,<br>
    <br>
    I think there is a misunderstanding (I assume you did not attend the
    meeting today).=A0 It has been clarified that SRTP allows the NULL
    encryption algorithm, and that this option will be available.<br><font =
color=3D"#888888">
    <br>
    Igor</font><div><div></div><div class=3D"h5"><br>
    <br>
    On 3/28/2012 11:49 AM, Roman Shpount wrote:
    <blockquote type=3D"cite">As I have mentioned before on this list I am =
strongly
      against making SRTP protection for RTP a requirement. I think this
      is an unnecessary requirement that serves little real purpose
      except feeding into some marketing message that most of the WebRTC
      users would not care about. Unless use of identity is also a
      requirement, requiring SRTP will provide security only in a very
      narrow sense of the word. At the same time I do believe that extra
      standard requirements will stifle innovation and=A0 will complicate
      new service or application creation.<br>
      <br>
      I have no objection to making DTLS-SRTP a required to implement
      protocol.<br clear=3D"all">
      _____________<br>
      Roman Shpount<br>
      <br>
      <br>
      <div class=3D"gmail_quote">On Wed, Mar 28, 2012 at 10:50 AM, Magnus
        Westerlund <span dir=3D"ltr">&lt;<a href=3D"mailto:magnus.westerlun=
d@ericsson.com" target=3D"_blank">magnus.westerlund@ericsson.com</a>&gt;</s=
pan>
        wrote:<br>
        <blockquote class=3D"gmail_quote" style=3D"margin:0pt 0pt 0pt 0.8ex=
;border-left:1px solid rgb(204,204,204);padding-left:1ex">
          WG,<br>
          <br>
          In todays RTCWEB WG meeting there was discussion around media
          security<br>
          mechanism. In this meeting there was some clear consensus in
          the<br>
          meeting which we would like to confirm on the list.<br>
          <br>
          The first was that there was overwhelming consensus that all
          RTP<br>
          packets SHALL be protected by SRTP.<br>
          <br>
          Secondly that no one objected against making DTLS-SRTP a
          mandatory to<br>
          implement and the default keying mechanism. Additional
          mechanisms are<br>
          not precluded.<br>
          <br>
          WG participants may state their position regarding these
          consensus calls<br>
          until 12th of April when the chairs will declare the final
          consensus. If<br>
          you where present in the meeting room and comment on this,
          please<br>
          indicate that.<br>
          <br>
          Best Regards<br>
          <br>
          Magnus Westerlund<br>
          For the WG chairs<br>
          <br>
          _______________________________________________<br>
          rtcweb mailing list<br>
          <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.=
org</a><br>
          <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=
=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
        </blockquote>
      </div>
      <br>
      <pre><fieldset></fieldset>
_______________________________________________
rtcweb mailing list
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
    </blockquote>
  </div></div></div>

<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br>

--0023544477cd5c31a904bc4fe9c0--

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On 3/28/12 6:10 PM, Kevin P. Fleming wrote:
> As an example, if the WG had chosen SDES-SRTP instead (even though it
> has its own set of faults), this concern would not be present, because
> that is already widely deployed, well understood and there are proven
> implementations available to choose from.
> 
> I'm not in any way lobbying against DTLS-SRTP being a MUST, though: I
> support that position.

That's true, if you consider that even the US National Security Agency
approved for Classified/Top-Secret use the SDES-SRTP schema
http://www.nsa.gov/ia/programs/mobility_program/index.shtml .

The reason, underlined by NSA for adoption, is that is a mature, secure
and widely diffused internet security protocol.

-- 
Fabio Pietrosanti
Founder, CTO

Tel: +39 02 911930893 + ext: 907
Mobile: +39 340 1801049
E-mail: fabio.pietrosanti@privatewave.com
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On Mar 28, 2012, at 6:15 PM, Roman Shpount wrote:

> On the related note, where is SRTP profile with NULL encoding defined? I =
have seen quite a few SRTP implementations but not a single one of them sup=
ported NULL codec in any of its profiles. =20

The RFC for SRTP itself included NULL cipher mode - see RFC 3711.
Note that it is even mandatory to implement, but not mandatory to use - so =
we can specify in RTCWeb it MUST NOT be used, I believe.

-hadriel



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--20cf3071cffa5ab77104bc504c43
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On Wed, Mar 28, 2012 at 12:35 PM, Hadriel Kaplan <HKaplan@acmepacket.com>wrote:

>
> On Mar 28, 2012, at 6:15 PM, Roman Shpount wrote:
>
> > On the related note, where is SRTP profile with NULL encoding defined? I
> have seen quite a few SRTP implementations but not a single one of them
> supported NULL codec in any of its profiles.
>
> The RFC for SRTP itself included NULL cipher mode - see RFC 3711.
> Note that it is even mandatory to implement, but not mandatory to use - so
> we can specify in RTCWeb it MUST NOT be used, I believe.
>
>
What was always getting me confused is that RFC 4568 provided no way to
specify that NULL cipher mode should be used. All crypto suites there are
AES CM or F8 based. This might be irrelevant if we are going to use DTLS
for cipher negotiation.

_____________
Roman Shpount

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<br clear=3D"all"><div class=3D"gmail_quote">On Wed, Mar 28, 2012 at 12:35 =
PM, Hadriel Kaplan <span dir=3D"ltr">&lt;<a href=3D"mailto:HKaplan@acmepack=
et.com">HKaplan@acmepacket.com</a>&gt;</span> wrote:<br><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex">
<div class=3D"im"><br>
On Mar 28, 2012, at 6:15 PM, Roman Shpount wrote:<br>
<br>
&gt; On the related note, where is SRTP profile with NULL encoding defined?=
 I have seen quite a few SRTP implementations but not a single one of them =
supported NULL codec in any of its profiles.<br>
<br>
</div>The RFC for SRTP itself included NULL cipher mode - see RFC 3711.<br>
Note that it is even mandatory to implement, but not mandatory to use - so =
we can specify in RTCWeb it MUST NOT be used, I believe.<br>
<font color=3D"#888888"></font><br></blockquote></div><br>What was always g=
etting me confused is that RFC 4568 provided no way to specify that NULL ci=
pher mode should be used. All crypto suites there are AES CM or F8 based. T=
his might be irrelevant if we are going to use DTLS for cipher negotiation.=
<br>
<br>_____________<br>Roman Shpount<br>
<br><br>

--20cf3071cffa5ab77104bc504c43--

From magnus.westerlund@ericsson.com  Wed Mar 28 09:47:09 2012
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Date: Wed, 28 Mar 2012 18:47:05 +0200
From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: [rtcweb] Additional keying mechanisms for media security for discussion
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WG,

In addition to the consensus in todays meeting (see other email) there
was no consensus established about if there should be additional keying
mechanisms.

We did a consensus call if in addition to DTLS-SRTP:

a) Security Description SHALL be mandatory to also implement

b) Security Description SHALL NOT be mandatory to also implement

This consensus call was clearly tied. We will continue to discuss this
with the goal to make a decision at a later point.

We chairs also deferred the question if EKT is a mechanism that shall or
be recommended to be supported. So please discuss this also.

Regards

Magnus Westerlund
WG chair


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From: "Dan Wing" <dwing@cisco.com>
To: "'Roman Shpount'" <roman@telurix.com>, <igor.faynberg@alcatel-lucent.com>
References: <4F732531.2030208@ericsson.com>	<CAD5OKxs6NHha2egNSTumEaHYJ0bB6qu_nfshmBM6dntx2n49HQ@mail.gmail.com>	<4F733492.9040601@alcatel-lucent.com> <CAD5OKxv8PhhwmjaDHqet1NBmJ+8ndKBc7p7fjC2vogE1wXT=sg@mail.gmail.com>
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Date: Wed, 28 Mar 2012 18:54:49 +0200
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Subject: Re: [rtcweb] Consensus call regarding media security
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> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Roman Shpount
> Sent: Wednesday, March 28, 2012 6:15 PM
> To: igor.faynberg@alcatel-lucent.com
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Consensus call regarding media security
> 
> Igor,
> 
> I could not attend the meeting today, but I am responding to the call
> of consent on the list, that did not mention that or any other
> conditions on SRTP encryption support. Does it also mean that the call
> can be established using plain RTP, or is DTLS handshake, even in order
> to setup NULL encruption, is still a requirement? If DTLS handshake is
> a requirement, I still would object to it, since this will require an
> extra implementation and interop step in anything communicating with
> WebRTC.

The opposite it also true.

Specifically, if Security Descriptions is required for RTCWEB endpoints,
it requires extra implementation and interop for all RTCWEB endpoints.

> On the related note, where is SRTP profile with NULL encoding defined?
> I have seen quite a few SRTP implementations but not a single one of
> them supported NULL codec in any of its profiles.  Also, we should be
> very careful not to support NULL encoding unless it is specifically
> enabled in API. Even though I do want to support unencrypted media, I
> do not want anything that allows to bid down from encrypted to
> unencrypted connection, for instance during DTLS negotiation, unless
> specifically enabled.

During their (D)TLS handshake, the (D)TLS client and server exchange
a list of cipher suites they support, and use the strongest supported
by both the client and server.  To negotiate Null, the ClientHello needs
to include the Null cipher suite and the (D)TLS server needs to choose 
Null -- if its policy allows choosing Null.  A (D)TLS server will 
usually not allow negotiating Null, and a (D)TLS client will usually 
not include Null in its list of cipher suites.

-d

> _____________
> Roman Shpount
> 
> 
> 
> On Wed, Mar 28, 2012 at 11:56 AM, Igor Faynberg <igor.faynberg@alcatel-
> lucent.com> wrote:
> 
> 
> 
> 	Roman,
> 
> 	I think there is a misunderstanding (I assume you did not attend
> the meeting today).  It has been clarified that SRTP allows the NULL
> encryption algorithm, and that this option will be available.
> 
> 	Igor
> 
> 
> 	On 3/28/2012 11:49 AM, Roman Shpount wrote:
> 
> 		As I have mentioned before on this list I am strongly
> against making SRTP protection for RTP a requirement. I think this is
> an unnecessary requirement that serves little real purpose except
> feeding into some marketing message that most of the WebRTC users would
> not care about. Unless use of identity is also a requirement, requiring
> SRTP will provide security only in a very narrow sense of the word. At
> the same time I do believe that extra standard requirements will stifle
> innovation and  will complicate new service or application creation.
> 
> 		I have no objection to making DTLS-SRTP a required to
> implement protocol.
> 		_____________
> 		Roman Shpount
> 
> 
> 
> 		On Wed, Mar 28, 2012 at 10:50 AM, Magnus Westerlund
> <magnus.westerlund@ericsson.com> wrote:
> 
> 
> 			WG,
> 
> 			In todays RTCWEB WG meeting there was discussion
> around media security
> 			mechanism. In this meeting there was some clear
> consensus in the
> 			meeting which we would like to confirm on the list.
> 
> 			The first was that there was overwhelming consensus
> that all RTP
> 			packets SHALL be protected by SRTP.
> 
> 			Secondly that no one objected against making DTLS-
> SRTP a mandatory to
> 			implement and the default keying mechanism.
> Additional mechanisms are
> 			not precluded.
> 
> 			WG participants may state their position regarding
> these consensus calls
> 			until 12th of April when the chairs will declare the
> final consensus. If
> 			you where present in the meeting room and comment on
> this, please
> 			indicate that.
> 
> 			Best Regards
> 
> 			Magnus Westerlund
> 			For the WG chairs
> 
> 			_______________________________________________
> 			rtcweb mailing list
> 			rtcweb@ietf.org
> 			https://www.ietf.org/mailman/listinfo/rtcweb
> 
> 
> 
> 
> 		_______________________________________________
> 		rtcweb mailing list
> 		rtcweb@ietf.org
> 		https://www.ietf.org/mailman/listinfo/rtcweb
> 
> 
> 	_______________________________________________
> 	rtcweb mailing list
> 	rtcweb@ietf.org
> 	https://www.ietf.org/mailman/listinfo/rtcweb
> 
> 
> 



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> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Richard L. Barnes
> Sent: Wednesday, March 28, 2012 12:44 PM
> To: Harald Alvestrand
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] SRTP and "marketing"
> 
> >> I didn't make it to the mic at the meeting today, but I wanted to
> >> express one concern about the possibility of making RTCWEB SRTP-
> only.
> >>
> >> Hadriel mentioned the "marketing value" of having always-on
> >> encryption, this idea that only supporting SRTP will make RTCWEB
> look
> >> like something secure and trustworthy. I'm concerned that this might
> >> not be the case, and in fact that being SRTP-only might effectively
> be
> >> an over-promise, in light of the fact the absence of universal
> >> authentication.
> >>
> >> Hadriel noted that the competitors to this technology are Skype and
> >> Flash, and it's worth considering the security situation with these
> >> technologies, because they kind of bracket RTCWEB. With Skype
> >> (assuming they've designed it properly), there is actually a
> universal
> >> authentication, under a single authority. So you really do know that
> >> you're talking to whatever Skype ID you intend to, and nobody else.
> >> With Flash, well, does anyone expect it to be secure anyway?
> >>
> >> What I'm concerned about in the RTCWEB context is that without a
> >> universal authentication/identity infrastructure, we will end up
> >> *promising* a secure call, but not *delivering* it. I haven't done
> the
> >> analysis, but it does not seem implausible to me that FireSheep-like
> >> vulnerabilities are lurking here.
> > If there are, we need to close them before we ship the specs.
> > Given reasonable practices (such as using only HTTPS for loading the
> > pages and JS libraries), if we can't deliver security (against known
> > attacks), we shouldn't ship the spec.
> 
> I agree that we should lock things down to the extent we can, and I
> think we will be able to do pretty well, using things such as you note.
> 
> But at base, without authentication, you cannot prevent MitM if someone
> is in the right place in the network.  You can just constrain the set
> of
> places from which an MitM can be launched.
> 
> 
> >> So ISTM the "marketing" argument carries with it some serious risks
> as
> >> well as some small possible benefit.
> > Only if we don't deliver.
> 
> Except that "deliver" in this case entails a global
> authentication/identity infrastructure.  And it seems unlikely that
> we'll deliver that in the short run.

We do need a foundation upon which an authentication/identity 
infrastructure can be built.  We know we need one.

That foundation is DTLS-SRTP, and not Security Descriptions.

-d



> 
> --Richard
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Basil Mohamed Gohar
> Sent: Wednesday, March 28, 2012 4:55 PM
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] Consensus call regarding media security
> 
> On 03/28/2012 10:50 AM, Magnus Westerlund wrote:
> > WG,
> >
> > In todays RTCWEB WG meeting there was discussion around media
> security
> > mechanism. In this meeting there was some clear consensus in the
> > meeting which we would like to confirm on the list.
> >
> > The first was that there was overwhelming consensus that all RTP
> > packets SHALL be protected by SRTP.
> >
> > Secondly that no one objected against making DTLS-SRTP a mandatory to
> > implement and the default keying mechanism. Additional mechanisms are
> > not precluded.
> >
> > WG participants may state their position regarding these consensus
> calls
> > until 12th of April when the chairs will declare the final consensus.
> If
> > you where present in the meeting room and comment on this, please
> > indicate that.
> >
> > Best Regards
> >
> > Magnus Westerlund
> > For the WG chairs
> I already brought-up my concerns in the other thread, so I'll summarize
> the core point I was making here.  Would using SRTP *require* a central
> authority for establishing authenticity, or can authenticity be
> established via a point-to-point means (e.g., how it's traditionally
> done via SSH [i.e., upon first connection or via previous key
> exchange])?

A central authority is not required.

DTLS-SRTP itself doesn't use the information in the DTLS certificates (the
information that might be present is ignored).  

Of course, if you want identity, then an identity service needs to exist.
But it is possible to operate DTLS-SRTP without identity, which still
provides value beyond Security Descriptions.  For example, because you
mentioned ssh, an 'easy' way to do DTLS-SRTP is to place the remote
peer's certificate fingerprint into your local address book.  No central
authority is needed, and you could get an alert if/when the remote peer's 
certificate changes.  A similar technique for HTTP is described in 
draft-ietf-websec-key-pinning.  A similar technique for ZRTP is
http://tools.ietf.org/html/rfc6189#section-12.

> This is about degrees of trust that the user is will to place upon
> various methods, of course.  I am stating that the option should exist
> for authenticity of an end point to be established outside of a central
> authority (e.g., key exchange via other means).

I agree that is valuable.

-d



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Subject: Re: [rtcweb] Additional keying mechanisms for media security for discussion
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(b)


On 3/28/12 6:47 PM, Magnus Westerlund wrote:
> WG,
>
> In addition to the consensus in todays meeting (see other email) there
> was no consensus established about if there should be additional keying
> mechanisms.
>
> We did a consensus call if in addition to DTLS-SRTP:
>
> a) Security Description SHALL be mandatory to also implement
>
> b) Security Description SHALL NOT be mandatory to also implement
>
> This consensus call was clearly tied. We will continue to discuss this
> with the goal to make a decision at a later point.
>
> We chairs also deferred the question if EKT is a mechanism that shall or
> be recommended to be supported. So please discuss this also.
>
> Regards
>
> Magnus Westerlund
> WG chair
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From ibc@aliax.net  Wed Mar 28 10:50:14 2012
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Subject: [rtcweb] About DTLS-SRTP-EKT and key changes
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Hi all, this is probably a very easy question:

In slide 34 in http://tools.ietf.org/agenda/83/slides/slides-83-rtcweb-3.pd=
f,
there is a new EKT key from the WebRTC endpoint, and in slide 35 there
is a re-INVITE from the SDES endpoints with (I assume) a new SDES key.

When do such key change occur? I assume it could occur in case a SIP
PBX performs a "hidden" transfer to other local phone by keeping
untouched the SIP dialog in the other leg (in this case the new SIP
destination would provide a different SDES key so it must be sent to
the WebRTC endpoint in the form of a new EKT key, am I right?).

But, does such a key change could occur in simpler cases as for
example "putting on hold"? I hope the answer is NOT.

Thanks a lot for any clarification.

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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2012/3/28 Roman Shpount <roman@telurix.com>:
> As I have mentioned before on this list I am strongly against making SRTP
> protection for RTP a requirement. I think this is an unnecessary requirem=
ent
> that serves little real purpose except feeding into some marketing messag=
e
> that most of the WebRTC users would not care about. Unless use of identit=
y
> is also a requirement, requiring SRTP will provide security only in a ver=
y
> narrow sense of the word. At the same time I do believe that extra standa=
rd
> requirements will stifle innovation and=C2=A0 will complicate new service=
 or
> application creation.

SRTP (with SDES so without identity authentication) is still much
better than plain RTP, right? If I'm in an airport connected to an
open WiFi network, but I use HTTPS/WSS for signaling from my WebRTC
browser, then I can be sure that no one in the airport can intercept
my media streams (using SRTP-SDES).

Of course this does not solve the fact that there could be some MiM
attacker somewhere in the signaling path, but NOT in the airport! What
is sure is that if I was using plain RTP then everyone in the open
WiFi network could intercept my media streams.

IMHO it's really clear that SRTP (even with SDES) is MUCH better than
plain RTP. And so far I have not heard any advantage fof allowing
plain RTP other than "it allows interoperability with my 5 years ago
SIP device".

So +1 for the voted consensus.

Regards.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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On 03/28/2012 01:56 PM, IÃ±aki Baz Castillo wrote:
> 2012/3/28 Roman Shpount <roman@telurix.com>:
>> As I have mentioned before on this list I am strongly against making SRTP
>> protection for RTP a requirement. I think this is an unnecessary requirement
>> that serves little real purpose except feeding into some marketing message
>> that most of the WebRTC users would not care about. Unless use of identity
>> is also a requirement, requiring SRTP will provide security only in a very
>> narrow sense of the word. At the same time I do believe that extra standard
>> requirements will stifle innovation and  will complicate new service or
>> application creation.
> SRTP (with SDES so without identity authentication) is still much
> better than plain RTP, right? If I'm in an airport connected to an
> open WiFi network, but I use HTTPS/WSS for signaling from my WebRTC
> browser, then I can be sure that no one in the airport can intercept
> my media streams (using SRTP-SDES).
>
> Of course this does not solve the fact that there could be some MiM
> attacker somewhere in the signaling path, but NOT in the airport! What
> is sure is that if I was using plain RTP then everyone in the open
> WiFi network could intercept my media streams.
>
> IMHO it's really clear that SRTP (even with SDES) is MUCH better than
> plain RTP. And so far I have not heard any advantage fof allowing
> plain RTP other than "it allows interoperability with my 5 years ago
> SIP device".
>
> So +1 for the voted consensus.
>
> Regards.
This captures the sentiment of what I was trying to convey.  So, +1 to
this explanation.

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Wed, 28 Mar 2012 20:01:50 +0200
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Subject: Re: [rtcweb] Additional keying mechanisms for media security for discussion
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2012/3/28 Magnus Westerlund <magnus.westerlund@ericsson.com>:
> We did a consensus call if in addition to DTLS-SRTP:
>
> a) Security Description SHALL be mandatory to also implement
>
> b) Security Description SHALL NOT be mandatory to also implement
>
> This consensus call was clearly tied. We will continue to discuss this
> with the goal to make a decision at a later point.

I vote for (a). Rationale: SRTP-SDES is proven to work and it's widely
implemented. It is not the perfect solution but it's easy to implement
and much better than plain RTP.

NOTE: Of course I also vote for SRTP-DTLS to be mandatory (for which
AFAIK there is already consensus: it MUST be implemented).



> We chairs also deferred the question if EKT is a mechanism that shall or
> be recommended to be supported. So please discuss this also.

That's still less tested than DTLS-SRTP, so I vote for "SHALL NOT be
mandatory to also implement" (at least for now).


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Hadriel Kaplan wrote:
>
> Actually, I was confused about that - it seems to me it's still an open
> item whether to allow null cipher or not.

As a browser vendor, I could see adding an about:config option to enable 
the null cipher for debugging purposes, but would expect that for the 
vast majority of users it will never be allowed.

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2012/3/28 Timothy B. Terriberry <tterriberry@mozilla.com>:
>> Actually, I was confused about that - it seems to me it's still an open
>> item whether to allow null cipher or not.
>
>
> As a browser vendor, I could see adding an about:config option to enable =
the
> null cipher for debugging purposes, but would expect that for the vast
> majority of users it will never be allowed.


And if such an about:config option is automatically reverted upon
browser restart, then much better.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Subject: [rtcweb] DTLS-SRTP implementation diffusion: Why not SDES-SRTP?
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Hi all,

i read that 80% of Sipit participant support SDES-SRTP but 0% support
DTLS-SRTP  https://www.sipit.net/SIPit29_summary .

At SIPit there were 34 attendees from 17 companies visiting from 12
countries with 25 distinct VoIP implementations.

For SDES-SRTP it's reported that:
**
All implementations supported SDES for SRTP key exchange.
Good interop with many different combinations of endpoints.
**

I do not really see which is the rationale in making DTLS-SRTP mandatory
while plain SRTP with SDES key exchange is already so well know and used.

Anyone can provide some very strong and valuable point about using
DTLS-SRTP (considering it's weak diffusion and incompatibility risks)
rather than standardizing the transport/communication of SDES key
exchange over a Web medium (HTTPS/HTTPS/Javascript)?

-- 
Fabio Pietrosanti
Founder, CTO

Tel: +39 02 911930893 + ext: 907
Mobile: +39 340 1801049
E-mail: fabio.pietrosanti@privatewave.com
Skype: fpietrosanti
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PrivateWave Italia S.p.A.
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Subject: Re: [rtcweb] DTLS-SRTP implementation diffusion: Why not SDES-SRTP?
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2012/3/28 Fabio Pietrosanti (naif) <lists@infosecurity.ch>:
> Hi all,
>
> i read that 80% of Sipit participant support SDES-SRTP but 0% support
> DTLS-SRTP =C2=A0https://www.sipit.net/SIPit29_summary .
>
> At SIPit there were 34 attendees from 17 companies visiting from 12
> countries with 25 distinct VoIP implementations.

Right, but this is rtcweb, not SIP.



> I do not really see which is the rationale in making DTLS-SRTP mandatory
> while plain SRTP with SDES key exchange is already so well know and used.

That's a good reason to *also* allow (and mandate) SDES-SRTP support
in WebRTC clients, much better than the interoperability with SIP
(again: this is rtcweb, not SIP world).


> Anyone can provide some very strong and valuable point about using
> DTLS-SRTP (considering it's weak diffusion and incompatibility risks)?

Lot of recent threads about this topic in this maillist. But also
check a recent presentation (yesterday in IETF Pairs):

http://tools.ietf.org/agenda/83/slides/slides-83-rtcweb-3.pdf




--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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From: Justin Uberti <juberti@google.com>
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On Wed, Mar 28, 2012 at 2:30 PM, Timothy B. Terriberry <
tterriberry@mozilla.com> wrote:

> Hadriel Kaplan wrote:
>
>>
>> Actually, I was confused about that - it seems to me it's still an open
>> item whether to allow null cipher or not.
>>
>
> As a browser vendor, I could see adding an about:config option to enable
> the null cipher for debugging purposes, but would expect that for the vast
> majority of users it will never be allowed.


As another browser vendor, this is also my expectation.

--20cf3074d9e2eb734904bc535b03
Content-Type: text/html; charset=UTF-8
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<br><br><div class=3D"gmail_quote">On Wed, Mar 28, 2012 at 2:30 PM, Timothy=
 B. Terriberry <span dir=3D"ltr">&lt;<a href=3D"mailto:tterriberry@mozilla.=
com">tterriberry@mozilla.com</a>&gt;</span> wrote:<br><blockquote class=3D"=
gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-=
left:1ex">

<div class=3D"im">Hadriel Kaplan wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<br>
Actually, I was confused about that - it seems to me it&#39;s still an open=
<br>
item whether to allow null cipher or not.<br>
</blockquote>
<br></div>
As a browser vendor, I could see adding an about:config option to enable th=
e null cipher for debugging purposes, but would expect that for the vast ma=
jority of users it will never be allowed.</blockquote><div><br></div><div>

As another browser vendor, this is also my expectation.</div></div>

--20cf3074d9e2eb734904bc535b03--

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--e89a8ff24801fff33404bc53a2db
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On Wed, Mar 28, 2012 at 1:56 PM, I=F1aki Baz Castillo <ibc@aliax.net> wrote=
:

> 2012/3/28 Roman Shpount <roman@telurix.com>:
> > As I have mentioned before on this list I am strongly against making SR=
TP
> > protection for RTP a requirement. I think this is an unnecessary
> requirement
> > that serves little real purpose except feeding into some marketing
> message
> > that most of the WebRTC users would not care about. Unless use of
> identity
> > is also a requirement, requiring SRTP will provide security only in a
> very
> > narrow sense of the word. At the same time I do believe that extra
> standard
> > requirements will stifle innovation and  will complicate new service or
> > application creation.
>
> SRTP (with SDES so without identity authentication) is still much
> better than plain RTP, right? If I'm in an airport connected to an
> open WiFi network, but I use HTTPS/WSS for signaling from my WebRTC
> browser, then I can be sure that no one in the airport can intercept
> my media streams (using SRTP-SDES).
>
> Of course this does not solve the fact that there could be some MiM
> attacker somewhere in the signaling path, but NOT in the airport! What
> is sure is that if I was using plain RTP then everyone in the open
> WiFi network could intercept my media streams.
>
> IMHO it's really clear that SRTP (even with SDES) is MUCH better than
> plain RTP. And so far I have not heard any advantage fof allowing
> plain RTP other than "it allows interoperability with my 5 years ago
> SIP device".
>
>
My main objection is that if an application developer does not take care to
develop a secure application, nothing you can do on the standard side will
make it a secure application. If I am building a public voice blog that
records a voice message that anybody can listen to on the web site security
is not needed. My assumption is that a fair number of applications would be
like this. So for such applications this is an unnecessary feature.

WebRTC will not exist in vacuum. It will communicate with other systems. It
is not limited to old SIP devices. It can be something new like server side
speech recognition that is integrated with web application. For such
application extra code and interop requirements to support security will
represent a real and significant cost. Any requirement, unless absolutely
necessary will create barriers to entry for new applications. I would like
to avoid as many of those as possible.
_____________
Roman Shpount

--e89a8ff24801fff33404bc53a2db
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<br clear=3D"all"><div class=3D"gmail_quote">On Wed, Mar 28, 2012 at 1:56 P=
M, I=F1aki Baz Castillo <span dir=3D"ltr">&lt;<a href=3D"mailto:ibc@aliax.n=
et">ibc@aliax.net</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote=
" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
2012/3/28 Roman Shpount &lt;<a href=3D"mailto:roman@telurix.com">roman@telu=
rix.com</a>&gt;:<br>
<div class=3D"im">&gt; As I have mentioned before on this list I am strongl=
y against making SRTP<br>
&gt; protection for RTP a requirement. I think this is an unnecessary requi=
rement<br>
&gt; that serves little real purpose except feeding into some marketing mes=
sage<br>
&gt; that most of the WebRTC users would not care about. Unless use of iden=
tity<br>
&gt; is also a requirement, requiring SRTP will provide security only in a =
very<br>
&gt; narrow sense of the word. At the same time I do believe that extra sta=
ndard<br>
&gt; requirements will stifle innovation and=A0 will complicate new service=
 or<br>
&gt; application creation.<br>
<br>
</div>SRTP (with SDES so without identity authentication) is still much<br>
better than plain RTP, right? If I&#39;m in an airport connected to an<br>
open WiFi network, but I use HTTPS/WSS for signaling from my WebRTC<br>
browser, then I can be sure that no one in the airport can intercept<br>
my media streams (using SRTP-SDES).<br>
<br>
Of course this does not solve the fact that there could be some MiM<br>
attacker somewhere in the signaling path, but NOT in the airport! What<br>
is sure is that if I was using plain RTP then everyone in the open<br>
WiFi network could intercept my media streams.<br>
<br>
IMHO it&#39;s really clear that SRTP (even with SDES) is MUCH better than<b=
r>
plain RTP. And so far I have not heard any advantage fof allowing<br>
plain RTP other than &quot;it allows interoperability with my 5 years ago<b=
r>
SIP device&quot;.<br>
<br></blockquote></div><br>My main objection is that if an application deve=
loper does not take care to develop a secure application, nothing you can d=
o on the standard side will make it a secure application. If I am building =
a public voice blog that records a voice message that anybody can listen to=
 on the web site security is not needed. My assumption is that a fair numbe=
r of applications would be like this. So for such applications this is an u=
nnecessary feature.<br>
<br>WebRTC will not exist in vacuum. It will communicate with other systems=
. It is not limited to old SIP devices. It can be something new like server=
 side speech recognition that is integrated with web application. For such =
application extra code and interop requirements to support security will re=
present a real and significant cost. Any requirement, unless absolutely nec=
essary will create barriers to entry for new applications. I would like to =
avoid as many of those as possible.=A0 <br>
_____________<br>Roman Shpount<br>
<br>

--e89a8ff24801fff33404bc53a2db--

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On 03/28/2012 04:41 PM, Roman Shpount wrote:
> My main objection is that if an application developer does not take
> care to develop a secure application, nothing you can do on the
> standard side will make it a secure application. If I am building a
> public voice blog that records a voice message that anybody can listen
> to on the web site security is not needed. My assumption is that a
> fair number of applications would be like this. So for such
> applications this is an unnecessary feature.
>
> WebRTC will not exist in vacuum. It will communicate with other
> systems. It is not limited to old SIP devices. It can be something new
> like server side speech recognition that is integrated with web
> application. For such application extra code and interop requirements
> to support security will represent a real and significant cost. Any
> requirement, unless absolutely necessary will create barriers to entry
> for new applications. I would like to avoid as many of those as
> possible. 
> _____________
> Roman Shpount
Roman,

You make a lot of good points.  However, the inverse is true as well -
namely, that is if encryption is not mandated, most implementations will
likely leave it out, and adoption of secured communications would be
stifled even longer.  I cannot speak about the implementation
difficulties, but I can speak from the user side that most people will
remain ignorant of the underlying technology and not know enough to
demand nor enable a feature if it is optional to implement and/or use.

As WebRTC is a new standard, requiring encryption will ensure that, at
least going forward, the important concept of encryption is widely
adopted correctly from the beginning.  Tacking it on later, no matter
how much it is emphasized, will be difficult or impossible.

The scope of WebRTC is broad enough to consider that we need to think
about what's best going forward with regards to its implementation. 
Security by default is one of the best practices in general, the support
from the browser community and others that are behind it will definitely
ensure that adoption is widespread enough to make it easy enough to
integrate into existing systems, as free software solutions will become
available shortly after the standard emerges.

From ibc@aliax.net  Wed Mar 28 14:12:18 2012
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Wed, 28 Mar 2012 23:11:56 +0200
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2012/3/28 Roman Shpount <roman@telurix.com>:
> My main objection is that if an application developer does not take care =
to
> develop a secure application, nothing you can do on the standard side wil=
l
> make it a secure application. If I am building a public voice blog that
> records a voice message that anybody can listen to on the web site securi=
ty
> is not needed. My assumption is that a fair number of applications would =
be
> like this. So for such applications this is an unnecessary feature.

In all those scenarios you mention, using security (SRTP) is not *bad*, is =
it?
The fact that in certain scenarios it could be not needed, it does not
mean that "no security" is better than "security".


> WebRTC will not exist in vacuum. It will communicate with other systems. =
It
> is not limited to old SIP devices. It can be something new like server si=
de
> speech recognition that is integrated with web application. For such
> application extra code and interop requirements to support security will
> represent a real and significant cost. Any requirement, unless absolutely
> necessary will create barriers to entry for new applications. I would lik=
e
> to avoid as many of those as possible.

RFC 3711 was created in 2004, 8 years ago!

Which "new" application you mean if it's not capable of implementing a
simple specification from 2004? how "new" is it? is it really new? or
is it a "SIP voicemail server" made in 2002?


Sorry but I still see *no* argument at all in favour of allowing plain
RTP in WebRTC. And AFAIK there is already consensus about it: plain
RTP is not allowed.


Regards.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Dan Wing <dwing@cisco.com>
Thread-Topic: [rtcweb] SRTP and "marketing"
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On Mar 28, 2012, at 6:59 PM, Dan Wing wrote:

> We do need a foundation upon which an authentication/identity=20
> infrastructure can be built.  We know we need one.
> That foundation is DTLS-SRTP, and not Security Descriptions.

Now you're starting to sound like a marketing guy.  ;)
What's next: "we'll build more synergy and have a unified platform with DTL=
S-SRTP"?

But more seriously, I don't understand this "foundation" argument.  We're g=
oing to have DTLS-SRTP.  No one's suggesting we don't have DTLS-SRTP.  All =
Browsers MUST implement DTLS-SRTP.  We'll have it for Browser-to-Browser, a=
nd for Browser-to-Gateway if the Gateway supports it.  We'll have the found=
ation.

Requiring it for Gateways would make sense if it offered some real advantag=
e, or didn't have any disadvantages.  There don't appear to be real advanta=
ges, while we know of disadvantages.  And gateways have no real means of of=
fering an end-to-end identity.  Why would you want to build a foundation on=
 air?

-hadriel


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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: "Richard L. Barnes" <rbarnes@bbn.com>
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On Mar 28, 2012, at 11:15 AM, Richard L. Barnes wrote:

> Hadriel noted that the competitors to this technology are Skype and Flash=
, and it's worth considering the security situation with these technologies=
, because they kind of bracket RTCWEB.  With Skype (assuming they've design=
ed it properly), there is actually a universal authentication, under a sing=
le authority.  So you really do know that you're talking to whatever Skype =
ID you intend to, and nobody else. With Flash, well, does anyone expect it =
to be secure anyway?

As far as I know, you don't actually "know" that with Skype.  You assume it=
, because you trust Skype.  They could forge whatever identity they wanted =
to, and they can insert a recording middlebox if they wanted to, afaict.  N=
o one is concerned about that.  They also have skype-in/skype-out to/from t=
he PSTN, and clearly in those cases they can assert whatever identity they =
want, and record it all.  Again no one is concerned.  Have you ever wondere=
d why no one freaks out?


> What I'm concerned about in the RTCWEB context is that without a universa=
l authentication/identity infrastructure, we will end up *promising* a secu=
re call, but not *delivering* it.  I haven't done the analysis, but it does=
 not seem implausible to me that FireSheep-like vulnerabilities are lurking=
 here.
> So ISTM the "marketing" argument carries with it some serious risks as we=
ll as some small possible benefit.


It was my understanding firesheep only works when the connection is HTTP, b=
ecause it sniffs the packets.  That's a real issue for RTP, not for SRTP (i=
n either SDES or DTLS cases).

Of course neither I nor anyone else can really foretell what the trade pres=
s will say - but I remember what they said about SIP back when a couple ARP=
-spoofing "attack" tools demonstrated how to intercept RTP and play it, sin=
ce I was in marketing at the time.  At the time, the articles were only adv=
ocating people should use "SRTP" instead.  They didn't care at all what the=
 key-exchange protocol was.

-hadriel


From Jim.Barnett@genesyslab.com  Wed Mar 28 15:31:37 2012
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Another point is that username/password authentication schemes aren't
really that secure - it's pretty easy to steal a username and password,
after all.  So there's not  really a binary
authenticated/unauthenticated switch.  It's more a matter of degree. =20

- Jim

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
Of Hadriel Kaplan
Sent: Wednesday, March 28, 2012 5:59 PM
To: Richard L. Barnes
Cc: <rtcweb@ietf.org>
Subject: Re: [rtcweb] SRTP and "marketing"


On Mar 28, 2012, at 11:15 AM, Richard L. Barnes wrote:

> Hadriel noted that the competitors to this technology are Skype and
Flash, and it's worth considering the security situation with these
technologies, because they kind of bracket RTCWEB.  With Skype (assuming
they've designed it properly), there is actually a universal
authentication, under a single authority.  So you really do know that
you're talking to whatever Skype ID you intend to, and nobody else. With
Flash, well, does anyone expect it to be secure anyway?

As far as I know, you don't actually "know" that with Skype.  You assume
it, because you trust Skype.  They could forge whatever identity they
wanted to, and they can insert a recording middlebox if they wanted to,
afaict.  No one is concerned about that.  They also have
skype-in/skype-out to/from the PSTN, and clearly in those cases they can
assert whatever identity they want, and record it all.  Again no one is
concerned.  Have you ever wondered why no one freaks out?


> What I'm concerned about in the RTCWEB context is that without a
universal authentication/identity infrastructure, we will end up
*promising* a secure call, but not *delivering* it.  I haven't done the
analysis, but it does not seem implausible to me that FireSheep-like
vulnerabilities are lurking here.
> So ISTM the "marketing" argument carries with it some serious risks as
well as some small possible benefit.


It was my understanding firesheep only works when the connection is
HTTP, because it sniffs the packets.  That's a real issue for RTP, not
for SRTP (in either SDES or DTLS cases).

Of course neither I nor anyone else can really foretell what the trade
press will say - but I remember what they said about SIP back when a
couple ARP-spoofing "attack" tools demonstrated how to intercept RTP and
play it, since I was in marketing at the time.  At the time, the
articles were only advocating people should use "SRTP" instead.  They
didn't care at all what the key-exchange protocol was.

-hadriel

_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

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Subject: Re: [rtcweb] SRTP and "marketing"
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On 3/28/2012 5:58 PM, Hadriel Kaplan wrote:
> On Mar 28, 2012, at 11:15 AM, Richard L. Barnes wrote:
>
>> Hadriel noted that the competitors to this technology are Skype and Flash, and it's worth considering the security situation with these technologies, because they kind of bracket RTCWEB.  With Skype (assuming they've designed it properly), there is actually a universal authentication, under a single authority.  So you really do know that you're talking to whatever Skype ID you intend to, and nobody else. With Flash, well, does anyone expect it to be secure anyway?
> As far as I know, you don't actually "know" that with Skype.  You assume it, because you trust Skype.  They could forge whatever identity they wanted to, and they can insert a recording middlebox if they wanted to, afaict.

And they do.  Very, very quietly I assume.  (I.e. they're certainly 
subject to CALEA in the US for on-net calls, and I assume have a deal 
with the FCC/etc to cover it.)  There have been a few public hints about 
this and similar deals with other countries.

>    No one is concerned about that.

Define "no one" :-)

> They also have skype-in/skype-out to/from the PSTN, and clearly in those cases they can assert whatever identity they want, and record it all.  Again no one is concerned.  Have you ever wondered why no one freaks out?

People don't freak out because a) few realize it, b) most who do realize 
it's government mandated on them, and fighting it would likely be a huge 
losing proposition for them ($$$$).  If I were a democracy activist in 
Iran or China other similar countries I would not assume Skype was safe 
(though in many of those countries it probably is, but probably not 
China).  Those are just my assumptions; small on-net companies and 
services can fly under the radar or avoid touching things that force 
them to comply (I did that at Worldgate; no PSTN access == no CALEA).

That doesn't mean I don't want to implement a truly secure 
communications ability.  Note that by "secure" here I mean "I can detect 
if the service MITMd me" -- without reading cert phrases and preferably 
without having to install client certs and somehow build a working PKI.

One idea we've had is a non-centralized WebRTC service built on 
distributed hash tables and a partial-mesh set of connections used to 
pass signaling on WebRTC (or other) data channels to set up calls, with 
some type of Id system to validate who you get connected to (since you 
can't really trust any of the nodes fully).  If an intermediary node can 
downgrade you to SDES and sniff the keys, there is no security.

An alternative would be to allow a hybrid: SDES at the start of a call 
and rekey over to DTLS-SRTP after the call is connected.  If something 
blocks the DTLS connection it would be treated as a 
gateway-to-legacy/PSTN call (i.e. show no security, though it's secure 
against non-in-path attackers); if the DTLS connection is allowed, we 
can use it to rekey to DTLS-SRTP and to verify identity.

This minimizes the risk of bid-down - the primary additional risk is 
that without initial SDES, to tap the signal you'd need to use a 
WebRTC-enabled B2BUA to accept the DTLS-SRTP connection and siphon it 
off.  Also, a "good" gateway in a pure DTLS-SRTP world might have a 
identity established ("gateway@L3.com"); with SDES it wouldn't by 
default (but it could if it wanted to by rekeying via DTLS also).  You 
also have a start-of-call risk where someone could see & hear until 
rekeying, and perhaps for a short while by blocking actual rekeying by 
dropping/corrupting DTLS packets, until the client either gives up or 
warns of no security.

I'm still strongly in favor of pure DTLS-SRTP - as Harald and others 
say, less moving parts and less things that need to be secured.  But 
this might be a useful compromise if we *need* to support SDES in some 
cases.

-- 
Randell Jesup
randell-ietf@jesup.org


From tterriberry@mozilla.com  Wed Mar 28 19:30:04 2012
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Subject: Re: [rtcweb] SRTP and "marketing"
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Hadriel Kaplan wrote:
> As far as I know, you don't actually "know" that with Skype.  You assume it,
> because you trust Skype.  They could forge whatever identity they wanted to,
> and they can insert a recording middlebox if they wanted to, afaict.
> No one is concerned about that.  They also have skype-in/skype-out to/from

This is, of course, not true. See, for example, 
http://www.technologyreview.com/web/38379/

The relevant bit, for those who don't want to read the whole article: 
'"We met using Mumble [which is open-source, uses digital certificate 
authentication, and is regarded by Takriz as more secure than Skype].' 
(brackets in the original).

Which is not to say that Tunisian rebels never used Skype (the article 
mentions several instances where they did), but it is clearly inaccurate 
to say no one was concerned about it.

From ietf@meetecho.com  Wed Mar 28 22:41:13 2012
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All,

the information below are also valid for today's meeting.

Cheers,
the Meetecho team


Il 27/03/2012 16:43, Meetecho IETF support ha scritto:
> Hi all,
>
> a virtual room has been reserved on the Meetecho system for Wednesday's
> RTCWEB WG meeting session.
>
> Access to the on-line session (including audio and video streams) will
> be available at:
> http://www.meetecho.com/ietf83/rtcweb
>
> The Meetecho session automatically logs you into the standard IETF
> jabber room. So, from there, you can have an integrated experience
> involving all media and allowing you to interact with the room.
> Remote participants might also send their own voice to the room, if they
> want to, by either calling a landline phone number, or using our
> embedded VoIP applet (in this last case they are *strongly* advised to
> use a headset).
>
> A tutorial of interactivity features of the tool can be found at:
> http://www.meetecho.com/ietf83/tutorials
>
> Cheers,
> the Meetecho team
>

From roman@telurix.com  Wed Mar 28 22:43:00 2012
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--047d7b2edf0331489504bc5b327c
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On Wed, Mar 28, 2012 at 10:30 PM, Timothy B. Terriberry <
tterriberry@mozilla.com> wrote:

> Hadriel Kaplan wrote:
>
>> As far as I know, you don't actually "know" that with Skype.  You assume
>> it,
>> because you trust Skype.  They could forge whatever identity they wanted
>> to,
>> and they can insert a recording middlebox if they wanted to, afaict.
>> No one is concerned about that.  They also have skype-in/skype-out to/from
>>
>
> This is, of course, not true. See, for example,
> http://www.technologyreview.**com/web/38379/<http://www.technologyreview.com/web/38379/>
>
> The relevant bit, for those who don't want to read the whole article: '"We
> met using Mumble [which is open-source, uses digital certificate
> authentication, and is regarded by Takriz as more secure than Skype].'
> (brackets in the original).
>
> Which is not to say that Tunisian rebels never used Skype (the article
> mentions several instances where they did), but it is clearly inaccurate to
> say no one was concerned about it.
>
>
And then there is an argument that Skype is a huge security risk exactly
due to the fact that it is using encrypted communications. There were hints
of using Skype communication protocol and Skype client to access files
remotely on the user computer or ability to open ports into a local network
using a Skype client running there (using Skype as a malicious VPN client).
Entire protocol and client are proprietary and were never validated by an
independent party as being secure in a sense of preventing unauthorized
remote access. This is one of the reasons why Skype is not allowed in a lot
of corporate or government networks. Even though this should not apply to
WebRTC since it is going to be an open standard with an intensive peer
review, it would still apply to the web browsers. If web browser software
is compromised due to some sort of attack or malicious plugin (or one of
the browser vendors snooping), and since browser applications will start
establishing unsolicited encrypted connections, there is no easy way to
determine that the browser is not doing something undesirable.  I actually
believe that if sufficient monitoring constraints are not build into a
browser (not to record but at least to monitor who the browser is
exchanging data with and using what protocols), WebRTC would be simply
disabled in most enterprises as a security risk.
_____________
Roman Shpount

--047d7b2edf0331489504bc5b327c
Content-Type: text/html; charset=ISO-8859-1
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<br clear=3D"all"><div class=3D"gmail_quote">On Wed, Mar 28, 2012 at 10:30 =
PM, Timothy B. Terriberry <span dir=3D"ltr">&lt;<a href=3D"mailto:tterriber=
ry@mozilla.com">tterriberry@mozilla.com</a>&gt;</span> wrote:<br><blockquot=
e class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc sol=
id;padding-left:1ex">
<div class=3D"im">Hadriel Kaplan wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
As far as I know, you don&#39;t actually &quot;know&quot; that with Skype. =
=A0You assume it,<br>
because you trust Skype. =A0They could forge whatever identity they wanted =
to,<br>
and they can insert a recording middlebox if they wanted to, afaict.<br>
No one is concerned about that. =A0They also have skype-in/skype-out to/fro=
m<br>
</blockquote>
<br></div>
This is, of course, not true. See, for example, <a href=3D"http://www.techn=
ologyreview.com/web/38379/" target=3D"_blank">http://www.technologyreview.<=
u></u>com/web/38379/</a><br>
<br>
The relevant bit, for those who don&#39;t want to read the whole article: &=
#39;&quot;We met using Mumble [which is open-source, uses digital certifica=
te authentication, and is regarded by Takriz as more secure than Skype].&#3=
9; (brackets in the original).<br>

<br>
Which is not to say that Tunisian rebels never used Skype (the article ment=
ions several instances where they did), but it is clearly inaccurate to say=
 no one was concerned about it.<div><div></div><div class=3D"h5"><br></div>
</div></blockquote><div><br>And then there is an argument that Skype is a h=
uge security risk exactly due to the fact that it is using encrypted commun=
ications. There were hints of using Skype communication protocol and Skype =
client to access files remotely on the user computer or ability to open por=
ts into a local network using a Skype client running there (using Skype as =
a malicious VPN client). Entire protocol and client are proprietary and wer=
e never validated by an independent party as being secure in a sense of pre=
venting unauthorized remote access. This is one of the reasons why Skype is=
 not allowed in a lot of corporate or government networks. Even though this=
 should not apply to WebRTC since it is going to be an open standard with a=
n intensive peer review, it would still apply to the web browsers. If web b=
rowser software is compromised due to some sort of attack or malicious plug=
in (or one of the browser vendors snooping), and since browser applications=
 will start establishing unsolicited encrypted connections, there is no eas=
y way to determine that the browser is not doing something undesirable.=A0 =
I actually believe that if sufficient monitoring constraints are not build =
into a browser (not to record but at least to monitor who the browser is ex=
changing data with and using what protocols), WebRTC would be simply disabl=
ed in most enterprises as a security risk.<br>
</div></div>_____________<br>Roman Shpount<br>
<br>

--047d7b2edf0331489504bc5b327c--

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Subject: Re: [rtcweb] Consensus call regarding media security
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On Wed, Mar 28, 2012 at 5:08 PM, Basil Mohamed Gohar <
abu_hurayrah@hidayahonline.org> wrote:

> The scope of WebRTC is broad enough to consider that we need to think
> about what's best going forward with regards to its implementation.
> Security by default is one of the best practices in general, the support
> from the browser community and others that are behind it will definitely
> ensure that adoption is widespread enough to make it easy enough to
> integrate into existing systems, as free software solutions will become
> available shortly after the standard emerges.
>
>
I would actually challenge the assumption that free software will emerge.
SRTP with SDES was released 8 years ago. The only open source library that
supported it until recently was libsrtp, which is a broken unusable peace
of code spaghetti (GPF crashes, two different replay DBs which do the same
thing, random number generator that stops working after generating few
thousand keys) that supports 2/3 of the protocol (no F8 support). Majority
of open source (and a large number of closed source) clients use this
library, but do not see any of those issues since very few real SRTP calls
are actually placed. DTLS-SRTP is two years old. Support for it was just
added to GNU TLS (unusable by many due to GNU license) and to OpenSSL
(using above mentioned libsrtp). There is little or no experience with
DTLS-SRTP interop. All of these things are not trivial.

My point is, if we require support for encryption implementation in the
standard and if 4 or 5 browser vendors will cooperate, we will get vast
majority of WebRTC clients security enabled. If everything surrounding
WebRTC will not use encryption, it is not a big loss since, in most cases,
it is not doing now anyway.
_____________
Roman Shpount

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<br clear=3D"all"><div class=3D"gmail_quote">On Wed, Mar 28, 2012 at 5:08 P=
M, Basil Mohamed Gohar <span dir=3D"ltr">&lt;<a href=3D"mailto:abu_hurayrah=
@hidayahonline.org">abu_hurayrah@hidayahonline.org</a>&gt;</span> wrote:<br=
><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1=
px #ccc solid;padding-left:1ex">
The scope of WebRTC is broad enough to consider that we need to think<br>
about what&#39;s best going forward with regards to its implementation.<br>
Security by default is one of the best practices in general, the support<br=
>
from the browser community and others that are behind it will definitely<br=
>
ensure that adoption is widespread enough to make it easy enough to<br>
integrate into existing systems, as free software solutions will become<br>
available shortly after the standard emerges.<br>
<div><div></div><br></div></blockquote></div><br>I would actually challenge=
 the assumption that free software will emerge. SRTP with SDES was released=
 8 years ago. The only open source library that supported it until recently=
 was libsrtp, which is a broken unusable peace of code spaghetti (GPF crash=
es, two different replay DBs which do the same thing, random number generat=
or that stops working after generating few thousand keys) that supports 2/3=
 of the protocol (no F8 support). Majority of open source (and a large numb=
er of closed source) clients use this library, but do not see any of those =
issues since very few real SRTP calls are actually placed. DTLS-SRTP is two=
 years old. Support for it was just added to GNU TLS (unusable by many due =
to GNU license) and to OpenSSL (using above mentioned libsrtp). There is li=
ttle or no experience with DTLS-SRTP interop. All of these things are not t=
rivial.<br>
<br>My point is, if we require support for encryption implementation in the=
 standard and if 4 or 5 browser vendors will cooperate, we will get vast ma=
jority of WebRTC clients security enabled. If everything surrounding WebRTC=
 will not use encryption, it is not a big loss since, in most cases, it is =
not doing now anyway.<br>
_____________<br>Roman Shpount<br>
<br>

--047d7b10d17b3cf37504bc5b6222--

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I agree with proposition #1 (SRTP) unconditionally.=20

With respect to proposition #2 (DTLS-SRTP)=2C perhaps the words "with detai=
ls to be worked out" should have been added. =20

I believe that the consensus achieved was only on a general direction=2C no=
t an endorsement of particular proposals.=20

Personally=2C I would like to have more specifics about the required featur=
es of DTLS-SRTP in the RTCWEB context.



> Date: Wed=2C 28 Mar 2012 16:50:25 +0200
> From: magnus.westerlund@ericsson.com
> To: rtcweb@ietf.org
> Subject: [rtcweb] Consensus call regarding media security
>=20
> WG=2C
>=20
> In todays RTCWEB WG meeting there was discussion around media security
> mechanism. In this meeting there was some clear consensus in the
> meeting which we would like to confirm on the list.
>=20
> The first was that there was overwhelming consensus that all RTP
> packets SHALL be protected by SRTP.
>=20
> Secondly that no one objected against making DTLS-SRTP a mandatory to
> implement and the default keying mechanism. Additional mechanisms are
> not precluded.
>=20
> WG participants may state their position regarding these consensus calls
> until 12th of April when the chairs will declare the final consensus. If
> you where present in the meeting room and comment on this=2C please
> indicate that.
>=20
> Best Regards
>=20
> Magnus Westerlund
> For the WG chairs
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
 		 	   		  =

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<body class=3D'hmmessage'><div dir=3D'ltr'>
I agree with proposition #1 (SRTP) unconditionally. <br><br>With respect to=
 proposition #2 (DTLS-SRTP)=2C perhaps the words "with details to be worked=
 out" should have been added.&nbsp=3B <br><br>I believe that the consensus =
achieved was only on a general direction=2C not an endorsement of particula=
r proposals. <br><br>Personally=2C I would like to have more specifics abou=
t the required features of DTLS-SRTP in the RTCWEB context.<br><br><br><br>=
<div><div id=3D"SkyDrivePlaceholder"></div>&gt=3B Date: Wed=2C 28 Mar 2012 =
16:50:25 +0200<br>&gt=3B From: magnus.westerlund@ericsson.com<br>&gt=3B To:=
 rtcweb@ietf.org<br>&gt=3B Subject: [rtcweb] Consensus call regarding media=
 security<br>&gt=3B <br>&gt=3B WG=2C<br>&gt=3B <br>&gt=3B In todays RTCWEB =
WG meeting there was discussion around media security<br>&gt=3B mechanism. =
In this meeting there was some clear consensus in the<br>&gt=3B meeting whi=
ch we would like to confirm on the list.<br>&gt=3B <br>&gt=3B The first was=
 that there was overwhelming consensus that all RTP<br>&gt=3B packets SHALL=
 be protected by SRTP.<br>&gt=3B <br>&gt=3B Secondly that no one objected a=
gainst making DTLS-SRTP a mandatory to<br>&gt=3B implement and the default =
keying mechanism. Additional mechanisms are<br>&gt=3B not precluded.<br>&gt=
=3B <br>&gt=3B WG participants may state their position regarding these con=
sensus calls<br>&gt=3B until 12th of April when the chairs will declare the=
 final consensus. If<br>&gt=3B you where present in the meeting room and co=
mment on this=2C please<br>&gt=3B indicate that.<br>&gt=3B <br>&gt=3B Best =
Regards<br>&gt=3B <br>&gt=3B Magnus Westerlund<br>&gt=3B For the WG chairs<=
br>&gt=3B <br>&gt=3B _______________________________________________<br>&gt=
=3B rtcweb mailing list<br>&gt=3B rtcweb@ietf.org<br>&gt=3B https://www.iet=
f.org/mailman/listinfo/rtcweb<br></div> 		 	   		  </div></body>
</html>=

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On Wed, Mar 28, 2012 at 5:11 PM, I=F1aki Baz Castillo <ibc@aliax.net> wrote=
:

> In all those scenarios you mention, using security (SRTP) is not *bad*, i=
s
> it?
> The fact that in certain scenarios it could be not needed, it does not
> mean that "no security" is better than "security".
>
> I actually think that you are confusing encryption with security. No
encryption often means better security. Bot nets use encryption to
communicate, it does not mean they enable user security. In a lot of cases
ability for the user to check what exactly they are communicating and to
whom is security. As a security conscious person I do not want encrypted
communication from my computer or my network unless I know exactly who it
is going to. If I do not have to I would prefer my communications to be
unencrypted.


> RFC 3711 was created in 2004, 8 years ago!
>
> Which "new" application you mean if it's not capable of implementing a
> simple specification from 2004? how "new" is it? is it really new? or
> is it a "SIP voicemail server" made in 2002?
>
> Well let me list them: FreeSwitch, Asterisk, linphone, etc. All of those
use libsrtp. For all of them SRTP is broken. Not a single one of them
bothered to check or only recently detected problems (
https://issues.asterisk.org/jira/browse/ASTERISK-16665).


>
> Sorry but I still see *no* argument at all in favour of allowing plain
> RTP in WebRTC. And AFAIK there is already consensus about it: plain
> RTP is not allowed.
>
> I do not see an argument why RTP should not be allowed. I think you are
arguing we should require a feature (SRTP) for the sake of requiring a
feature.
_____________
Roman Shpount

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<br clear=3D"all"><div class=3D"gmail_quote">On Wed, Mar 28, 2012 at 5:11 P=
M, I=F1aki Baz Castillo <span dir=3D"ltr">&lt;<a href=3D"mailto:ibc@aliax.n=
et">ibc@aliax.net</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote=
" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
In all those scenarios you mention, using security (SRTP) is not *bad*, is =
it?<br>
The fact that in certain scenarios it could be not needed, it does not<br>
mean that &quot;no security&quot; is better than &quot;security&quot;.<br>
<div class=3D"im"><br></div></blockquote><div>I actually think that you are=
 confusing encryption with security. No encryption often means better secur=
ity. Bot nets use encryption to communicate, it does not mean they enable u=
ser security. In a lot of cases ability for the user to check what exactly =
they are communicating and to whom is security. As a security conscious per=
son I do not want encrypted communication from my computer or my network un=
less I know exactly who it is going to. If I do not have to I would prefer =
my communications to be unencrypted. <br>
=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0pt 0pt 0pt 0.8e=
x;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div class=3D"im=
">

</div>RFC 3711 was created in 2004, 8 years ago!<br>
<br>
Which &quot;new&quot; application you mean if it&#39;s not capable of imple=
menting a<br>
simple specification from 2004? how &quot;new&quot; is it? is it really new=
? or<br>
is it a &quot;SIP voicemail server&quot; made in 2002?<br>
<br></blockquote><div>Well let me list them: FreeSwitch, Asterisk, linphone=
, etc. All of those use libsrtp. For all of them SRTP is broken. Not a sing=
le one of them bothered to check or only recently detected problems (<a hre=
f=3D"https://issues.asterisk.org/jira/browse/ASTERISK-16665">https://issues=
.asterisk.org/jira/browse/ASTERISK-16665</a>).<br>
=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0pt 0pt 0pt 0.8e=
x;border-left:1px solid rgb(204,204,204);padding-left:1ex">
<br>
Sorry but I still see *no* argument at all in favour of allowing plain<br>
RTP in WebRTC. And AFAIK there is already consensus about it: plain<br>
RTP is not allowed.<br>
<div><div></div><div class=3D"h5"><br></div></div></blockquote><div>I do no=
t see an argument why RTP should not be allowed. I think you are arguing we=
 should require a feature (SRTP) for the sake of requiring a feature. <br>
</div></div>_____________<br>Roman Shpount<br>

--e89a8ff24801b821cb04bc5b8056--

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Subject: Re: [rtcweb] Consensus call regarding media security
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--047d7b10cde36a3dc104bc5b86e1
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On Wed, Mar 28, 2012 at 5:11 PM, I=F1aki Baz Castillo <ibc@aliax.net> wrote=
:

> And AFAIK there is already consensus about it: plain
> RTP is not allowed.
>
>
AFAIK there is a consensus call on this exact issue right now. Nothing was
reached yet.
_____________
Roman Shpount

--047d7b10cde36a3dc104bc5b86e1
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<br clear=3D"all"><div class=3D"gmail_quote">On Wed, Mar 28, 2012 at 5:11 P=
M, I=F1aki Baz Castillo <span dir=3D"ltr">&lt;<a href=3D"mailto:ibc@aliax.n=
et">ibc@aliax.net</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote=
" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
And AFAIK there is already consensus about it: plain<br>
RTP is not allowed.<br>
<div><div></div><br></div></blockquote><div><br>AFAIK there is a consensus =
call on this exact issue right now. Nothing was reached yet.<br></div></div=
>_____________<br>Roman Shpount<br>

--047d7b10cde36a3dc104bc5b86e1--

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On Fri, Mar 23, 2012 at 11:19 PM, Hadriel Kaplan <HKaplan@acmepacket.com>wrote:

>  I'm not trying to be pedantic or difficult here, or trying to dismiss the
> use-case.  I know there are many such "super-tight" networks, but I just
> don't see how we can reasonably accommodate their use-case in this WG.  And
> I'm pointing out that we really are following their policies, inasmuch as
> WebRTC media will fail to escape their network - I would be far more
> worried if we somehow bypassed their policies instead.  At least this way
> it's similar behavior they get for unknown/unauthorized applications, which
> is arguably what going to random websites and running downloaded javascript
> is equivalent to.
>

Sorry for the late response, but enterprise policy does not have to be
WebRTC on or off. It can be video is not allowed from the enterprise
network but audio is ok. It can be that enterprise would only allow to
communicate with SDP signed by some predefined list of identity providers.
It can be that only certain amount of bandwidth can be used for WebRTC
traffic. WebRTC media communications can only be allowed with certain IP
networks. All of those policies are common with SIP and enterprise session
border controllers. Right now there is no granular control similar to the
control that you would get by running  SIP enabled firewall. I would
consider this a serious regression comparing to current SIP based
communications and would really like this addressed.
_____________
Roman Shpount

--047d7b3395a13a19f304bc5bb163
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<br clear=3D"all"><div class=3D"gmail_quote">On Fri, Mar 23, 2012 at 11:19 =
PM, Hadriel Kaplan <span dir=3D"ltr">&lt;<a href=3D"mailto:HKaplan@acmepack=
et.com">HKaplan@acmepacket.com</a>&gt;</span> wrote:<br><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex">




<div style=3D"word-wrap:break-word"><div>
I&#39;m not trying to be pedantic or difficult here, or trying to dismiss t=
he use-case. =A0I know there are many such &quot;super-tight&quot; networks=
, but I just don&#39;t see how we can reasonably accommodate their use-case=
 in this WG. =A0And I&#39;m pointing out that we really
 are following their policies, inasmuch as WebRTC media will fail to escape=
 their network - I would be far more worried if we somehow bypassed their p=
olicies instead. =A0At least this way it&#39;s similar behavior they get fo=
r unknown/unauthorized applications, which
 is arguably what going to random websites and running downloaded javascrip=
t is equivalent to.</div>
<div></div></div></blockquote><div><br>Sorry for the late response, but ent=
erprise policy does not have to be WebRTC on or off. It can be video is not=
 allowed from the enterprise network but audio is ok. It can be that enterp=
rise would only allow to communicate with SDP signed by some predefined lis=
t of identity providers. It can be that only certain amount of bandwidth ca=
n be used for WebRTC traffic. WebRTC media communications can only be allow=
ed with certain IP networks. All of those policies are common with SIP and =
enterprise session border controllers. Right now there is no granular contr=
ol similar to the control that you would get by running=A0 SIP enabled fire=
wall. I would consider this a serious regression comparing to current SIP b=
ased communications and would really like this addressed.<br>
</div></div>_____________<br>Roman Shpount<br>

--047d7b3395a13a19f304bc5bb163--

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Date: Thu, 29 Mar 2012 02:20:49 -0400
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From: Roman Shpount <roman@telurix.com>
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Subject: Re: [rtcweb] Resolving RTP/SDES question in Paris
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On Fri, Mar 23, 2012 at 2:39 PM, Harald Alvestrand <harald@alvestrand.no>wrote:

> **
> It seems to me that you are arguing that the scenarios in section 4.1 of
> the use cases document do not cover that specific case, and I think you are
> right in that; the list is:
>
>    The following considerations are applicable to all use cases:
>    o  Clients can be on IPv4-only
>    o  Clients can be on IPv6-only
>    o  Clients can be on dual-stack
>    o  Clients can be on wideband (10s of Mbits/sec)
>    o  Clients can be on narrowband (10s to 100s of Kbits/sec)
>    o  Clients can be on variable-media-quality networks (wireless)
>    o  Clients can be on congested networks
>    o  Clients can be on firewalled networks with no UDP allowed
>    o  Clients can be on networks with cone NAT
>    o  Clients can be on networks with symmetric NAT
>
> Now, there are two ways to interpret this omission:
>
> - The WG did not think of that use case when the list was created
> - The WG does not want that use case on the list because it constrains the
> solution space too much
>
> If (re)opening this issue, I think I'd find myself in the "do not want
> that use case" camp.
>
>
I do not recall this use case ever being discussed on the working group, so
I would assume the current situation is due to WG not thinking about this
case when the list was created.
_____________
Roman Shpount

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<br clear=3D"all"><div class=3D"gmail_quote">On Fri, Mar 23, 2012 at 2:39 P=
M, Harald Alvestrand <span dir=3D"ltr">&lt;<a href=3D"mailto:harald@alvestr=
and.no">harald@alvestrand.no</a>&gt;</span> wrote:<br><blockquote class=3D"=
gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-=
left:1ex">
<u></u>

 =20
   =20
   =20
 =20
  <div bgcolor=3D"#ffffff" text=3D"#000000">
    It seems to me that you are arguing that the scenarios in section
    4.1 of the use cases document do not cover that specific case, and I
    think you are right in that; the list is:<br>
    <br>
    =A0=A0 The following considerations are applicable to all use cases:<br=
>
    =A0=A0 o=A0 Clients can be on IPv4-only<br>
    =A0=A0 o=A0 Clients can be on IPv6-only<br>
    =A0=A0 o=A0 Clients can be on dual-stack<br>
    =A0=A0 o=A0 Clients can be on wideband (10s of Mbits/sec)<br>
    =A0=A0 o=A0 Clients can be on narrowband (10s to 100s of Kbits/sec)<br>
    =A0=A0 o=A0 Clients can be on variable-media-quality networks (wireless=
)<br>
    =A0=A0 o=A0 Clients can be on congested networks<br>
    =A0=A0 o=A0 Clients can be on firewalled networks with no UDP allowed<b=
r>
    =A0=A0 o=A0 Clients can be on networks with cone NAT<br>
    =A0=A0 o=A0 Clients can be on networks with symmetric NAT<br>
    <br>
    Now, there are two ways to interpret this omission:<br>
    <br>
    - The WG did not think of that use case when the list was created<br>
    - The WG does not want that use case on the list because it
    constrains the solution space too much<br>
    <br>
    If (re)opening this issue, I think I&#39;d find myself in the &quot;do =
not
    want that use case&quot; camp.<br><font color=3D"#888888">
    </font><br></div></blockquote><div><br>I do not recall this use case ev=
er being discussed on the working group, so I would assume the current situ=
ation is due to WG not thinking about this case when the list was created. =
<br>
</div></div>_____________<br>Roman Shpount<br>

--047d7b15aee9a8430604bc5bb9db--

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Subject: Re: [rtcweb] SRTP and "marketing"
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On 3/28/12 11:58 PM, Hadriel Kaplan wrote:
> 
> On Mar 28, 2012, at 6:59 PM, Dan Wing wrote:
> 
>> We do need a foundation upon which an authentication/identity 
>> infrastructure can be built.  We know we need one.
>> That foundation is DTLS-SRTP, and not Security Descriptions.
> 
> Now you're starting to sound like a marketing guy.  ;)
> What's next: "we'll build more synergy and have a unified platform with DTLS-SRTP"?
> 
> But more seriously, I don't understand this "foundation" argument.  We're going to have DTLS-SRTP.  No one's suggesting we don't have DTLS-SRTP.  

I'm no-one, but i would strongly argue to use SDES-SRTP considering that
forcing the world to implement a non-used new standard (DTLS-SRTP) for a
new not-yet-implemented new standard (WebRTC) it's a fault.

> All Browsers MUST implement DTLS-SRTP.  We'll have it for Browser-to-Browser, and for Browser-to-Gateway if the Gateway supports it.  We'll have the foundation.
> 
> Requiring it for Gateways would make sense if it offered some real advantage, or didn't have any disadvantages.  There don't appear to be real advantages, while we know of disadvantages.  And gateways have no real means of offering an end-to-end identity.  Why would you want to build a foundation on air?

We need to build a new protocol on the foundation of existing protocol.

DTLS-SRTP doesn't exists because no-one use it, there are no diffused
implementation and no interoperability testing done.

DTLS-SRTP require a *huge effort* to set it up.

SDES-SRTP does not require such *huge effort* and will unleash the
advantage of an existing ecosystem of application and protocol stacks
already there.

While i understand Mr. Wing points related to Identity, i think that
Identity will be guaranteed not at "media level" but from existing
transport that is HTTP/HTTPS and all the W3C protocol that run on top of
the Web (Federated Authority) to handle Authorization and Authentication.

Reinventing something new it's imho just wrong.

-- 
Fabio Pietrosanti
Founder, CTO

Tel: +39 02 911930893 + ext: 907
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On 3/29/12 12:31 AM, Jim Barnett wrote:
> Another point is that username/password authentication schemes aren't
> really that secure - it's pretty easy to steal a username and password,
> after all.  So there's not  really a binary
> authenticated/unauthenticated switch.  It's more a matter of degree.  

Why we would like to solve this problem at WebRTC level, while there's a
stockpile of standard, best-practice and future perspective to
authenticate/authorize users over Web secured (TLS) transport?

VoIP for Browser imply that we have a Browser and a Web application
(server & client side javascript).
Imho we should re-use what's already there trying to avoid adding more
complexity.

-- 
Fabio Pietrosanti
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On 3/29/12 7:42 AM, Roman Shpount wrote:
> I actually believe that if sufficient
> monitoring constraints are not build into a browser (not to record but
> at least to monitor who the browser is exchanging data with and using
> what protocols), WebRTC would be simply disabled in most enterprises as
> a security risk.

Your concern would be addressed by the use of SDES-SRTP rather than
DTLS-SRTP.

SDES-SRTP would provide, in the context of WebRTC, the transport of SDES
key over HTTP(S) and so would let all existing methods for HTTP/HTTPS
inspection to works fine.

Technology for inspection of HTTP/HTTPS traffic already exists, are
widely deployed and so if we transport keying material over HTTP (with
SDES-SRTP), all Enterprises will already have their existing
infrastructure in-place.


-- 
Fabio Pietrosanti
Founder, CTO

Tel: +39 02 911930893 + ext: 907
Mobile: +39 340 1801049
E-mail: fabio.pietrosanti@privatewave.com
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PrivateWave Italia S.p.A.
Via Gaetano Giardino 1 - 20123 Milano - Italy
www.privatewave.com

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On 3/28/12 8:01 PM, Basil Mohamed Gohar wrote:
>> IMHO it's really clear that SRTP (even with SDES) is MUCH better than
>> plain RTP. And so far I have not heard any advantage fof allowing
>> plain RTP other than "it allows interoperability with my 5 years ago
>> SIP device".
>>
>> So +1 for the voted consensus.
>>
>> Regards.
> This captures the sentiment of what I was trying to convey.  So, +1 to
> this explanation.
+1

I totally support this initiative.

If we would allow plain RTP any people with just using free too such as
Cain&Abel http://www.oxid.it/cain.html can do Mitm+RTP-recording+Playing
with a couple of click.

We do not really want to see such kind of reviews and widespread of
dumb-proof hacking tools/video/howto to show "your grandma how easy is
to intercept your neighborhood calls".

That's just a bad reputation that will hit that newly born technology.

-- 
Fabio Pietrosanti
Founder, CTO

Tel: +39 02 911930893 + ext: 907
Mobile: +39 340 1801049
E-mail: fabio.pietrosanti@privatewave.com
Skype: fpietrosanti
Linkedin: http://linkedin.com/in/secret

PrivateWave Italia S.p.A.
Via Gaetano Giardino 1 - 20123 Milano - Italy
www.privatewave.com

From roman@telurix.com  Wed Mar 28 23:45:59 2012
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--047d7b15a68b70d2b504bc5c13fd
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On Thu, Mar 29, 2012 at 2:27 AM, Fabio Pietrosanti (naif) <
lists@infosecurity.ch> wrote:

> On 3/29/12 7:42 AM, Roman Shpount wrote:
> > I actually believe that if sufficient
> > monitoring constraints are not build into a browser (not to record but
> > at least to monitor who the browser is exchanging data with and using
> > what protocols), WebRTC would be simply disabled in most enterprises as
> > a security risk.
>
> Your concern would be addressed by the use of SDES-SRTP rather than
> DTLS-SRTP.
>
> SDES-SRTP would provide, in the context of WebRTC, the transport of SDES
> key over HTTP(S) and so would let all existing methods for HTTP/HTTPS
> inspection to works fine.
>
> Technology for inspection of HTTP/HTTPS traffic already exists, are
> widely deployed and so if we transport keying material over HTTP (with
> SDES-SRTP), all Enterprises will already have their existing
> infrastructure in-place.
>

I am not sure I would agree with this. In order for what you propose to
work there is a need for a reliable mechanism to extract SDP from
HTTP/HTTPS traffic. This is not as simple as you would think, since
JavaScript application can use any custom protocol to transmit the SDP
data. It can, for instance, even implement encryption in JavaScript that
will encode SDP using user supplied key, which would make SDP extraction
impossible. In order to implement such inspection we need to have some sort
of hook in the browser that allows an external entity to intercept and
examine all the SDP being passed to and from WebRTC. Without such hook
there is no benefit to SDES. With such hook there is no downside to DTLS.
_____________
Roman Shpount

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<br clear=3D"all"><div class=3D"gmail_quote">On Thu, Mar 29, 2012 at 2:27 A=
M, Fabio Pietrosanti (naif) <span dir=3D"ltr">&lt;<a href=3D"mailto:lists@i=
nfosecurity.ch">lists@infosecurity.ch</a>&gt;</span> wrote:<br><blockquote =
class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid=
;padding-left:1ex">
<div class=3D"im">On 3/29/12 7:42 AM, Roman Shpount wrote:<br>
&gt; I actually believe that if sufficient<br>
&gt; monitoring constraints are not build into a browser (not to record but=
<br>
&gt; at least to monitor who the browser is exchanging data with and using<=
br>
&gt; what protocols), WebRTC would be simply disabled in most enterprises a=
s<br>
&gt; a security risk.<br>
<br>
</div>Your concern would be addressed by the use of SDES-SRTP rather than<b=
r>
DTLS-SRTP.<br>
<br>
SDES-SRTP would provide, in the context of WebRTC, the transport of SDES<br=
>
key over HTTP(S) and so would let all existing methods for HTTP/HTTPS<br>
inspection to works fine.<br>
<br>
Technology for inspection of HTTP/HTTPS traffic already exists, are<br>
widely deployed and so if we transport keying material over HTTP (with<br>
SDES-SRTP), all Enterprises will already have their existing<br>
infrastructure in-place.<br></blockquote><div><br>I am not sure I would agr=
ee with this. In order for what you propose to work there is a need for a r=
eliable mechanism to extract SDP from HTTP/HTTPS traffic. This is not as si=
mple as you would think, since JavaScript application can use any custom pr=
otocol to transmit the SDP data. It can, for instance, even implement encry=
ption in JavaScript that will encode SDP using user supplied key, which wou=
ld make SDP extraction impossible. In order to implement such inspection we=
 need to have some sort of hook in the browser that allows an external enti=
ty to intercept and examine all the SDP being passed to and from WebRTC. Wi=
thout such hook there is no benefit to SDES. With such hook there is no dow=
nside to DTLS.<br>
</div></div>_____________<br>Roman Shpount<br>
<br>

--047d7b15a68b70d2b504bc5c13fd--

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Subject: Re: [rtcweb] DTLS-SRTP implementation diffusion: Why not SDES-SRTP?
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On 3/28/12 9:53 PM, IÃ±aki Baz Castillo wrote:
> 2012/3/28 Fabio Pietrosanti (naif) <lists@infosecurity.ch>:
>> Hi all,
>>
>> i read that 80% of Sipit participant support SDES-SRTP but 0% support
>> DTLS-SRTP  https://www.sipit.net/SIPit29_summary .
>>
>> At SIPit there were 34 attendees from 17 companies visiting from 12
>> countries with 25 distinct VoIP implementations.
> 
> Right, but this is rtcweb, not SIP.
> 
> 
> 
>> I do not really see which is the rationale in making DTLS-SRTP mandatory
>> while plain SRTP with SDES key exchange is already so well know and used.
> 
> That's a good reason to *also* allow (and mandate) SDES-SRTP support
> in WebRTC clients, much better than the interoperability with SIP
> (again: this is rtcweb, not SIP world).

That's true, but it's also true that the "rtcweb world" will strictly
inter-operate with the "sip world".

It would be reasonable to expect that current existing PBX software
would evolve also with support for Rtcweb, to provide Web phone systems.

In particular all opensource software will setup the path for the
adoption of the standard, as we know history will repeat.

It appear me as natural behavior of diffusion of implementation, and for
that reason i see the need to "easily" inter-operate with the SIP world
is a key value point.

Creating an incompatible "media format" would require a lot of more
effort because the "amount of compatibility testing to be done with
DTLS-SRTP will be significantly higher than SDES-SRTP" .

So it would provide an advantage for the *few vendor* supporting it,
practically introducing a *technological entrance barrier*.

This is a bad practice already see in other standard bodies.

> 
> 
>> Anyone can provide some very strong and valuable point about using
>> DTLS-SRTP (considering it's weak diffusion and incompatibility risks)?
> 
> Lot of recent threads about this topic in this maillist. But also
> check a recent presentation (yesterday in IETF Pairs):
> 
> http://tools.ietf.org/agenda/83/slides/slides-83-rtcweb-3.pdf

I would like to strongly argue against the SLIDE 3 statement that
"DTLS-SRTP meets RTCWEB's technical requirements" .

DTLS-SRTP doesn't meet the RTCWEB's technical requirements because:

- It does NOT provide inter-operation with existing SIP endpoints

This is confirmed by the October 2011 SIPit data, with 80% of
participants supporting, with good interoperability, SDES but with 0%
supporting DTLS-SRTP .

In order to try to "try to improve it's non-interoperability issue" the
DTLS-SRTP is re-proposing him-selves as DTLS-SRTP-EKT .

To speaking about the fact that even EKT draft explain that:
" Today, Security Descriptions [RFC4568] is used for distributing SRTP
   keys in several different IP PBX systems and is expected to be used
   by 3GPP's Long Term Evolution (LTE). "

http://tools.ietf.org/html/draft-ietf-avt-srtp-ekt-01#section-6.1

So:
- Internt is using SDES-SRTP
- 3GPP LTE is using SDES-SRTP
- WebRTC is going to use DTLS-SRTP

I do not really see how we can rationally accept to follow this
different direction.

I mean, we are discussing about a "new standard" that's based on "new
technologies" rather than using existing, widely implemented technology.

Do we really understand how much effort we are going to cause on the
overall technological and security ecosystem by selecting DTLS-SRTP
rather than SDES-SRTP?

All US Federal Government will not be able to use WebRTC because NSA
standardized SDES-SRTP for use in Classified communication:
http://www.nsa.gov/ia/programs/mobility_program/index.shtml

The argument that DTLS is *more secure* must face the reality that
no-one is using it and that SDES-SRTP is *widely diffused and
interoperable*.

All Internet operators will have to introduce Protocol Gateway.
All mobile operators will have to introduce Protocol Gateway.

All that subjects, if using just SDES-SRTP, would just need to "update
the software they already use to run VoIP infrastructure" with no need
to handle modification to the Media for Interworking.

So definitely the choice to go for DTLS-SRTP is imho a wrong choice,
against any rationale for the diffusion of WebRTC standard, introducing
artificially complexity where it may be possible to keep it simple.

-- 
Fabio Pietrosanti
Founder, CTO

Tel: +39 02 911930893 + ext: 907
Mobile: +39 340 1801049
E-mail: fabio.pietrosanti@privatewave.com
Skype: fpietrosanti
Linkedin: http://linkedin.com/in/secret

PrivateWave Italia S.p.A.
Via Gaetano Giardino 1 - 20123 Milano - Italy
www.privatewave.com

From magnus.westerlund@ericsson.com  Thu Mar 29 00:09:49 2012
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Subject: [rtcweb] Poll for dates for Interim Meeting before IETF 84
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WG,

We have started a doodle poll for an Interim meeting before IETF 84 in
Vancouver. Locations considered so far are Stockholm, Boston Area or
Silicon Valley. The meeting is planned to be a 2 day meeting. Please
indicate which of the suggest times that works for you.


http://doodle.com/nm3pp69znr3286cy


Cheers

Magnus Westerlund
WG chair


From martin.thomson@gmail.com  Thu Mar 29 00:39:01 2012
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...is a matter for API, not a subject for the IETF.

Release of "resources" is the only feature that PRANSWER provides.
This is the business of the contract between application and
application host (browser).

--Martin

From magnus.westerlund@ericsson.com  Thu Mar 29 00:57:15 2012
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Subject: [rtcweb] Regarding cloning of PeerConnections when multiple answers are received
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WG, as individual

As just discussed in the meeting Justin and Hadriel didn't think that
you could clone PeerConnections properly. I do believe it is possible.

If the API provides a method where you can clone a current
peerConnection you can actually share the transport layer information at
one end-point for multiple peer connections assuming that the other
end-points transport addresses are different between the different
peerConnections being cloned.

This works as the complete ICE username is the concatenation of both
sides, as long as an secondary answer contains a different username and
has candidates with different transport addresses (address+port). Then
end-point A receiving the second answer can have the ICE stack accept
both usernames and if connectivity checks works then you create
different 5-tuple filter to divide any media streams to the right peer
connections media handler.

When you clone a PC then must allocate the media decoding resources for
each cloning. And that is something the cloning function may block on.


Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


From ashwin.shirvanthe@gmail.com  Thu Mar 29 01:36:11 2012
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Subject: [rtcweb] Discussion on Congestion Control
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Hi,

At the iccrg presentations at IETF there were a good number of people
who were interested in working on congestion control in RTCWeb. I am
interested in working in this area and I have expressed my interest to
a few people working on RTCWeb at Mozilla. I would like to know if
people are interested in discussing the problems in this area. I am
assuming that the people interested in this area would attend the
conex wg meeting today. The conex presentation ends at 19:40 in room
253. We could meet in this room after the conex presentation and if
required move to another room.

Topics that I would like to discuss include
1. Is RTCWeb [ http://tools.ietf.org/html/draft-alvestrand-rtcweb-congestion-01
]  being too conservative with respect to increasing the rate of
sending packets.
2. How to address issues related to other delay based congestion
control algorithms like Vegas.
3. Lessons learned from LEDBAT.

Please feel free to add other topics.

Regards,
Ashwin

From ietf@meetecho.com  Thu Mar 29 01:36:27 2012
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Cc: Team Meetecho <team@meetecho.com>
Subject: [rtcweb] Meetecho session recording available
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Dear all,

the full recording (synchronized video, audio, slides and jabber room)
of RTCWEB session at IETF-83 (Wednesday) is available.

You can watch it by either clicking the proper link on the remote 
participation page 
(http://www.ietf.org/meeting/83/remote-participation.html#Meetecho), or 
by directly accessing the following URL:
http://ietf83.conf.meetecho.com/index.php/Recorded_Sessions#RTCWEB_IETF83

For the chair(s): please feel free to put the link to the recording in 
the minutes, if you think this might be useful.

In case of problems with the playout, just drop an e-mail to
team@meetecho.com.

Cheers,
the Meetecho team

-- 
Meetecho s.r.l.
Web Conferencing and Collaboration Tools
www.meetecho.com

From magnus.westerlund@ericsson.com  Thu Mar 29 01:44:22 2012
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Subject: Re: [rtcweb] Poll for dates for Interim Meeting before IETF 84
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WG,

Please respond to this poll no later than Friday the 13th of April.

Cheers

Magnus

On 2012-03-29 09:09, Magnus Westerlund wrote:
> WG,
> 
> We have started a doodle poll for an Interim meeting before IETF 84 in
> Vancouver. Locations considered so far are Stockholm, Boston Area or
> Silicon Valley. The meeting is planned to be a 2 day meeting. Please
> indicate which of the suggest times that works for you.
> 
> 
> http://doodle.com/nm3pp69znr3286cy
> 
> 
> Cheers
> 
> Magnus Westerlund
> WG chair
> 
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> 


-- 

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


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Subject: [rtcweb] Support of SDES in WebRTC
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On the topic of SDES vs DTLS-SRTP a very nice analysis has been done on:
http://www.potaroo.net/ietf/idref/draft-ohlsson-rtcweb-sdes-support/

Considering how vague are the security implementation consideration of
DTLS-SRTP and how widely diffused in SDES-SRTP, i'm wondering whenever
we should not mandate the use of SDES-SRTP.

That way we would be able to see:

- All implementation will start with SDES (simpler and quicker)
- For who want enhanced security, greater implementation and
interoperability complexity, implement also DTLS-SRTP

But for the industry a full cycle of:
- Implementation
- Stabilization
- Interoperability

for DTLS-SRTP (that nobody use) may require from 2 up to 4-5 years of time.

If we goes for SDES-SRTP by default, as early choice, probably within
1-2 years at maximum we can expect the overall technological ecosystem
to be stable and interoperable.

Because it will be built on top of existing stable framework.

We may save dozens millions of USD of investments costs for the industry
and gain some hundreds millions of users year in advance, just because
we would save several years of time.

I think that those consideration are valuable enough to strongly
consider arguments in making SDES-SRTP a preferred, mandatory
implementation for WebRTC.

-- 
Fabio Pietrosanti
Founder, CTO

Tel: +39 02 85961748 (direct)
Mobile: +39 340 1801049
E-mail: fabio.pietrosanti@privatewave.com
Skype: fpietrosanti
Linkedin: http://linkedin.com/in/secret

PrivateWave Italia S.p.A.
Via Gaetano Giardino 1 - 20123 Milano - Italy
www.privatewave.com

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From: Varun Singh <vsingh.ietf@gmail.com>
Date: Thu, 29 Mar 2012 11:00:30 +0200
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Cc: Randell Jesup <randell@jesup.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>, Patrick McManus <pmcmanus@mozilla.com>
Subject: Re: [rtcweb] Discussion on Congestion Control
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Hi Ashwin,

Most of the discussion related to congestion control for RTCWEB takes
place at rtp-congestion@alvestrand.no. Will be happy to join this
meeting.

Cheers,
Varun

On Thu, Mar 29, 2012 at 10:36, Ashwin Rao <ashwin.shirvanthe@gmail.com> wro=
te:
> Hi,
>
> At the iccrg presentations at IETF there were a good number of people
> who were interested in working on congestion control in RTCWeb. I am
> interested in working in this area and I have expressed my interest to
> a few people working on RTCWeb at Mozilla. I would like to know if
> people are interested in discussing the problems in this area. I am
> assuming that the people interested in this area would attend the
> conex wg meeting today. The conex presentation ends at 19:40 in room
> 253. We could meet in this room after the conex presentation and if
> required move to another room.
>
> Topics that I would like to discuss include
> 1. Is RTCWeb [ http://tools.ietf.org/html/draft-alvestrand-rtcweb-congest=
ion-01
> ] =A0being too conservative with respect to increasing the rate of
> sending packets.
> 2. How to address issues related to other delay based congestion
> control algorithms like Vegas.
> 3. Lessons learned from LEDBAT.
>
> Please feel free to add other topics.
>
> Regards,
> Ashwin
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb



--=20
http://www.netlab.tkk.fi/~varun/

From otto.wittner@uninett.no  Thu Mar 29 02:57:20 2012
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Subject: [rtcweb] Mandatory codec on condition?
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This might be a totally ridiculous suggestion, but I give it a shot anyway.

Is there any way that two codecs could be specified as mandatory:
 - one unconditionally mandatory because it is guaranteed royalty free
(e.g. h261)
 - one conditionally mandatory, that is mandatory as long as it stays
royalty free (e.g. VP8).

I'm unsure if "conditionally manadatory" really makes sense, I admit.

-- 
Otto J Wittner   Tel 73557945
UNINETT AS       Mob 99550566

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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Randell Jesup <randell-ietf@jesup.org>
Thread-Topic: [rtcweb] SRTP and "marketing"
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On Mar 29, 2012, at 2:54 AM, Randell Jesup wrote:

>>   No one is concerned about that.
>=20
> Define "no one" :-)

There are Billions of people on the planet.  Obviously I don't mean the num=
ber who don't care is 0. :)


> That doesn't mean I don't want to implement a truly secure communications=
 ability.  Note that by "secure" here I mean "I can detect if the service M=
ITMd me" -- without reading cert phrases and preferably without having to i=
nstall client certs and somehow build a working PKI.
> One idea we've had is a non-centralized WebRTC service built on distribut=
ed hash tables and a partial-mesh set of connections used to pass signaling=
 on WebRTC (or other) data channels to set up calls, with some type of Id s=
ystem to validate who you get connected to (since you can't really trust an=
y of the nodes fully).  If an intermediary node can downgrade you to SDES a=
nd sniff the keys, there is no security.

If you use SDES, no one should be claiming you *have* end-to-end security. =
 You have security from the browser to the HTTPS server, but nothing more. =
 The browser shouldn't be showing a lock icon, for example (and it shouldn'=
t for DTLS-SRTP either).  Since the media potentially goes into the PSTN, i=
t can be no more secure than that.  No matter how much stronger we make the=
 Browser-to-Gateway link, the end-to-end can't exceed the PSTN's level of s=
ecurity.  It's reasonably sufficient security for everyday use, but it won'=
t let you bypass the government.  And using DTLS-SRTP to a gateway won't ei=
ther.  And arguably even using IdP won't if the government can force the Id=
P to lie.


> An alternative would be to allow a hybrid: SDES at the start of a call an=
d rekey over to DTLS-SRTP after the call is connected.  If something blocks=
 the DTLS connection it would be treated as a gateway-to-legacy/PSTN call (=
i.e. show no security, though it's secure against non-in-path attackers); i=
f the DTLS connection is allowed, we can use it to rekey to DTLS-SRTP and t=
o verify identity.

Why bother?  I assume the concern is a bid-down.  Let's imagine we were dea=
ling with a malicious website, and the website javascript removed DTLS-SRTP=
 from SDP but put in SDES, so that browser believes it's talking to a gatew=
ay and uses SDES.  The malicious website now has your SRTP key and can decr=
ypt your media, assuming it has some access to your media.  So you build a =
better browser and try an active check for DTLS-SRTP in the media plane reg=
ardless of SDP.  If your browser is popular, why wouldn't the malicious web=
site just insert a DTLS-SRTP b2bua gateway, and still be able to listen in?=
 =20

As far as I can see, there is no bid-down issue created by adding SDES - it=
 was already there.  It was already possible to insert a DTLS-SRTP B2BUA an=
d be malicious.  Yes, we may have made it slightly cheaper to be malicious,=
 but it was already possible.  The only thing that prevents a malicious web=
site's DTLS-SRTP B2BUA from working is if then humans verify the fingerprin=
ts or having an IdP you can trust.  Even if you *do* verify the fingerprint=
s and find they do NOT match, all you know is it went through one or more g=
ateways in the middle - you don't know it's malicious - it could just be go=
ing through the PSTN legitimately.

With SDES the browser can indicate right-off-the-bat that the call is not e=
nd-to-end secure, without having to show the fingerprint and having the use=
r read it out.


> This minimizes the risk of bid-down - the primary additional risk is that=
 without initial SDES, to tap the signal you'd need to use a WebRTC-enabled=
 B2BUA to accept the DTLS-SRTP connection and siphon it off.  Also, a "good=
" gateway in a pure DTLS-SRTP world might have a identity established ("gat=
eway@L3.com"); with SDES it wouldn't by default (but it could if it wanted =
to by rekeying via DTLS also). =20

Yeah I've been trying to figure out if there's some advantage to having a g=
ateway be able to indicate "gateway@telco.com", once there is an IdP model.=
  I would think it's a major pita to deploy such a thing on gateways, unles=
s there's a very popular IdP they could know everyone would trust.  It woul=
d actually be easier to just give the gateway a real TLS cert for DTLS to u=
se, from a major CA, since gateways are the types of systems real certs wou=
ld be reasonably possible to install on.


> I'm still strongly in favor of pure DTLS-SRTP - as Harald and others say,=
 less moving parts and less things that need to be secured.  But this might=
 be a useful compromise if we *need* to support SDES in some cases.

I'm trying to be convinced that DTLS-SRTP is worth the complexity, cost and=
 downsides on the gateways, but I haven't found it yet.  I think people are=
 willing to be convinced, myself included, but I have yet to hear a real ad=
vantage for DTLS-SRTP in the gateway scenario.  Everything put forward toda=
y for DTLS-SRTP applied to the browser-to-browser scenarios, but not gatewa=
ys. =20

-hadriel


From Ranjit.Avasarala@Polycom.com  Thu Mar 29 04:08:39 2012
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From: "Avasarala, Ranjit" <Ranjit.Avasarala@Polycom.com>
To: Otto J Wittner <otto.wittner@uninett.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Thu, 29 Mar 2012 19:08:29 +0800
Thread-Topic: [rtcweb] Mandatory codec on condition?
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By two codecs you mean 1 audio codec and 1 video codec or 2 audio and 2 vid=
eo codecs?

Regards
Ranjit


-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Otto J Wittner
Sent: Thursday, March 29, 2012 3:27 PM
To: rtcweb@ietf.org
Subject: [rtcweb] Mandatory codec on condition?

This might be a totally ridiculous suggestion, but I give it a shot anyway.

Is there any way that two codecs could be specified as mandatory:
 - one unconditionally mandatory because it is guaranteed royalty free (e.g=
. h261)
 - one conditionally mandatory, that is mandatory as long as it stays royal=
ty free (e.g. VP8).

I'm unsure if "conditionally manadatory" really makes sense, I admit.

--=20
Otto J Wittner   Tel 73557945
UNINETT AS       Mob 99550566
_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

From martin.thomson@gmail.com  Thu Mar 29 04:26:00 2012
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Subject: [rtcweb] IdP and universal trust
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On 29 March 2012 08:08, Hadriel Kaplan <HKaplan@acmepacket.com> wrote:
> Yeah I've been trying to figure out if there's some advantage to having a=
 gateway be able to indicate "gateway@telco.com", once there is an IdP mode=
l. =C2=A0I would think it's a major pita to deploy such a thing on gateways=
, unless there's a very popular IdP they could know everyone would trust. =
=C2=A0It would actually be easier to just give the gateway a real TLS cert =
for DTLS to use, from a major CA, since gateways are the types of systems r=
eal certs would be reasonably possible to install on.

This is an important point, and I think that perhaps you missed the
distinction. Iff third party assertions are permitted, then the
problem that you refer to - namely, finding someone that others trust
- is somewhat difficult.  On the other hand, it is also true that you
can gain multiple assertions and increase the chances of finding an
acceptable IdP.

What seems more likely is that IdPs are only going to be authoritative
for their own domain.  The overall story is a lot easier to tell.
This has the nice quality that you don't have to trust the IdP, you
only have to trust their CA.

--Martin

From otto.wittner@uninett.no  Thu Mar 29 04:33:30 2012
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Two of each if necessary.

O2

On 03/29/2012 01:08 PM, Avasarala, Ranjit wrote:
> By two codecs you mean 1 audio codec and 1 video codec or 2 audio and 2 video codecs?
> 
> Regards
> Ranjit
> 
> 
> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of Otto J Wittner
> Sent: Thursday, March 29, 2012 3:27 PM
> To: rtcweb@ietf.org
> Subject: [rtcweb] Mandatory codec on condition?
> 
> This might be a totally ridiculous suggestion, but I give it a shot anyway.
> 
> Is there any way that two codecs could be specified as mandatory:
>  - one unconditionally mandatory because it is guaranteed royalty free (e.g. h261)
>  - one conditionally mandatory, that is mandatory as long as it stays royalty free (e.g. VP8).
> 
> I'm unsure if "conditionally manadatory" really makes sense, I admit.
> 

-- 
Otto J Wittner   Tel 73557945
UNINETT AS       Mob 99550566

From ietf@meetecho.com  Thu Mar 29 04:43:09 2012
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Cc: Team Meetecho <team@meetecho.com>
Subject: Re: [rtcweb] Meetecho session recording available
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Hi again,

the recording of Thursday's session is available as well:
http://ietf83.conf.meetecho.com/index.php/Recorded_Sessions#RTCWEB_II_IETF83

Cheers,
the Meetecho team

Il 29/03/2012 10:36, Meetecho IETF support ha scritto:
> Dear all,
>
> the full recording (synchronized video, audio, slides and jabber room)
> of RTCWEB session at IETF-83 (Wednesday) is available.
>
> You can watch it by either clicking the proper link on the remote
> participation page
> (http://www.ietf.org/meeting/83/remote-participation.html#Meetecho), or
> by directly accessing the following URL:
> http://ietf83.conf.meetecho.com/index.php/Recorded_Sessions#RTCWEB_IETF83
>
> For the chair(s): please feel free to put the link to the recording in
> the minutes, if you think this might be useful.
>
> In case of problems with the playout, just drop an e-mail to
> team@meetecho.com.
>
> Cheers,
> the Meetecho team
>

From ibc@aliax.net  Thu Mar 29 04:58:40 2012
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Subject: Re: [rtcweb] DTLS-SRTP implementation diffusion: Why not SDES-SRTP?
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2012/3/29 Fabio Pietrosanti (naif) <lists@infosecurity.ch>:

> That's true, but it's also true that the "rtcweb world" will strictly
> inter-operate with the "sip world".
>
> It would be reasonable to expect that current existing PBX software
> would evolve also with support for Rtcweb, to provide Web phone systems.

Right, but it also reasonable to expect that those SIP PBX do
implement some security, at least SDES-SRTP.

NOTE: I do know you are not talking about allowing plain RTP in WebRTC ;)



> In particular all opensource software will setup the path for the
> adoption of the standard, as we know history will repeat.
>
> It appear me as natural behavior of diffusion of implementation, and for
> that reason i see the need to "easily" inter-operate with the SIP world
> is a key value point.
>
> Creating an incompatible "media format" would require a lot of more
> effort because the "amount of compatibility testing to be done with
> DTLS-SRTP will be significantly higher than SDES-SRTP" .

I agree.



>> http://tools.ietf.org/agenda/83/slides/slides-83-rtcweb-3.pdf
>
> I would like to strongly argue against the SLIDE 3 statement that
> "DTLS-SRTP meets RTCWEB's technical requirements" .
>
> DTLS-SRTP doesn't meet the RTCWEB's technical requirements because:
>
> - It does NOT provide inter-operation with existing SIP endpoints
>
> This is confirmed by the October 2011 SIPit data, with 80% of
> participants supporting, with good interoperability, SDES but with 0%
> supporting DTLS-SRTP .
>
> In order to try to "try to improve it's non-interoperability issue" the
> DTLS-SRTP is re-proposing him-selves as DTLS-SRTP-EKT .
>
> To speaking about the fact that even EKT draft explain that:
> " Today, Security Descriptions [RFC4568] is used for distributing SRTP
> =C2=A0 keys in several different IP PBX systems and is expected to be use=
d
> =C2=A0 by 3GPP's Long Term Evolution (LTE). "
>
> http://tools.ietf.org/html/draft-ietf-avt-srtp-ekt-01#section-6.1
>
> So:
> - Internt is using SDES-SRTP
> - 3GPP LTE is using SDES-SRTP
> - WebRTC is going to use DTLS-SRTP
>
> I do not really see how we can rationally accept to follow this
> different direction.

Well, AFAIK the decision for WebRTC is one of the following options
(assuming that plain RTP is not allowed, for which there was consensus
yesterday in Paris):

1) Mandate DTLS-SRTP(-EKT?) implementation and dissalow SDES-SRTP.

2) Mandate DTLS-SRTP(-EKT?) implementation and allow SDES-SRTP (optional).

3) Mandate DTLS-SRTP(-EKT?) implementation and also SDES-SRTP.


I vote for option (3) so WebRTC can *already* interop with well tested
SDES-SRTP environments. Is option (3) good for you?




> The argument that DTLS is *more secure* must face the reality that
> no-one is using it and that SDES-SRTP is *widely diffused and
> interoperable*.
>
> All Internet operators will have to introduce Protocol Gateway.
> All mobile operators will have to introduce Protocol Gateway.

It's worse than that. If you read slides 34 and 35 in
http://tools.ietf.org/agenda/83/slides/slides-83-rtcweb-3.pdf, the
"Media Gateway" MUST be also a *signaling* B2BUA (cannot be a proxy)
since it needs to generate re-INVITE in one leg (for crypto key
updates) !!

I hope I will not need that annoying stuff in order to make WebRTC to
interoperate with SIP networks (assuming SDES-SRTP in those SIP
networks, of course).




> So definitely the choice to go for DTLS-SRTP is imho a wrong choice,
> against any rationale for the diffusion of WebRTC standard, introducing
> artificially complexity where it may be possible to keep it simple.

Worse, it's not just choosing a non tested standard, but creating a
new specification. The slide 39 says:

------------------------
Do part of DTLS handshake in ICE connectivity checks
=E2=80=93 draft-thomson-rtcweb-ice-dtls
-----------------------

So if that *new* spec is mandatory for WebRTC, then forget
interoperability with any other network.


Regards.

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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On 3/29/12 8:18 AM, Roman Shpount wrote:
>
> On Fri, Mar 23, 2012 at 11:19 PM, Hadriel Kaplan <HKaplan@acmepacket.com
> <mailto:HKaplan@acmepacket.com>> wrote:
>
>     I'm not trying to be pedantic or difficult here, or trying to
>     dismiss the use-case.  I know there are many such "super-tight"
>     networks, but I just don't see how we can reasonably accommodate
>     their use-case in this WG.  And I'm pointing out that we really are
>     following their policies, inasmuch as WebRTC media will fail to
>     escape their network - I would be far more worried if we somehow
>     bypassed their policies instead.  At least this way it's similar
>     behavior they get for unknown/unauthorized applications, which is
>     arguably what going to random websites and running downloaded
>     javascript is equivalent to.
>
>
> Sorry for the late response, but enterprise policy does not have to be
> WebRTC on or off. It can be video is not allowed from the enterprise
> network but audio is ok. It can be that enterprise would only allow to
> communicate with SDP signed by some predefined list of identity
> providers. It can be that only certain amount of bandwidth can be used
> for WebRTC traffic. WebRTC media communications can only be allowed with
> certain IP networks. All of those policies are common with SIP and
> enterprise session border controllers. Right now there is no granular
> control similar to the control that you would get by running  SIP
> enabled firewall. I would consider this a serious regression comparing
> to current SIP based communications and would really like this addressed.


This sounds like an excellent use case for host-based controls, and an 
excellent example of the limitations of network-based controls.

--Richard

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--f46d043bde66c8a18e04bc608a47
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@Herald: thanks.

>>For communications that did not need low latency, the WG felt that the
RTCWEB requirements were not different from the requirements of any other
application, so those could be fulfilled with other types of interfaces,
including server-mediated file transfers.

What about use cases of static(non-live) video streaming from browser to
browser, this would require low latency and servers-mediated file transfers
would only add up to the latency, right?. A user would like to share his
video without actually having to upload the content to a server.

>> The purpose of the "use cases and requirements" is to find a minimum set
of things that the protocol suite has to be able to do in order to be "good
enough", not to delineate everything the protocol can be used for.

Agree, but the use cases for the data API, does look promising, and it
didn't seem to have much attention in comparison to live video streaming
use cases.

--f46d043bde66c8a18e04bc608a47
Content-Type: text/html; charset=ISO-8859-1
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<div>@Herald: thanks.</div><div><br></div><div>&gt;&gt;For communications t=
hat did not need low latency, the WG felt that the RTCWEB requirements were=
 not different from the requirements of any other application, so those cou=
ld be fulfilled with other types of interfaces, including server-mediated f=
ile transfers.<br>

</div><div><br></div>What about use cases of static(non-live) video streami=
ng from browser to browser, this would require low latency and servers-medi=
ated file transfers would only add up to the latency, right?. A user would =
like to share his video without actually having to upload the content to a =
server.<div>
<br>
</div><div>&gt;&gt;
The purpose of the &quot;use cases and requirements&quot; is to find a mini=
mum set of things that the protocol suite has to be able to do in order to =
be &quot;good enough&quot;, not to delineate everything the protocol can be=
 used for.=A0</div>
<div><br></div><div>Agree, but the use cases for the data API, does look pr=
omising, and it didn&#39;t seem to have much attention in=A0comparison=A0to=
 live video streaming use cases.</div><div><br></div><div><br></div><div><b=
r>
</div><div><br></div>
<div><br></div><div><br><br>
</div><div><br></div>

--f46d043bde66c8a18e04bc608a47--

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Subject: Re: [rtcweb] DTLS-SRTP implementation diffusion: Why not SDES-SRTP?
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+1

Igor

On 3/29/2012 7:58 AM, IÃ±aki Baz Castillo wrote:
> 2012/3/29 Fabio Pietrosanti (naif)<lists@infosecurity.ch>:
>
>> That's true, but it's also true that the "rtcweb world" will strictly
>> inter-operate with the "sip world".
>>
>> It would be reasonable to expect that current existing PBX software
>> would evolve also with support for Rtcweb, to provide Web phone systems.
> Right, but it also reasonable to expect that those SIP PBX do
> implement some security, at least SDES-SRTP.
>
> NOTE: I do know you are not talking about allowing plain RTP in WebRTC ;)
>
>
>
>> In particular all opensource software will setup the path for the
>> adoption of the standard, as we know history will repeat.
>>
>> It appear me as natural behavior of diffusion of implementation, and for
>> that reason i see the need to "easily" inter-operate with the SIP world
>> is a key value point.
>>
>> Creating an incompatible "media format" would require a lot of more
>> effort because the "amount of compatibility testing to be done with
>> DTLS-SRTP will be significantly higher than SDES-SRTP" .
> I agree.
>
>
>
>>> http://tools.ietf.org/agenda/83/slides/slides-83-rtcweb-3.pdf
>> I would like to strongly argue against the SLIDE 3 statement that
>> "DTLS-SRTP meets RTCWEB's technical requirements" .
>>
>> DTLS-SRTP doesn't meet the RTCWEB's technical requirements because:
>>
>> - It does NOT provide inter-operation with existing SIP endpoints
>>
>> This is confirmed by the October 2011 SIPit data, with 80% of
>> participants supporting, with good interoperability, SDES but with 0%
>> supporting DTLS-SRTP .
>>
>> In order to try to "try to improve it's non-interoperability issue" the
>> DTLS-SRTP is re-proposing him-selves as DTLS-SRTP-EKT .
>>
>> To speaking about the fact that even EKT draft explain that:
>> " Today, Security Descriptions [RFC4568] is used for distributing SRTP
>>    keys in several different IP PBX systems and is expected to be used
>>    by 3GPP's Long Term Evolution (LTE). "
>>
>> http://tools.ietf.org/html/draft-ietf-avt-srtp-ekt-01#section-6.1
>>
>> So:
>> - Internt is using SDES-SRTP
>> - 3GPP LTE is using SDES-SRTP
>> - WebRTC is going to use DTLS-SRTP
>>
>> I do not really see how we can rationally accept to follow this
>> different direction.
> Well, AFAIK the decision for WebRTC is one of the following options
> (assuming that plain RTP is not allowed, for which there was consensus
> yesterday in Paris):
>
> 1) Mandate DTLS-SRTP(-EKT?) implementation and dissalow SDES-SRTP.
>
> 2) Mandate DTLS-SRTP(-EKT?) implementation and allow SDES-SRTP (optional).
>
> 3) Mandate DTLS-SRTP(-EKT?) implementation and also SDES-SRTP.
>
>
> I vote for option (3) so WebRTC can *already* interop with well tested
> SDES-SRTP environments. Is option (3) good for you?
>
>
>
>
>> The argument that DTLS is *more secure* must face the reality that
>> no-one is using it and that SDES-SRTP is *widely diffused and
>> interoperable*.
>>
>> All Internet operators will have to introduce Protocol Gateway.
>> All mobile operators will have to introduce Protocol Gateway.
> It's worse than that. If you read slides 34 and 35 in
> http://tools.ietf.org/agenda/83/slides/slides-83-rtcweb-3.pdf, the
> "Media Gateway" MUST be also a *signaling* B2BUA (cannot be a proxy)
> since it needs to generate re-INVITE in one leg (for crypto key
> updates) !!
>
> I hope I will not need that annoying stuff in order to make WebRTC to
> interoperate with SIP networks (assuming SDES-SRTP in those SIP
> networks, of course).
>
>
>
>
>> So definitely the choice to go for DTLS-SRTP is imho a wrong choice,
>> against any rationale for the diffusion of WebRTC standard, introducing
>> artificially complexity where it may be possible to keep it simple.
> Worse, it's not just choosing a non tested standard, but creating a
> new specification. The slide 39 says:
>
> ------------------------
> Do part of DTLS handshake in ICE connectivity checks
> â€“ draft-thomson-rtcweb-ice-dtls
> -----------------------
>
> So if that *new* spec is mandatory for WebRTC, then forget
> interoperability with any other network.
>
>
> Regards.
>

From ibc@aliax.net  Thu Mar 29 05:15:06 2012
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Thu, 29 Mar 2012 14:14:44 +0200
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Consensus call regarding media security
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2012/3/29 Roman Shpount <roman@telurix.com>:
>
> On Wed, Mar 28, 2012 at 5:11 PM, I=C3=B1aki Baz Castillo <ibc@aliax.net> =
wrote:
>>
>> In all those scenarios you mention, using security (SRTP) is not *bad*, =
is
>> it?
>> The fact that in certain scenarios it could be not needed, it does not
>> mean that "no security" is better than "security".
>>
> I actually think that you are confusing encryption with security.

Roman, you still don't want to see my simple example. Could you please
reply to the following two questions rather than avoiding them?:

1) Why is better to use plain RTP in an open WiFi network (airport)
rather than using SDES-SRTP?

2) Why is plain RTP better than SDES-SRTP in any other case?




> No encryption often means better security.

Sorry, I don't agree. Encryption is not enough, of course, but it's
better than nothing (and again: the open WiFi network in the airport).



> Bot nets use encryption to
> communicate, it does not mean they enable user security.

1) User identity authentication/validation and confidentiality (encryption)=
.

2) Just confidentiality (encryption).

3) Nothing.


1 is better than 2 and 2 is better than 3.



> In a lot of cases
> ability for the user to check what exactly they are communicating and to
> whom is security. As a security conscious person I do not want encrypted
> communication from my computer or my network unless I know exactly who it=
 is
> going to. If I do not have to I would prefer my communications to be
> unencrypted.

Sure, but you are advanced user and I hope you are not talking in
behalf of the 500 millions of Facebook users in the world.
So you will be able to enter into your browser about://config and set
a null crypto key for SRTP (so you get plain RTP and can debug it or
whatever). This is confirmed to be the expectation of two browser
vendors.




>> RFC 3711 was created in 2004, 8 years ago!
>>
>> Which "new" application you mean if it's not capable of implementing a
>> simple specification from 2004? how "new" is it? is it really new? or
>> is it a "SIP voicemail server" made in 2002?
>>
> Well let me list them: FreeSwitch, Asterisk, linphone, etc. All of those =
use
> libsrtp. For all of them SRTP is broken. Not a single one of them bothere=
d
> to check or only recently detected problems
> (https://issues.asterisk.org/jira/browse/ASTERISK-16665).

I expect that, once SRTP becomes widely extended (thanks to WebRTC),
developers will really want to fix their SRTP support, including
libsrtp. But we cannot make a *new spec less secure than it can be
just due a to a bug in a opensource library.



>> Sorry but I still see *no* argument at all in favour of allowing plain
>> RTP in WebRTC. And AFAIK there is already consensus about it: plain
>> RTP is not allowed.
>>
> I do not see an argument why RTP should not be allowed. I think you are
> arguing we should require a feature (SRTP) for the sake of requiring a
> feature.

I still don't understand why everyone in an airport should be able to
intercept my audio/video communication with a simple RTP sniffer.


Regards.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From dean.willis@softarmor.com  Thu Mar 29 05:15:37 2012
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Subject: [rtcweb] Proposal for H.263 baseline codec
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--e89a8ff1cb265e71b504bc60ae18
Content-Type: text/plain; charset=ISO-8859-1

In today's meeting I made a plea for a mandatory baseline codec that is
approachable to the small developer without the cost and IPR risks of
either H.264 or VP8.

I believe that some sort of baseline codec is required to jump start the
protocol. It's OK if this isn't the best possible codec. Market forces will
assure that commercial implementations converge on a higher quality when
the market requires it.

While I proposed today that H.261 could serve as a baseline, we know this
would be "extremely basic". In other words, it might not pass the "good
enough to use" test required for a successful launch.

Stephane and several others suggested H.263 as an alternative during the
after-meeting chat. While it is not as IPR-safe a choice as H.261, it is
relatively mature and the known challenges have generally failed.
Implementations are reasonably available and should make it possible for
smaller implementers, students, and hobbyists to play ball. It's not all
that network efficient and is outright piggy at higher resolutions, but it
probably works "well enough to use."

So I propose we set H.263 as the baseline ( I expect a bit of profiling may
be necessary to further qualify the baseline) and run with that for now. If
the situation changes, we can always replace it before final publication.
--
Dean

--e89a8ff1cb265e71b504bc60ae18
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In today&#39;s meeting I made a plea for a mandatory baseline codec that is=
 approachable to the small developer without the cost and IPR risks of eith=
er H.264 or VP8.<div><br></div><div>I believe that some sort of baseline co=
dec is required to jump start the protocol. It&#39;s OK if this isn&#39;t t=
he best possible codec. Market forces will assure that commercial implement=
ations converge on a higher quality when the market requires it.</div>
<div><br></div><div>While I proposed today that H.261 could serve as a base=
line, we know this would be &quot;extremely basic&quot;. In other words, it=
 might not pass the &quot;good enough to use&quot; test required for a succ=
essful launch.</div>
<div><br></div><div>Stephane and several others suggested H.263 as an alter=
native during the after-meeting chat. While it is not as IPR-safe a choice =
as H.261, it is relatively mature and the known challenges have generally f=
ailed. Implementations are reasonably available and should make it possible=
 for smaller implementers, students, and hobbyists to play ball. It&#39;s n=
ot all that network efficient and is outright piggy at higher resolutions, =
but it probably works &quot;well enough to use.&quot;<span></span></div>
<div><br></div><div>So I propose we set H.263 as the baseline ( I expect a =
bit of profiling may be necessary to further qualify the baseline) and run =
with that for now. If the situation changes, we can always replace it befor=
e final publication.</div>
<div>--</div><div>Dean</div>

--e89a8ff1cb265e71b504bc60ae18--

From lorenzo@meetecho.com  Thu Mar 29 05:29:37 2012
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From: Lorenzo Miniero <lorenzo@meetecho.com>
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+1

Lorenzo


Il giorno Thu, 29 Mar 2012 14:15:35 +0200
Dean Willis <dean.willis@softarmor.com> ha scritto:

> In today's meeting I made a plea for a mandatory baseline codec that
> is approachable to the small developer without the cost and IPR risks
> of either H.264 or VP8.
> 
> I believe that some sort of baseline codec is required to jump start
> the protocol. It's OK if this isn't the best possible codec. Market
> forces will assure that commercial implementations converge on a
> higher quality when the market requires it.
> 
> While I proposed today that H.261 could serve as a baseline, we know
> this would be "extremely basic". In other words, it might not pass
> the "good enough to use" test required for a successful launch.
> 
> Stephane and several others suggested H.263 as an alternative during
> the after-meeting chat. While it is not as IPR-safe a choice as
> H.261, it is relatively mature and the known challenges have
> generally failed. Implementations are reasonably available and should
> make it possible for smaller implementers, students, and hobbyists to
> play ball. It's not all that network efficient and is outright piggy
> at higher resolutions, but it probably works "well enough to use."
> 
> So I propose we set H.263 as the baseline ( I expect a bit of
> profiling may be necessary to further qualify the baseline) and run
> with that for now. If the situation changes, we can always replace it
> before final publication. --
> Dean



-- 
Lorenzo Miniero, COB

Meetecho s.r.l.
Web Conferencing and Collaboration Tools
http://www.meetecho.com

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Hi,

I support Dean's proposal to make H.263 as the mandatory-to-implement codec=
. I'm not expert on profiles, but someone mentioned H.263 profile 0 as some=
thing suitable for our needs (smallest IPR risk).

I also believe that even making H.261 as the mandatory-to-implement would b=
e better than going for the forced pseudo-random selection between H.264 or=
 VP8.

Markus


From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 ext Dean Willis
Sent: 29 March, 2012 15:16
To: rtcweb@ietf.org
Subject: [rtcweb] Proposal for H.263 baseline codec

In today's meeting I made a plea for a mandatory baseline codec that is app=
roachable to the small developer without the cost and IPR risks of either H=
.264 or VP8.

I believe that some sort of baseline codec is required to jump start the pr=
otocol. It's OK if this isn't the best possible codec. Market forces will a=
ssure that commercial implementations converge on a higher quality when the=
 market requires it.

While I proposed today that H.261 could serve as a baseline, we know this w=
ould be "extremely basic". In other words, it might not pass the "good enou=
gh to use" test required for a successful launch.

Stephane and several others suggested H.263 as an alternative during the af=
ter-meeting chat. While it is not as IPR-safe a choice as H.261, it is rela=
tively mature and the known challenges have generally failed. Implementatio=
ns are reasonably available and should make it possible for smaller impleme=
nters, students, and hobbyists to play ball. It's not all that network effi=
cient and is outright piggy at higher resolutions, but it probably works "w=
ell enough to use."

So I propose we set H.263 as the baseline ( I expect a bit of profiling may=
 be necessary to further qualify the baseline) and run with that for now. I=
f the situation changes, we can always replace it before final publication.
--
Dean

--_000_E44893DD4E290745BB608EB23FDDB7621AFB9A008AM1MPN1042mgdn_
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<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Hi,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">I support Dean&#8217;s pr=
oposal to make H.263 as the mandatory-to-implement codec. I&#8217;m not exp=
ert on profiles, but someone mentioned H.263 profile 0 as something
 suitable for our needs (smallest IPR risk).<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">I also believe that even =
making H.261 as the mandatory-to-implement would be better than going for t=
he forced pseudo-random selection between H.264 or VP8.<o:p></o:p></span></=
p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Markus
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0cm 0cm 0cm =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm =
0cm 0cm">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> rtcweb-b=
ounces@ietf.org [mailto:rtcweb-bounces@ietf.org]
<b>On Behalf Of </b>ext Dean Willis<br>
<b>Sent:</b> 29 March, 2012 15:16<br>
<b>To:</b> rtcweb@ietf.org<br>
<b>Subject:</b> [rtcweb] Proposal for H.263 baseline codec<o:p></o:p></span=
></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">In today's meeting I made a plea for a mandatory bas=
eline codec that is approachable to the small developer without the cost an=
d IPR risks of either H.264 or VP8.<o:p></o:p></p>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">I believe that some sort of baseline codec is requir=
ed to jump start the protocol. It's OK if this isn't the best possible code=
c. Market forces will assure that commercial implementations converge on a =
higher quality when the market requires
 it.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">While I proposed today that H.261 could serve as a b=
aseline, we know this would be &quot;extremely basic&quot;. In other words,=
 it might not pass the &quot;good enough to use&quot; test required for a s=
uccessful launch.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Stephane and several others suggested H.263 as an al=
ternative during the after-meeting chat. While it is not as IPR-safe a choi=
ce as H.261, it is relatively mature and the known challenges have generall=
y failed. Implementations are reasonably
 available and should make it possible for smaller implementers, students, =
and hobbyists to play ball. It's not all that network efficient and is outr=
ight piggy at higher resolutions, but it probably works &quot;well enough t=
o use.&quot;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">So I propose we set H.263 as the baseline ( I expect=
 a bit of profiling may be necessary to further qualify the baseline) and r=
un with that for now. If the situation changes, we can always replace it be=
fore final publication.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">--<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Dean<o:p></o:p></p>
</div>
</div>
</div>
</body>
</html>

--_000_E44893DD4E290745BB608EB23FDDB7621AFB9A008AM1MPN1042mgdn_--

From magnus.westerlund@ericsson.com  Thu Mar 29 05:35:41 2012
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On 2012-03-29 08:02, Bernard Aboba wrote:
> I agree with proposition #1 (SRTP) unconditionally.
> 
> With respect to proposition #2 (DTLS-SRTP), perhaps the words "with
> details to be worked out" should have been added. 
> 
> I believe that the consensus achieved was only on a general direction,
> not an endorsement of particular proposals.
> 
> Personally, I would like to have more specifics about the required
> features of DTLS-SRTP in the RTCWEB context.

I hope someone that knows the details can elaborate on this. I thought
DTLS-SRTP has a core that you will need to implement. Then there is
clearly a question of crypto algorithms to be supported. But that also
applies to SRTP where we also need to select which crypto suites that
are to be implemented if any in addtion to the MITM. The WG will need to
select these details as part of the next steps.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
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Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
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From bernard_aboba@hotmail.com  Thu Mar 29 05:45:51 2012
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Subject: Re: [rtcweb] IdP and universal trust
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There is a more basic problem here, which is that "legacy" scenarios are typ=
ically about reaching an E.164 number.   Given that IdP (or PKI for that mat=
ter) cannot be used to prove ownership of an E.164 number, such as +14255551=
212@example.com, it is difficult to see how identity can be provided by any m=
echanism for the legacy cases, and MITM attacks seem intrinsic to the scenar=
io.  For example, +14255551212@example.com is equivalent to +14255551212@evi=
ltwin.example.net so that first-party IdPs are not authoritative for E.164 n=
umbers, and MITM attacks have long been acknowledged as possible within the S=
IP DTLS/SRTP framework (e.g. eviltwin.example.net can change the E.164 URI a=
nd resign the message via RFC 4474 without being detected).



On Mar 29, 2012, at 13:26, "Martin Thomson" <martin.thomson@gmail.com> wrote=
:

> On 29 March 2012 08:08, Hadriel Kaplan <HKaplan@acmepacket.com> wrote:
>> Yeah I've been trying to figure out if there's some advantage to having a=
 gateway be able to indicate "gateway@telco.com", once there is an IdP model=
.  I would think it's a major pita to deploy such a thing on gateways, unles=
s there's a very popular IdP they could know everyone would trust.  It would=
 actually be easier to just give the gateway a real TLS cert for DTLS to use=
, from a major CA, since gateways are the types of systems real certs would b=
e reasonably possible to install on.
>=20
> This is an important point, and I think that perhaps you missed the
> distinction. Iff third party assertions are permitted, then the
> problem that you refer to - namely, finding someone that others trust
> - is somewhat difficult.  On the other hand, it is also true that you
> can gain multiple assertions and increase the chances of finding an
> acceptable IdP.
>=20
> What seems more likely is that IdPs are only going to be authoritative
> for their own domain.  The overall story is a lot easier to tell.
> This has the nice quality that you don't have to trust the IdP, you
> only have to trust their CA.
>=20
> --Martin
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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+1

On Mar 29, 2012, at 14:29, "Lorenzo Miniero" <lorenzo@meetecho.com> wrote:

> +1
> 
> Lorenzo
> 
> 
> Il giorno Thu, 29 Mar 2012 14:15:35 +0200
> Dean Willis <dean.willis@softarmor.com> ha scritto:
> 
>> In today's meeting I made a plea for a mandatory baseline codec that
>> is approachable to the small developer without the cost and IPR risks
>> of either H.264 or VP8.
>> 
>> I believe that some sort of baseline codec is required to jump start
>> the protocol. It's OK if this isn't the best possible codec. Market
>> forces will assure that commercial implementations converge on a
>> higher quality when the market requires it.
>> 
>> While I proposed today that H.261 could serve as a baseline, we know
>> this would be "extremely basic". In other words, it might not pass
>> the "good enough to use" test required for a successful launch.
>> 
>> Stephane and several others suggested H.263 as an alternative during
>> the after-meeting chat. While it is not as IPR-safe a choice as
>> H.261, it is relatively mature and the known challenges have
>> generally failed. Implementations are reasonably available and should
>> make it possible for smaller implementers, students, and hobbyists to
>> play ball. It's not all that network efficient and is outright piggy
>> at higher resolutions, but it probably works "well enough to use."
>> 
>> So I propose we set H.263 as the baseline ( I expect a bit of
>> profiling may be necessary to further qualify the baseline) and run
>> with that for now. If the situation changes, we can always replace it
>> before final publication. --
>> Dean
> 
> 
> 
> -- 
> Lorenzo Miniero, COB
> 
> Meetecho s.r.l.
> Web Conferencing and Collaboration Tools
> http://www.meetecho.com
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From p.ohanlon@gmail.com  Thu Mar 29 05:48:42 2012
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Date: Thu, 29 Mar 2012 14:48:35 +0200
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On 29 Mar 2012, at 14:30, <Markus.Isomaki@nokia.com> wrote:

> Hi,
> =20
> I support Dean=92s proposal to make H.263 as the =
mandatory-to-implement codec. I=92m not expert on profiles, but someone =
mentioned H.263 profile 0 as something suitable for our needs (smallest =
IPR risk).
> =20
> I also believe that even making H.261 as the mandatory-to-implement =
would be better than going for the forced pseudo-random selection =
between H.264 or VP8.
> =20
One of the key limitations with H.261 is it's small range of available =
supported resolutions (QCIF, Max:CIF(352x288)) - so if there is a low =
IPR risk for an H.263(+?) configuration then it would be preferable.=20

The whole issue of which Profile and levels one might support for these =
codecs is another issue as I think they lead to differing IPR =
dependencies.

Piers.

> Markus
> =20
> =20
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On =
Behalf Of ext Dean Willis
> Sent: 29 March, 2012 15:16
> To: rtcweb@ietf.org
> Subject: [rtcweb] Proposal for H.263 baseline codec
> =20
> In today's meeting I made a plea for a mandatory baseline codec that =
is approachable to the small developer without the cost and IPR risks of =
either H.264 or VP8.
> =20
> I believe that some sort of baseline codec is required to jump start =
the protocol. It's OK if this isn't the best possible codec. Market =
forces will assure that commercial implementations converge on a higher =
quality when the market requires it.
> =20
> While I proposed today that H.261 could serve as a baseline, we know =
this would be "extremely basic". In other words, it might not pass the =
"good enough to use" test required for a successful launch.
> =20
> Stephane and several others suggested H.263 as an alternative during =
the after-meeting chat. While it is not as IPR-safe a choice as H.261, =
it is relatively mature and the known challenges have generally failed. =
Implementations are reasonably available and should make it possible for =
smaller implementers, students, and hobbyists to play ball. It's not all =
that network efficient and is outright piggy at higher resolutions, but =
it probably works "well enough to use."
> =20
> So I propose we set H.263 as the baseline ( I expect a bit of =
profiling may be necessary to further qualify the baseline) and run with =
that for now. If the situation changes, we can always replace it before =
final publication.
> --
> Dean
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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<html><head></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
"><br><div><div>On 29 Mar 2012, at 14:30, &lt;<a =
href=3D"mailto:Markus.Isomaki@nokia.com">Markus.Isomaki@nokia.com</a>&gt; =
wrote:</div><br class=3D"Apple-interchange-newline"><blockquote =
type=3D"cite"><span class=3D"Apple-style-span" style=3D"border-collapse: =
separate; font-family: Helvetica; font-style: normal; font-variant: =
normal; font-weight: normal; letter-spacing: normal; line-height: =
normal; orphans: 2; text-align: -webkit-auto; text-indent: 0px; =
text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; =
-webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; font-size: medium; "><div =
lang=3D"EN-US" link=3D"blue" vlink=3D"purple"><div class=3D"WordSection1" =
style=3D"page: WordSection1; "><div style=3D"margin-top: 0cm; =
margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span style=3D"font-size: =
11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); =
">Hi,<o:p></o:p></span></div><div style=3D"margin-top: 0cm; =
margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span style=3D"font-size: =
11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); =
"><o:p>&nbsp;</o:p></span></div><div style=3D"margin-top: 0cm; =
margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span style=3D"font-size: =
11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); ">I =
support Dean=92s proposal to make H.263 as the mandatory-to-implement =
codec. I=92m not expert on profiles, but someone mentioned H.263 profile =
0 as something suitable for our needs (smallest IPR =
risk).<o:p></o:p></span></div><div style=3D"margin-top: 0cm; =
margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span style=3D"font-size: =
11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); =
"><o:p>&nbsp;</o:p></span></div><div style=3D"margin-top: 0cm; =
margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span style=3D"font-size: =
11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); ">I =
also believe that even making H.261 as the mandatory-to-implement would =
be better than going for the forced pseudo-random selection between =
H.264 or VP8.<o:p></o:p></span></div><div style=3D"margin-top: 0cm; =
margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span style=3D"font-size: =
11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); =
"><o:p>&nbsp;</o:p></span></div></div></div></span></blockquote>One of =
the key limitations with H.261 is it's small range of available =
supported resolutions (QCIF, Max:CIF(352x288))&nbsp;- so if there is a =
low IPR risk for an H.263(+?) configuration then it would be =
preferable.&nbsp;</div><div><br></div><div>The whole issue of which =
Profile and levels one might support for these codecs is another issue =
as I think they lead to differing IPR =
dependencies.</div><div><br></div><div>Piers.</div><div><br><blockquote =
type=3D"cite"><span class=3D"Apple-style-span" style=3D"border-collapse: =
separate; font-family: Helvetica; font-style: normal; font-variant: =
normal; font-weight: normal; letter-spacing: normal; line-height: =
normal; orphans: 2; text-align: -webkit-auto; text-indent: 0px; =
text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; =
-webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; font-size: medium; "><div =
lang=3D"EN-US" link=3D"blue" vlink=3D"purple"><div class=3D"WordSection1" =
style=3D"page: WordSection1; "><div style=3D"margin-top: 0cm; =
margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span style=3D"font-size: =
11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); =
">Markus<o:p></o:p></span></div><div style=3D"margin-top: 0cm; =
margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span style=3D"font-size: =
11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); =
"><o:p>&nbsp;</o:p></span></div><div style=3D"margin-top: 0cm; =
margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span style=3D"font-size: =
11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); =
"><o:p>&nbsp;</o:p></span></div><div style=3D"border-top-style: none; =
border-right-style: none; border-bottom-style: none; border-width: =
initial; border-color: initial; border-left-style: solid; =
border-left-color: blue; border-left-width: 1.5pt; padding-top: 0cm; =
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style=3D"border-right-style: none; border-bottom-style: none; =
border-left-style: none; border-width: initial; border-color: initial; =
border-top-style: solid; border-top-color: rgb(181, 196, 223); =
border-top-width: 1pt; padding-top: 3pt; padding-right: 0cm; =
padding-bottom: 0cm; padding-left: 0cm; "><div style=3D"margin-top: 0cm; =
margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; "><b><span =
style=3D"font-size: 10pt; font-family: Tahoma, sans-serif; =
">From:</span></b><span style=3D"font-size: 10pt; font-family: Tahoma, =
sans-serif; "><span class=3D"Apple-converted-space">&nbsp;</span><a =
href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a> =
[mailto:rtcweb-bounces@ietf.org]<span =
class=3D"Apple-converted-space">&nbsp;</span><b>On Behalf Of<span =
class=3D"Apple-converted-space">&nbsp;</span></b>ext Dean =
Willis<br><b>Sent:</b><span =
class=3D"Apple-converted-space">&nbsp;</span>29 March, 2012 =
15:16<br><b>To:</b><span class=3D"Apple-converted-space">&nbsp;</span><a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br><b>Subject:</b><spa=
n class=3D"Apple-converted-space">&nbsp;</span>[rtcweb] Proposal for =
H.263 baseline codec<o:p></o:p></span></div></div></div><div =
style=3D"margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; =
margin-bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New =
Roman', serif; "><o:p>&nbsp;</o:p></div><div style=3D"margin-top: 0cm; =
margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; ">In today's meeting I made =
a plea for a mandatory baseline codec that is approachable to the small =
developer without the cost and IPR risks of either H.264 or =
VP8.<o:p></o:p></div><div><div style=3D"margin-top: 0cm; margin-right: =
0cm; margin-left: 0cm; margin-bottom: 0.0001pt; font-size: 12pt; =
font-family: 'Times New Roman', serif; =
"><o:p>&nbsp;</o:p></div></div><div><div style=3D"margin-top: 0cm; =
margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; ">I believe that some sort =
of baseline codec is required to jump start the protocol. It's OK if =
this isn't the best possible codec. Market forces will assure that =
commercial implementations converge on a higher quality when the market =
requires it.<o:p></o:p></div></div><div><div style=3D"margin-top: 0cm; =
margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; =
"><o:p>&nbsp;</o:p></div></div><div><div style=3D"margin-top: 0cm; =
margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; ">While I proposed today =
that H.261 could serve as a baseline, we know this would be "extremely =
basic". In other words, it might not pass the "good enough to use" test =
required for a successful launch.<o:p></o:p></div></div><div><div =
style=3D"margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; =
margin-bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New =
Roman', serif; "><o:p>&nbsp;</o:p></div></div><div><div =
style=3D"margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; =
margin-bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New =
Roman', serif; ">Stephane and several others suggested H.263 as an =
alternative during the after-meeting chat. While it is not as IPR-safe a =
choice as H.261, it is relatively mature and the known challenges have =
generally failed. Implementations are reasonably available and should =
make it possible for smaller implementers, students, and hobbyists to =
play ball. It's not all that network efficient and is outright piggy at =
higher resolutions, but it probably works "well enough to =
use."<o:p></o:p></div></div><div><div style=3D"margin-top: 0cm; =
margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; =
"><o:p>&nbsp;</o:p></div></div><div><div style=3D"margin-top: 0cm; =
margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; ">So I propose we set H.263 =
as the baseline ( I expect a bit of profiling may be necessary to =
further qualify the baseline) and run with that for now. If the =
situation changes, we can always replace it before final =
publication.<o:p></o:p></div></div><div><div style=3D"margin-top: 0cm; =
margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; =
">--<o:p></o:p></div></div><div><div style=3D"margin-top: 0cm; =
margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; =
">Dean<o:p></o:p></div></div></div></div>_________________________________=
______________<br>rtcweb mailing list<br><a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>https://www.ietf.or=
g/mailman/listinfo/rtcweb</div></span></blockquote></div><br></body></html=
>=

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From fluffy@iii.ca  Thu Mar 29 05:54:03 2012
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To: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: Re: [rtcweb] Poll for dates for Interim Meeting before IETF 84
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Could we have the poll choices to indicate the location of the meeting?

The problem is, and I suspect that this is a problem for most people, =
depending on the location, my travel time dramatically changes the =
number of days I need to have clear on my calendar.=20



On Mar 29, 2012, at 9:09 , Magnus Westerlund wrote:

> WG,
>=20
> We have started a doodle poll for an Interim meeting before IETF 84 in
> Vancouver. Locations considered so far are Stockholm, Boston Area or
> Silicon Valley. The meeting is planned to be a 2 day meeting. Please
> indicate which of the suggest times that works for you.
>=20
>=20
> http://doodle.com/nm3pp69znr3286cy
>=20
>=20
> Cheers
>=20
> Magnus Westerlund
> WG chair
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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VGhlcmUgYXJlIGEgbnVtYmVyIG9mIHBvdGVudGlhbCB2YXJpYW50czoNCg0KYS4gRFRMUy1TUlRQ
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From oscar.ohlsson@ericsson.com  Thu Mar 29 06:19:16 2012
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From: Oscar Ohlsson <oscar.ohlsson@ericsson.com>
To: "Fabio Pietrosanti (naif)" <lists@infosecurity.ch>, "<rtcweb@ietf.org>" <rtcweb@ietf.org>
Date: Thu, 29 Mar 2012 15:19:11 +0200
Thread-Topic: [rtcweb] Support of SDES in WebRTC
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Subject: Re: [rtcweb] Support of SDES in WebRTC
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Hi Fabio,

My assumption has always been the following:

- DTLS-SRTP is the default
- DTLS-SRTP + identity can be turned on via the JavaScript API if the webap=
p wishes to do so
- SDES can be turned on by a manipulated SDP offer/answer provided the enti=
re webapp was retrieved over HTTPS

BTW I'm also in favor of _only_ allowing the SRTP NULL encryption algorithm=
 in browser debug/developer mode.

Regards,

Oscar Ohlsson


> -----Original Message-----
> From: rtcweb-bounces@ietf.org=20
> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Fabio Pietrosanti (naif)
> Sent: Thursday, March 29, 2012 10:54 AM
> To: <rtcweb@ietf.org>
> Subject: [rtcweb] Support of SDES in WebRTC
>=20
> On the topic of SDES vs DTLS-SRTP a very nice analysis has=20
> been done on:
> http://www.potaroo.net/ietf/idref/draft-ohlsson-rtcweb-sdes-support/
>=20
> Considering how vague are the security implementation=20
> consideration of DTLS-SRTP and how widely diffused in=20
> SDES-SRTP, i'm wondering whenever we should not mandate the=20
> use of SDES-SRTP.
>=20
> That way we would be able to see:
>=20
> - All implementation will start with SDES (simpler and quicker)
> - For who want enhanced security, greater implementation and=20
> interoperability complexity, implement also DTLS-SRTP
>=20
> But for the industry a full cycle of:
> - Implementation
> - Stabilization
> - Interoperability
>=20
> for DTLS-SRTP (that nobody use) may require from 2 up to 4-5=20
> years of time.
>=20
> If we goes for SDES-SRTP by default, as early choice, probably within
> 1-2 years at maximum we can expect the overall technological=20
> ecosystem to be stable and interoperable.
>=20
> Because it will be built on top of existing stable framework.
>=20
> We may save dozens millions of USD of investments costs for=20
> the industry and gain some hundreds millions of users year in=20
> advance, just because we would save several years of time.
>=20
> I think that those consideration are valuable enough to=20
> strongly consider arguments in making SDES-SRTP a preferred,=20
> mandatory implementation for WebRTC.
>=20
> --
> Fabio Pietrosanti
> Founder, CTO
>=20
> Tel: +39 02 85961748 (direct)
> Mobile: +39 340 1801049
> E-mail: fabio.pietrosanti@privatewave.com
> Skype: fpietrosanti
> Linkedin: http://linkedin.com/in/secret
>=20
> PrivateWave Italia S.p.A.
> Via Gaetano Giardino 1 - 20123 Milano - Italy=20
> www.privatewave.com _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> =

From abu_hurayrah@hidayahonline.org  Thu Mar 29 06:19:41 2012
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Subject: Re: [rtcweb] Proposal for H.263 baseline codec
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On 03/29/2012 08:15 AM, Dean Willis wrote:
> In today's meeting I made a plea for a mandatory baseline codec that
> is approachable to the small developer without the cost and IPR risks
> of either H.264 or VP8.
>
> I believe that some sort of baseline codec is required to jump start
> the protocol. It's OK if this isn't the best possible codec. Market
> forces will assure that commercial implementations converge on a
> higher quality when the market requires it.
>
> While I proposed today that H.261 could serve as a baseline, we know
> this would be "extremely basic". In other words, it might not pass the
> "good enough to use" test required for a successful launch.
>
> Stephane and several others suggested H.263 as an alternative during
> the after-meeting chat. While it is not as IPR-safe a choice as H.261,
> it is relatively mature and the known challenges have generally
> failed. Implementations are reasonably available and should make it
> possible for smaller implementers, students, and hobbyists to play
> ball. It's not all that network efficient and is outright piggy at
> higher resolutions, but it probably works "well enough to use."
>
> So I propose we set H.263 as the baseline ( I expect a bit of
> profiling may be necessary to further qualify the baseline) and run
> with that for now. If the situation changes, we can always replace it
> before final publication.
> --
> Dean
Claiming that VP8 is riskier than a known IPR-enforced format is a
specious claim considering its already widely implemented usage across
the web on numerous sites and in hardware.  Additionally, a patent pool
was called for over a year ago, and nothing has been made public about this.

No technology can be completely free from the specter of patent trolls
and spurious software patents, so these arguments that VP8 is not
suitable based on this could be applied to anyone.  The fact of the
matter is that the only known patents on VP8 are owned now by Google and
they have been licensed in a way compatible with all appropriate uses of
the technology, including WebRTC.

While I appreciate the sentiment of this suggestion, I am of the opinion
that choosing a known-restricted format over VP8 will be
counter-productive to the adoption of the WebRTC standard and will, in
fact, restrict possible implementations.

From martin.thomson@gmail.com  Thu Mar 29 06:30:08 2012
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On 29 March 2012 14:46, Bernard Aboba <bernard_aboba@hotmail.com> wrote:
>=C2=A0For example, +14255551212@example.com is equivalent to +14255551212@=
eviltwin.example.net

Actually, if third party assertions were prohibited, these would be
considered different, though I suspect that a user would have some
difficulty making the distinction.

But I take your point.

From stewe@stewe.org  Thu Mar 29 06:48:03 2012
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From: Stephan Wenger <stewe@stewe.org>
To: Piers O'Hanlon <p.ohanlon@gmail.com>, "Markus.Isomaki@nokia.com" <Markus.Isomaki@nokia.com>
Thread-Topic: [rtcweb] Proposal for H.263 baseline codec
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Date: Thu, 29 Mar 2012 13:47:54 +0000
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Subject: Re: [rtcweb] Proposal for H.263 baseline codec
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--_000_CB9A2910852BCstewesteweorg_
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Hi,

A few random data points about H.263.

I write this as someone who sat through the meetings which defined H.263+ a=
nd H.263++ and with a vague recollection who proposed what.  I'm leaning my=
self fairly far out of the window here, so let me emphasize that this is no=
t a formal risk analysis, no legal advice, and that it is instead today's s=
napshot of what I would think is a sensible position for a risk-averse impl=
ementer, which may be very different tomorrow.  As usual, you and your lega=
l folks will have to perform your own risk analysis.  For those unfamiliar =
with the ITU tool set, the ITU patent database can be found here http://www=
.itu.int/ipr/IPRSearch.aspx?iprtype=3DPS and can be searched for Recommenda=
tion H.263.  The content of this database probably needs to be interpreted =
in the context of the then valid policies, disclosure and declaration requi=
rements, and practices in the ITU, which were IMO and at the time, consider=
ably less strict than the disclosure requirements observed in the IETF toda=
y.

H.263 is not "profiled" in the sense the MPEG video codecs are.  There is a=
n Annex X defining profiles and levels, but this Annex was added several ye=
ars after the first revision of H.263, almost as an afterthought.  The prof=
iles defined therein are not necessarily the mode combinations that are (or=
 have been) in practical use.

Compared to modern video codecs, H.263 is trivial to implement.  I would su=
ggest that the availability or non-availability of certain modes and option=
s in today's software base (be it open source or proprietary) should not th=
e key driver in profiling H.263.  Compared to the implementation effort of =
the rest of the rtcweb stack, an H.263 implementation is cheap to have.

The reference implementation for H.263+ was maintained by the University of=
 British Columbia and later by UB Video, and access to it for the public wa=
s removed around 1999.  UB Video also sold a close-to-real-time (on the mac=
hines of the time) H.263+ encoder and decoder library.  On today's machines=
, I would expect this software, after some optimization reflecting "modern"=
 instructions sets, to handle 720p encoding and decoding on most machines. =
 UB Video was acquired by Scientific Atlanta, which in turn was acquired by=
 Cisco.  AFAIK, today, Cisco owns the rights to all this software.  I could=
 likely establish contacts to those people inside Cisco who may still have =
a copy of the software, if I were approached by Cisco folks not so fortunat=
e.

Personally, I would not expect an IPR risk to go up significantly (compared=
 to H.263 baseline 1996) when using the PLUSPTYPE header, which would (amon=
g other things) allow for flexible picture sizes.  The same is IMHO true fo=
r some of the optional modes of H.263+ that are "bug fixes" to the 1996 ver=
sion of the spec, for example Annexes I (advanced intra), J (deblocking fil=
ter), K (slices), and S (VLC bug fix) , as well as 1996 Annex D (unrestrict=
ed motion).  Annex I helps for Intra pictures, J is good for subjective qua=
lity and not such a computational complexity bear as Annex F (overlapped bl=
ock motion compensation), you want slices for MTU size matching if your pic=
ture size goes beyond CIF, and S is a bug fix.

The majority of the above coding tools do not require tricky encoder mode d=
ecision steps; the are simply alternate or additional syntax for baseline t=
ools and functionalities.

Regards,
Stephan



From: Piers O'Hanlon <p.ohanlon@gmail.com<mailto:p.ohanlon@gmail.com>>
Date: Thu, 29 Mar 2012 14:48:35 +0200
To: <Markus.Isomaki@nokia.com<mailto:Markus.Isomaki@nokia.com>>
Cc: <rtcweb@ietf.org<mailto:rtcweb@ietf.org>>, <dean.willis@softarmor.com<m=
ailto:dean.willis@softarmor.com>>
Subject: Re: [rtcweb] Proposal for H.263 baseline codec


On 29 Mar 2012, at 14:30, <Markus.Isomaki@nokia.com<mailto:Markus.Isomaki@n=
okia.com>> wrote:

Hi,

I support Dean=92s proposal to make H.263 as the mandatory-to-implement cod=
ec. I=92m not expert on profiles, but someone mentioned H.263 profile 0 as =
something suitable for our needs (smallest IPR risk).

I also believe that even making H.261 as the mandatory-to-implement would b=
e better than going for the forced pseudo-random selection between H.264 or=
 VP8.

One of the key limitations with H.261 is it's small range of available supp=
orted resolutions (QCIF, Max:CIF(352x288)) - so if there is a low IPR risk =
for an H.263(+?) configuration then it would be preferable.

The whole issue of which Profile and levels one might support for these cod=
ecs is another issue as I think they lead to differing IPR dependencies.

Piers.

Markus


From: rtcweb-bounces@ietf.org<mailto:rtcweb-bounces@ietf.org> [mailto:rtcwe=
b-bounces@ietf.org] On Behalf Of ext Dean Willis
Sent: 29 March, 2012 15:16
To: rtcweb@ietf.org<mailto:rtcweb@ietf.org>
Subject: [rtcweb] Proposal for H.263 baseline codec

In today's meeting I made a plea for a mandatory baseline codec that is app=
roachable to the small developer without the cost and IPR risks of either H=
.264 or VP8.

I believe that some sort of baseline codec is required to jump start the pr=
otocol. It's OK if this isn't the best possible codec. Market forces will a=
ssure that commercial implementations converge on a higher quality when the=
 market requires it.

While I proposed today that H.261 could serve as a baseline, we know this w=
ould be "extremely basic". In other words, it might not pass the "good enou=
gh to use" test required for a successful launch.

Stephane and several others suggested H.263 as an alternative during the af=
ter-meeting chat. While it is not as IPR-safe a choice as H.261, it is rela=
tively mature and the known challenges have generally failed. Implementatio=
ns are reasonably available and should make it possible for smaller impleme=
nters, students, and hobbyists to play ball. It's not all that network effi=
cient and is outright piggy at higher resolutions, but it probably works "w=
ell enough to use."

So I propose we set H.263 as the baseline ( I expect a bit of profiling may=
 be necessary to further qualify the baseline) and run with that for now. I=
f the situation changes, we can always replace it before final publication.
--
Dean
_______________________________________________
rtcweb mailing list
rtcweb@ietf.org<mailto:rtcweb@ietf.org>
https://www.ietf.org/mailman/listinfo/rtcweb

_______________________________________________ rtcweb mailing list rtcweb@=
ietf.org<mailto:rtcweb@ietf.org> https://www.ietf.org/mailman/listinfo/rtcw=
eb

--_000_CB9A2910852BCstewesteweorg_
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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3DWindows-1=
252">
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space; color: rgb(0, 0, 0); font-size: 14px; font-fami=
ly: Calibri, sans-serif; ">
<div>Hi,</div>
<div><br>
</div>
<div>A few random data points about H.263.</div>
<div><br>
</div>
<div>I write this as someone who sat through the meetings which defined H.2=
63&#43; and H.263&#43;&#43; and with a vague recollection who proposed what=
. &nbsp;I'm leaning myself fairly far out of the window here, so let me emp=
hasize that this is not a formal risk analysis, no
 legal advice, and that it is instead today's snapshot of what I would thin=
k is a sensible position for a risk-averse implementer, which may be very d=
ifferent tomorrow. &nbsp;As usual, you and your legal folks will have to pe=
rform your own risk analysis. &nbsp;For those
 unfamiliar with the ITU tool set, the ITU patent database can be found her=
e&nbsp;<a href=3D"http://www.itu.int/ipr/IPRSearch.aspx?iprtype=3DPS">http:=
//www.itu.int/ipr/IPRSearch.aspx?iprtype=3DPS</a>&nbsp;and can be searched =
for Recommendation H.263. &nbsp;The content of this database
 probably needs to be interpreted in the context of the then valid policies=
, disclosure and declaration requirements, and practices in the ITU, which =
were IMO and at the time, considerably less strict than the disclosure requ=
irements observed in the IETF today.</div>
<div><br>
</div>
<div>H.263 is not &quot;profiled&quot; in the sense the MPEG video codecs a=
re. &nbsp;There is an Annex X defining profiles and levels, but this Annex =
was added several years after the first revision of H.263, almost as an aft=
erthought. &nbsp;The profiles defined therein are not
 necessarily the mode combinations that are (or have been) in practical use=
.</div>
<div><br>
</div>
<div>Compared to modern video codecs, H.263 is trivial to implement. &nbsp;=
I would suggest that the availability or non-availability of certain modes =
and options in today's software base (be it open source or proprietary) sho=
uld not the key driver in profiling H.263.
 &nbsp;Compared to the implementation effort of the rest of the rtcweb stac=
k, an H.263 implementation is cheap to have. &nbsp;</div>
<div><br>
</div>
<div>The reference implementation for H.263&#43; was maintained by the Univ=
ersity of British Columbia and later by UB Video, and access to it for the =
public was removed around 1999. &nbsp;UB Video also sold a close-to-real-ti=
me (on the machines of the time) H.263&#43; encoder
 and decoder library. &nbsp;On today's machines, I would expect this softwa=
re, after some optimization reflecting &quot;modern&quot; instructions sets=
, to handle 720p encoding and decoding on most machines. &nbsp;UB Video was=
 acquired by Scientific Atlanta, which in turn was acquired
 by Cisco. &nbsp;AFAIK, today, Cisco owns the rights to all this software. =
&nbsp;I could likely establish contacts to those people inside Cisco who ma=
y still have a copy of the software, if I were approached by Cisco folks no=
t so fortunate.</div>
<div><br>
</div>
<div>Personally, I would not expect an IPR risk to go up significantly (com=
pared to H.263 baseline 1996) when using the PLUSPTYPE header, which would =
(among other things) allow for flexible picture sizes. &nbsp;The same is IM=
HO true for some of the optional modes
 of H.263&#43; that are &quot;bug fixes&quot; to the 1996 version of the sp=
ec, for example Annexes I (advanced intra), J (deblocking filter), K (slice=
s), and S (VLC bug fix) , as well as 1996 Annex D (unrestricted motion). &n=
bsp;Annex I helps for Intra pictures, J is good for
 subjective quality and not such a computational complexity bear as Annex F=
 (overlapped block motion compensation), you want slices for MTU size match=
ing if your picture size goes beyond CIF, and S is a bug fix.&nbsp;&nbsp;</=
div>
<div><br>
</div>
<div>The majority of the above coding tools do not require tricky encoder m=
ode decision steps; the are simply alternate or additional syntax for basel=
ine tools and functionalities.</div>
<div><br>
</div>
<div>Regards,</div>
<div>Stephan</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<div style=3D"font-family:Calibri; font-size:11pt; text-align:left; color:b=
lack; BORDER-BOTTOM: medium none; BORDER-LEFT: medium none; PADDING-BOTTOM:=
 0in; PADDING-LEFT: 0in; PADDING-RIGHT: 0in; BORDER-TOP: #b5c4df 1pt solid;=
 BORDER-RIGHT: medium none; PADDING-TOP: 3pt">
<span style=3D"font-weight:bold">From: </span>Piers O'Hanlon &lt;<a href=3D=
"mailto:p.ohanlon@gmail.com">p.ohanlon@gmail.com</a>&gt;<br>
<span style=3D"font-weight:bold">Date: </span>Thu, 29 Mar 2012 14:48:35 &#4=
3;0200<br>
<span style=3D"font-weight:bold">To: </span>&lt;<a href=3D"mailto:Markus.Is=
omaki@nokia.com">Markus.Isomaki@nokia.com</a>&gt;<br>
<span style=3D"font-weight:bold">Cc: </span>&lt;<a href=3D"mailto:rtcweb@ie=
tf.org">rtcweb@ietf.org</a>&gt;, &lt;<a href=3D"mailto:dean.willis@softarmo=
r.com">dean.willis@softarmor.com</a>&gt;<br>
<span style=3D"font-weight:bold">Subject: </span>Re: [rtcweb] Proposal for =
H.263 baseline codec<br>
</div>
<div><br>
</div>
<div>
<div style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line=
-break: after-white-space; ">
<br>
<div>
<div>On 29 Mar 2012, at 14:30, &lt;<a href=3D"mailto:Markus.Isomaki@nokia.c=
om">Markus.Isomaki@nokia.com</a>&gt; wrote:</div>
<br class=3D"Apple-interchange-newline">
<blockquote type=3D"cite"><span class=3D"Apple-style-span" style=3D"border-=
collapse: separate; font-family: Helvetica; font-style: normal; font-varian=
t: normal; font-weight: normal; letter-spacing: normal; line-height: normal=
; orphans: 2; text-align: -webkit-auto; text-indent: 0px; text-transform: n=
one; white-space: normal; widows: 2; word-spacing: 0px; -webkit-border-hori=
zontal-spacing: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-dec=
orations-in-effect: none; -webkit-text-size-adjust: auto; -webkit-text-stro=
ke-width: 0px; font-size: medium; ">
<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div class=3D"WordSection1" style=3D"page: WordSection1; ">
<div style=3D"margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; margin-=
bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; "=
>
<span style=3D"font-size: 11pt; color: rgb(31, 73, 125); font-family: Calib=
ri, sans-serif; ">Hi,<o:p></o:p></span></div>
<div style=3D"margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; margin-=
bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; "=
>
<span style=3D"font-size: 11pt; color: rgb(31, 73, 125); font-family: Calib=
ri, sans-serif; "><o:p>&nbsp;</o:p></span></div>
<div style=3D"margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; margin-=
bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; "=
>
<span style=3D"font-size: 11pt; color: rgb(31, 73, 125); font-family: Calib=
ri, sans-serif; ">I support Dean=92s proposal to make H.263 as the mandator=
y-to-implement codec. I=92m not expert on profiles, but someone mentioned H=
.263 profile 0 as something suitable for
 our needs (smallest IPR risk).<o:p></o:p></span></div>
<div style=3D"margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; margin-=
bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; "=
>
<span style=3D"font-size: 11pt; color: rgb(31, 73, 125); font-family: Calib=
ri, sans-serif; "><o:p>&nbsp;</o:p></span></div>
<div style=3D"margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; margin-=
bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; "=
>
<span style=3D"font-size: 11pt; color: rgb(31, 73, 125); font-family: Calib=
ri, sans-serif; ">I also believe that even making H.261 as the mandatory-to=
-implement would be better than going for the forced pseudo-random selectio=
n between H.264 or VP8.<o:p></o:p></span></div>
<div style=3D"margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; margin-=
bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; "=
>
<span style=3D"font-size: 11pt; color: rgb(31, 73, 125); font-family: Calib=
ri, sans-serif; "><o:p>&nbsp;</o:p></span></div>
</div>
</div>
</span></blockquote>
One of the key limitations with H.261 is it's small range of available supp=
orted resolutions (QCIF, Max:CIF(352x288))&nbsp;- so if there is a low IPR =
risk for an H.263(&#43;?) configuration then it would be preferable.&nbsp;<=
/div>
<div><br>
</div>
<div>The whole issue of which Profile and levels one might support for thes=
e codecs is another issue as I think they lead to differing IPR dependencie=
s.</div>
<div><br>
</div>
<div>Piers.</div>
<div><br>
<blockquote type=3D"cite"><span class=3D"Apple-style-span" style=3D"border-=
collapse: separate; font-family: Helvetica; font-style: normal; font-varian=
t: normal; font-weight: normal; letter-spacing: normal; line-height: normal=
; orphans: 2; text-align: -webkit-auto; text-indent: 0px; text-transform: n=
one; white-space: normal; widows: 2; word-spacing: 0px; -webkit-border-hori=
zontal-spacing: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-dec=
orations-in-effect: none; -webkit-text-size-adjust: auto; -webkit-text-stro=
ke-width: 0px; font-size: medium; ">
<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div class=3D"WordSection1" style=3D"page: WordSection1; ">
<div style=3D"margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; margin-=
bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; "=
>
<span style=3D"font-size: 11pt; color: rgb(31, 73, 125); font-family: Calib=
ri, sans-serif; ">Markus<o:p></o:p></span></div>
<div style=3D"margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; margin-=
bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; "=
>
<span style=3D"font-size: 11pt; color: rgb(31, 73, 125); font-family: Calib=
ri, sans-serif; "><o:p>&nbsp;</o:p></span></div>
<div style=3D"margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; margin-=
bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; "=
>
<span style=3D"font-size: 11pt; color: rgb(31, 73, 125); font-family: Calib=
ri, sans-serif; "><o:p>&nbsp;</o:p></span></div>
<div style=3D"border-top-style: none; border-right-style: none; border-bott=
om-style: none; border-width: initial; border-color: initial; border-left-s=
tyle: solid; border-left-color: blue; border-left-width: 1.5pt; padding-top=
: 0cm; padding-right: 0cm; padding-bottom: 0cm; padding-left: 4pt; ">
<div>
<div style=3D"border-right-style: none; border-bottom-style: none; border-l=
eft-style: none; border-width: initial; border-color: initial; border-top-s=
tyle: solid; border-top-color: rgb(181, 196, 223); border-top-width: 1pt; p=
adding-top: 3pt; padding-right: 0cm; padding-bottom: 0cm; padding-left: 0cm=
; ">
<div style=3D"margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; margin-=
bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; "=
>
<b><span style=3D"font-size: 10pt; font-family: Tahoma, sans-serif; ">From:=
</span></b><span style=3D"font-size: 10pt; font-family: Tahoma, sans-serif;=
 "><span class=3D"Apple-converted-space">&nbsp;</span><a href=3D"mailto:rtc=
web-bounces@ietf.org">rtcweb-bounces@ietf.org</a>
 [<a href=3D"mailto:rtcweb-bounces@ietf.org">mailto:rtcweb-bounces@ietf.org=
</a>]<span class=3D"Apple-converted-space">&nbsp;</span><b>On Behalf Of<spa=
n class=3D"Apple-converted-space">&nbsp;</span></b>ext Dean Willis<br>
<b>Sent:</b><span class=3D"Apple-converted-space">&nbsp;</span>29 March, 20=
12 15:16<br>
<b>To:</b><span class=3D"Apple-converted-space">&nbsp;</span><a href=3D"mai=
lto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<b>Subject:</b><span class=3D"Apple-converted-space">&nbsp;</span>[rtcweb] =
Proposal for H.263 baseline codec<o:p></o:p></span></div>
</div>
</div>
<div style=3D"margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; margin-=
bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; "=
>
<o:p>&nbsp;</o:p></div>
<div style=3D"margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; margin-=
bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; "=
>
In today's meeting I made a plea for a mandatory baseline codec that is app=
roachable to the small developer without the cost and IPR risks of either H=
.264 or VP8.<o:p></o:p></div>
<div>
<div style=3D"margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; margin-=
bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; "=
>
<o:p>&nbsp;</o:p></div>
</div>
<div>
<div style=3D"margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; margin-=
bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; "=
>
I believe that some sort of baseline codec is required to jump start the pr=
otocol. It's OK if this isn't the best possible codec. Market forces will a=
ssure that commercial implementations converge on a higher quality when the=
 market requires it.<o:p></o:p></div>
</div>
<div>
<div style=3D"margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; margin-=
bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; "=
>
<o:p>&nbsp;</o:p></div>
</div>
<div>
<div style=3D"margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; margin-=
bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; "=
>
While I proposed today that H.261 could serve as a baseline, we know this w=
ould be &quot;extremely basic&quot;. In other words, it might not pass the =
&quot;good enough to use&quot; test required for a successful launch.<o:p><=
/o:p></div>
</div>
<div>
<div style=3D"margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; margin-=
bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; "=
>
<o:p>&nbsp;</o:p></div>
</div>
<div>
<div style=3D"margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; margin-=
bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; "=
>
Stephane and several others suggested H.263 as an alternative during the af=
ter-meeting chat. While it is not as IPR-safe a choice as H.261, it is rela=
tively mature and the known challenges have generally failed. Implementatio=
ns are reasonably available and
 should make it possible for smaller implementers, students, and hobbyists =
to play ball. It's not all that network efficient and is outright piggy at =
higher resolutions, but it probably works &quot;well enough to use.&quot;<o=
:p></o:p></div>
</div>
<div>
<div style=3D"margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; margin-=
bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; "=
>
<o:p>&nbsp;</o:p></div>
</div>
<div>
<div style=3D"margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; margin-=
bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; "=
>
So I propose we set H.263 as the baseline ( I expect a bit of profiling may=
 be necessary to further qualify the baseline) and run with that for now. I=
f the situation changes, we can always replace it before final publication.=
<o:p></o:p></div>
</div>
<div>
<div style=3D"margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; margin-=
bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; "=
>
--<o:p></o:p></div>
</div>
<div>
<div style=3D"margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; margin-=
bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; "=
>
Dean<o:p></o:p></div>
</div>
</div>
</div>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.o=
rg/mailman/listinfo/rtcweb</a></div>
</span></blockquote>
</div>
<br>
</div>
</div>
_______________________________________________ rtcweb mailing list <a href=
=3D"mailto:rtcweb@ietf.org">
rtcweb@ietf.org</a> <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb=
">https://www.ietf.org/mailman/listinfo/rtcweb</a>
</span>
</body>
</html>

--_000_CB9A2910852BCstewesteweorg_--

From bernard_aboba@hotmail.com  Thu Mar 29 06:54:05 2012
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To: Oscar Ohlsson <oscar.ohlsson@ericsson.com>
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Subject: Re: [rtcweb] Support of SDES in WebRTC
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Oscar said:

> - SDES can be turned on by a manipulated SDP offer/answer provided the ent=
ire webapp was retrieved over HTTPS

[BA[ Unless SDES is indicated as supported in the SDP blob provided by creat=
eAnswer(), "manipulating" the SDP offer to add it probably isn't a good idea=
.=20

That's one reason why we need understand the precise semantics of createAnsw=
er().  To (mis)quote Christer:  does createAnswer() indicate a (subset) of w=
hat the browser can do, what it would prefer to do, or all that it is capabl=
e of doing?=20=

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From: "Dan Wing" <dwing@cisco.com>
To: "'Hadriel Kaplan'" <HKaplan@acmepacket.com>
References: <4F72D6B3.40803@bbn.com> <4F72E453.7070204@alvestrand.no> <4F72EB53.5000409@bbn.com> <0bf301cd0d04$22d53200$687f9600$@com> <00052A1F-CE65-4A53-9B7D-261E1CC75426@acmepacket.com>
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Subject: Re: [rtcweb] SRTP and "marketing"
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> -----Original Message-----
> From: Hadriel Kaplan [mailto:HKaplan@acmepacket.com]
> Sent: Wednesday, March 28, 2012 11:59 PM
> To: Dan Wing
> Cc: Richard L. Barnes; Harald Alvestrand; <rtcweb@ietf.org>
> Subject: Re: [rtcweb] SRTP and "marketing"
> 
> 
> On Mar 28, 2012, at 6:59 PM, Dan Wing wrote:
> 
> > We do need a foundation upon which an authentication/identity
> > infrastructure can be built.  We know we need one.
> > That foundation is DTLS-SRTP, and not Security Descriptions.
> 
> Now you're starting to sound like a marketing guy.  ;)
> What's next: "we'll build more synergy and have a unified platform with
> DTLS-SRTP"?
> 
> But more seriously, I don't understand this "foundation" argument.
> We're going to have DTLS-SRTP.  No one's suggesting we don't have DTLS-
> SRTP.  All Browsers MUST implement DTLS-SRTP.  We'll have it for
> Browser-to-Browser, and for Browser-to-Gateway if the Gateway supports
> it.  We'll have the foundation.
> 
> Requiring it for Gateways would make sense if it offered some real
> advantage, or didn't have any disadvantages.  There don't appear to be
> real advantages, while we know of disadvantages.  And gateways have no
> real means of offering an end-to-end identity.  Why would you want to
> build a foundation on air?

Even without any sort of identity at all, DTLS-SRTP provides superior
media security compared with Security Descriptions -- even if one (or 
both) endpoints are gateways to RTP or are gateways to SDESC-SRTP.

-d



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--20cf300515066c0fdd04bc6225b8
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On Thu, Mar 29, 2012 at 1:56 AM, Roman Shpount <roman@telurix.com> wrote:

>
> On Wed, Mar 28, 2012 at 5:08 PM, Basil Mohamed Gohar <
> abu_hurayrah@hidayahonline.org> wrote:
>
>> The scope of WebRTC is broad enough to consider that we need to think
>> about what's best going forward with regards to its implementation.
>> Security by default is one of the best practices in general, the support
>> from the browser community and others that are behind it will definitely
>> ensure that adoption is widespread enough to make it easy enough to
>> integrate into existing systems, as free software solutions will become
>> available shortly after the standard emerges.
>>
>>
> I would actually challenge the assumption that free software will emerge.
> SRTP with SDES was released 8 years ago. The only open source library that
> supported it until recently was libsrtp, which is a broken unusable peace
> of code spaghetti (GPF crashes, two different replay DBs which do the same
> thing, random number generator that stops working after generating few
> thousand keys) that supports 2/3 of the protocol (no F8 support). Majority
> of open source (and a large number of closed source) clients use this
> library, but do not see any of those issues since very few real SRTP calls
> are actually placed. DTLS-SRTP is two years old. Support for it was just
> added to GNU TLS (unusable by many due to GNU license) and to OpenSSL
> (using above mentioned libsrtp). There is little or no experience with
> DTLS-SRTP interop. All of these things are not trivial.
>
>
This is FUD. Google+ Hangouts uses libsrtp for all of its calls, and over
the billions of minutes of call time to date, we haven't seen any crash
bugs that could be blamed on libsrtp. And we track this stuff pretty
closely.



> My point is, if we require support for encryption implementation in the
> standard and if 4 or 5 browser vendors will cooperate, we will get vast
> majority of WebRTC clients security enabled. If everything surrounding
> WebRTC will not use encryption, it is not a big loss since, in most cases,
> it is not doing now anyway.
>

--20cf300515066c0fdd04bc6225b8
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<br><br><div class=3D"gmail_quote">On Thu, Mar 29, 2012 at 1:56 AM, Roman S=
hpount <span dir=3D"ltr">&lt;<a href=3D"mailto:roman@telurix.com">roman@tel=
urix.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=
=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">

<div class=3D"im"><br clear=3D"all"><div class=3D"gmail_quote">On Wed, Mar =
28, 2012 at 5:08 PM, Basil Mohamed Gohar <span dir=3D"ltr">&lt;<a href=3D"m=
ailto:abu_hurayrah@hidayahonline.org" target=3D"_blank">abu_hurayrah@hidaya=
honline.org</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
The scope of WebRTC is broad enough to consider that we need to think<br>
about what&#39;s best going forward with regards to its implementation.<br>
Security by default is one of the best practices in general, the support<br=
>
from the browser community and others that are behind it will definitely<br=
>
ensure that adoption is widespread enough to make it easy enough to<br>
integrate into existing systems, as free software solutions will become<br>
available shortly after the standard emerges.<br>
<div><div></div><br></div></blockquote></div><br></div>I would actually cha=
llenge the assumption that free software will emerge. SRTP with SDES was re=
leased 8 years ago. The only open source library that supported it until re=
cently was libsrtp, which is a broken unusable peace of code spaghetti (GPF=
 crashes, two different replay DBs which do the same thing, random number g=
enerator that stops working after generating few thousand keys) that suppor=
ts 2/3 of the protocol (no F8 support). Majority of open source (and a larg=
e number of closed source) clients use this library, but do not see any of =
those issues since very few real SRTP calls are actually placed. DTLS-SRTP =
is two years old. Support for it was just added to GNU TLS (unusable by man=
y due to GNU license) and to OpenSSL (using above mentioned libsrtp). There=
 is little or no experience with DTLS-SRTP interop. All of these things are=
 not trivial.<br>


<br></blockquote><div><br></div><div>This is FUD. Google+ Hangouts uses lib=
srtp for all of its calls, and over the billions of minutes of call time to=
 date, we haven&#39;t seen any crash bugs that could be blamed on libsrtp. =
And we track this stuff pretty closely.</div>

<div><br></div><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"=
margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">My point is,=
 if we require support for encryption implementation in the standard and if=
 4 or 5 browser vendors will cooperate, we will get vast majority of WebRTC=
 clients security enabled. If everything surrounding WebRTC will not use en=
cryption, it is not a big loss since, in most cases, it is not doing now an=
yway.<br>

</blockquote></div>

--20cf300515066c0fdd04bc6225b8--

From bernard_aboba@hotmail.com  Thu Mar 29 07:02:51 2012
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While they would be perceived as different with first party IdP, a signed as=
sertion is still meaningless without some proof that the E.164 number is act=
ually owned by the domain making the assertion.=20

On Mar 29, 2012, at 15:30, "Martin Thomson" <martin.thomson@gmail.com> wrote=
:

> On 29 March 2012 14:46, Bernard Aboba <bernard_aboba@hotmail.com> wrote:
>>  For example, +14255551212@example.com is equivalent to +14255551212@evil=
twin.example.net
>=20
> Actually, if third party assertions were prohibited, these would be
> considered different, though I suspect that a user would have some
> difficulty making the distinction.
>=20
> But I take your point.

From xiphmont@gmail.com  Thu Mar 29 07:09:34 2012
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Subject: [rtcweb] Proposal for Theora baseline codec
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If we're suggesting ten+ year-old codecs with low patent risk, let's
choose an obviously higher performance example.  Dare I say it...
Theora?

I think we've gotten off track.

Twelve years ago, oblique threats were made against the nascent Vorbis
by Thomson and the whole world decided it was a patent risk.  It never
was.  Fool me once, shame on you.

In the mid 2000s, people started picking on Theora the same way, and a
few years ago Steve Jobs [and Larry Horn, et al] informed the world he
was coming after us. Nothing came of it.  Fool me twice, shame on me.

Now we have the same substance-less muttering about VP8, suggesting
that it's better to use leftover crumbs from 20 years ago.

Fool me three times, I should begin to wonder if I have the requisite
capacity to be engaging in technical endeavors.

Monty
Xiph.Org

From andrew.hutton@siemens-enterprise.com  Thu Mar 29 07:10:29 2012
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From: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>
To: Cullen Jennings <fluffy@iii.ca>, Magnus Westerlund <magnus.westerlund@ericsson.com>
Date: Thu, 29 Mar 2012 16:10:25 +0200
Thread-Topic: [rtcweb] Poll for dates for Interim Meeting before IETF 84
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+1=20

The location has a big relationship to the date and even day of the week I =
would choose.

Andy

> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Cullen Jennings
> Sent: 29 March 2012 14:54
> To: Magnus Westerlund
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Poll for dates for Interim Meeting before IETF 84
>=20
>=20
> Could we have the poll choices to indicate the location of the meeting?
>=20
> The problem is, and I suspect that this is a problem for most people,
> depending on the location, my travel time dramatically changes the
> number of days I need to have clear on my calendar.
>=20
>=20
>=20
> On Mar 29, 2012, at 9:09 , Magnus Westerlund wrote:
>=20
> > WG,
> >
> > We have started a doodle poll for an Interim meeting before IETF 84
> in
> > Vancouver. Locations considered so far are Stockholm, Boston Area or
> > Silicon Valley. The meeting is planned to be a 2 day meeting. Please
> > indicate which of the suggest times that works for you.
> >
> >
> > http://doodle.com/nm3pp69znr3286cy
> >
> >
> > Cheers
> >
> > Magnus Westerlund
> > WG chair
> >
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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From: Stephan Wenger <stewe@stewe.org>
To: Basil Mohamed Gohar <abu_hurayrah@hidayahonline.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Proposal for H.263 baseline codec
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Please see inline.
Stephan

On 3.29.2012 15:19 , "Basil Mohamed Gohar"
<abu_hurayrah@hidayahonline.org> wrote:

>On 03/29/2012 08:15 AM, Dean Willis wrote:
>>[=8A]
>>
>> So I propose we set H.263 as the baseline ( I expect a bit of
>> profiling may be necessary to further qualify the baseline) and run
>> with that for now. If the situation changes, we can always replace it
>> before final publication.
>> --
>> Dean
>Claiming that VP8 is riskier than a known IPR-enforced format is a
>specious claim considering its already widely implemented usage across
>the web on numerous sites and in hardware.  Additionally, a patent pool
>was called for over a year ago, and nothing has been made public about
>this.

These things take time.  The formation of the MPEG-2 patent pool took 4+
years, the formation of the H.264 pool almost as long.  VP8 may be a
particularly hard-to-deal-with technology, because it is not a formal
standard, which may provide rightholders options to enforce their patents
in ways not available if they would have been involved in a standards
setting process (see below), which can make the negotiation of the pool
more complex (can't re-use previous pool agreements as templates, may have
to check for antitrust compliance, and so on).

The most commonly cited timeline for a widely in use technology to be
"save" from a patent viewpoint, based on equitable defenses such as laches
(in the US) is six years.  In some countries of significant size, this
time is longer, and in others, equitable defenses do not exist.  (Very
briefly, and perhaps incorrectly put, those equitable defenses allow a
defendant to argue successfully that a patent cannot be enforced as the
right holder knew that the patent claim was likely being infringed, and
did not enforce the patent.).

>No technology can be completely free from the specter of patent trolls
>and spurious software patents, so these arguments that VP8 is not
>suitable based on this could be applied to anyone.  The fact of the
>matter is that the only known patents on VP8 are owned now by Google and
>they have been licensed in a way compatible with all appropriate uses of
>the technology, including WebRTC.

The second part of your sentence may or may not be true, depending on your
relationship with google, your willingness to use the webm implementation
in unchanged form, and other factors.  Please see the webm license
conditions, which AFAIK can be found here:
http://www.webmproject.org/license/additional/

In addition, as I pointed out in the meeting, the use of a video codec
created by a body such as MPEG or ITU-T SG16 has the advantage of that the
patents of all participating players are available at least under
Reasonable and Non Discriminatory (RAND) terms.  This may sound like a Bad
Thing if you operate under a business model that prevents you to pay
anything for patent licenses, but it is surely a Good Thing if you are
willing to dish out a moderate amount of money for a license.  RAND
recently has gotten teeth, not so much in terms of the monetary
compensation aspect, but in terms of difficulty (if not unavailability) to
obtain injunctive relive, among others.  H.26x and the MPEG standards
benefit from RAND commitments, VP8, AFAIK, does not.

Stephan

>
>While I appreciate the sentiment of this suggestion, I am of the opinion
>that choosing a known-restricted format over VP8 will be
>counter-productive to the adoption of the WebRTC standard and will, in
>fact, restrict possible implementations.
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb
>



From basilgohar@librevideo.org  Thu Mar 29 07:17:48 2012
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Date: Thu, 29 Mar 2012 10:17:40 -0400
From: Basil Mohamed Gohar <basilgohar@librevideo.org>
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Subject: Re: [rtcweb] Proposal for Theora baseline codec
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On 03/29/2012 10:09 AM, Monty Montgomery wrote:
> If we're suggesting ten+ year-old codecs with low patent risk, let's
> choose an obviously higher performance example.  Dare I say it...
> Theora?
>
> I think we've gotten off track.
>
> Twelve years ago, oblique threats were made against the nascent Vorbis
> by Thomson and the whole world decided it was a patent risk.  It never
> was.  Fool me once, shame on you.
>
> In the mid 2000s, people started picking on Theora the same way, and a
> few years ago Steve Jobs [and Larry Horn, et al] informed the world he
> was coming after us. Nothing came of it.  Fool me twice, shame on me.
>
> Now we have the same substance-less muttering about VP8, suggesting
> that it's better to use leftover crumbs from 20 years ago.
>
> Fool me three times, I should begin to wonder if I have the requisite
> capacity to be engaging in technical endeavors.
>
> Monty
> Xiph.Org
+1

VP8 is the only option that makes any sense for WebRTC going forward. 
It is high performant, it has optimizations for realtime usage (in the
reference implementation), it is already implemented in a wide variety
of software (Skype, browsers [Mozilla Firefox, Google Chrome, Opera]),
and it is also implemented in hardware (as well as supported by the
Android OS).

No one can say something will never be the target of a lawsuit.  But
there *are* legal protections against organizations that assert patents
against a standard after its adoption when they had the ability to make
the standards body aware of it ahead of time (estoppel).

From roman@telurix.com  Thu Mar 29 07:20:49 2012
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From: Roman Shpount <roman@telurix.com>
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--047d7b2edf030a2c5704bc626dec
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On Thu, Mar 29, 2012 at 8:14 AM, I=F1aki Baz Castillo <ibc@aliax.net> wrote=
:

> Roman, you still don't want to see my simple example. Could you please
> reply to the following two questions rather than avoiding them?:
>
> 1) Why is better to use plain RTP in an open WiFi network (airport)
> rather than using SDES-SRTP?
>

I did not say that use of plain RTP is better in an open WiFi. what  am
saying that if I am communicating with a large number of applications it is
no worse. For me, anything that can cause protocol not to work or to be
used for some sort of an attack goes under MUST. Anything that is nice to
have should be under SHOULD. From my point of view security is a SHOULD.


> 2) Why is plain RTP better than SDES-SRTP in any other case?
>

In case of an enterprise network, their security concerns are not about
user privacy (it is not end user equipment in a first place) but adherence
to corporate standards or ability to monitor. In such cases SRTP is a
liability, not an asset. In Flash, in HTTP, in email, encryption is
optional. If you have anything where encryption is mandatory while
communicating with an external party, it is not acceptable in a lot of
enterprise or government environments. My goal is to have WebRTC
functionality available in as many places as possible, even if it means
giving up encryption in some cases.

Developing new applications is harder. If you are building a speech
recognition engine, you often need to add media support. Adding encrypted
media support is much harder. I want people to build new things with the
minimal effort, even just to educate themselves or to try new things. This
is what makes progress possible. The more things we pile up in front of
this, the harder it would be for people to do new things.


> > No encryption often means better security.
>
> Sorry, I don't agree. Encryption is not enough, of course, but it's
> better than nothing (and again: the open WiFi network in the airport).
>
> > Bot nets use encryption to
> > communicate, it does not mean they enable user security.
>
> 1) User identity authentication/validation and confidentiality
> (encryption).
>
> 2) Just confidentiality (encryption).
>
> 3) Nothing.
>
>
> 1 is better than 2 and 2 is better than 3.
>
>
>
> > In a lot of cases
> > ability for the user to check what exactly they are communicating and t=
o
> > whom is security. As a security conscious person I do not want encrypte=
d
> > communication from my computer or my network unless I know exactly who
> it is
> > going to. If I do not have to I would prefer my communications to be
> > unencrypted.
>
> Sure, but you are advanced user and I hope you are not talking in
> behalf of the 500 millions of Facebook users in the world.
> So you will be able to enter into your browser about://config and set
> a null crypto key for SRTP (so you get plain RTP and can debug it or
> whatever). This is confirmed to be the expectation of two browser
> vendors.
>
> And you are missing a point completely. If I am operating a corporate
network with 50,000 computers, do I need to go into about://config for all
of them? If I am put into this situation, I would just disable WebRTC
completely and call it security. This is exactly what I am trying to avoid.


>
> >> RFC 3711 was created in 2004, 8 years ago!
> >>
> >> Which "new" application you mean if it's not capable of implementing a
> >> simple specification from 2004? how "new" is it? is it really new? or
> >> is it a "SIP voicemail server" made in 2002?
> >>
> > Well let me list them: FreeSwitch, Asterisk, linphone, etc. All of thos=
e
> use
> > libsrtp. For all of them SRTP is broken. Not a single one of them
> bothered
> > to check or only recently detected problems
> > (https://issues.asterisk.org/jira/browse/ASTERISK-16665).
>
> I expect that, once SRTP becomes widely extended (thanks to WebRTC),
> developers will really want to fix their SRTP support, including
> libsrtp. But we cannot make a *new spec less secure than it can be
> just due a to a bug in a opensource library.
>

The issue is most of the developers would just look at this as a hurdle to
interop with WebRTC, instead of the enabling the security. If security
implementation is not taken seriously, you often end up with something that
can communicate but not really secure. For instance you can use a bad
algorithm to generate encryption keys and end up with keys that are easily
predictable. Result -- encrypted but unsecured traffic. Encryption should
be something that application developers can turn on or off, but unless
they want it, forcing support for it is a fulls quest.


>
> >> Sorry but I still see *no* argument at all in favour of allowing plain
> >> RTP in WebRTC. And AFAIK there is already consensus about it: plain
> >> RTP is not allowed.
> >>
> > I do not see an argument why RTP should not be allowed. I think you are
> > arguing we should require a feature (SRTP) for the sake of requiring a
> > feature.
>
> I still don't understand why everyone in an airport should be able to
> intercept my audio/video communication with a simple RTP sniffer.
>

There are enough cases where I or other users do not care if this happens,
so I think that encryption should be a SHOULD.
_____________
Roman Shpount

--047d7b2edf030a2c5704bc626dec
Content-Type: text/html; charset=ISO-8859-1
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On Thu, Mar 29, 2012 at 8:14 AM, I=F1aki Baz Castillo <span dir=3D"ltr">&lt=
;<a href=3D"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt;</span> wrote:<br><d=
iv class=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"margin:=
0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Roman, you still don&#39;t want to see my simple example. Could you please<=
br>
reply to the following two questions rather than avoiding them?:<br>
<br>
1) Why is better to use plain RTP in an open WiFi network (airport)<br>
rather than using SDES-SRTP?<br></blockquote><div><br>I did not say that us=
e of plain RTP is better in an open WiFi. what=A0 am saying that if I am co=
mmunicating with a large number of applications it is no worse. For me, any=
thing that can cause protocol not to work or to be used for some sort of an=
 attack goes under MUST. Anything that is nice to have should be under SHOU=
LD. From my point of view security is a SHOULD.<br>
</div><div>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0pt 0=
pt 0pt 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
2) Why is plain RTP better than SDES-SRTP in any other case?<br></blockquot=
e><div><br>In case of an enterprise network, their security concerns are no=
t about user privacy (it is not end user equipment in a first place) but ad=
herence to corporate standards or ability to monitor. In such cases SRTP is=
 a liability, not an asset. In Flash, in HTTP, in email, encryption is opti=
onal. If you have anything where encryption is mandatory while communicatin=
g with an external party, it is not acceptable in a lot of enterprise or go=
vernment environments. My goal is to have WebRTC functionality available in=
 as many places as possible, even if it means giving up encryption in some =
cases.<br>
<br>Developing new applications is harder. If you are building a speech rec=
ognition engine, you often need to add media support. Adding encrypted medi=
a support is much harder. I want people to build new things with the minima=
l effort, even just to educate themselves or to try new things. This is wha=
t makes progress possible. The more things we pile up in front of this, the=
 harder it would be for people to do new things.<br>
<br></div><blockquote class=3D"gmail_quote" style=3D"margin:0pt 0pt 0pt 0.8=
ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
<div class=3D"im">
<br>
&gt; No encryption often means better security.<br>
<br>
</div>Sorry, I don&#39;t agree. Encryption is not enough, of course, but it=
&#39;s<br>
better than nothing (and again: the open WiFi network in the airport).<br>
<div class=3D"im"><br>
&gt; Bot nets use encryption to<br>
&gt; communicate, it does not mean they enable user security.<br>
<br>
</div>1) User identity authentication/validation and confidentiality (encry=
ption).<br>
<br>
2) Just confidentiality (encryption).<br>
<br>
3) Nothing.<br>
<br>
<br>
1 is better than 2 and 2 is better than 3.<br>
<div class=3D"im"><br>
<br>
<br>
&gt; In a lot of cases<br>
&gt; ability for the user to check what exactly they are communicating and =
to<br>
&gt; whom is security. As a security conscious person I do not want encrypt=
ed<br>
&gt; communication from my computer or my network unless I know exactly who=
 it is<br>
&gt; going to. If I do not have to I would prefer my communications to be<b=
r>
&gt; unencrypted.<br>
<br>
</div>Sure, but you are advanced user and I hope you are not talking in<br>
behalf of the 500 millions of Facebook users in the world.<br>
So you will be able to enter into your browser about://config and set<br>
a null crypto key for SRTP (so you get plain RTP and can debug it or<br>
whatever). This is confirmed to be the expectation of two browser<br>
vendors.<br>
<div class=3D"im"><br></div></blockquote><div>And you are missing a point c=
ompletely. If I am operating a corporate network with 50,000 computers, do =
I need to go into about://config for all of them? If I am put into this sit=
uation, I would just disable WebRTC completely and call it security. This i=
s exactly what I am trying to avoid.<br>
=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0pt 0pt 0pt 0.8e=
x;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div class=3D"im=
">
<br>
&gt;&gt; RFC 3711 was created in 2004, 8 years ago!<br>
&gt;&gt;<br>
&gt;&gt; Which &quot;new&quot; application you mean if it&#39;s not capable=
 of implementing a<br>
&gt;&gt; simple specification from 2004? how &quot;new&quot; is it? is it r=
eally new? or<br>
&gt;&gt; is it a &quot;SIP voicemail server&quot; made in 2002?<br>
&gt;&gt;<br>
&gt; Well let me list them: FreeSwitch, Asterisk, linphone, etc. All of tho=
se use<br>
&gt; libsrtp. For all of them SRTP is broken. Not a single one of them both=
ered<br>
&gt; to check or only recently detected problems<br>
&gt; (<a href=3D"https://issues.asterisk.org/jira/browse/ASTERISK-16665" ta=
rget=3D"_blank">https://issues.asterisk.org/jira/browse/ASTERISK-16665</a>)=
.<br>
<br>
</div>I expect that, once SRTP becomes widely extended (thanks to WebRTC),<=
br>
developers will really want to fix their SRTP support, including<br>
libsrtp. But we cannot make a *new spec less secure than it can be<br>
just due a to a bug in a opensource library.<br></blockquote><div><br>The i=
ssue is most of the developers would just look at this as a hurdle to inter=
op with WebRTC, instead of the enabling the security. If security implement=
ation is not taken seriously, you often end up with something that can comm=
unicate but not really secure. For instance you can use a bad algorithm to =
generate encryption keys and end up with keys that are easily predictable. =
Result -- encrypted but unsecured traffic. Encryption should be something t=
hat application developers can turn on or off, but unless they want it, for=
cing support for it is a fulls quest.<br>
=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0pt 0pt 0pt 0.8e=
x;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div class=3D"im=
">
<br>
&gt;&gt; Sorry but I still see *no* argument at all in favour of allowing p=
lain<br>
&gt;&gt; RTP in WebRTC. And AFAIK there is already consensus about it: plai=
n<br>
&gt;&gt; RTP is not allowed.<br>
&gt;&gt;<br>
&gt; I do not see an argument why RTP should not be allowed. I think you ar=
e<br>
&gt; arguing we should require a feature (SRTP) for the sake of requiring a=
<br>
&gt; feature.<br>
<br>
</div>I still don&#39;t understand why everyone in an airport should be abl=
e to<br>
intercept my audio/video communication with a simple RTP sniffer.<br></bloc=
kquote><br>There are enough cases where I or other users do not care if thi=
s happens, so I think that encryption should be a SHOULD.<br>_____________<=
br>
</div>Roman Shpount<br>
<br><br>

--047d7b2edf030a2c5704bc626dec--

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On 03/29/2012 10:20 AM, Stephan Wenger wrote:
> The second part of your sentence may or may not be true, depending on your
> relationship with google, your willingness to use the webm implementation
> in unchanged form, and other factors.  Please see the webm license
> conditions, which AFAIK can be found here:
> http://www.webmproject.org/license/additional/
Correct.  I think you are referring to this part, explicitly:
> If you or your agent or exclusive licensee institute or order or agree
> to the institution of patent litigation against any entity (including
> a cross-claim or counterclaim in a lawsuit) alleging that this
> implementation of VP8 or any code incorporated within this
> implementation of VP8 constitutes direct or contributory patent
> infringement, or inducement of patent infringement, then any patent
> rights granted to you under this License for this implementation of
> VP8 shall terminate as of the date such litigation is filed.
Perhaps I assumed that that is a very reasonable part of the license. 
That is, if you are suing someone alleging a patent infringement within
VP8, you are no longer granted the license to use VP8's patented
technologies that Google owns.

From roman@telurix.com  Thu Mar 29 07:31:17 2012
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--047d7b3395a170f41304bc6293b0
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On Thu, Mar 29, 2012 at 10:00 AM, Justin Uberti <juberti@google.com> wrote:

> This is FUD. Google+ Hangouts uses libsrtp for all of its calls, and over
> the billions of minutes of call time to date, we haven't seen any crash
> bugs that could be blamed on libsrtp. And we track this stuff pretty
> closely.
>
>
> This is not a FUD. Even with 100s of millions of secured minutes we are
pushing we see new libsrtp related problems on a weekly basis. I gave you a
reference to the Asterisk bug. This bug is addressed in sourceforge, but
present in the download library (1.4.4) that is included in a lot of the
products. This bug guarantees a crash in case RTCP and even small packet
loss are present.  You can try to run crypto_get_random in a loop until it
generates an error. Simple and easy to reproduce bug. There is probably
more, but it is outside of the scope of this list. If you want to do your
users a favor -- swap this lib out from your code. You probably have a much
better crypto utilities (random, AES) in other code you use, such as
OpenSSL. The rest is trivial to re-implement and it will make the result
product faster and more secure.
_____________
Roman Shpount

--047d7b3395a170f41304bc6293b0
Content-Type: text/html; charset=ISO-8859-1
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<br clear=3D"all"><div class=3D"gmail_quote">On Thu, Mar 29, 2012 at 10:00 =
AM, Justin Uberti <span dir=3D"ltr">&lt;<a href=3D"mailto:juberti@google.co=
m">juberti@google.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_q=
uote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1e=
x">
This is FUD. Google+ Hangouts uses libsrtp for all of its calls, and over t=
he billions of minutes of call time to date, we haven&#39;t seen any crash =
bugs that could be blamed on libsrtp. And we track this stuff pretty closel=
y.<div class=3D"gmail_quote">
<div class=3D"im">

<div><br></div><br></div></div></blockquote><div>This is not a FUD. Even wi=
th 100s of millions of secured minutes we are pushing we see new libsrtp re=
lated problems on a weekly basis. I gave you a reference to the Asterisk bu=
g. This bug is addressed in sourceforge, but present in the download librar=
y (1.4.4) that is included in a lot of the products. This bug guarantees a =
crash in case RTCP and even small packet loss are present.=A0 You can try t=
o run crypto_get_random in a loop until it generates an error. Simple and e=
asy to reproduce bug. There is probably more, but it is outside of the scop=
e of this list. If you want to do your users a favor -- swap this lib out f=
rom your code. You probably have a much better crypto utilities (random, AE=
S) in other code you use, such as OpenSSL. The rest is trivial to re-implem=
ent and it will make the result product faster and more secure.<br>
</div></div>_____________<br>Roman Shpount<br>
<br>

--047d7b3395a170f41304bc6293b0--

From stewe@stewe.org  Thu Mar 29 07:33:52 2012
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From: Stephan Wenger <stewe@stewe.org>
To: Monty Montgomery <xiphmont@gmail.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Proposal for Theora baseline codec
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Hi Monty,
Couldn't it be that no one went after Vorbis and Theora because the
rightholders had no real incentive to do so?  For example, because the
products implementing Vorbis, in their vast majority, also included MP3
and/or AAC, and the rightholders would have to deal with patent exhaustion
arguments and whatnot after having gotten their money; and because theora
deployment never reached a critical mass because it was a) not good enough
technically, and b) felt to be too risky?
I would turn your argument around: the (comparatively speaking) lack of
commercial success of vorbis and theora, and the success of royalty
bearing standard codecs, suggest to me only that no one was fooled, except
perhaps the enthusiastic followers of those folks who keep pushing out one
"free" codec after another...
Stephan
 =20

On 3.29.2012 16:09 , "Monty Montgomery" <xiphmont@gmail.com> wrote:

>If we're suggesting ten+ year-old codecs with low patent risk, let's
>choose an obviously higher performance example.  Dare I say it...
>Theora?
>
>I think we've gotten off track.
>
>Twelve years ago, oblique threats were made against the nascent Vorbis
>by Thomson and the whole world decided it was a patent risk.  It never
>was.  Fool me once, shame on you.
>
>In the mid 2000s, people started picking on Theora the same way, and a
>few years ago Steve Jobs [and Larry Horn, et al] informed the world he
>was coming after us. Nothing came of it.  Fool me twice, shame on me.
>
>Now we have the same substance-less muttering about VP8, suggesting
>that it's better to use leftover crumbs from 20 years ago.
>
>Fool me three times, I should begin to wonder if I have the requisite
>capacity to be engaging in technical endeavors.
>
>Monty
>Xiph.Org
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb
>



From keith.drage@alcatel-lucent.com  Thu Mar 29 07:35:00 2012
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: Basil Mohamed Gohar <basilgohar@librevideo.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
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> VP8 is the only option that makes any sense for WebRTC going forward.
> It is high performant, it has optimizations for realtime usage (in the
> reference implementation), it is already implemented in a wide variety
> of software (Skype, browsers [Mozilla Firefox, Google Chrome, Opera]),
> and it is also implemented in hardware (as well as supported by the
> Android OS).
>

So are you saying that Adam's presentation today was incorrect?

Quoting:

"H.264:

More deployed hardware acceleration
Well known, clearly identified patent pool; royalties due for some uses
Quality, compression ratio, and complexity approximately equal to VP8

VP8:

Can be hardware accelerated, but current deployment is very low
No specific patents asserted1, no one collecting royalties
Quality, compression ratio, and complexity approximately equal to
H.264"

If so I think you need to be a little more technically specific as to why y=
ou see one better than the other.

Regards

Keith

> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
> Of Basil Mohamed Gohar
> Sent: 29 March 2012 15:18
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] Proposal for Theora baseline codec
>=20
> On 03/29/2012 10:09 AM, Monty Montgomery wrote:
> > If we're suggesting ten+ year-old codecs with low patent risk, let's
> > choose an obviously higher performance example.  Dare I say it...
> > Theora?
> >
> > I think we've gotten off track.
> >
> > Twelve years ago, oblique threats were made against the nascent Vorbis
> > by Thomson and the whole world decided it was a patent risk.  It never
> > was.  Fool me once, shame on you.
> >
> > In the mid 2000s, people started picking on Theora the same way, and a
> > few years ago Steve Jobs [and Larry Horn, et al] informed the world he
> > was coming after us. Nothing came of it.  Fool me twice, shame on me.
> >
> > Now we have the same substance-less muttering about VP8, suggesting
> > that it's better to use leftover crumbs from 20 years ago.
> >
> > Fool me three times, I should begin to wonder if I have the requisite
> > capacity to be engaging in technical endeavors.
> >
> > Monty
> > Xiph.Org
> +1
>=20
> VP8 is the only option that makes any sense for WebRTC going forward.
> It is high performant, it has optimizations for realtime usage (in the
> reference implementation), it is already implemented in a wide variety
> of software (Skype, browsers [Mozilla Firefox, Google Chrome, Opera]),
> and it is also implemented in hardware (as well as supported by the
> Android OS).
>=20
> No one can say something will never be the target of a lawsuit.  But
> there *are* legal protections against organizations that assert patents
> against a standard after its adoption when they had the ability to make
> the standards body aware of it ahead of time (estoppel).
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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From: Basil Mohamed Gohar <basilgohar@librevideo.org>
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On 03/29/2012 10:34 AM, DRAGE, Keith (Keith) wrote:
>> VP8 is the only option that makes any sense for WebRTC going forward.
>> It is high performant, it has optimizations for realtime usage (in the
>> reference implementation), it is already implemented in a wide variety
>> of software (Skype, browsers [Mozilla Firefox, Google Chrome, Opera]),
>> and it is also implemented in hardware (as well as supported by the
>> Android OS).
>>
> So are you saying that Adam's presentation today was incorrect?
>
> Quoting:
>
> "H.264:
>
> More deployed hardware acceleration
> Well known, clearly identified patent pool; royalties due for some uses
> Quality, compression ratio, and complexity approximately equal to VP8
>
> VP8:
>
> Can be hardware accelerated, but current deployment is very low
> No specific patents asserted1, no one collecting royalties
> Quality, compression ratio, and complexity approximately equal to
> H.264"
>
> If so I think you need to be a little more technically specific as to why you see one better than the other.
>
> Regards
>
> Keith
I'm not saying that.

What I am saying, and what I left out of that short list above, is that
VP8's royalty-free basis and patent grant from the only known owners of
patents on the technology, *in addition* to its technical suitability,
make it the only choice that makes any sense.

-- 
Libre Video
http://librevideo.org


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> Hi Monty,
> Couldn't it be that no one went after Vorbis and Theora because the
> rightholders had no real incentive to do so? =A0For example, because the
[...]

The opposing argument neatly boils down to 'no one ever got fired for
buying from IBM'.  That specific sentiment seems quaint now, but the
point is still relevant.

Monty
Xiph.Org

From basilgohar@librevideo.org  Thu Mar 29 07:47:26 2012
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On 03/29/2012 10:33 AM, Stephan Wenger wrote:
> Hi Monty,
> Couldn't it be that no one went after Vorbis and Theora because the
> rightholders had no real incentive to do so?  For example, because the
> products implementing Vorbis, in their vast majority, also included MP3
> and/or AAC, and the rightholders would have to deal with patent exhaustion
> arguments and whatnot after having gotten their money; and because theora
> deployment never reached a critical mass because it was a) not good enough
> technically, and b) felt to be too risky?
> I would turn your argument around: the (comparatively speaking) lack of
> commercial success of vorbis and theora, and the success of royalty
> bearing standard codecs, suggest to me only that no one was fooled, except
> perhaps the enthusiastic followers of those folks who keep pushing out one
> "free" codec after another...
> Stephan
>
Stephan,

Monty already pointed out that sabre-rattling already happened.  Had
there been any meat to the threats, then it would have been ridiculous
not to have pursued them.  Mozilla implements both Theora and Vorbis in
browser, and do not implement MPEG-related codecs, and they're a
lucrative target (they're getting around $1 billion from Google in the
coming few years).  They *do* have a commercial entity that could be the
target, as well.

The real threat was that these formats posed a risk to the other formats
monopolistic adoption, and the FUD that was spread every time these
formats were presented in a standard or appeared poised to be adopted
more widely, the sabre-rattling began again, and no real threats
materialized.

It's arguing by saying, "You never can be sure...".

-- 
Libre Video
http://librevideo.org


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From: Stephan Wenger <stewe@stewe.org>
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On 3.29.2012 16:24 , "Basil Mohamed Gohar" <basilgohar@librevideo.org>
wrote:

>On 03/29/2012 10:20 AM, Stephan Wenger wrote:
>> The second part of your sentence may or may not be true, depending on
>>your
>> relationship with google, your willingness to use the webm
>>implementation
>> in unchanged form, and other factors.  Please see the webm license
>> conditions, which AFAIK can be found here:
>> http://www.webmproject.org/license/additional/
>Correct.  I think you are referring to this part, explicitly:
>> If you or your agent or exclusive licensee institute or order or agree
>> to the institution of patent litigation against any entity (including
>> a cross-claim or counterclaim in a lawsuit) alleging that this
>> implementation of VP8 or any code incorporated within this
>> implementation of VP8 constitutes direct or contributory patent
>> infringement, or inducement of patent infringement, then any patent
>> rights granted to you under this License for this implementation of
>> VP8 shall terminate as of the date such litigation is filed.
>Perhaps I assumed that that is a very reasonable part of the license.
>That is, if you are suing someone alleging a patent infringement within
>VP8, you are no longer granted the license to use VP8's patented
>technologies that Google owns.

Yes, that's one issue.  Call it personal preference for different type of
reciprocity conditions :-)  (I could rant about it for hours, but let's
continue to pretend that this is mostly a technical mailing list)

The other issue, though (the fact that the license grant extends only to
the VP8 implementation as provided by google, and does not extent to
derivative works such as hardware implementations) should be moderately
alarming even for an open source person.  With respect to this clause, I
will note that I criticized the licensing conditions in private and in
public (IETF mike) several times, months ago, and nothing happened.
Suggests to me one of three things: (1) google is a large company and
decisions take time, or (2) google's legal is currently occupied with
other stuff, or (3) that the choice of language is intentional, and
intended to prevent forks.  Take your pick.
Stephan
 =20
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb
>



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On 03/29/2012 10:53 AM, Stephan Wenger wrote:
>
> On 3.29.2012 16:24 , "Basil Mohamed Gohar" <basilgohar@librevideo.org>
> wrote:
>
>> On 03/29/2012 10:20 AM, Stephan Wenger wrote:
>>> The second part of your sentence may or may not be true, depending on
>>> your
>>> relationship with google, your willingness to use the webm
>>> implementation
>>> in unchanged form, and other factors.  Please see the webm license
>>> conditions, which AFAIK can be found here:
>>> http://www.webmproject.org/license/additional/
>> Correct.  I think you are referring to this part, explicitly:
>>> If you or your agent or exclusive licensee institute or order or agree
>>> to the institution of patent litigation against any entity (including
>>> a cross-claim or counterclaim in a lawsuit) alleging that this
>>> implementation of VP8 or any code incorporated within this
>>> implementation of VP8 constitutes direct or contributory patent
>>> infringement, or inducement of patent infringement, then any patent
>>> rights granted to you under this License for this implementation of
>>> VP8 shall terminate as of the date such litigation is filed.
>> Perhaps I assumed that that is a very reasonable part of the license.
>> That is, if you are suing someone alleging a patent infringement within
>> VP8, you are no longer granted the license to use VP8's patented
>> technologies that Google owns.
> Yes, that's one issue.  Call it personal preference for different type of
> reciprocity conditions :-)  (I could rant about it for hours, but let's
> continue to pretend that this is mostly a technical mailing list)
>
> The other issue, though (the fact that the license grant extends only to
> the VP8 implementation as provided by google, and does not extent to
> derivative works such as hardware implementations) should be moderately
> alarming even for an open source person.  With respect to this clause, I
> will note that I criticized the licensing conditions in private and in
> public (IETF mike) several times, months ago, and nothing happened.
> Suggests to me one of three things: (1) google is a large company and
> decisions take time, or (2) google's legal is currently occupied with
> other stuff, or (3) that the choice of language is intentional, and
> intended to prevent forks.  Take your pick.
> Stephan
Stephan,

If the license wording is meant to limit the patent grant only to
software implementations, then I will raise this issue myself, because
that's a glaring hole and huge setback to the adoption of the format. 
If I find anything out about this, I'll report back here.  The fact that
there are already hardware implementations, though, seems to imply that
in reality, this is not the case.  Google has been open to changing the
licensing situation of VP8 in the past when issues like this were
brought up.

I do not, however, want this to be a point of FUD.  I would welcome
anyone from Google on this list authorized to speak about this to
clarify this ASAP.  A public statement like that should allow us to
discount this worry.

-- 
Libre Video
http://librevideo.org


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From: Stephan Wenger <stewe@stewe.org>
To: Basil Mohamed Gohar <basilgohar@librevideo.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Proposal for Theora baseline codec
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HI Basil,
Really interesting discussion.  Top-posting this time, as I have to run.
Yes, saber-rattling was present.  Is present.  It worked in the past,
plus/minus Mozilla, didn't it?  That's the whole idea behind
sabre-rattling.  It's better than sabre-swinging.  More diligently put,
warning of potential infringement is better than to punish an infringer.
Cheaper, and serves the purpose.
And thinking of a Mozilla as a juicy target (based on the numbers you
cited) seems to me a bit of a far fetch, considering a) the percentage of
multimedia codecs on a web browser's value proposition even today (let
alone 5-10 years ago), and b) the monetary damages asked in high-profile
patent lawsuits today.
Stephan

On 3.29.2012 16:47 , "Basil Mohamed Gohar" <basilgohar@librevideo.org>
wrote:

>On 03/29/2012 10:33 AM, Stephan Wenger wrote:
>> Hi Monty,
>> Couldn't it be that no one went after Vorbis and Theora because the
>> rightholders had no real incentive to do so?  For example, because the
>> products implementing Vorbis, in their vast majority, also included MP3
>> and/or AAC, and the rightholders would have to deal with patent
>>exhaustion
>> arguments and whatnot after having gotten their money; and because
>>theora
>> deployment never reached a critical mass because it was a) not good
>>enough
>> technically, and b) felt to be too risky?
>> I would turn your argument around: the (comparatively speaking) lack of
>> commercial success of vorbis and theora, and the success of royalty
>> bearing standard codecs, suggest to me only that no one was fooled,
>>except
>> perhaps the enthusiastic followers of those folks who keep pushing out
>>one
>> "free" codec after another...
>> Stephan
>>
>Stephan,
>
>Monty already pointed out that sabre-rattling already happened.  Had
>there been any meat to the threats, then it would have been ridiculous
>not to have pursued them.  Mozilla implements both Theora and Vorbis in
>browser, and do not implement MPEG-related codecs, and they're a
>lucrative target (they're getting around $1 billion from Google in the
>coming few years).  They *do* have a commercial entity that could be the
>target, as well.
>
>The real threat was that these formats posed a risk to the other formats
>monopolistic adoption, and the FUD that was spread every time these
>formats were presented in a standard or appeared poised to be adopted
>more widely, the sabre-rattling began again, and no real threats
>materialized.
>
>It's arguing by saying, "You never can be sure...".
>
>--=20
>Libre Video
>http://librevideo.org
>
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb
>



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On 03/29/2012 11:03 AM, Stephan Wenger wrote:
> HI Basil,
> Really interesting discussion.  Top-posting this time, as I have to run.
> Yes, saber-rattling was present.  Is present.  It worked in the past,
> plus/minus Mozilla, didn't it?  That's the whole idea behind
> sabre-rattling.  It's better than sabre-swinging.  More diligently put,
> warning of potential infringement is better than to punish an infringer.
> Cheaper, and serves the purpose.
> And thinking of a Mozilla as a juicy target (based on the numbers you
> cited) seems to me a bit of a far fetch, considering a) the percentage of
> multimedia codecs on a web browser's value proposition even today (let
> alone 5-10 years ago), and b) the monetary damages asked in high-profile
> patent lawsuits today.
> Stephan
I disagree.  If there was a real, defensible infringement, then engaging
in a license agreement would be the far wiser approach than to warn of
infringement.  This would be the tactic when a technology one has
patents on can be displaced by another which may-or-may-not infringe. 
Being replaced by another technology is a real threat to revenue.  Being
replaced by a technology that one can also extract licenses from, is not
a threat.  Therefore, sabre-rattling only makes sense when there's the
real threat to ones existing revenue stream, and one doubts litigation
will be in their favor.
> On 3.29.2012 16:47 , "Basil Mohamed Gohar" <basilgohar@librevideo.org>
> wrote:
>
>> On 03/29/2012 10:33 AM, Stephan Wenger wrote:
>>> Hi Monty,
>>> Couldn't it be that no one went after Vorbis and Theora because the
>>> rightholders had no real incentive to do so?  For example, because the
>>> products implementing Vorbis, in their vast majority, also included MP3
>>> and/or AAC, and the rightholders would have to deal with patent
>>> exhaustion
>>> arguments and whatnot after having gotten their money; and because
>>> theora
>>> deployment never reached a critical mass because it was a) not good
>>> enough
>>> technically, and b) felt to be too risky?
>>> I would turn your argument around: the (comparatively speaking) lack of
>>> commercial success of vorbis and theora, and the success of royalty
>>> bearing standard codecs, suggest to me only that no one was fooled,
>>> except
>>> perhaps the enthusiastic followers of those folks who keep pushing out
>>> one
>>> "free" codec after another...
>>> Stephan
>>>
>> Stephan,
>>
>> Monty already pointed out that sabre-rattling already happened.  Had
>> there been any meat to the threats, then it would have been ridiculous
>> not to have pursued them.  Mozilla implements both Theora and Vorbis in
>> browser, and do not implement MPEG-related codecs, and they're a
>> lucrative target (they're getting around $1 billion from Google in the
>> coming few years).  They *do* have a commercial entity that could be the
>> target, as well.
>>
>> The real threat was that these formats posed a risk to the other formats
>> monopolistic adoption, and the FUD that was spread every time these
>> formats were presented in a standard or appeared poised to be adopted
>> more widely, the sabre-rattling began again, and no real threats
>> materialized.
>>
>> It's arguing by saying, "You never can be sure...".
>>
>> -- 
>> Libre Video
>> http://librevideo.org
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>


-- 
Libre Video
http://librevideo.org


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On 03/29/2012 10:57 AM, Basil Mohamed Gohar wrote:
> On 03/29/2012 10:53 AM, Stephan Wenger wrote:
>> On 3.29.2012 16:24 , "Basil Mohamed Gohar" <basilgohar@librevideo.org>
>> wrote:
>>
>>> On 03/29/2012 10:20 AM, Stephan Wenger wrote:
>>>> The second part of your sentence may or may not be true, depending on
>>>> your
>>>> relationship with google, your willingness to use the webm
>>>> implementation
>>>> in unchanged form, and other factors.  Please see the webm license
>>>> conditions, which AFAIK can be found here:
>>>> http://www.webmproject.org/license/additional/
>>> Correct.  I think you are referring to this part, explicitly:
>>>> If you or your agent or exclusive licensee institute or order or agree
>>>> to the institution of patent litigation against any entity (including
>>>> a cross-claim or counterclaim in a lawsuit) alleging that this
>>>> implementation of VP8 or any code incorporated within this
>>>> implementation of VP8 constitutes direct or contributory patent
>>>> infringement, or inducement of patent infringement, then any patent
>>>> rights granted to you under this License for this implementation of
>>>> VP8 shall terminate as of the date such litigation is filed.
>>> Perhaps I assumed that that is a very reasonable part of the license.
>>> That is, if you are suing someone alleging a patent infringement within
>>> VP8, you are no longer granted the license to use VP8's patented
>>> technologies that Google owns.
>> Yes, that's one issue.  Call it personal preference for different type of
>> reciprocity conditions :-)  (I could rant about it for hours, but let's
>> continue to pretend that this is mostly a technical mailing list)
>>
>> The other issue, though (the fact that the license grant extends only to
>> the VP8 implementation as provided by google, and does not extent to
>> derivative works such as hardware implementations) should be moderately
>> alarming even for an open source person.  With respect to this clause, I
>> will note that I criticized the licensing conditions in private and in
>> public (IETF mike) several times, months ago, and nothing happened.
>> Suggests to me one of three things: (1) google is a large company and
>> decisions take time, or (2) google's legal is currently occupied with
>> other stuff, or (3) that the choice of language is intentional, and
>> intended to prevent forks.  Take your pick.
>> Stephan
> Stephan,
>
> If the license wording is meant to limit the patent grant only to
> software implementations, then I will raise this issue myself, because
> that's a glaring hole and huge setback to the adoption of the format. 
> If I find anything out about this, I'll report back here.  The fact that
> there are already hardware implementations, though, seems to imply that
> in reality, this is not the case.  Google has been open to changing the
> licensing situation of VP8 in the past when issues like this were
> brought up.
>
> I do not, however, want this to be a point of FUD.  I would welcome
> anyone from Google on this list authorized to speak about this to
> clarify this ASAP.  A public statement like that should allow us to
> discount this worry.
>
A quicker follow-up than a requested, and by someone from outside of Google.

The first license you linked to appears to cover the libvpx
implementation itself.  It's solely a software implementation, obviously.

The VP8 bitstream is licensed separately, here:

http://www.webmproject.org/license/bitstream/

And that appears to make no distinction between software or hardware. 
It relates to the VP8 format (a.k.a., the bitstream) itself.  If I'm
wrong, hopefully someone can correct me.

-- 
Libre Video
http://librevideo.org


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From: Roman Shpount <roman@telurix.com>
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Subject: Re: [rtcweb] Consensus call regarding media security
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--047d7b2edf0343c72004bc633144
Content-Type: text/plain; charset=ISO-8859-1

I wanted to clarify what i actually want to see as security/encryption
policy for WebRTC.

1. SRTP-DTLS is required to be supported and is offered by default with no
option to bid down to anything else.
2. From HTTPS session SDES SRTP can be enabled via an API method
3. From HTTP session RTP can be enabled via an API method

I only want to allow RTP from HTTP initiated session, since I see no
benefit of supporting SDES SRTP there. I only want to allow SDES SRTP from
HTTPS initiated sessions, since SDES SRTP provides no security benefits
when session description is transmitted over clear channel.

With my proposal, if an application provider does not care about security,
they will use HTTP/RTP. If they care about security, they will use
HTTPS/DTLS-SRTP or SDES-SRTP, if interop or weather forces them to do so.
Also, this way we do not provide false sense of security if someone decides
to start SDES-SRTP session from HTTP.

Is someone can explain to me how this is less secure, even in the airport
wifi comparing to SDES-RTP from HTTP session, please do so.

P.S. I do know why this is less secure then DTLS-SRTP only option.
_____________
Roman Shpount

--047d7b2edf0343c72004bc633144
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

I wanted to clarify what i actually want to see as security/encryption poli=
cy for WebRTC.<br><br>1. SRTP-DTLS is required to be supported and is offer=
ed by default with no option to bid down to anything else.<br>2. From HTTPS=
 session SDES SRTP can be enabled via an API method<br>
3. From HTTP session RTP can be enabled via an API method<br><br>I only wan=
t to allow RTP from HTTP initiated session, since I see no benefit of suppo=
rting SDES SRTP there. I only want to allow SDES SRTP from HTTPS initiated =
sessions, since SDES SRTP provides no security benefits when session descri=
ption is transmitted over clear channel.<br>
<br>With my proposal, if an application provider does not care about securi=
ty, they will use HTTP/RTP. If they care about security, they will use HTTP=
S/DTLS-SRTP or SDES-SRTP, if interop or weather forces them to do so. Also,=
 this way we do not provide false sense of security if someone decides to s=
tart SDES-SRTP session from HTTP.<br>
<br>Is someone can explain to me how this is less secure, even in the airpo=
rt wifi comparing to SDES-RTP from HTTP session, please do so.<br><br>P.S. =
I do know why this is less secure then DTLS-SRTP only option.<br clear=3D"a=
ll">
_____________<br>Roman Shpount<br>
<br><br>

--047d7b2edf0343c72004bc633144--

From jmvalin@mozilla.com  Thu Mar 29 08:16:35 2012
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Proposal for Theora baseline codec
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On 29/03/12 10:33 AM, Stephan Wenger wrote:
> Couldn't it be that no one went after Vorbis and Theora because the
> rightholders had no real incentive to do so?  For example, because the
> products implementing Vorbis, in their vast majority, also included MP3
> and/or AAC, and the rightholders would have to deal with patent exhaustion
> arguments and whatnot after having gotten their money; and because theora
> deployment never reached a critical mass because it was a) not good enough
> technically, and b) felt to be too risky?

Whatever the reason might have applied to Vorbis, it seems like would
also apply to VP8. I've also heard similar arguments about "nobody going
after Speex mainly because nobody with deep pockets is using it". That's
despite the fact that Speex is being shipped by companies like
Microsoft, Google, Apple and CISCO. You would have thought someone would
have sued if they had something. Similarly, Vorbis is also shipped by
(at least) Microsoft and Google and we haven't heard of anything. In
fact, the only codec related patent lawsuits/threats I'm aware of all
involve ITU/MPEG codecs with established patent pools around them. So if
anything, I'd say VP8 might be safer from unexpected risks than H.264

	Jean-Marc


> On 3.29.2012 16:09 , "Monty Montgomery" <xiphmont@gmail.com> wrote:
> 
>> If we're suggesting ten+ year-old codecs with low patent risk, let's
>> choose an obviously higher performance example.  Dare I say it...
>> Theora?
>>
>> I think we've gotten off track.
>>
>> Twelve years ago, oblique threats were made against the nascent Vorbis
>> by Thomson and the whole world decided it was a patent risk.  It never
>> was.  Fool me once, shame on you.
>>
>> In the mid 2000s, people started picking on Theora the same way, and a
>> few years ago Steve Jobs [and Larry Horn, et al] informed the world he
>> was coming after us. Nothing came of it.  Fool me twice, shame on me.
>>
>> Now we have the same substance-less muttering about VP8, suggesting
>> that it's better to use leftover crumbs from 20 years ago.
>>
>> Fool me three times, I should begin to wonder if I have the requisite
>> capacity to be engaging in technical endeavors.
>>
>> Monty
>> Xiph.Org
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
> 
> 
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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Subject: Re: [rtcweb] IdP and universal trust
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On 29 March 2012 16:03, Bernard Aboba <bernard_aboba@hotmail.com> wrote:
> While they would be perceived as different with first party IdP, a signed assertion is still meaningless without some proof that the E.164 number is actually owned by the domain making the assertion.

Yes, if you infer that it is an E.164.  I wouldn't have inferred that
it was an E.164 number because I know that you can't make those sorts
of assertions - it's just a string @ a domain.

From andrew.hutton@siemens-enterprise.com  Thu Mar 29 08:19:37 2012
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To: Magnus Westerlund <magnus.westerlund@ericsson.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Thu, 29 Mar 2012 17:19:34 +0200
Thread-Topic: [rtcweb] Consensus call regarding media security
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I agree that there was clear consensus on mandating the use of SRTP but it =
was not clear to me what the consensus is regarding the use of SRTP with a =
null cipher. Does the statement "there was overwhelming consensus that all =
RTP packets SHALL be protected by SRTP" mean that the null cipher will not =
be allowed?

Regards
Andy



> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Magnus Westerlund
> Sent: 28 March 2012 16:50
> To: rtcweb@ietf.org
> Subject: [rtcweb] Consensus call regarding media security
>=20
> WG,
>=20
> In todays RTCWEB WG meeting there was discussion around media security
> mechanism. In this meeting there was some clear consensus in the
> meeting which we would like to confirm on the list.
>=20
> The first was that there was overwhelming consensus that all RTP
> packets SHALL be protected by SRTP.
>=20
> Secondly that no one objected against making DTLS-SRTP a mandatory to
> implement and the default keying mechanism. Additional mechanisms are
> not precluded.
>=20
> WG participants may state their position regarding these consensus
> calls
> until 12th of April when the chairs will declare the final consensus.
> If
> you where present in the meeting room and comment on this, please
> indicate that.
>=20
> Best Regards
>=20
> Magnus Westerlund
> For the WG chairs
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From ibc@aliax.net  Thu Mar 29 08:21:25 2012
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Thu, 29 Mar 2012 17:21:04 +0200
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To: Roman Shpount <roman@telurix.com>
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Subject: Re: [rtcweb] Consensus call regarding media security
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2012/3/29 Roman Shpount <roman@telurix.com>:
> 1. SRTP-DTLS is required to be supported and is offered by default with n=
o
> option to bid down to anything else.
> 2. From HTTPS session SDES SRTP can be enabled via an API method
> 3. From HTTP session RTP can be enabled via an API method
>
> I only want to allow RTP from HTTP initiated session, since I see no bene=
fit
> of supporting SDES SRTP there. I only want to allow SDES SRTP from HTTPS
> initiated sessions, since SDES SRTP provides no security benefits when
> session description is transmitted over clear channel.
>
> With my proposal, if an application provider does not care about security=
,
> they will use HTTP/RTP. If they care about security, they will use
> HTTPS/DTLS-SRTP or SDES-SRTP, if interop or weather forces them to do so.
> Also, this way we do not provide false sense of security if someone decid=
es
> to start SDES-SRTP session from HTTP.
>
> Is someone can explain to me how this is less secure, even in the airport
> wifi comparing to SDES-RTP from HTTP session, please do so.


You miss something:

1) I access some web using HTTPS with a valid certificate and so.

2) Some HTML and JavaScript is got.

3) The JS code opens a *non* secure WebSocket connection to some other
server (same domain or not, it does not matter here).

4) WebRTC signaling goes through the WebSocket connection (so forget
the initial HTTPS hyper-secure-and-verified connection).

5) SDES-SRTP is negotiated and accepted by both peers.

6) So signaling can be intercepted (unsecure WebSocket connection),
and therefore also the SDES keys. Interception possible regardless the
initial communication was HTTPS.

7) FAIL.




--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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2012/3/29 Hutton, Andrew <andrew.hutton@siemens-enterprise.com>:
> I agree that there was clear consensus on mandating the use of SRTP but i=
t was not clear to me what the consensus is regarding the use of SRTP with =
a null cipher. Does the statement "there was overwhelming consensus that al=
l RTP packets SHALL be protected by SRTP" mean that the null cipher will no=
t be allowed?

IMHO it's very easy:

- The JavaScript WebRTC API MUST NOT be able to set a null cipher (never).

- The browser MAY include an option in about://config ("SRTP: user
null cipher for debugging purposes").

- Such an option is reverted (so dissabled) upon browser restart.

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From basilgohar@librevideo.org  Thu Mar 29 08:30:21 2012
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On 03/29/2012 11:23 AM, IÃ±aki Baz Castillo wrote:
> 2012/3/29 Hutton, Andrew <andrew.hutton@siemens-enterprise.com>:
>> I agree that there was clear consensus on mandating the use of SRTP but it was not clear to me what the consensus is regarding the use of SRTP with a null cipher. Does the statement "there was overwhelming consensus that all RTP packets SHALL be protected by SRTP" mean that the null cipher will not be allowed?
> IMHO it's very easy:
>
> - The JavaScript WebRTC API MUST NOT be able to set a null cipher (never).
>
> - The browser MAY include an option in about://config ("SRTP: user
> null cipher for debugging purposes").
>
> - Such an option is reverted (so dissabled) upon browser restart.
>
I think the standard need not specify that NULL cipher is allowed to
enable the debugging feature for browsers or other implementations. 
It's reasonable to assume that, when testing and debugging, one may
break the spec in the process.  Leaving it out of the standard is the
safest way to prevent it from being used in a production environment.

-- 
Libre Video
http://librevideo.org


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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Thu, 29 Mar 2012 17:33:26 +0200
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Subject: Re: [rtcweb] Support of SDES in WebRTC
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2012/3/29 Oscar Ohlsson <oscar.ohlsson@ericsson.com>:
> Hi Fabio,
>
> My assumption has always been the following:
>
> - DTLS-SRTP is the default
> - DTLS-SRTP + identity can be turned on via the JavaScript API if the web=
app wishes to do so
> - SDES can be turned on by a manipulated SDP offer/answer provided the en=
tire webapp was retrieved over HTTPS

Please check this mail in which I explain that retrieving the web app
by means of HTTPS means nothing:

  http://www.ietf.org/mail-archive/web/rtcweb/current/msg03914.html



--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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--047d7b2edf032e0d9104bc63736f
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On Thu, Mar 29, 2012 at 11:21 AM, I=F1aki Baz Castillo <ibc@aliax.net> wrot=
e:

> You miss something:
>
> 1) I access some web using HTTPS with a valid certificate and so.
>
> 2) Some HTML and JavaScript is got.
>
> 3) The JS code opens a *non* secure WebSocket connection to some other
> server (same domain or not, it does not matter here).
>
> 4) WebRTC signaling goes through the WebSocket connection (so forget
> the initial HTTPS hyper-secure-and-verified connection).
>
> 5) SDES-SRTP is negotiated and accepted by both peers.
>
> 6) So signaling can be intercepted (unsecure WebSocket connection),
> and therefore also the SDES keys. Interception possible regardless the
> initial communication was HTTPS.
>
> 7) FAIL.
>

First of  all, HTTP WebSocket connection are normally not allowed from
HTTPS initiated sessions (or generate a warning).

Second, my point was that SDES-SRTP is no more secure then plain RTP when
signaling is transmitted over clear channel. You are saying the same thing.
If SDES-SRTP is allowed, there is no harm in allowing plain RTP from HTTP,
since, as far as security is concerned, there is no difference.

> _____________
Roman Shpount

--047d7b2edf032e0d9104bc63736f
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<br clear=3D"all"><div class=3D"gmail_quote">On Thu, Mar 29, 2012 at 11:21 =
AM, I=F1aki Baz Castillo <span dir=3D"ltr">&lt;<a href=3D"mailto:ibc@aliax.=
net">ibc@aliax.net</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quot=
e" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">

You miss something:<br>
<br>
1) I access some web using HTTPS with a valid certificate and so.<br>
<br>
2) Some HTML and JavaScript is got.<br>
<br>
3) The JS code opens a *non* secure WebSocket connection to some other<br>
server (same domain or not, it does not matter here).<br>
<br>
4) WebRTC signaling goes through the WebSocket connection (so forget<br>
the initial HTTPS hyper-secure-and-verified connection).<br>
<br>
5) SDES-SRTP is negotiated and accepted by both peers.<br>
<br>
6) So signaling can be intercepted (unsecure WebSocket connection),<br>
and therefore also the SDES keys. Interception possible regardless the<br>
initial communication was HTTPS.<br>
<br>
7) FAIL.<br></blockquote><div>=A0<br>First of=A0 all, HTTP WebSocket connec=
tion are normally not allowed from HTTPS initiated sessions (or generate a =
warning).<br><br>Second, my point was that SDES-SRTP is no more secure then=
 plain RTP when signaling is transmitted over clear channel. You are saying=
 the same thing. If SDES-SRTP is allowed, there is no harm in allowing plai=
n RTP from HTTP, since, as far as security is concerned, there is no differ=
ence.<br>
</div><blockquote class=3D"gmail_quote" style=3D"margin:0pt 0pt 0pt 0.8ex;b=
order-left:1px solid rgb(204,204,204);padding-left:1ex">
</blockquote></div>_____________<br>Roman Shpount<br>

--047d7b2edf032e0d9104bc63736f--

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2012/3/29 Roman Shpount <roman@telurix.com>:
> First of=C2=A0 all, HTTP WebSocket connection are normally not allowed fr=
om HTTPS
> initiated sessions (or generate a warning).

Ok, I didn't check that.


> Second, my point was that SDES-SRTP is no more secure then plain RTP when
> signaling is transmitted over clear channel. You are saying the same thin=
g.

Right:)


> If SDES-SRTP is allowed, there is no harm in allowing plain RTP from HTTP=
,
> since, as far as security is concerned, there is no difference.

The only I meant is that the signaling must be secured. Otherwise,
using SDES-SRTP adds no security at all. But the signaling could be a
different connection (i.e. WebSocket) different from the HTTP(s)
connection used to retrieve the web.

Let's assume this example:

- The browser access the web via plain HTTP.

- The JS opens two websocket connections:
  1) First one for the WebRTC signaling, a secure WS connection.
  2) Second one for other realtime stuff, a NON secure WS connection.


If the SDES key is carried over the first WS connection (wss) then
it's ok, it provides confidenciality. But if the SDES key is carried
over the second WS connection (ws, so unsecure) then it's
interceptable and therefore also the SRTP.

Question: can the WebRTC stack figure which WS connection is being
used for WebRTC signaling?
Autoanswer: NO, since the signaling protocol is up to the app developer.

So Houston we have a problem if we try to make requirements in which
SDES-SRTP can be used. I'm in favour of mandating SDES-SRTP in WebRTC,
but we need to solve this issue, and it is not easy (AFAIK).


Regards.

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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On 29/03/12 11:03 AM, Stephan Wenger wrote:
>  That's the whole idea behind
> sabre-rattling.  It's better than sabre-swinging.  More diligently put,
> warning of potential infringement is better than to punish an infringer.
> Cheaper, and serves the purpose.

Actually, I thought it was the other way around. Those who actually have
valid claims swing without warning to maximize damages. Those who cannot
swing, rattle and scare others in the direction of codecs for which they
hold patents.

	Jean-Marc

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Subject: Re: [rtcweb] Support of SDES in WebRTC
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On 3/29/12 5:33 PM, IÃ±aki Baz Castillo wrote:
> 2012/3/29 Oscar Ohlsson <oscar.ohlsson@ericsson.com>:
>> Hi Fabio,
>>
>> My assumption has always been the following:
>>
>> - DTLS-SRTP is the default
>> - DTLS-SRTP + identity can be turned on via the JavaScript API if the webapp wishes to do so
>> - SDES can be turned on by a manipulated SDP offer/answer provided the entire webapp was retrieved over HTTPS
> 
> Please check this mail in which I explain that retrieving the web app
> by means of HTTPS means nothing:
> 
>   http://www.ietf.org/mail-archive/web/rtcweb/current/msg03914.html

Yeah, but that's an issue that relate to general web and Javascript
trust issue.

Javascript trust issue is a problem that's already discussed and
evaluated in other area and context of w3c, there are several solutions
on-going.

This kind of issue is the same considered for Javascript encryption.

Trusted javascript can be verified/loaded today trough plug-ins and by
trusting the 'application delivery source', so that's not an issue.

>From 0 up to 100 of security with Javascript you can stay at 0 or reach
100, depending on the way you deploy and use web.

Imho making SDES SDP offer/answer available to Javascript it's a great
achievement as it will be able to unlock the creativity of the
technological ecosystem.

People will probably manipulate SDP, sign it, re-encrypt it, pass it
trough OOB channels via CORS or as many other options we may envision.



-- 
Fabio Pietrosanti
Founder, CTO

Tel: +39 02 85961748 (direct)
Mobile: +39 340 1801049
E-mail: fabio.pietrosanti@privatewave.com
Skype: fpietrosanti
Linkedin: http://linkedin.com/in/secret

PrivateWave Italia S.p.A.
Via Gaetano Giardino 1 - 20123 Milano - Italy
www.privatewave.com

From pravindran@sonusnet.com  Thu Mar 29 10:01:54 2012
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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Consensus call regarding media security
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WebRTC trust model has to be considered as one of the main factor for decid=
ing the key mechanism. AFAIK, SDES does not fit into WebRTC as Dr.Evil HTTP=
S RTCWeb server must be trusted in case of SDES. There is no means to track=
 or analyze whether Dr.Evil involves in monitoring or recording or terminat=
e the media traffic.  It will be good in case whoever advocate for SDES exp=
lain how SDES fits within WebRTC trust model.

Thanks
Partha

>-----Original Message-----
>From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
>Of Magnus Westerlund
>Sent: Wednesday, March 28, 2012 8:20 PM
>To: rtcweb@ietf.org
>Subject: [rtcweb] Consensus call regarding media security
>
>WG,
>
>In todays RTCWEB WG meeting there was discussion around media security
>mechanism. In this meeting there was some clear consensus in the meeting
>which we would like to confirm on the list.
>
>The first was that there was overwhelming consensus that all RTP packets
>SHALL be protected by SRTP.
>
>Secondly that no one objected against making DTLS-SRTP a mandatory to
>implement and the default keying mechanism. Additional mechanisms are
>not precluded.
>
>WG participants may state their position regarding these consensus calls
>until 12th of April when the chairs will declare the final consensus. If
>you where present in the meeting room and comment on this, please
>indicate that.
>
>Best Regards
>
>Magnus Westerlund
>For the WG chairs
>
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb

From silviapfeiffer1@gmail.com  Thu Mar 29 10:13:25 2012
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From: Silvia Pfeiffer <silviapfeiffer1@gmail.com>
Date: Thu, 29 Mar 2012 10:12:52 -0700
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Very interesting discussion. Out of personal curiosity, I have some
questions inline.

On Thu, Mar 29, 2012 at 7:20 AM, Stephan Wenger <stewe@stewe.org> wrote:
>
> The most commonly cited timeline for a widely in use technology to be
> "save" from a patent viewpoint, based on equitable defenses such as lache=
s
> (in the US) is six years.

Very interesting to know the usually used limit on the number of
years. IIUC that would make Speex, Vorbis and Theora "save" codecs,
since they were released in 2003, 2000, and 2004 respectively?


> In addition, as I pointed out in the meeting, the use of a video codec
> created by a body such as MPEG or ITU-T SG16 has the advantage of that th=
e
> patents of all participating players are available at least under
> Reasonable and Non Discriminatory (RAND) terms. =A0This may sound like a =
Bad
> Thing if you operate under a business model that prevents you to pay
> anything for patent licenses, but it is surely a Good Thing if you are
> willing to dish out a moderate amount of money for a license.

So, what is the situation with H.263 and patent licensing? Is what
Larry wrote in 2003 in this email still valid?
http://lists.mpegif.org/pipermail/mp4-tech/2003-October/002771.html
My Web search hasn't turned up any more up-to-date information on
this, so it seems there are several patents associated that are still
valid for H.263 and there is no patent pool for it, thus requiring
every user of H.263 to come to separate license arrangements with each
patent holder?

Note: These are just questions of clarification - I am not implying
any codec preferences in this email.

Regards,
Silvia.

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
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2012/3/29 Fabio Pietrosanti (naif) <lists@infosecurity.ch>:
> Yeah, but that's an issue that relate to general web and Javascript
> trust issue.
>
> Javascript trust issue is a problem that's already discussed and
> evaluated in other area and context of w3c, there are several solutions
> on-going.

It's nice to know that the issue is already being addressed.


> Imho making SDES SDP offer/answer available to Javascript it's a great
> achievement as it will be able to unlock the creativity of the
> technological ecosystem.

Sure, I agree.


Regards.

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From lists@infosecurity.ch  Thu Mar 29 10:31:51 2012
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On 3/29/12 7:02 PM, Ravindran, Parthasarathi wrote:
> WebRTC trust model has to be considered as one of the main factor for deciding the key mechanism. AFAIK, SDES does not fit into WebRTC as Dr.Evil HTTPS RTCWeb server must be trusted in case of SDES. There is no means to track or analyze whether Dr.Evil involves in monitoring or recording or terminate the media traffic.  It will be good in case whoever advocate for SDES explain how SDES fits within WebRTC trust model.

Sure!

From:
http://datatracker.ietf.org/doc/draft-ietf-rtcweb-security-arch/?include_text=1

"   The basic assumption of this architecture is that network resources
   exist in a hierarchy of trust, rooted in the browser, which serves as
   the user's TRUSTED COMPUTING BASE (TCB).  Any security property which
   the user wishes to have enforced must be ultimately guaranteed by the
   browser (or transitively by some property the browser verifies)."

So, it means that if the browser already have a hierarchy of trust to
use TLS for HTTPS, then SDES-SRTP will inherit the trust-properties of
the HTTPS website from which it's loaded.

It seems to me quite easy to fit SDES-SRTP into the browser model, as it
allow you to assure that the communication path between the client and
the server is secure.

Do you expect WebRTC to be only peer-to-peer/client-to-client?

I sincerly expect *a lot* of communications to goes trough SIP/RTP media
proxy for security purpose, for billing purposes, for value added
service purpose.

SDES-SRTP provide a very reliable and simple way to let a WebRTC peer to
establish security with the server, assuming that it already have
established security trough HTTPS/TLS that's a consolidate trust method.


-- 
Fabio Pietrosanti
Founder, CTO

Tel: +39 02 85961748 (direct)
Mobile: +39 340 1801049
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www.privatewave.com

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2012/3/29 Ravindran, Parthasarathi <pravindran@sonusnet.com>:
> WebRTC trust model has to be considered as one of the main factor for dec=
iding the key mechanism. AFAIK, SDES does not fit into WebRTC as Dr.Evil HT=
TPS RTCWeb server must be trusted in case of SDES. There is no means to tra=
ck or analyze whether Dr.Evil involves in monitoring or recording or termin=
ate the media traffic. =C2=A0It will be good in case whoever advocate for S=
DES explain how SDES fits within WebRTC trust model.

If Dr. Evil attaks my back webpage and owns it, and then I visit it
(HTTPS with valid certificate) and enter my back credentials... for me
that is much worse than the case you describe. Should we drop HTTPS
then because it does not fit 100% "security" requirements?

BTW: previously you wanted to allow plain RTP in WebRTC... and now
DTLS-SRTP is the only valid solution? :)

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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2012/3/29 Fabio Pietrosanti (naif) <lists@infosecurity.ch>:
> From:
> http://datatracker.ietf.org/doc/draft-ietf-rtcweb-security-arch/?include_=
text=3D1
>
> " =C2=A0 The basic assumption of this architecture is that network resour=
ces
> =C2=A0 exist in a hierarchy of trust, rooted in the browser, which serves=
 as
> =C2=A0 the user's TRUSTED COMPUTING BASE (TCB). =C2=A0Any security proper=
ty which
> =C2=A0 the user wishes to have enforced must be ultimately guaranteed by =
the
> =C2=A0 browser (or transitively by some property the browser verifies)."
>
> So, it means that if the browser already have a hierarchy of trust to
> use TLS for HTTPS, then SDES-SRTP will inherit the trust-properties of
> the HTTPS website from which it's loaded.
>
> It seems to me quite easy to fit SDES-SRTP into the browser model, as it
> allow you to assure that the communication path between the client and
> the server is secure.
>
> Do you expect WebRTC to be only peer-to-peer/client-to-client?
>
> I sincerly expect *a lot* of communications to goes trough SIP/RTP media
> proxy for security purpose, for billing purposes, for value added
> service purpose.
>
> SDES-SRTP provide a very reliable and simple way to let a WebRTC peer to
> establish security with the server, assuming that it already have
> established security trough HTTPS/TLS that's a consolidate trust method.

+2  (I don't use "+1" since I don't use Google+).

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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On 3/29/12 2:35 PM, Magnus Westerlund wrote:
> On 2012-03-29 08:02, Bernard Aboba wrote:
>> I agree with proposition #1 (SRTP) unconditionally.
>>
>> With respect to proposition #2 (DTLS-SRTP), perhaps the words "with
>> details to be worked out" should have been added. 
>>
>> I believe that the consensus achieved was only on a general direction,
>> not an endorsement of particular proposals.
>>
>> Personally, I would like to have more specifics about the required
>> features of DTLS-SRTP in the RTCWEB context.
> 
> I hope someone that knows the details can elaborate on this. I thought
> DTLS-SRTP has a core that you will need to implement. Then there is
> clearly a question of crypto algorithms to be supported. But that also
> applies to SRTP where we also need to select which crypto suites that
> are to be implemented if any in addtion to the MITM. The WG will need to
> select these details as part of the next steps.

Hi Magnus,

imho the selection of a crypto algorithms is not the first priority (it
will probably goes with some well known AES-stuff), as the first
priority is to define which will be the Key Management system to be used
for DTLS-SRTP.

Given the SRTP standard,the security and trust model radically change
depending on the key exchange system details.

With DTLS-SRTP the key exchange and security/trust model seems quite
opaque and there's not a specific guideline to implement it and/or to
satisfy a specific security requirement.

While the SDES-SRTP method satisfy  all the 4.1.1.* Calling method
requirement of
http://tools.ietf.org/html/draft-ietf-rtcweb-security-02#section-4.1.1 ,
because it would just rely on existing TLS/HTTPS method.

So it would not even enter into the context/discussion of trust with the
server, that has been already managed by the existing TLS/HTTPS.

I did not really understand how DTLS-SRTP would like to manage the
so-many-complicated trust scenario that TLS/HTTPS already manage.

Imho everything should be simplified with two context:
- Do you need end-to-site (TLS equivalent) security?
  SDES-SRTP + TLS/HTTPS

- Do you need end-to-end (ZRTP/PGP equivalent) security?
  DTLS-SRTP with user-driven verification schema (such as Short
Authentication String / Fingerprint / Key pinning)

-- 
Fabio Pietrosanti
Founder, CTO

Tel: +39 02 85961748 (direct)
Mobile: +39 340 1801049
E-mail: fabio.pietrosanti@privatewave.com
Skype: fpietrosanti
Linkedin: http://linkedin.com/in/secret

PrivateWave Italia S.p.A.
Via Gaetano Giardino 1 - 20123 Milano - Italy
www.privatewave.com

From lists@infosecurity.ch  Thu Mar 29 12:53:40 2012
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Subject: [rtcweb] SDES-SRTP as a platform for multiple key management
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Hi all,

i've been thinking that one of the very interesting elements about the
support of SDES-SRTP, is that, other than providing compatibility with
existing telephony ecosystem, it may allow the implementation of custom
key managegement systems.

Basically if WebRTC would introduce support for SDES-SRTP and w3c would
define API to handle SDES SDP call keys, it would become possible to
further implement in Javascript additional key management systems.

For example someone may implement a javascript application to be
provided from an https source or browser extension additional to
implement OpenPGPJS based identity verification (http://openpgpjs.org/)
or integration with DH based key exchange
(https://github.com/kaepora/cryptocat/).

So basically a side-effect of introducing SDES-SRTP, could be to let
HTML5 application developers, to effectively be able to implement custom
security mechanisms for voice applications.

-- 
Fabio Pietrosanti
Founder, CTO

Tel: +39 02 85961748 (direct)
Mobile: +39 340 1801049
E-mail: fabio.pietrosanti@privatewave.com
Skype: fpietrosanti
Linkedin: http://linkedin.com/in/secret

PrivateWave Italia S.p.A.
Via Gaetano Giardino 1 - 20123 Milano - Italy
www.privatewave.com

From Markus.Isomaki@nokia.com  Thu Mar 29 13:23:38 2012
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From: <Markus.Isomaki@nokia.com>
To: <rtcweb@ietf.org>
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Hi,

Some comments on the data channel proposal:

1. Congestion control: If no real-time sessions are on, TCP-style loss-driv=
en congestion control is fine. However, if voice/video session is up, such =
congestion control would be disasterous to the real-time traffic quality if=
 sendind or receiving more than a slight amount of data. Especially in mobi=
le access networks the user would bloat his/her own buffers very easily. In=
 that case, some less-than-best-effort congestion control algorithm =E1 la =
LEDBAT would be required. It would be best to make this a MUST requirement =
on implementations, otherwise we will get a lot of crappy video even if the=
 codec was better than H.261 :-)

2. Multihoming and interface switching: I don't suggest going for the multi=
path support on the SCTP level. I think it would be best to deal with this =
through ICE in the same way as for RTP, i.e. adding and removing candidates=
 as interfaces become available or go. Or let the application handle it by =
creating a new data channel and continue from the breakpoint. The most impo=
rtant requirement is that the application gets notified about these changes=
 downstrairs so it can act.=20

3. HTTP tunneling: In practice we are going to need HTTP tunneling last-res=
ort option for the data channel as well. If doing so, what will the protoco=
l stack look like? Is it SCTP/DTLS/UDP/HTTP/TLS/TCP? Or can we collapse som=
e of these layers. I think we'd better.

Markus=20





From ekr@rtfm.com  Thu Mar 29 13:28:26 2012
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From: Eric Rescorla <ekr@rtfm.com>
Date: Thu, 29 Mar 2012 22:27:44 +0200
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On Thu, Mar 29, 2012 at 10:23 PM,  <Markus.Isomaki@nokia.com> wrote:
> Hi,
>
> Some comments on the data channel proposal:
>
> 1. Congestion control: If no real-time sessions are on, TCP-style loss-dr=
iven congestion control is fine. However, if voice/video session is up, suc=
h congestion control would be disasterous to the real-time traffic quality =
if sendind or receiving more than a slight amount of data. Especially in mo=
bile access networks the user would bloat his/her own buffers very easily. =
In that case, some less-than-best-effort congestion control algorithm =E1 l=
a LEDBAT would be required. It would be best to make this a MUST requiremen=
t on implementations, otherwise we will get a lot of crappy video even if t=
he codec was better than H.261 :-)
>
> 2. Multihoming and interface switching: I don't suggest going for the mul=
tipath support on the SCTP level. I think it would be best to deal with thi=
s through ICE in the same way as for RTP, i.e. adding and removing candidat=
es as interfaces become available or go. Or let the application handle it b=
y creating a new data channel and continue from the breakpoint. The most im=
portant requirement is that the application gets notified about these chang=
es downstrairs so it can act.
>
> 3. HTTP tunneling: In practice we are going to need HTTP tunneling last-r=
esort option for the data channel as well. If doing so, what will the proto=
col stack look like? Is it SCTP/DTLS/UDP/HTTP/TLS/TCP? Or can we collapse s=
ome of these layers. I think we'd better.

I don't think I understand point 3. Why would we need HTTP tunneling?
If we can bring
up a bidirectional UDP channel, then why would we need to run HTTP over it?

-Ekr

From Markus.Isomaki@nokia.com  Thu Mar 29 13:36:59 2012
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From: <Markus.Isomaki@nokia.com>
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Thread-Topic: [rtcweb] Data channel comments and questions
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Hi,

Eric Rescorla [mailto:ekr@rtfm.com] wrote:
>>
>> 3. HTTP tunneling: In practice we are going to need HTTP tunneling last-
>resort option for the data channel as well. If doing so, what will the pro=
tocol
>stack look like? Is it SCTP/DTLS/UDP/HTTP/TLS/TCP? Or can we collapse some
>of these layers. I think we'd better.
>
>I don't think I understand point 3. Why would we need HTTP tunneling?
>If we can bring
>up a bidirectional UDP channel, then why would we need to run HTTP over it=
?
>

Ah, sorry for not being clear. I mean a situation where we can't setup the =
UDP channel, if one endpoint is for instance in a corporate network from wh=
ere only HTTP or HTTPS is allowed. So presumably we'd have to support some =
kind of HTTP or TLS encapsulation/tunneling for the data channel as well, f=
rom the endpoint to some kind of relay. But how would the end-to-end data c=
hannel look like then? I.e. what would be sent over the tunnel?

>-Ekr

Markus

From ekr@rtfm.com  Thu Mar 29 13:48:59 2012
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Subject: Re: [rtcweb] Data channel comments and questions
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On Thu, Mar 29, 2012 at 10:36 PM,  <Markus.Isomaki@nokia.com> wrote:
> Hi,
>
> Eric Rescorla [mailto:ekr@rtfm.com] wrote:
>>>
>>> 3. HTTP tunneling: In practice we are going to need HTTP tunneling last=
-
>>resort option for the data channel as well. If doing so, what will the pr=
otocol
>>stack look like? Is it SCTP/DTLS/UDP/HTTP/TLS/TCP? Or can we collapse som=
e
>>of these layers. I think we'd better.
>>
>>I don't think I understand point 3. Why would we need HTTP tunneling?
>>If we can bring
>>up a bidirectional UDP channel, then why would we need to run HTTP over i=
t?
>>
>
> Ah, sorry for not being clear. I mean a situation where we can't setup th=
e UDP channel, if one endpoint is for instance in a corporate network from =
where only HTTP or HTTPS is allowed. So presumably we'd have to support som=
e kind of HTTP or TLS encapsulation/tunneling for the data channel as well,=
 from the endpoint to some kind of relay. But how would the end-to-end data=
 channel look like then? I.e. what would be sent over the tunnel?

I had sort of assumed that people would use that HTTP/HTTPS encapsulation o=
nly
to a relay which emitted UDP, so that that encapsulation wouldn't be end-to=
-end
in the general case.

-Ekr

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Hi,

Eric Rescorla <ekr@rtfm.com> wrote:
>
>I had sort of assumed that people would use that HTTP/HTTPS encapsulation
>only to a relay which emitted UDP, so that that encapsulation wouldn't be
>end-to-end in the general case.
>

Makes sense and now that I think of it, it's actually quite clear that it c=
an be done in the same way as with TURN. But SCTP/DTLS would be e2e in any =
case, and they'd be tunneled over HTTP/HTTPS (and TCP). Well, perhaps that'=
s not so bad for the last resort case.

>-Ekr

Markus

From Markus.Isomaki@nokia.com  Thu Mar 29 14:04:47 2012
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>
>Ah, sorry for not being clear. I mean a situation where we can't setup the=
 UDP
>channel, if one endpoint is for instance in a corporate network from where
>only HTTP or HTTPS is allowed. So presumably we'd have to support some kin=
d
>of HTTP or TLS encapsulation/tunneling for the data channel as well, from =
the
>endpoint to some kind of relay. But how would the end-to-end data channel
>look like then? I.e. what would be sent over the tunnel?
>

There's by the way at least one other use case where UDP-based data channel=
 sucks. If you want to keep it up for a long time for some infrequent messa=
ging, keeping the NAT state up by a keepalive every ~30s. would drain the b=
attery on a mobile device. To solve that case some kind of TCP-based varian=
t would be needed, at least between the mobile endpoint and a relay. TURN-T=
CP, I guess. If the other end was still using UDP and doing frequent keepal=
ives, I suppose the relay would unfortunately forward them as well causing =
the same ill effect.=20

I'm not sure if there are apps that would need to keep the data channel up =
infrequently used, though. Or should they just use the "signaling" path (HT=
TP, websockets) for that type of purposes.=20

Markus=20

From oscar.ohlsson@ericsson.com  Thu Mar 29 14:13:43 2012
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From: Oscar Ohlsson <oscar.ohlsson@ericsson.com>
To: =?iso-8859-1?Q?I=F1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Thu, 29 Mar 2012 23:13:30 +0200
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Hi,

That's why I wrote "the entire webapp" below. If it was not clear I meant t=
hat the

- main HTML page
- all external CSS files, JavaScript files, images, etc=20
- all XmlHttpRequests
- all WebSocket connections

are protected with TLS. This is obviously verifiable and it's a feature sup=
ported by all modern browsers (no mixed content).=20

/Oscar



> -----Original Message-----
> From: I=F1aki Baz Castillo [mailto:ibc@aliax.net]=20
> Sent: Thursday, March 29, 2012 5:33 PM
> To: Oscar Ohlsson
> Cc: Fabio Pietrosanti (naif); <rtcweb@ietf.org>
> Subject: Re: [rtcweb] Support of SDES in WebRTC
>=20
> 2012/3/29 Oscar Ohlsson <oscar.ohlsson@ericsson.com>:
> > Hi Fabio,
> >
> > My assumption has always been the following:
> >
> > - DTLS-SRTP is the default
> > - DTLS-SRTP + identity can be turned on via the JavaScript=20
> API if the=20
> > webapp wishes to do so
> > - SDES can be turned on by a manipulated SDP offer/answer=20
> provided the=20
> > entire webapp was retrieved over HTTPS
>=20
> Please check this mail in which I explain that retrieving the=20
> web app by means of HTTPS means nothing:
>=20
>   http://www.ietf.org/mail-archive/web/rtcweb/current/msg03914.html
>=20
>=20
>=20
> --
> I=F1aki Baz Castillo
> <ibc@aliax.net>
> =

From oscar.ohlsson@ericsson.com  Thu Mar 29 14:23:11 2012
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Subject: Re: [rtcweb] SDES-SRTP as a platform for multiple key management
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Hi,

While I'm in favour of SDES I don't believe this is a particularly strong a=
rgument. To provide any real security gain, any additional signing or encry=
ption that you may think of has to be performed outside of the webapp's con=
trol.

Regards,

Oscar

> -----Original Message-----
> From: rtcweb-bounces@ietf.org=20
> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Fabio Pietrosanti (naif)
> Sent: Thursday, March 29, 2012 9:54 PM
> To: <rtcweb@ietf.org>
> Subject: [rtcweb] SDES-SRTP as a platform for multiple key management
>=20
> Hi all,
>=20
> i've been thinking that one of the very interesting elements=20
> about the support of SDES-SRTP, is that, other than providing=20
> compatibility with existing telephony ecosystem, it may allow=20
> the implementation of custom key managegement systems.
>=20
> Basically if WebRTC would introduce support for SDES-SRTP and=20
> w3c would define API to handle SDES SDP call keys, it would=20
> become possible to further implement in Javascript additional=20
> key management systems.
>=20
> For example someone may implement a javascript application to=20
> be provided from an https source or browser extension=20
> additional to implement OpenPGPJS based identity verification=20
> (http://openpgpjs.org/) or integration with DH based key=20
> exchange (https://github.com/kaepora/cryptocat/).
>=20
> So basically a side-effect of introducing SDES-SRTP, could be to let
> HTML5 application developers, to effectively be able to=20
> implement custom security mechanisms for voice applications.
>=20
> --
> Fabio Pietrosanti
> Founder, CTO
>=20
> Tel: +39 02 85961748 (direct)
> Mobile: +39 340 1801049
> E-mail: fabio.pietrosanti@privatewave.com
> Skype: fpietrosanti
> Linkedin: http://linkedin.com/in/secret
>=20
> PrivateWave Italia S.p.A.
> Via Gaetano Giardino 1 - 20123 Milano - Italy=20
> www.privatewave.com _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> =

From ibc@aliax.net  Thu Mar 29 14:54:05 2012
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
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Subject: Re: [rtcweb] Support of SDES in WebRTC
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2012/3/29 Oscar Ohlsson <oscar.ohlsson@ericsson.com>:
> Hi,
>
> That's why I wrote "the entire webapp" below. If it was not clear I meant=
 that the
>
> - main HTML page
> - all external CSS files, JavaScript files, images, etc
> - all XmlHttpRequests
> - all WebSocket connections
>
> are protected with TLS. This is obviously verifiable and it's a feature s=
upported by all modern browsers (no mixed content).

Good point.

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Oscar Ohlsson [oscar.ohlsson@ericsson.com] wrote:
> That's why I wrote "the entire webapp" below. If it was not clear I meant=
 that the
> - main HTML page
> - all external CSS files, JavaScript files, images, etc
> - all XmlHttpRequests
> - all WebSocket connections
> are protected with TLS. This is obviously verifiable and it's a feature s=
upported by all modern browsers (no mixed content).

Even this is insufficient.

First, even if it is in principle possible to provide adequate
cryptographic security inside the JS applications a great many of them
won't (for many reasons). We can't eliminate inconsistency in the security
of webrtc applications=97 but we can certainly make it hard to go below
a certain level of security especially by accident.   In particular,
if JS applications provide only authentication services (e.g. by giving
them access to hashes of the ephemeral session keys, but not the session
keys themselves) then a cryptographically-incompetent application can
only fail the user by failing to provide the authentication it promised
over and above the platform baseline capability.

With HTTPS applications are currently free to do dumb things (mixed
content, esp scripts) but at least the browser can detected this and
alerts the user with varying degrees of loudness.  In a SDES/SRTP
world the browser will not be generally able to detect insecure
behavior by applications=97 creating something of a lemon market
for webrtc based application security. As a resutl its important to
narrow the amount of insecurity possible.

This is effectively a repeat of the arguments against supporting
plaintext. If plaintext is supported it will be commonly used because it's
easiest, because users can't easily tell how insecure their connection
is, and because users often don't have a good feel for the threat model
(e.g. things like firesheep were _very_ surprising to most people)...
Because of this the platform should provide the highest level of security
that can reasonably be provided in the platform, and that means (among
other things) perfect-forward-secrecy=97 while allowing applications to
add things like authentication on top (because while strong ephemerally
keyed crypto can be done very low in the stack, meaningful authentication
needs layer-7/8 hooks).

There is also the application/library split issue=97 if a vulnerability
is found in a common negotiation procedure what secures users
better? Updating a few easily identified browsers / softphone apps?
or a much larger base of webapps, many of which would be run by security
ignorant/clueless people?  (and at least if the webserver is clueful but
the client operator is not the app could yell about the insecure client,
the other way around doesn't work as well).  =97   Really, cryptographic
negotiation is not properly an application feature, it belongs lower in
the stack, and many applications that roll their own crypto have done
a poor job of it.

It's also inadequate on purely technical grounds: Javascript provides
no mechanism for working with mlocked memory,  no mechanism to ensure
that garbage collected data gets zeroized.  Your crypto app in JS could
easily have its long term keying material pulled out of free ram by
malware long after it runs, or pulled off the disk from swap.
The breakneck pace of fancy JIT systems makes it seem unlikely to me
that javascript will provide for that any time soon.




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Subject: Re: [rtcweb] SRTP and "marketing"
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On 3/29/2012 2:08 AM, Hadriel Kaplan wrote:
> On Mar 29, 2012, at 2:54 AM, Randell Jesup wrote:
>
>> That doesn't mean I don't want to implement a truly secure communications ability.  Note that by "secure" here I mean "I can detect if the service MITMd me" -- without reading cert phrases and preferably without having to install client certs and somehow build a working PKI.

>> One idea we've had is a non-centralized WebRTC service built on distributed hash tables and a partial-mesh set of connections used to pass signaling on WebRTC (or other) data channels to set up calls, with some type of Id system to validate who you get connected to (since you can't really trust any of the nodes fully).  If an intermediary node can downgrade you to SDES and sniff the keys, there is no security.
> If you use SDES, no one should be claiming you *have* end-to-end security.  You have security from the browser to the HTTPS server, but nothing more.  The browser shouldn't be showing a lock icon, for example (and it shouldn't for DTLS-SRTP either).  Since the media potentially goes into the PSTN, it can be no more secure than that.  No matter how much stronger we make the Browser-to-Gateway link, the end-to-end can't exceed the PSTN's level of security.  It's reasonably sufficient security for everyday use, but it won't let you bypass the government.  And using DTLS-SRTP to a gateway won't either.  And arguably even using IdP won't if the government can force the IdP to lie.


Total agreement.  SDES cannot be touted as secure.  It's more secure 
against sniffers; that's all.

>> An alternative would be to allow a hybrid: SDES at the start of a call and rekey over to DTLS-SRTP after the call is connected.  If something blocks the DTLS connection it would be treated as a gateway-to-legacy/PSTN call (i.e. show no security, though it's secure against non-in-path attackers); if the DTLS connection is allowed, we can use it to rekey to DTLS-SRTP and to verify identity.
> Why bother?  I assume the concern is a bid-down.  Let's imagine we were dealing with a malicious website, and the website javascript removed DTLS-SRTP from SDP but put in SDES, so that browser believes it's talking to a gateway and uses SDES.  The malicious website now has your SRTP key and can decrypt your media, assuming it has some access to your media.  So you build a better browser and try an active check for DTLS-SRTP in the media plane regardless of SDP.  If your browser is popular, why wouldn't the malicious website just insert a DTLS-SRTP b2bua gateway, and still be able to listen in?

This allows avoiding a gateway ('burning gateway' argument) with PSTN 
calls while minimizing the risk of bid-down.  Note, correctly, that you 
need Identity+DTLS-SRTP to be secure against in-signaling-path attackers 
(providers or governments).  That's not of no benefit.  It's not of real 
benefit to J. Random Salesman, but even non-political people may worry 
about someone watching in more than they worry about listening in.  
(#include lawsuits over schools using laptop computers to take snapshots 
from student computers at home, etc, etc).

> As far as I can see, there is no bid-down issue created by adding SDES - it was already there.  It was already possible to insert a DTLS-SRTP B2BUA and be malicious.  Yes, we may have made it slightly cheaper to be malicious, but it was already possible.  The only thing that prevents a malicious website's DTLS-SRTP B2BUA from working is if then humans verify the fingerprints or having an IdP you can trust.  Even if you *do* verify the fingerprints and find they do NOT match, all you know is it went through one or more gateways in the middle - you don't know it's malicious - it could just be going through the PSTN legitimately.

Video doesn't go through the PSTN...  That's a kinda obvious downgrade.  
And only people who care will note the insecurity; part of why it must 
be dead-simple but not overly intrusive (no doorhangers, etc).  This is 
W3C land though, mostly.  This just comes back to the argument over 
whether users care.  Some do, and most of them are non-technical.  
Identity doesn't provide perfect security.  It does provide better 
security and at least makes any tapper be concerned that they're raising 
red flags, if the person notices/cares.  I guarantee the Tunisian 
rebels/activists mentioned by Tim would care.


> With SDES the browser can indicate right-off-the-bat that the call is not end-to-end secure, without having to show the fingerprint and having the user read it out.

True :-)

>> This minimizes the risk of bid-down - the primary additional risk is that without initial SDES, to tap the signal you'd need to use a WebRTC-enabled B2BUA to accept the DTLS-SRTP connection and siphon it off.  Also, a "good" gateway in a pure DTLS-SRTP world might have a identity established ("gateway@L3.com"); with SDES it wouldn't by default (but it could if it wanted to by rekeying via DTLS also).
> Yeah I've been trying to figure out if there's some advantage to having a gateway be able to indicate "gateway@telco.com", once there is an IdP model.  I would think it's a major pita to deploy such a thing on gateways, unless there's a very popular IdP they could know everyone would trust.  It would actually be easier to just give the gateway a real TLS cert for DTLS to use, from a major CA, since gateways are the types of systems real certs would be reasonably possible to install on.
>
>
>> I'm still strongly in favor of pure DTLS-SRTP - as Harald and others say, less moving parts and less things that need to be secured.  But this might be a useful compromise if we *need* to support SDES in some cases.
> I'm trying to be convinced that DTLS-SRTP is worth the complexity, cost and downsides on the gateways, but I haven't found it yet.  I think people are willing to be convinced, myself included, but I have yet to hear a real advantage for DTLS-SRTP in the gateway scenario.  Everything put forward today for DTLS-SRTP applied to the browser-to-browser scenarios, but not gateways.

Whereas I come at it from "I care about browser-to-browser cases and 
avoiding doing something that hurts them, and I don't care about 
security (other than sniffers) for gateway calls".  :-)



-- 
Randell Jesup
randell-ietf@jesup.org


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Subject: Re: [rtcweb] SRTP and "marketing"
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Randell Jesup wrote:
>> Everything put forward today for DTLS-SRTP applied to the
>> browser-to-browser scenarios, but not gateways.
>
> Whereas I come at it from "I care about browser-to-browser cases and
> avoiding doing something that hurts them, and I don't care about
> security (other than sniffers) for gateway calls". :-)

And the difficulty is, a browser has no way to tell (a priori) when the 
other end is another browser or a gateway.

From roman@telurix.com  Thu Mar 29 20:30:00 2012
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On Thu, Mar 29, 2012 at 5:59 PM, Gregory Maxwell <gmaxwell@juniper.net>wrot=
e:

> Even this is insufficient.
>
> First, even if it is in principle possible to provide adequate
> cryptographic security inside the JS applications a great many of them
> won't (for many reasons). We can't eliminate inconsistency in the securit=
y
> of webrtc applications=97 but we can certainly make it hard to go below
> a certain level of security especially by accident.   In particular,
> if JS applications provide only authentication services (e.g. by giving
> them access to hashes of the ephemeral session keys, but not the session
> keys themselves) then a cryptographically-incompetent application can
> only fail the user by failing to provide the authentication it promised
> over and above the platform baseline capability.
>
> With HTTPS applications are currently free to do dumb things (mixed
> content, esp scripts) but at least the browser can detected this and
> alerts the user with varying degrees of loudness.  In a SDES/SRTP
> world the browser will not be generally able to detect insecure
> behavior by applications=97 creating something of a lemon market
> for webrtc based application security. As a resutl its important to
> narrow the amount of insecurity possible.
>
> This is effectively a repeat of the arguments against supporting
> plaintext. If plaintext is supported it will be commonly used because it'=
s
> easiest, because users can't easily tell how insecure their connection
> is, and because users often don't have a good feel for the threat model
> (e.g. things like firesheep were _very_ surprising to most people)...
> Because of this the platform should provide the highest level of security
> that can reasonably be provided in the platform, and that means (among
> other things) perfect-forward-secrecy=97 while allowing applications to
> add things like authentication on top (because while strong ephemerally
> keyed crypto can be done very low in the stack, meaningful authentication
> needs layer-7/8 hooks).
>
> There is also the application/library split issue=97 if a vulnerability
> is found in a common negotiation procedure what secures users
> better? Updating a few easily identified browsers / softphone apps?
> or a much larger base of webapps, many of which would be run by security
> ignorant/clueless people?  (and at least if the webserver is clueful but
> the client operator is not the app could yell about the insecure client,
> the other way around doesn't work as well).  =97   Really, cryptographic
> negotiation is not properly an application feature, it belongs lower in
> the stack, and many applications that roll their own crypto have done
> a poor job of it.
>
> It's also inadequate on purely technical grounds: Javascript provides
> no mechanism for working with mlocked memory,  no mechanism to ensure
> that garbage collected data gets zeroized.  Your crypto app in JS could
> easily have its long term keying material pulled out of free ram by
> malware long after it runs, or pulled off the disk from swap.
> The breakneck pace of fancy JIT systems makes it seem unlikely to me
> that javascript will provide for that any time soon.
>
>
I guess what you are saying (and I believe everybody on this list agree) is
that SDES-SRTP in combination with JavaScript signaling negotiation does
not and cannot provide any meaningful security above preventing simple
network sniffing.
_____________
Roman Shpount

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<br clear=3D"all"><div class=3D"gmail_quote">On Thu, Mar 29, 2012 at 5:59 P=
M, Gregory Maxwell <span dir=3D"ltr">&lt;<a href=3D"mailto:gmaxwell@juniper=
.net">gmaxwell@juniper.net</a>&gt;</span> wrote:<br><blockquote class=3D"gm=
ail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-le=
ft:1ex">
Even this is insufficient.<br>
<br>
First, even if it is in principle possible to provide adequate<br>
cryptographic security inside the JS applications a great many of them<br>
won&#39;t (for many reasons). We can&#39;t eliminate inconsistency in the s=
ecurity<br>
of webrtc applications=97 but we can certainly make it hard to go below<br>
a certain level of security especially by accident. =A0 In particular,<br>
if JS applications provide only authentication services (e.g. by giving<br>
them access to hashes of the ephemeral session keys, but not the session<br=
>
keys themselves) then a cryptographically-incompetent application can<br>
only fail the user by failing to provide the authentication it promised<br>
over and above the platform baseline capability.<br>
<br>
With HTTPS applications are currently free to do dumb things (mixed<br>
content, esp scripts) but at least the browser can detected this and<br>
alerts the user with varying degrees of loudness. =A0In a SDES/SRTP<br>
world the browser will not be generally able to detect insecure<br>
behavior by applications=97 creating something of a lemon market<br>
for webrtc based application security. As a resutl its important to<br>
narrow the amount of insecurity possible.<br>
<br>
This is effectively a repeat of the arguments against supporting<br>
plaintext. If plaintext is supported it will be commonly used because it&#3=
9;s<br>
easiest, because users can&#39;t easily tell how insecure their connection<=
br>
is, and because users often don&#39;t have a good feel for the threat model=
<br>
(e.g. things like firesheep were _very_ surprising to most people)...<br>
Because of this the platform should provide the highest level of security<b=
r>
that can reasonably be provided in the platform, and that means (among<br>
other things) perfect-forward-secrecy=97 while allowing applications to<br>
add things like authentication on top (because while strong ephemerally<br>
keyed crypto can be done very low in the stack, meaningful authentication<b=
r>
needs layer-7/8 hooks).<br>
<br>
There is also the application/library split issue=97 if a vulnerability<br>
is found in a common negotiation procedure what secures users<br>
better? Updating a few easily identified browsers / softphone apps?<br>
or a much larger base of webapps, many of which would be run by security<br=
>
ignorant/clueless people? =A0(and at least if the webserver is clueful but<=
br>
the client operator is not the app could yell about the insecure client,<br=
>
the other way around doesn&#39;t work as well). =A0=97 =A0 Really, cryptogr=
aphic<br>
negotiation is not properly an application feature, it belongs lower in<br>
the stack, and many applications that roll their own crypto have done<br>
a poor job of it.<br>
<br>
It&#39;s also inadequate on purely technical grounds: Javascript provides<b=
r>
no mechanism for working with mlocked memory, =A0no mechanism to ensure<br>
that garbage collected data gets zeroized. =A0Your crypto app in JS could<b=
r>
easily have its long term keying material pulled out of free ram by<br>
malware long after it runs, or pulled off the disk from swap.<br>
The breakneck pace of fancy JIT systems makes it seem unlikely to me<br>
that javascript will provide for that any time soon.<br>
<div><div></div><br></div></blockquote></div><br>I guess what you are sayin=
g (and I believe everybody on this list agree) is that SDES-SRTP in combina=
tion with JavaScript signaling negotiation does not and cannot provide any =
meaningful security above preventing simple network sniffing. <br>
_____________<br>Roman Shpount<br>
<br>

--e89a8ff1c65a7415ad04bc6d7467--

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Sender: Fabio Pietrosanti <naif@infosecurity.ch>
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Subject: Re: [rtcweb] SDES-SRTP as a platform for multiple key management
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Hi Oscar,

i totally agree that it's not one of the stronger argument, however when
a technology is open and will let third party to work on it, third party
WILL work on it applying their creativity.

I am confident that we would see inside the crypto-nerds environments a
lot of JS-based key management extensions to extend SDES with customer
crypto semantic.

Then, for security purpose, this must be done outside the webapp, and
this can be done in many way such as:
- Javascript written browser plugin
- Javascript pinned to Browser Cache
- Locally loaded javascript

Javascript encryption is the next frontieer for Web security and there's
a lot of work from that side also from browser manufacturer (DOMCrypt,
JS-signing, JS RNG, etc) to provide secure crypto.

We are in 2012 and if we speak of an argument that Browser+Crypto we
just cannot not consider that Javascript will have a role in that
context, one way or another way.

Fabio

On 3/29/12 11:23 PM, Oscar Ohlsson wrote:
> Hi,
> 
> While I'm in favour of SDES I don't believe this is a particularly strong argument. To provide any real security gain, any additional signing or encryption that you may think of has to be performed outside of the webapp's control.
> 
> Regards,
> 
> Oscar
> 
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org 
>> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Fabio Pietrosanti (naif)
>> Sent: Thursday, March 29, 2012 9:54 PM
>> To: <rtcweb@ietf.org>
>> Subject: [rtcweb] SDES-SRTP as a platform for multiple key management
>>
>> Hi all,
>>
>> i've been thinking that one of the very interesting elements 
>> about the support of SDES-SRTP, is that, other than providing 
>> compatibility with existing telephony ecosystem, it may allow 
>> the implementation of custom key managegement systems.
>>
>> Basically if WebRTC would introduce support for SDES-SRTP and 
>> w3c would define API to handle SDES SDP call keys, it would 
>> become possible to further implement in Javascript additional 
>> key management systems.
>>
>> For example someone may implement a javascript application to 
>> be provided from an https source or browser extension 
>> additional to implement OpenPGPJS based identity verification 
>> (http://openpgpjs.org/) or integration with DH based key 
>> exchange (https://github.com/kaepora/cryptocat/).
>>
>> So basically a side-effect of introducing SDES-SRTP, could be to let
>> HTML5 application developers, to effectively be able to 
>> implement custom security mechanisms for voice applications.
>>
>> --
>> Fabio Pietrosanti
>> Founder, CTO
>>
>> Tel: +39 02 85961748 (direct)
>> Mobile: +39 340 1801049
>> E-mail: fabio.pietrosanti@privatewave.com
>> Skype: fpietrosanti
>> Linkedin: http://linkedin.com/in/secret
>>
>> PrivateWave Italia S.p.A.
>> Via Gaetano Giardino 1 - 20123 Milano - Italy 
>> www.privatewave.com _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb


From stefan.lk.hakansson@ericsson.com  Fri Mar 30 01:59:42 2012
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From: Stefan Hakansson LK <stefan.lk.hakansson@ericsson.com>
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Subject: [rtcweb] Interaction between MediaStream API and signaling
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The JS API has deals with MediaStreams (this is what you send and 
receive using PeerConnection from an application perspective).

A browser receiving RTP streams, needs side info to be able to assemble 
those RTP streams into MediaStreams in a correct way. The current model 
is that this is signaled using SDP exchanges (where Haralds MSID 
proposal would tell which MediaStream an RTP stream belongs to).

As I brought up at the mike yesterday, I think we may have a race 
condition for the responder.

For the initiator side browser, this is clear: once an (PR-)ANSWER is 
received, the responder has received the SDP, and hence can map incoming 
RTP streams into MediaStreams.

But for the responder side this is less clear to me. Imagine 
applications where the responder just mirrors the initiator - if one of 
the parties adds a MediaStream to PeerConnection, the other end would 
add the corresponding MediaStream.

This can happen any time in the session, so ICE can very well be up and 
running. One example could be that the data channel is used for text 
chat, when one side clicks a button to start video. And the application 
can have asked for permission to use all input devices earlier, so no 
user interaction may be involved.

In this situation the responder's (added) RTP streams can very well 
arrive before the ANSWER if I understand correctly. I think we need to 
find a way to handle this. One way is to add an "ACK" that indicates to 
the responder that the initiator has received the ANSWER, but I'm not 
sure that is the best way.

Stefan

From HKaplan@acmepacket.com  Fri Mar 30 02:43:53 2012
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On Mar 30, 2012, at 2:58 AM, Timothy B. Terriberry wrote:

> And the difficulty is, a browser has no way to tell (a priori) when the o=
ther end is another browser or a gateway.

You mean before the browser receives an offer/answer?  Obviously it won't, =
but why does that matter? (I must be missing something obvious)

-hadriel


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--047d7b33923db44d5d04bc732c11
Content-Type: text/plain; charset=ISO-8859-1

Note that even when you think you know the landscape of the IPR on
something, you can still get nasty surprises.

A quote from todays Rethink that brings this out:

"However, several key cases are now reaching final judgements, and
injunctions are more likely to be granted in that event. Motorola - soon to
be owned by Google - has asked for a ban on Microsoft's Xbox 360, Windows
and Internet Explorer products, which it claims infringe on its
H.264-related patents, if Motorola wins a case to be heard in Mannheim,
Germany on April 17. According to IPR expert Florian Mueller of FOSS
Patents, the judge is expected to favour Motorola."

--047d7b33923db44d5d04bc732c11
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<p>Note that even when you think you know the landscape of the IPR on somet=
hing, you can still get nasty surprises.</p>
<p>A quote from todays Rethink that brings this out:</p>
<p>&quot;However, several key cases are now reaching final judgements, and =
injunctions are more likely to be granted in that event. Motorola - soon to=
 be owned by Google - has asked for a ban on Microsoft&#39;s Xbox 360, Wind=
ows and Internet Explorer products, which it claims infringe on its H.264-r=
elated patents, if Motorola wins a case to be heard in Mannheim, Germany on=
 April 17. According to IPR expert Florian Mueller of FOSS Patents, the jud=
ge is expected to favour Motorola.&quot;</p>


--047d7b33923db44d5d04bc732c11--

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On 03/30/2012 06:19 AM, Dean Willis wrote:
>
> Note that even when you think you know the landscape of the IPR on
> something, you can still get nasty surprises.
>
> A quote from todays Rethink that brings this out:
>
> "However, several key cases are now reaching final judgements, and
> injunctions are more likely to be granted in that event. Motorola -
> soon to be owned by Google - has asked for a ban on Microsoft's Xbox
> 360, Windows and Internet Explorer products, which it claims infringe
> on its H.264-related patents, if Motorola wins a case to be heard in
> Mannheim, Germany on April 17. According to IPR expert Florian Mueller
> of FOSS Patents, the judge is expected to favour Motorola."
>
>
Anyone reporter who thinks Florian Mueller is an IPR expert is clueless,
demented, or grinding an axe.
                                - Jim


From basilgohar@librevideo.org  Fri Mar 30 08:07:46 2012
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On 03/30/2012 09:56 AM, Jim Gettys wrote:
> On 03/30/2012 06:19 AM, Dean Willis wrote:
>> Note that even when you think you know the landscape of the IPR on
>> something, you can still get nasty surprises.
>>
>> A quote from todays Rethink that brings this out:
>>
>> "However, several key cases are now reaching final judgements, and
>> injunctions are more likely to be granted in that event. Motorola -
>> soon to be owned by Google - has asked for a ban on Microsoft's Xbox
>> 360, Windows and Internet Explorer products, which it claims infringe
>> on its H.264-related patents, if Motorola wins a case to be heard in
>> Mannheim, Germany on April 17. According to IPR expert Florian Mueller
>> of FOSS Patents, the judge is expected to favour Motorola."
>>
>>
> Anyone reporter who thinks Florian Mueller is an IPR expert is clueless,
> demented, or grinding an axe.
>                                 - Jim
+1

-- 
Libre Video
http://librevideo.org


From gmaxwell@juniper.net  Fri Mar 30 08:39:29 2012
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From: Gregory Maxwell <gmaxwell@juniper.net>
To: Randell Jesup <randell-ietf@jesup.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Fri, 30 Mar 2012 08:37:57 -0700
Thread-Topic: [rtcweb] SRTP and "marketing"
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Randell Jesup [randell-ietf@jesup.org] wrote:
> That's not of no benefit. It's not of real
> benefit to J. Random Salesman, but even non-political people may worry
> about someone watching in more than they worry about listening in.
> (#include lawsuits over schools using laptop computers to take snapshots
> from student computers at home, etc, etc).

A point which needs to be emphasized is that undetectable attacks are
not at all the same thing as detectable attacks: Even when the chance
of detection is somewhat low, if the cost of detection is high the
possibility of it can be an effective deterrent.

Assuming 'hardcoded' ephemeral key agreement, Users Comparing
Tools->PageInfo->Media->session fingerprint =97 especially if documented
as a best practice in HowToBeParanoid documents=97 would be likely to
detect network level mass interception, even if the chance of detection
on a case by case is very low.  This makes the development and deployment
of _covert_ mass surveillance/censorship infrastructure, the sort with
the greatest negative human rights value, less attractive from a cost
benefit perspective.

(Of course, it does nothing for non-secret monitoring=97 but when the
monitoring is not secret the users of protcol have the most important
information they need in order to make an informed choice about how
they communicate, what public policies they live under, who they have
providing their network services, etc)

In computer and cryptographic security we're often concerned with absolute
protection. This is because absolute security is often achievable in this
area,  a luxury few other kinds of security have. But where absolute
security can't be provided (e.g. we _can't_ prevent MITM without hard to
provide and often missing identity service), relative security still has
value. When considering the real world context of attacks always being
a cost benefit trade-off the removal of nearly-undetectable attacks is as
an enormous relative improvement as is the removal of passive sniffing.


From salvatore.loreto@ericsson.com  Fri Mar 30 12:15:12 2012
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Hi Markus,

see in line

On 3/29/12 10:23 PM, Markus.Isomaki@nokia.com wrote:
> Hi,
>
> Some comments on the data channel proposal:
>
> 1. Congestion control: If no real-time sessions are on, TCP-style loss-driven congestion control is fine. However, if voice/video session is up, such congestion control would be disasterous to the real-time traffic quality if sendind or receiving more than a slight amount of data. Especially in mobile access networks the user would bloat his/her own buffers very easily. In that case, some less-than-best-effort congestion control algorithm á la LEDBAT would be required. It would be best to make this a MUST requirement on implementations, otherwise we will get a lot of crappy video even if the codec was better than H.261 :-)
good point worth to discuss

the congestion control algorithm in the actual SCTP open source userland 
implementation is a TCP-friendly congestion control.
As I said during the presentation we can change it if we see the reason 
not to use it (and you have already some good one)...
however it should be clear that at moment SCTP does not have any way to 
negotiate the congestion control algorithm to use in an association.

So whatever we choose it will be the one used in all the possible 
scenarios (e.g. voice/video session up or down, etc).

>
> 2. Multihoming and interface switching: I don't suggest going for the multipath support on the SCTP level. I think it would be best to deal with this through ICE in the same way as for RTP, i.e. adding and removing candidates as interfaces become available or go. Or let the application handle it by creating a new data channel and continue from the breakpoint. The most important requirement is that the application gets notified about these changes downstrairs so it can act.
that is exact the proposal in the data-channel draft.
Anyway you can not use multihoming because of the fact that SCTP runs on 
top of DTLS
>
>
> 3. HTTP tunneling: In practice we are going to need HTTP tunneling last-resort option for the data channel as well. If doing so, what will the protocol stack look like? Is it SCTP/DTLS/UDP/HTTP/TLS/TCP? Or can we collapse some of these layers. I think we'd better.

this is something to figure it out.
Eric proposal make sense to me;
another possibility would be use WebSocket when you have to tunneling 
data over HTTP/HTTPS.

Salvatore

-- 
Salvatore Loreto, PhD
www.sloreto.com


From randell-ietf@jesup.org  Fri Mar 30 13:46:55 2012
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On 3/30/2012 3:15 PM, Salvatore Loreto wrote:
> On 3/29/12 10:23 PM, Markus.Isomaki@nokia.com wrote:
>> Some comments on the data channel proposal:
>>
>> 1. Congestion control: If no real-time sessions are on, TCP-style
>> loss-driven congestion control is fine. However, if voice/video
>> session is up, such congestion control would be disasterous to the
>> real-time traffic quality if sendind or receiving more than a slight
>> amount of data. Especially in mobile access networks the user would
>> bloat his/her own buffers very easily. In that case, some
>> less-than-best-effort congestion control algorithm á la LEDBAT would
>> be required. It would be best to make this a MUST requirement on
>> implementations, otherwise we will get a lot of crappy video even if
>> the codec was better than H.261 :-)
> good point worth to discuss
>
> the congestion control algorithm in the actual SCTP open source userland
> implementation is a TCP-friendly congestion control.
> As I said during the presentation we can change it if we see the reason
> not to use it (and you have already some good one)...

Correct; please also see the presentation from the rtcweb Interim 
meeting in February that I gave; one issue it covered was congestion 
control.

We definitely want to both be able to balance bandwidth used by data 
channels versus media and to avoid data channels causing congestion that 
forces the media channels to degrade.

This could be done by:

a) developing a 'merged' congestion control regime for the media 
channels and the data channels.  As we're already getting a "total 
bandwidth" notification from the receiver, that calculation could 
include data channels, and let the sender allocate bits to media and the 
data channels according to that information.  In this scenario, the data 
channel code itself would be slaved to a wider congestion control.

b) run separate congestion control for the data channels based on a 
low-delay variant of classic TCP, such as Cx-TCP.  It so happens that 
the FreeBSD SCTP implementation includes a low-delay congestion 
algorithm based on an ancestor of the current Cx-TCP work.

Cx-TCP (for links see the Interim slides) runs in a low-delay fashion 
unless it's fighting with long-lived/saturating TCP flows, where it will 
transition into a loss-based behavior (with longer delay), and 
transition back if the TCP flows drop below bottleneck bandwidth.

c) run classic loss-based congestion control (the default), but put an 
external bandwidth limiter on it controlled by the main application and 
congestion code, limiting the data channels to a specified amount or 
fraction of the media channels.

d) run classic loss-based congestion control and let it compete with 
media; don't send large amounts of data, or pace it in the application. 
  Or live with the data channel killing your media.

> however it should be clear that at moment SCTP does not have any way to
> negotiate the congestion control algorithm to use in an association.

Correct, though we could fix that - and we could give the JS control and 
let the app decide, since data channels are largely within-app 
proprietary connections.


>> 2. Multihoming and interface switching: I don't suggest going for the
>> multipath support on the SCTP level. I think it would be best to deal
>> with this through ICE in the same way as for RTP, i.e. adding and
>> removing candidates as interfaces become available or go. Or let the
>> application handle it by creating a new data channel and continue from
>> the breakpoint. The most important requirement is that the application
>> gets notified about these changes downstrairs so it can act.
> that is exact the proposal in the data-channel draft.
> Anyway you can not use multihoming because of the fact that SCTP runs on
> top of DTLS

Correct, though there was some in-room discussion of running it over 
multiple DTLS connections to support smooth handoff across interfaces 
(think mobile and 4g<->WiFi switchover).


-- 
Randell Jesup
randell-ietf@jesup.org

From randell-ietf@jesup.org  Fri Mar 30 14:42:46 2012
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On 3/30/2012 4:59 AM, Stefan Hakansson LK wrote:
> The JS API has deals with MediaStreams (this is what you send and 
> receive using PeerConnection from an application perspective).
>
> A browser receiving RTP streams, needs side info to be able to 
> assemble those RTP streams into MediaStreams in a correct way. The 
> current model is that this is signaled using SDP exchanges (where 
> Haralds MSID proposal would tell which MediaStream an RTP stream 
> belongs to).
>
> As I brought up at the mike yesterday, I think we may have a race 
> condition for the responder.
>
> For the initiator side browser, this is clear: once an (PR-)ANSWER is 
> received, the responder has received the SDP, and hence can map 
> incoming RTP streams into MediaStreams.
>
> But for the responder side this is less clear to me. Imagine 
> applications where the responder just mirrors the initiator - if one 
> of the parties adds a MediaStream to PeerConnection, the other end 
> would add the corresponding MediaStream.
>
> This can happen any time in the session, so ICE can very well be up 
> and running. One example could be that the data channel is used for 
> text chat, when one side clicks a button to start video. And the 
> application can have asked for permission to use all input devices 
> earlier, so no user interaction may be involved.
>
> In this situation the responder's (added) RTP streams can very well 
> arrive before the ANSWER if I understand correctly. 

Yes.  Just like in SIP.  And so when you send an OFFER (or modified 
re-OFFER), you must be ready to receive data per that offer even if no 
ANSWER has been received - just like in SIP.  And if its a re-offer, you 
need to accept the old, and accept the new (though you could probably 
use reception of obviously new-OFFER media to turn off 
decoding/rendering old-OFFER in preparation for the ANSWER).

The flip side of this is the responder has to infer when the sender 
switches over to the result of the ANSWER from the media.  For example:

A                                      B
<--- H.261 --->
re-OFFER(VP8) --->
<-- ANSWER(VP8) (delayed in reception)
<-----------VP8            (A should infer that B ANSWERed and accepted VP8)
  ----------> H.261
<-- ANSWER(VP8) (received)
<--------VP8----------> (B should infer by reception of VP8 that ANSWER 
was received)

(Personally, I hate inferences, but without a 3 (or 4) way handshake, 
you have to).  If you switches of codecs are staged, then this isn't 
(much) of a problem.  Either leave old codec on the list, or leave it on 
the list until accept, and then re-OFFER to remove the un-used codec.

One problem is what to do in the switchover window when you might get a 
mixture of old and new media, especially if you moved them to different 
ports and so can't count on RTP sequence re-ordering to un-mix them; in 
the past I dealt with that (and long codec-switch times) by locking out 
codec changes for a fraction of a second after I do one.  Not a huge 
deal, however.

My apologies if I've missed something in JSEP; I've been heads-down 
enough in Data Channels and bring-up that I could have a disconnect here 
and be saying something silly.

> I think we need to find a way to handle this. One way is to add an 
> "ACK" that indicates to the responder that the initiator has received 
> the ANSWER, but I'm not sure that is the best way.

If you need to know that, you need a SIP-style ACK.

-- 
Randell Jesup
randell-ietf@jesup.org


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On 3/29/2012 5:59 PM, Gregory Maxwell wrote:
> Oscar Ohlsson [oscar.ohlsson@ericsson.com] wrote:
>> That's why I wrote "the entire webapp" below. If it was not clear I meant that the
>> - main HTML page
>> - all external CSS files, JavaScript files, images, etc
>> - all XmlHttpRequests
>> - all WebSocket connections
>> are protected with TLS. This is obviously verifiable and it's a feature supported by all modern browsers (no mixed content).
>
> Even this is insufficient.

[long nice analysis of security in JS vs in the stack deleted]

>  Really, cryptographic
> negotiation is not properly an application feature, it belongs lower in
> the stack, and many applications that roll their own crypto have done
> a poor job of it.
>
> It's also inadequate on purely technical grounds: Javascript provides
> no mechanism for working with mlocked memory,  no mechanism to ensure
> that garbage collected data gets zeroized.  Your crypto app in JS could
> easily have its long term keying material pulled out of free ram by
> malware long after it runs, or pulled off the disk from swap.
> The breakneck pace of fancy JIT systems makes it seem unlikely to me
> that javascript will provide for that any time soon.

I rarely do this sort of reply for obvious reasons, but:

I agree!  :-)

-- 
Randell Jesup
randell-ietf@jesup.org

From stefan.lk.hakansson@ericsson.com  Fri Mar 30 22:01:51 2012
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On 03/30/2012 04:26 PM, Jim Barnett wrote:
> Doesn't the basic model require that the offer be ready to receive media
> as soon as it sends the offer?  If that's the case then the offer must
> be ready and able to map RTP streams to media streams (using the mapping
> specified in the SDP for its offer) as soon as it sends the offer.

Yes, the basic SDP o/a model requires that the offerer is ready to 
receive media as soon as the offer is sent. But as I tried to explain 
(but obviously failed to) is that there is an additional constraint 
here: the JS application deals with MediaStreams, which often has more 
than one track, e.g. one video and one audio track. A track in turn 
would correspond to an RTP stream. And you can have several concurrent 
MediaStreams in action.

Say now that the offerer, before getting an ANSWER with info, receives 
three RTP streams; two video and one audio. The offerer would in this 
situation not have a clue how these belong together. One could be video 
from a front facing camera (that is supposed to be displayed on a large 
video screen), the other video RTP stream could contain the video from 
the rear facing camera and is supposed to together with the audio RTP 
stream be one MediaStream that is supposed to be played in sync in a 
thumbnail video display.

Without the offerer having the info about what these RTP streams 
represent it can't do anything sensible.

Stefan

>
> - Jim
>
> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
> Of Stefan Hakansson LK
> Sent: Friday, March 30, 2012 5:00 AM
> To: rtcweb@ietf.org
> Subject: [rtcweb] Interaction between MediaStream API and signaling
>
> The JS API has deals with MediaStreams (this is what you send and
> receive using PeerConnection from an application perspective).
>
> A browser receiving RTP streams, needs side info to be able to assemble
> those RTP streams into MediaStreams in a correct way. The current model
> is that this is signaled using SDP exchanges (where Haralds MSID
> proposal would tell which MediaStream an RTP stream belongs to).
>
> As I brought up at the mike yesterday, I think we may have a race
> condition for the responder.
>
> For the initiator side browser, this is clear: once an (PR-)ANSWER is
> received, the responder has received the SDP, and hence can map incoming
> RTP streams into MediaStreams.
>
> But for the responder side this is less clear to me. Imagine
> applications where the responder just mirrors the initiator - if one of
> the parties adds a MediaStream to PeerConnection, the other end would
> add the corresponding MediaStream.
>
> This can happen any time in the session, so ICE can very well be up and
> running. One example could be that the data channel is used for text
> chat, when one side clicks a button to start video. And the application
> can have asked for permission to use all input devices earlier, so no
> user interaction may be involved.
>
> In this situation the responder's (added) RTP streams can very well
> arrive before the ANSWER if I understand correctly. I think we need to
> find a way to handle this. One way is to add an "ACK" that indicates to
> the responder that the initiator has received the ANSWER, but I'm not
> sure that is the best way.
>
> Stefan
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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On 03/30/2012 11:39 PM, Randell Jesup wrote:
> On 3/30/2012 4:59 AM, Stefan Hakansson LK wrote:
>> The JS API has deals with MediaStreams (this is what you send and
>> receive using PeerConnection from an application perspective).
>>
>> A browser receiving RTP streams, needs side info to be able to
>> assemble those RTP streams into MediaStreams in a correct way. The
>> current model is that this is signaled using SDP exchanges (where
>> Haralds MSID proposal would tell which MediaStream an RTP stream
>> belongs to).
>>
>> As I brought up at the mike yesterday, I think we may have a race
>> condition for the responder.
>>
>> For the initiator side browser, this is clear: once an (PR-)ANSWER is
>> received, the responder has received the SDP, and hence can map
>> incoming RTP streams into MediaStreams.
>>
>> But for the responder side this is less clear to me. Imagine
>> applications where the responder just mirrors the initiator - if one
>> of the parties adds a MediaStream to PeerConnection, the other end
>> would add the corresponding MediaStream.
>>
>> This can happen any time in the session, so ICE can very well be up
>> and running. One example could be that the data channel is used for
>> text chat, when one side clicks a button to start video. And the
>> application can have asked for permission to use all input devices
>> earlier, so no user interaction may be involved.
>>
>> In this situation the responder's (added) RTP streams can very well
>> arrive before the ANSWER if I understand correctly.
>
> Yes.  Just like in SIP.  And so when you send an OFFER (or modified
> re-OFFER), you must be ready to receive data per that offer even if no
> ANSWER has been received - just like in SIP.  And if its a re-offer, you
> need to accept the old, and accept the new (though you could probably
> use reception of obviously new-OFFER media to turn off
> decoding/rendering old-OFFER in preparation for the ANSWER).
>
> The flip side of this is the responder has to infer when the sender
> switches over to the result of the ANSWER from the media.  For example:
>
> A                                      B
> <--- H.261 --->
> re-OFFER(VP8) --->
> <-- ANSWER(VP8) (delayed in reception)
> <-----------VP8            (A should infer that B ANSWERed and accepted VP8)
>    ---------->  H.261
> <-- ANSWER(VP8) (received)
> <--------VP8---------->  (B should infer by reception of VP8 that ANSWER
> was received)
>
> (Personally, I hate inferences, but without a 3 (or 4) way handshake,
> you have to).  If you switches of codecs are staged, then this isn't
> (much) of a problem.  Either leave old codec on the list, or leave it on
> the list until accept, and then re-OFFER to remove the un-used codec.

I think I understand what you mean, and this would work fine as long as 
you just switch codecs that are used in already set-up MediaStreams.

But if A in this case, as part of re-OFFERING the session, not only 
offers a new codec (VP8) for the already flowing video but also adds a 
new outgoing video stream (e.g. front cam), and then (without receiving 
the ANSWER - delayed in reception) starts receiving VP8 video it could 
not really know if this VP8 video is new video from the responders front 
cam or just a new codec for the existing (back cam) video from the 
responder to the sender.

>
> One problem is what to do in the switchover window when you might get a
> mixture of old and new media, especially if you moved them to different
> ports and so can't count on RTP sequence re-ordering to un-mix them; in
> the past I dealt with that (and long codec-switch times) by locking out
> codec changes for a fraction of a second after I do one.  Not a huge
> deal, however.
>
> My apologies if I've missed something in JSEP; I've been heads-down
> enough in Data Channels and bring-up that I could have a disconnect here
> and be saying something silly.

Actually I don't think this is very JSEP related; it is the generic 
problem that the browser receiving RTP streams need some side info about 
them before being able to do anything sensible with them.

>
>> I think we need to find a way to handle this. One way is to add an
>> "ACK" that indicates to the responder that the initiator has received
>> the ANSWER, but I'm not sure that is the best way.
>
> If you need to know that, you need a SIP-style ACK.

As explained, I do think we need to know that.

>


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On 03/31/2012 07:17 AM, Stefan Hakansson LK wrote:
> On 03/30/2012 11:39 PM, Randell Jesup wrote:
>> On 3/30/2012 4:59 AM, Stefan Hakansson LK wrote:
>>> The JS API has deals with MediaStreams (this is what you send and
>>> receive using PeerConnection from an application perspective).
>>>
>>> A browser receiving RTP streams, needs side info to be able to
>>> assemble those RTP streams into MediaStreams in a correct way. The
>>> current model is that this is signaled using SDP exchanges (where
>>> Haralds MSID proposal would tell which MediaStream an RTP stream
>>> belongs to).
>>>
>>> As I brought up at the mike yesterday, I think we may have a race
>>> condition for the responder.
>>>
>>> For the initiator side browser, this is clear: once an (PR-)ANSWER is
>>> received, the responder has received the SDP, and hence can map
>>> incoming RTP streams into MediaStreams.
>>>
>>> But for the responder side this is less clear to me. Imagine
>>> applications where the responder just mirrors the initiator - if one
>>> of the parties adds a MediaStream to PeerConnection, the other end
>>> would add the corresponding MediaStream.
>>>
>>> This can happen any time in the session, so ICE can very well be up
>>> and running. One example could be that the data channel is used for
>>> text chat, when one side clicks a button to start video. And the
>>> application can have asked for permission to use all input devices
>>> earlier, so no user interaction may be involved.
>>>
>>> In this situation the responder's (added) RTP streams can very well
>>> arrive before the ANSWER if I understand correctly.
>>
>> Yes.  Just like in SIP.  And so when you send an OFFER (or modified
>> re-OFFER), you must be ready to receive data per that offer even if no
>> ANSWER has been received - just like in SIP.  And if its a re-offer, you
>> need to accept the old, and accept the new (though you could probably
>> use reception of obviously new-OFFER media to turn off
>> decoding/rendering old-OFFER in preparation for the ANSWER).
>>
>> The flip side of this is the responder has to infer when the sender
>> switches over to the result of the ANSWER from the media.  For example:
>>
>> A                                      B
>> <--- H.261 --->
>> re-OFFER(VP8) --->
>> <-- ANSWER(VP8) (delayed in reception)
>> <-----------VP8            (A should infer that B ANSWERed and accepted VP8)
>>     ---------->   H.261
>> <-- ANSWER(VP8) (received)
>> <--------VP8---------->   (B should infer by reception of VP8 that ANSWER
>> was received)
>>
>> (Personally, I hate inferences, but without a 3 (or 4) way handshake,
>> you have to).  If you switches of codecs are staged, then this isn't
>> (much) of a problem.  Either leave old codec on the list, or leave it on
>> the list until accept, and then re-OFFER to remove the un-used codec.
>
> I think I understand what you mean, and this would work fine as long as
> you just switch codecs that are used in already set-up MediaStreams.
>
> But if A in this case, as part of re-OFFERING the session, not only
> offers a new codec (VP8) for the already flowing video but also adds a
> new outgoing video stream (e.g. front cam), and then (without receiving
> the ANSWER - delayed in reception) starts receiving VP8 video it could
> not really know if this VP8 video is new video from the responders front
> cam or just a new codec for the existing (back cam) video from the
> responder to the sender.

This may have been a very bad example. Probably you can tell them apart 
on the SSRC. But even so, the A browser won't know what the VP8 stream 
(if it has an unknown SSRC) represents without receiving the ANSWER.

>
>>
>> One problem is what to do in the switchover window when you might get a
>> mixture of old and new media, especially if you moved them to different
>> ports and so can't count on RTP sequence re-ordering to un-mix them; in
>> the past I dealt with that (and long codec-switch times) by locking out
>> codec changes for a fraction of a second after I do one.  Not a huge
>> deal, however.
>>
>> My apologies if I've missed something in JSEP; I've been heads-down
>> enough in Data Channels and bring-up that I could have a disconnect here
>> and be saying something silly.
>
> Actually I don't think this is very JSEP related; it is the generic
> problem that the browser receiving RTP streams need some side info about
> them before being able to do anything sensible with them.
>
>>
>>> I think we need to find a way to handle this. One way is to add an
>>> "ACK" that indicates to the responder that the initiator has received
>>> the ANSWER, but I'm not sure that is the best way.
>>
>> If you need to know that, you need a SIP-style ACK.
>
> As explained, I do think we need to know that.
>
>>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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Subject: Re: [rtcweb] DTLS-SRTP implementation diffusion: Why not SDES-SRTP?
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On 3/29/2012 3:05 AM, Fabio Pietrosanti (naif) wrote:
> On 3/28/12 9:53 PM, IÃ±aki Baz Castillo wrote:
>> 2012/3/28 Fabio Pietrosanti (naif)<lists@infosecurity.ch>:
>>> Hi all,
>>>
>>> i read that 80% of Sipit participant support SDES-SRTP but 0% support
>>> DTLS-SRTP  https://www.sipit.net/SIPit29_summary .
>>>
>>> At SIPit there were 34 attendees from 17 companies visiting from 12
>>> countries with 25 distinct VoIP implementations.
>>
>> Right, but this is rtcweb, not SIP.
>>
>>
>>
>>> I do not really see which is the rationale in making DTLS-SRTP mandatory
>>> while plain SRTP with SDES key exchange is already so well know and used.
>>
>> That's a good reason to *also* allow (and mandate) SDES-SRTP support
>> in WebRTC clients, much better than the interoperability with SIP
>> (again: this is rtcweb, not SIP world).
>
> That's true, but it's also true that the "rtcweb world" will strictly
> inter-operate with the "sip world".

The meaning of this statement is unclear.  If you mean "some rtcweb 
clients will connect to SIP gateways", or "some rtcweb clients will 
translate JSEP into SIP and send it to a SIP server", then that's true. 
  Some (many) rtcweb clients will never talk to any sip equipment.

> It would be reasonable to expect that current existing PBX software
> would evolve also with support for Rtcweb, to provide Web phone systems.

Which is no problem, the PBX can act as a WebRTC server, and provide the 
WebRTC JS app to the browsers, and also act as a gateway to SIP. 
There's no reason it can't terminate DTLS-SRTP for calls that connect to 
SIP devices.

> In particular all opensource software will setup the path for the
> adoption of the standard, as we know history will repeat.
>
> It appear me as natural behavior of diffusion of implementation, and for
> that reason i see the need to "easily" inter-operate with the SIP world
> is a key value point.

Being able to gateway to SIP and to PSTN are both useful abilities for 
certain (popular) classes of potential apps.  Note the earlier 
discussions showed that one requirement (consent) makes it hard (though 
not impossible) to directly connect WebRTC clients to PSTN gateways or 
to other SIP devices, even if you allow SDES.  If consent forces you 
through a gateway (or some device that knows WebRTC), DTLS-SRTP-only 
becomes a much smaller hump.

> Creating an incompatible "media format" would require a lot of more
> effort because the "amount of compatibility testing to be done with
> DTLS-SRTP will be significantly higher than SDES-SRTP" .
>
> So it would provide an advantage for the *few vendor* supporting it,
> practically introducing a *technological entrance barrier*.

Given the expectation that there will be a quality open-source version 
available (and Mozilla's implementation is of course open source), the 
barrier should be quite low.

> This is a bad practice already see in other standard bodies.
>
>>
>>
>>> Anyone can provide some very strong and valuable point about using
>>> DTLS-SRTP (considering it's weak diffusion and incompatibility risks)?
>>
>> Lot of recent threads about this topic in this maillist. But also
>> check a recent presentation (yesterday in IETF Pairs):
>>
>> http://tools.ietf.org/agenda/83/slides/slides-83-rtcweb-3.pdf
>
> I would like to strongly argue against the SLIDE 3 statement that
> "DTLS-SRTP meets RTCWEB's technical requirements" .
>
> DTLS-SRTP doesn't meet the RTCWEB's technical requirements because:
>
> - It does NOT provide inter-operation with existing SIP endpoints

That is not an rtcweb requirement.  The only mention of SIP in the 
use-case and requirements doc is the ability to use SIP to provide 
federation (F27).

> This is confirmed by the October 2011 SIPit data, with 80% of
> participants supporting, with good interoperability, SDES but with 0%
> supporting DTLS-SRTP .

As discussed, this is an only partially-relevant fact. SIPit is 
self-selected; most devices there are not representative of average 
devices in use, and devices deployed may not enable options used in 
SIPit tests.

> In order to try to "try to improve it's non-interoperability issue" the
> DTLS-SRTP is re-proposing him-selves as DTLS-SRTP-EKT .

I wasn't particularly enthused about EKT.

> To speaking about the fact that even EKT draft explain that:
> " Today, Security Descriptions [RFC4568] is used for distributing SRTP
>     keys in several different IP PBX systems and is expected to be used
>     by 3GPP's Long Term Evolution (LTE). "
>
> http://tools.ietf.org/html/draft-ietf-avt-srtp-ekt-01#section-6.1

I have no belief that LTE devices will ever connect directly to WebRTC 
devices directly (even if they're the same device); that's just not how 
they work.

> So:
> - Internt is using SDES-SRTP
> - 3GPP LTE is using SDES-SRTP
> - WebRTC is going to use DTLS-SRTP
>
> I do not really see how we can rationally accept to follow this
> different direction.
>
> I mean, we are discussing about a "new standard" that's based on "new
> technologies" rather than using existing, widely implemented technology.
>
> Do we really understand how much effort we are going to cause on the
> overall technological and security ecosystem by selecting DTLS-SRTP
> rather than SDES-SRTP?
>
> All US Federal Government will not be able to use WebRTC because NSA
> standardized SDES-SRTP for use in Classified communication:
> http://www.nsa.gov/ia/programs/mobility_program/index.shtml

They realize SDES is fundamentally insecure, and secure it by running it 
over a VPN link to their protected servers (and my understanding is that 
they consider even that a compromise).  And the devices must be TOTALLY 
locked down to any unencrypted network access.  These will not be using 
WebRTC anytime soon, under any circumstance.

> The argument that DTLS is *more secure* must face the reality that
> no-one is using it and that SDES-SRTP is *widely diffused and
> interoperable*.

Widely diffused and interoperable may be (mostly) true, though I'll tell 
you that probably 99.x% of *deployed* VoIP devices do not *use* SDES 
(even if they support it), so arguments about deployment numbers are 
mostly irrelevant.  Not totally irrelevant, of course.

> All Internet operators will have to introduce Protocol Gateway.

Due to the 'consent' requirement and other issues, this is likely 
anyways.  And who are the "internet operators" in your estimation?

> All mobile operators will have to introduce Protocol Gateway.

They will anyways I believe.

> All that subjects, if using just SDES-SRTP, would just need to "update
> the software they already use to run VoIP infrastructure" with no need
> to handle modification to the Media for Interworking.
>
> So definitely the choice to go for DTLS-SRTP is imho a wrong choice,
> against any rationale for the diffusion of WebRTC standard, introducing
> artificially complexity where it may be possible to keep it simple.

There certainly is an argument for SDES.  I don't agree about what it 
leads us to, but it's a reasoned argument.  However, many of your 
assertions here I would not accept.

-- 
Randell Jesup
randell-ietf@jesup.org

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