
From alan.b.johnston@gmail.com  Wed Jun  1 05:38:06 2011
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Date: Wed, 1 Jun 2011 07:10:01 -0500
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From: Alan Johnston <alan.b.johnston@gmail.com>
To: Justin Uberti <juberti@google.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>, Phil Zimmermann <prz@mit.edu>
Subject: Re: [rtcweb] Fwd: I-D Action: draft-johnston-rtcweb-media-privacy-00.txt
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Justin,

Thanks for your comment.  You are correct, the mapping of any
offer/answer exchange will need to be mappable to SDP, so this is
possible.

- Alan -

On Tue, May 31, 2011 at 10:41 PM, Justin Uberti <juberti@google.com> wrote:
> Interesting read. One comment - I believe the statement in this draft
> "In the RTCWEB architecture, there is no standardization of the signaling
> protocol possible, and hence this [SRTP SDES] approach can not be used"
> is not completely accurate. While the syntax of the signaling protocol (e=
.g.
> HTTP, XMPP, SIP) is left up to the web application, we must have a clear
> definition of the protocol semantics, as well as the format of session
> descriptions (as understood by the browser), so that they can be mapped t=
o
> SDP / XMPP if desired.
> Therefore there is no reason we cannot include SRTP keying information in
> these session descriptions.
>
> On Tue, May 31, 2011 at 8:21 PM, Alan Johnston <alan.b.johnston@gmail.com=
>
> wrote:
>>
>> All,
>>
>> Here is a new draft with some thoughts on media privacy in RTCWEB. =A0We
>> discuss the challenge of providing media privacy for browser users
>> when there is no standardization of the signaling protocol. =A0We also
>> discuss the need to sometimes allow trusted Man in the Middle (MitM)
>> servers to provide media services. We also discuss how ZRTP ("Media
>> Path Key Agreement for Unicast Secure RTP" RFC 6189) meets these
>> requirements.
>>
>> Comments most welcome.
>>
>> - Alan -
>>
>>
>> ---------- Forwarded message ----------
>> From: =A0<internet-drafts@ietf.org>
>> Date: Tue, May 31, 2011 at 10:15 PM
>> Subject: I-D Action: draft-johnston-rtcweb-media-privacy-00.txt
>> To: i-d-announce@ietf.org
>>
>>
>> A New Internet-Draft is available from the on-line Internet-Drafts
>> directories.
>>
>> =A0 =A0 =A0 =A0Title =A0 =A0 =A0 =A0 =A0 : RTCWEB Media Privacy
>> =A0 =A0 =A0 =A0Author(s) =A0 =A0 =A0 : Alan Johnston
>> =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0Philip Zimmermann
>> =A0 =A0 =A0 =A0Filename =A0 =A0 =A0 =A0: draft-johnston-rtcweb-media-pri=
vacy-00.txt
>> =A0 =A0 =A0 =A0Pages =A0 =A0 =A0 =A0 =A0 : 7
>> =A0 =A0 =A0 =A0Date =A0 =A0 =A0 =A0 =A0 =A0: 2011-05-31
>>
>> =A0 RTCWEB is the joint effort between the IETF and the W3C to add real-
>> =A0 time voice, video, and communication capabilities to browsers. =A0Th=
is
>> =A0 document looks at the requirements for media privacy and existing
>> =A0 mechanisms.
>>
>>
>> A URL for this Internet-Draft is:
>>
>> http://www.ietf.org/internet-drafts/draft-johnston-rtcweb-media-privacy-=
00.txt
>>
>> Internet-Drafts are also available by anonymous FTP at:
>> ftp://ftp.ietf.org/internet-drafts/
>>
>> This Internet-Draft can be retrieved at:
>>
>> ftp://ftp.ietf.org/internet-drafts/draft-johnston-rtcweb-media-privacy-0=
0.txt
>> _______________________________________________
>> I-D-Announce mailing list
>> I-D-Announce@ietf.org
>> https://www.ietf.org/mailman/listinfo/i-d-announce
>> Internet-Draft directories: http://www.ietf.org/shadow.html
>> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

From john.lazzaro@gmail.com  Thu Jun  2 11:56:48 2011
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From: John Lazzaro <john.lazzaro@gmail.com>
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Subject: [rtcweb] rtcweb use cases for audio/rtp-midi
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Hi everyone,

Just a note to let everyone know I'm
monitoring this list, in case there is
interest in adding audio/rtp-midi to the
list of supported payload formats for rtcweb.

I can see a few reasons why audio/rtp-midi
support may be a good idea:

[1] There are now hundreds of iOS apps that
use rtp-midi via Apple's Network MIDI protocol,
to let iPads and iPhones serve as touch-screen
musical instrument and mixing console controllers.
One could imagine wanting platform-independent
versions of these applications in web browsers.

[2] Network musical performance sessions, as
described in the introduction section to RFC 4695
(soon to be obsoleted by RFC 6295).

[3] Both games and telephony have, over the years,
found it useful to have a simple General MIDI or
DLS synthesis engine around, to create low-bitrate
music in lieu of streaming audio files.  One could
imagine web applications benefitting from this as well,
and over-the-network MIDI control being a requirement.

[4] Platforms such as ChromeOS, which only have
a web browser, would benefit from having standardized
real-time MIDI control as a web standard, even if
network use is not of interest.  In this use case,
one would want to plug a MIDI keyboard into the
ChromeOS device USB port, and use the keyboard
to control a music synthesizer locally running as a
web app -- the sort of infrastructure that would need
to be around to make apps [1], [2], [3] above work best.

--

That all said, its not clear to me that it makes sense
for me to write an I-D describing this as a use case
for the first-generation rtcweb.  There are so many
challenges to reaching consensus and interoperability
for plain-vanilla audio-video teleconferencing, that I don't
think adding "nice-to-haves" like MIDI to the manifest
is prudent. So I was planning to just watch and wait.

But, if anyone out there thinks waiting is not a good
idea, feel free to speak up, and if there's sufficient
interest I'll write up an I-D on the topic.  Thanks,

-- 
John Lazzaro
http://www.cs.berkeley.edu/~lazzaro
john [dot] lazzaro [at] gmail [dot] com

From harald@alvestrand.no  Sat Jun  4 22:55:47 2011
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Subject: Re: [rtcweb] rtcweb use cases for audio/rtp-midi
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This (midi in browsers) seems to be closely related to what the W3C 
"audio" group is doing.
Are you in contact with them?

                        Harald

On 06/02/11 20:56, John Lazzaro wrote:
> Hi everyone,
>
> Just a note to let everyone know I'm
> monitoring this list, in case there is
> interest in adding audio/rtp-midi to the
> list of supported payload formats for rtcweb.
>
> I can see a few reasons why audio/rtp-midi
> support may be a good idea:
>
> [1] There are now hundreds of iOS apps that
> use rtp-midi via Apple's Network MIDI protocol,
> to let iPads and iPhones serve as touch-screen
> musical instrument and mixing console controllers.
> One could imagine wanting platform-independent
> versions of these applications in web browsers.
>
> [2] Network musical performance sessions, as
> described in the introduction section to RFC 4695
> (soon to be obsoleted by RFC 6295).
>
> [3] Both games and telephony have, over the years,
> found it useful to have a simple General MIDI or
> DLS synthesis engine around, to create low-bitrate
> music in lieu of streaming audio files.  One could
> imagine web applications benefitting from this as well,
> and over-the-network MIDI control being a requirement.
>
> [4] Platforms such as ChromeOS, which only have
> a web browser, would benefit from having standardized
> real-time MIDI control as a web standard, even if
> network use is not of interest.  In this use case,
> one would want to plug a MIDI keyboard into the
> ChromeOS device USB port, and use the keyboard
> to control a music synthesizer locally running as a
> web app -- the sort of infrastructure that would need
> to be around to make apps [1], [2], [3] above work best.
>
> --
>
> That all said, its not clear to me that it makes sense
> for me to write an I-D describing this as a use case
> for the first-generation rtcweb.  There are so many
> challenges to reaching consensus and interoperability
> for plain-vanilla audio-video teleconferencing, that I don't
> think adding "nice-to-haves" like MIDI to the manifest
> is prudent. So I was planning to just watch and wait.
>
> But, if anyone out there thinks waiting is not a good
> idea, feel free to speak up, and if there's sufficient
> interest I'll write up an I-D on the topic.  Thanks,
>


From harald@alvestrand.no  Sun Jun  5 01:31:50 2011
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From: Harald Alvestrand <harald@alvestrand.no>
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Subject: [rtcweb] Fwd: I-D Action: draft-alvestrand-rtcweb-overview-00.txt
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This is a multi-part message in MIME format.
--------------000603010305050308010704
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Content-Transfer-Encoding: 7bit

This is my contribution to the interim meeting on Wednesday, and what 
I'll expect to be discussing under the heading of "architecture".

I will ask the chairs to adopt this as a working group document.

                       Harald

-------- Original Message --------
Subject: 	I-D Action: draft-alvestrand-rtcweb-overview-00.txt
Date: 	Sat, 04 Jun 2011 23:15:16 -0700
From: 	internet-drafts@ietf.org
Reply-To: 	internet-drafts@ietf.org
To: 	i-d-announce@ietf.org



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

	Title           : Overview: Real Time Protocols for Brower-based Applications
	Author(s)       : Harald T. Alvestrand
	Filename        : draft-alvestrand-rtcweb-overview-00.txt
	Pages           : 14
	Date            : 2011-06-04

    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.



A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-alvestrand-rtcweb-overview-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-alvestrand-rtcweb-overview-00.txt
_______________________________________________
I-D-Announce mailing list
I-D-Announce@ietf.org
https://www.ietf.org/mailman/listinfo/i-d-announce
Internet-Draft directories: http://www.ietf.org/shadow.html
or ftp://ftp.ietf.org/ietf/1shadow-sites.txt



--------------000603010305050308010704
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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">
    This is my contribution to the interim meeting on Wednesday, and
    what I'll expect to be discussing under the heading of
    "architecture".<br>
    <br>
    I will ask the chairs to adopt this as a working group document.<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>
    -------- 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>I-D Action: draft-alvestrand-rtcweb-overview-00.txt</td>
        </tr>
        <tr>
          <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Date: </th>
          <td>Sat, 04 Jun 2011 23:15:16 -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">Reply-To:
          </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:i-d-announce@ietf.org">i-d-announce@ietf.org</a></td>
        </tr>
      </tbody>
    </table>
    <br>
    <br>
    <pre>A New Internet-Draft is available from the on-line Internet-Drafts directories.

	Title           : Overview: Real Time Protocols for Brower-based Applications
	Author(s)       : Harald T. Alvestrand
	Filename        : draft-alvestrand-rtcweb-overview-00.txt
	Pages           : 14
	Date            : 2011-06-04

   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.



A URL for this Internet-Draft is:
<a class="moz-txt-link-freetext" href="http://www.ietf.org/internet-drafts/draft-alvestrand-rtcweb-overview-00.txt">http://www.ietf.org/internet-drafts/draft-alvestrand-rtcweb-overview-00.txt</a>

Internet-Drafts are also available by anonymous FTP at:
<a class="moz-txt-link-freetext" href="ftp://ftp.ietf.org/internet-drafts/">ftp://ftp.ietf.org/internet-drafts/</a>

This Internet-Draft can be retrieved at:
<a class="moz-txt-link-freetext" href="ftp://ftp.ietf.org/internet-drafts/draft-alvestrand-rtcweb-overview-00.txt">ftp://ftp.ietf.org/internet-drafts/draft-alvestrand-rtcweb-overview-00.txt</a>
_______________________________________________
I-D-Announce mailing list
<a class="moz-txt-link-abbreviated" href="mailto:I-D-Announce@ietf.org">I-D-Announce@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/i-d-announce">https://www.ietf.org/mailman/listinfo/i-d-announce</a>
Internet-Draft directories: <a class="moz-txt-link-freetext" href="http://www.ietf.org/shadow.html">http://www.ietf.org/shadow.html</a>
or <a class="moz-txt-link-freetext" href="ftp://ftp.ietf.org/ietf/1shadow-sites.txt">ftp://ftp.ietf.org/ietf/1shadow-sites.txt</a>

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

--------------000603010305050308010704--

From john.lazzaro@gmail.com  Sun Jun  5 09:21:13 2011
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Date: Sun, 5 Jun 2011 09:21:07 -0700
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From: John Lazzaro <john.lazzaro@gmail.com>
To: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] rtcweb use cases for audio/rtp-midi
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On Sat, Jun 4, 2011 at 10:55 PM, Harald Alvestrand <harald@alvestrand.no> wrote:
> This (midi in browsers) seems to be closely related to what the W3C "audio"
> group is doing.
> Are you in contact with them?

I've looked at their web pages a few times since
they began, but don't tracked it closely.  Here's
how it relates to audio/rtp-midi.

Basically, the session description parameters for
audio/rtp-midi lets a session define the renderer
(i.e. the way that MIDI should be turned into audio
output in the browser) in a extensible way.  And so,
one could do a short standards-track extension RFC
that describes  how to say "Here is Javascript that
embodies the rendering algorithm. Run it in the
browser using the W3C audio group tools for
sending audio samples to the client output".  Or
an equivalent for technologies like NaCl, for
situations where Javascript is too slow.

This approach is analogous to dispensing with
the native voice codec for telephony, and instead
send along the voice decoding algorithm in
Javascript to be run in the browser. With the
same advantages and disadvantages, to a
first order.

The alternative is to build MIDI renderers into the
browsers, in the same way we build native voice
codecs into browsers.  The RTP MIDI RFC describes
how to specify three standardized MIDI renderers,
offering increasing levels of programmability and
normative output: General MIDI, DLS2, and MPEG 4
Structured Audio.  Alternatively, the mechanism allows
for extensions to specify other renderers, be they
standardized or proprietary, via standards-track
extension RFCs.

Note that everything I discussed above covers the
[2], [3], and [4] use cases in my original post. The
[1] use case involves the user on the client manipulating
GUI elements (graphical depictions of piano keys, audio
volume faders, etc) that are sent as MIDI commands over
audio/rtp-midi to a device that the user wishes to control --
anything from a hardware music synthesizer, to a lighting
panel, to a PC running emulations of some hardware device.
In that case, there's no audio rendering specified.  Most of
the iOS apps that use RTP MIDI are of this type.

-- 
John Lazzaro
http://www.cs.berkeley.edu/~lazzaro
john [dot] lazzaro [at] gmail [dot] com

From csp@csperkins.org  Mon Jun  6 01:29:40 2011
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This is our contribution to the RTC-Web interim meeting. I'd also like =
to ask the chairs to consider this as a working group document.

Colin



Begin forwarded message:
> From: internet-drafts@ietf.org
> Date: 5 June 2011 22:32:19 GMT+01:00
> To: i-d-announce@ietf.org
> Subject: I-D Action: draft-perkins-rtcweb-rtp-usage-01.txt
> Reply-To: internet-drafts@ietf.org
>=20
> A New Internet-Draft is available from the on-line Internet-Drafts =
directories.
>=20
> 	Title           : RTP Requirements for RTC-Web
> 	Author(s)       : Colin Perkins
>                          Magnus Westerlund
>                          Joerg Ott
> 	Filename        : draft-perkins-rtcweb-rtp-usage-01.txt
> 	Pages           : 19
> 	Date            : 2011-06-05
>=20
>   This memo discusses use of RTP in the context of the RTC-Web
>   activity.  It discusses important features of RTP that need to be
>   considered by other parts of the RTC-Web framework, describes which
>   RTP profile to use in this environment, and outlines what RTP
>   extensions should be supported.
>=20
>=20
> A URL for this Internet-Draft is:
> =
http://www.ietf.org/internet-drafts/draft-perkins-rtcweb-rtp-usage-01.txt
>=20
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>=20
> This Internet-Draft can be retrieved at:
> =
ftp://ftp.ietf.org/internet-drafts/draft-perkins-rtcweb-rtp-usage-01.txt
> _______________________________________________
> I-D-Announce mailing list
> I-D-Announce@ietf.org
> https://www.ietf.org/mailman/listinfo/i-d-announce
> Internet-Draft directories: http://www.ietf.org/shadow.html
> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt

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




From christer.holmberg@ericsson.com  Mon Jun  6 01:32:58 2011
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, "hardie@ipinfusion.com" <hardie@ipinfusion.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Mon, 6 Jun 2011 10:32:54 +0200
Thread-Topic: [rtcweb] Contributions for the Interim meeting
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Hi,

We haven't submitted a new version of the draft for the intermim meeting, b=
ut the current version can be found at:

http://tools.ietf.org/id/draft-holmberg-rtcweb-ucreqs-01.txt

Regards,

Christer=20

> -----Original Message-----
> From: rtcweb-bounces@ietf.org=20
> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Christer Holmberg
> Sent: 10. toukokuuta 2011 17:51
> To: hardie@ipinfusion.com; rtcweb@ietf.org
> Subject: Re: [rtcweb] Contributions for the Interim meeting
>=20
>=20
> Hi,
>=20
> We are willing to continue the work on the use-case &=20
> requirements draft (draft-holmberg-rtcweb-ucreqs), if the=20
> group thinks thinks it's useful.
>=20
> Regards,
>=20
> Christer
>=20
> =20
>=20
> > -----Original Message-----
> > From: rtcweb-bounces@ietf.org
> > [mailto:rtcweb-bounces@ietf.org] On Behalf Of Ted Hardie
> > Sent: 4. toukokuuta 2011 3:08
> > To: rtcweb@ietf.org
> > Subject: [rtcweb] Contributions for the Interim meeting
> >=20
> > As you will have seen from Cullen's announcement, we are now=20
> > officially a working group and starting to collect data on the=20
> > possible timing of an interim meeting.
> >=20
> > For the interim meeting, we want to start work on the chartered=20
> > deliverables.  If you are interested in being an editor for=20
> one of the=20
> > working group docs, the chairs would like you to prepare=20
> something for=20
> > the Interim as a draft-00 of your take.  If you can, raise=20
> your hand=20
> > on the list to say you're doing so, so as to let others=20
> know to talk=20
> > to you about collaborating.  If you are not able to attend=20
> any of the=20
> > times put forward for the interim, you can still volunteer=20
> to edit a=20
> > document--it just means you'll have to have the -00 done=20
> early enough=20
> > for the chairs to find a stuckee for feedback coming in at the=20
> > interim.
> >=20
> > regards,
> >=20
> > Ted, Cullen, and Magnus
> > _______________________________________________
> > 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
> =

From jdrosen@jdrosen.net  Mon Jun  6 10:27:49 2011
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Date: Mon, 06 Jun 2011 13:27:47 -0400
From: Jonathan Rosenberg <jdrosen@jdrosen.net>
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Subject: [rtcweb] Fwd: I-D Action: draft-kaufman-rtcweb-traversal-00.txt
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Hi folks,

Matthew and I did a writeup on a proposed model for FW/NAT traversal, 
emphasizing browser minimalisim.

Thanks,
Jonathan R.

-------- Original Message --------
Subject: I-D Action: draft-kaufman-rtcweb-traversal-00.txt
Date: Mon, 06 Jun 2011 10:24:10 -0700
From: internet-drafts@ietf.org
Reply-To: internet-drafts@ietf.org
To: i-d-announce@ietf.org

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

	Title           : NAT Traversal Requirements for RTCWEB
	Author(s)       : Matthew Kaufman
                           Jonathan Rosenberg
	Filename        : draft-kaufman-rtcweb-traversal-00.txt
	Pages           : 8
	Date            : 2011-06-06

    This document describes a minimal set of requirements to enable NAT
    traversal (and satisfy one of the security requirements) for media
    channels within browser-based real-time communications (RTCWEB).


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

Cullen and I did a write up on the protocol requirements.  It more or less
attempts to consolidate a lot of the work that has already been done to
date.

Regards,

-Cary Bran



-------- Original Message --------
From: internet-drafts@ietf.org=A0
To: i-d-announce@ietf.org=A0
Reply-to: internet-drafts@ietf.org=A0
Subject: I-D Action: draft-cbran-rtcweb-protocols-00.txt=A0
X-RSN: 1/0/935/37055/40429=A0
=A0
A New Internet-Draft is available from the on-line Internet-Drafts
directories.=A0
=A0
Title           : RTC-Web Communications Protocols=A0
Author(s)       : Cary Bran=A0
 Cullen Jennings=A0
Filename        : draft-cbran-rtcweb-protocols-00.txt=A0
Pages           : 21=A0
Date            : 2011-06-05=A0
=A0
 The real time communications web (RTC-Web) will enable applications=A0
 such as web browsers to natively support real time interactive voice=A0
 and video.  This document outlines the communication protocols for=A0
 realizing RTC-Web functionality within applications such as web=A0
 browsers.  In addition to communications protocols, this document=A0
 proposes a set of application programming interface requirements for=A0
 controlling the protocol stack.=A0
=A0
=A0
A URL for this Internet-Draft is:=A0
http://www.ietf.org/internet-drafts/draft-cbran-rtcweb-protocols-00.txt=A0
=A0
Internet-Drafts are also available by anonymous FTP at:=A0
ftp://ftp.ietf.org/internet-drafts/=A0
=A0
This Internet-Draft can be retrieved at:=A0
ftp://ftp.ietf.org/internet-drafts/draft-cbran-rtcweb-protocols-00.txt=A0
_______________________________________________=A0
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<FONT FACE=3D"Lucida Grande"><SPAN STYLE=3D'font-size:11pt'>Hi,<BR>
<BR>
Cullen and I did a write up on the protocol requirements. &nbsp;It more or =
less attempts to consolidate a lot of the work that has already been done to=
 date.<BR>
<BR>
Regards,<BR>
<BR>
-Cary Bran<BR>
<BR>
<BR>
<BR>
</SPAN><FONT SIZE=3D"2"><SPAN STYLE=3D'font-size:10pt'>-------- Original Messag=
e --------<BR>
</SPAN></FONT><SPAN STYLE=3D'font-size:11pt'>From: <a href=3D"internet-drafts@i=
etf.org">internet-drafts@ietf.org</a>=A0<BR>
To: <a href=3D"i-d-announce@ietf.org">i-d-announce@ietf.org</a>=A0<BR>
Reply-to: <a href=3D"internet-drafts@ietf.org">internet-drafts@ietf.org</a>=A0<=
BR>
Subject: I-D Action: draft-cbran-rtcweb-protocols-00.txt=A0<BR>
X-RSN: 1/0/935/37055/40429=A0<BR>
=A0<BR>
A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.=A0<BR>
=A0<BR>
Title &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;: RTC-Web=
 Communications Protocols=A0<BR>
Author(s) &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;: Cary Bran=A0<BR>
&nbsp;Cullen Jennings=A0<BR>
Filename &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;: draft-cbran-rtcweb-pro=
tocols-00.txt=A0<BR>
Pages &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;: 21=A0<BR>
Date &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;: 20=
11-06-05=A0<BR>
=A0<BR>
&nbsp;The real time communications web (RTC-Web) will enable applications=A0<=
BR>
&nbsp;such as web browsers to natively support real time interactive voice=A0=
<BR>
&nbsp;and video. &nbsp;This document outlines the communication protocols f=
or=A0<BR>
&nbsp;realizing RTC-Web functionality within applications such as web=A0<BR>
&nbsp;browsers. &nbsp;In addition to communications protocols, this documen=
t=A0<BR>
&nbsp;proposes a set of application programming interface requirements for=A0=
<BR>
&nbsp;controlling the protocol stack.=A0<BR>
=A0<BR>
=A0<BR>
A URL for this Internet-Draft is:=A0<BR>
<a href=3D"http://www.ietf.org/internet-drafts/draft-cbran-rtcweb-protocols-0=
0.txt">http://www.ietf.org/internet-drafts/draft-cbran-rtcweb-protocols-00.t=
xt</a>=A0<BR>
=A0<BR>
Internet-Drafts are also available by anonymous FTP at:=A0<BR>
<a href=3D"ftp://ftp.ietf.org/internet-drafts/">ftp://ftp.ietf.org/internet-d=
rafts/</a>=A0<BR>
=A0<BR>
This Internet-Draft can be retrieved at:=A0<BR>
<a href=3D"ftp://ftp.ietf.org/internet-drafts/draft-cbran-rtcweb-protocols-00=
.txt">ftp://ftp.ietf.org/internet-drafts/draft-cbran-rtcweb-protocols-00.txt=
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Architecture:  30 minutes
draft-alvestrand-rtcweb-overview (Harald covering all drafts - 15 min)=20=

draft-cbran-rtcweb-protocols =20
draft-holmberg-rtcweb-ucreqs
draft-rosenberg-rtcweb-framework
discussion 15 min
focus on identifying what are the big architecture decisions that are =
needed more than actually debating what they choices we should make.

Use Cases & Solution Requirements: 45 minutes
draft-holmberg-rtcweb-ucreqs (Christer - 15 min)=20
discussion (30 min)

Security: 30 minutes
draft-rescorla-rtcweb-security (Rescorla 15 min )
discussion 15 min

NAT Traversal: 30 minutes
draft-kaufman-rtcweb-traversal (Kaufman 10 min)
20 min discussion

RTP Usage: 30 Minutes
draft-perkins-rtcweb-rtp-usage (Colin - 15 min )=20
15 min discussion

ZRTP=20
draft-johnston-rtcweb-media-privacy (Alan J - 5 min)
- focus on topic of are we going to have the "ZRTP or not" debate=20
5 min discussion

Non audio/video data: 25 min
frame issues (chairs 5 min  )
- draft-holmberg-rtcweb-ucreqs=20
- draft-cbran-rtcweb-protocols
discussion 20 min=20
* make sure we agree on requirements
* are focused on unreliable data ?

Any other business: 15 minutes

Discussion of when next interim meeting is.

30 minutes reserved for other use



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Subject: [rtcweb] Conference bridge info for tomorrows virtual interim meeting
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Conference bridge information is=20

Topic: RTC Web WG=20
Date: Wednesday, June 8, 2011=20
Time: 7:30 am, Pacific Daylight Time (San Francisco, GMT-07:00)=20
Meeting Number: 204 987 300=20
Meeting Password: ietf=20

------------------------------------------------------=20
To start the online meeting=20
-------------------------------------------------------=20
1. Go to =
https://cisco.webex.com/ciscosales/j.php?ED=3D166273742&UID=3D482186897&PW=
=3DNMTAyNWI0Yzdl&RT=3DMiM0=20
2. Log in to your account.=20
3. Click "Start Now".=20
4. Follow the instructions that appear on your screen.=20

-------------------------------------------------------=20
To join the teleconference only=20
-------------------------------------------------------=20

0. Find a local phone number from =
http://cisco.com/en/US/about/doing_business/conferencing/index.html=20

A few numbers are=20

San Jose, CA: +1.408.525.6800 RTP: +1.919.392.3330=20
US/Canada: +1.866.432.9903 United Kingdom: +44.20.8824.0117=20
India: +91.80.4350.1111 Germany: +49.619.6773.9002=20
Japan: +81.3.5763.9394 China: +86.10.8515.5666=20

1. Dial into Cisco WebEx=20

2. Follow the prompts to enter the Meeting Number (l204 987 300) =
followed by the # sign.=20

-------------------------------------------------------=20
For assistance=20
-------------------------------------------------------=20
1. Go to https://cisco.webex.com/ciscosales/mc=20
2. On the left navigation bar, click "Support".=20
To add this meeting to your calendar program (for example Microsoft =
Outlook), click this link:=20
=
https://cisco.webex.com/ciscosales/j.php?ED=3D166273742&UID=3D482186897&IC=
S=3DMS&LD=3D1&RD=3D2&ST=3D1&SHA2=3DIGsAhT8DkosTrdfDP4EqSGFzCAm14Z1H9uq-AYQ=
xgv0=3D=20

To check whether you have the appropriate players installed for UCF =
(Universal Communications Format) rich media files, go to =
https://cisco.webex.com/ciscosales/systemdiagnosis.php=20

If something goes really wrong, SMS my mobile phone at +1 408 421 9990



From likepeng@huawei.com  Tue Jun  7 23:03:47 2011
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Date: Wed, 08 Jun 2011 06:02:42 +0000
From: Likepeng <likepeng@huawei.com>
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Subject: Re: [rtcweb] Conference bridge info for tomorrows virtual interim	meeting
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Hi Cullen, 

Can you do us a favor to send a link for the audio record after the conference call? It is quite late for me to join this call.

I remember that last time for CoRE, Cullen has sent a link for the record of the conference call. 

And also I see the record for the RTC-Web F2F meeting, it is quite good.

Thanks,
Kind Regards
Kepeng
-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of Cullen Jennings
Sent: Wednesday, June 08, 2011 12:12 AM
To: rtcweb@ietf.org
Subject: [rtcweb] Conference bridge info for tomorrows virtual interim meeting

Conference bridge information is 

Topic: RTC Web WG 
Date: Wednesday, June 8, 2011 
Time: 7:30 am, Pacific Daylight Time (San Francisco, GMT-07:00) 
Meeting Number: 204 987 300 
Meeting Password: ietf 

------------------------------------------------------ 
To start the online meeting 
------------------------------------------------------- 
1. Go to https://cisco.webex.com/ciscosales/j.php?ED=166273742&UID=482186897&PW=NMTAyNWI0Yzdl&RT=MiM0 
2. Log in to your account. 
3. Click "Start Now". 
4. Follow the instructions that appear on your screen. 

------------------------------------------------------- 
To join the teleconference only 
------------------------------------------------------- 

0. Find a local phone number from http://cisco.com/en/US/about/doing_business/conferencing/index.html 

A few numbers are 

San Jose, CA: +1.408.525.6800 RTP: +1.919.392.3330 
US/Canada: +1.866.432.9903 United Kingdom: +44.20.8824.0117 
India: +91.80.4350.1111 Germany: +49.619.6773.9002 
Japan: +81.3.5763.9394 China: +86.10.8515.5666 

1. Dial into Cisco WebEx 

2. Follow the prompts to enter the Meeting Number (l204 987 300) followed by the # sign. 

------------------------------------------------------- 
For assistance 
------------------------------------------------------- 
1. Go to https://cisco.webex.com/ciscosales/mc 
2. On the left navigation bar, click "Support". 
To add this meeting to your calendar program (for example Microsoft Outlook), click this link: 
https://cisco.webex.com/ciscosales/j.php?ED=166273742&UID=482186897&ICS=MS&LD=1&RD=2&ST=1&SHA2=IGsAhT8DkosTrdfDP4EqSGFzCAm14Z1H9uq-AYQxgv0= 

To check whether you have the appropriate players installed for UCF (Universal Communications Format) rich media files, go to https://cisco.webex.com/ciscosales/systemdiagnosis.php 

If something goes really wrong, SMS my mobile phone at +1 408 421 9990


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

From fluffy@cisco.com  Wed Jun  8 06:34:14 2011
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Subject: [rtcweb] Presentations for meeting today
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I have been uploading them to the wiki page at 

http://trac.tools.ietf.org/wg/rtcweb/trac/wiki

Scroll to the bottom of that web page and you can find them. 

Still missing a few ...


From igor.faynberg@alcatel-lucent.com  Fri Jun 10 15:01:31 2011
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Date: Fri, 10 Jun 2011 18:01:23 -0400
From: Igor Faynberg <igor.faynberg@alcatel-lucent.com>
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Subject: [rtcweb] How do we implement SIP event subscription in RTCweb?
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A question.

When SIP is implemented in an application within the present RTCweb 
architecture, how exactly could we use WebSockets for SUBSCRIBE/NOTIFY?

It appears to me that the only way to achieve that is to open a 
persistent connection.  Am I right? 

If so, is it not a problem, given the potential duration of such 
connections and potential tie-up of network resources?

If not, how else could that be done?

With many thanks,

Igor

From juberti@google.com  Fri Jun 10 15:29:50 2011
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Subject: Re: [rtcweb] How do we implement SIP event subscription in RTCweb?
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On Fri, Jun 10, 2011 at 3:01 PM, Igor Faynberg <
igor.faynberg@alcatel-lucent.com> wrote:

> A question.
>
> When SIP is implemented in an application within the present RTCweb
> architecture, how exactly could we use WebSockets for SUBSCRIBE/NOTIFY?
>
> It appears to me that the only way to achieve that is to open a persistent
> connection.  Am I right?
>

Yes, that is correct. This is how asynchronous web applications work today.


> If so, is it not a problem, given the potential duration of such
> connections and potential tie-up of network resources?
>

An unused TCP/WebSockets connection consumes very few resources.


> If not, how else could that be done?
>
> With many thanks,
>
> Igor
> ______________________________**_________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/**listinfo/rtcweb<https://www.ietf.org/mailman/listinfo/rtcweb>
>

--0016e649c74c51394504a5631b45
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<br><br><div class=3D"gmail_quote">On Fri, Jun 10, 2011 at 3:01 PM, Igor Fa=
ynberg <span dir=3D"ltr">&lt;<a href=3D"mailto:igor.faynberg@alcatel-lucent=
.com">igor.faynberg@alcatel-lucent.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;">

A question.<br>
<br>
When SIP is implemented in an application within the present RTCweb archite=
cture, how exactly could we use WebSockets for SUBSCRIBE/NOTIFY?<br>
<br>
It appears to me that the only way to achieve that is to open a persistent =
connection. =C2=A0Am I right? <br></blockquote><div><br></div><div>Yes, tha=
t is correct. This is how asynchronous web applications work today.</div><d=
iv>

=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bo=
rder-left:1px #ccc solid;padding-left:1ex;">
If so, is it not a problem, given the potential duration of such connection=
s and potential tie-up of network resources?<br></blockquote><div><br></div=
><div>An unused TCP/WebSockets connection consumes very few resources.</div=
>

<div><br></div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex=
;border-left:1px #ccc solid;padding-left:1ex;">
<br>
If not, how else could that be done?<br>
<br>
With many thanks,<br>
<br>
Igor<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>

--0016e649c74c51394504a5631b45--

From Ranjit.Avasarala@Polycom.com  Sat Jun 11 02:09:48 2011
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From: "Avasarala, Ranjit" <Ranjit.Avasarala@Polycom.com>
To: "igor.faynberg@alcatel-lucent.com" <igor.faynberg@alcatel-lucent.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Sat, 11 Jun 2011 17:09:41 +0800
Thread-Topic: [rtcweb] How do we implement SIP event subscription in RTCweb?
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Subject: Re: [rtcweb] How do we implement SIP event subscription in RTCweb?
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Hi

Any idea how the initial SIP signaling would take place between an RTC web =
application and SIP server?

Regards
Ranjit

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Igor Faynberg
Sent: Saturday, June 11, 2011 3:31 AM
To: rtcweb@ietf.org
Subject: [rtcweb] How do we implement SIP event subscription in RTCweb?

A question.

When SIP is implemented in an application within the present RTCweb=20
architecture, how exactly could we use WebSockets for SUBSCRIBE/NOTIFY?

It appears to me that the only way to achieve that is to open a=20
persistent connection.  Am I right?=20

If so, is it not a problem, given the potential duration of such=20
connections and potential tie-up of network resources?

If not, how else could that be done?

With many thanks,

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

From bernard_aboba@hotmail.com  Sat Jun 11 13:43:11 2011
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Subject: [rtcweb] Review of draft-kaufman-rtcweb-traversal
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Review of draft-kaufman-rtcweb-traversal

Overall=2C a very good document.=20

Comments below:

Section 2.2.  Hampers Innovation

   One of the benefits of ICE is that it allows local implementation
   flexibility in the way candidates are gathered=2C offered and
   prioritized.  However=2C once ICE is baked into the browser=2C it is no
   longer possible for that innovation to take place - or at least=2C it
   leaves the hands of the voice application providers.  To date=2C there
   has been variability in this aspect of implementation=2C with different
   providers tuning it to tweak their needs and deployments.

[BA] It might be pointed out this flexibility has already proven useful
to enable support for HTTP failover as well as in implementations of
ICE supporting IPV6.  Given the wide variety of IPv6 deployments in=20
progress=2C consideration experimentation and tweaking has been required=20
to handle the wide variety of conditions that can be experienced
(e.g. IPv6 tunnels of various flavors=2C network devices or services=20
with partial or no IPv6 support=2C  missing IPv6 routes=2C NAT 64=2C etc.).=
=20

2.3.  Unneccesary Cost in some Cases

   There is a broad array of use cases for VoIP.  It is used for
   everything from consumer Internet services (like Skype) to small
   business phone systems.  Though clearly global consumer Internet
   services require the kind of traversal technology provided by full
   ICE=2C it is not needed in other cases.  One such use case is=2C in fact=
=2C
   enterprise telephony=2C where users make calls within the confines of
   their corporate network=2C and remote access is supported through VPN.
   Today=2C VoIP endpoints in these environments do not generally use ICE.

[BA] ICE has in fact proven quite popular in enterprise deployments=20
involving remote users (e.g. mobile devices)=2C so I'd reword the
last sentence to say "For example=2C in an enterprise deployment where
users make calls primarily within a corporate network without deployed=20
NATs=2C VoIP endpoints will not require ICE."=20

3.  Proposed Model

   For security purposes=2C the browser will refuse to send=2C or accept=2C
   media to or from a peer to which a STUN transaction has not completed
   successfully.  This ensures that the browser cannot be used as a DoS
   tool to launch a voice hammer attack.

[BA] This condition is necessary but not sufficient.  For example=2C it
would not make sense for the browser to authorize media to be sent to
an endpoint which completed a STUN transaction without ICE extensions.
If this were allowed=2C then it would be possible to DoS a STUN server.=20

   With this model=2C there is now a great deal of flexibility in how NAT
   traversal can be done.  Some of the models which can now be supported
   are:

   ICE in Javascript:  A full ICE implementation is possible in
      Javascript itself.  Because the implementation resides in
      Javasript=2C it is trivially changed at any time.

   Server-Based ICE:  A full ICE implementation can execute in the
      server=2C using remote-control commands to inform the browser to
      send STUN transactions=2C and passing the results from the browser
      back to the server.  In essence - MGCP for ICE.

   STUN-Only:  For deployments where the peer is always publically
      reachable from clients - such as enterprises or PSTN termination
      services - the Javascript can do a single STUN transaction to
      create a permission in the browser=2C and then proceed to send
      media.

   Non-ICE:  Protocols similar to ICE=2C but not otherwise compliant=2C can
      also be implemented.  Negotiation of which NAT traversal mechanism
      is needed=2C is done by the application outside of the browser.

[BA] There are some interoperability implications to these
approaches that need to be explored.  For example=2C the ICE
Javascript approach should enable interoperability
between different browsers attached to the same service.  However=2C
it is not clear to me that this will necessarily work for=20
inter-service communication (e.g. the clients could have=20
incompatible ICE Javascript libraries).=20

4.4.  Receipt of Response

   Upon receipt of a valid STUN transaction response from the responding
   client=2C the initiating client MUST call a callback function that
   delivers the attribute/value pairs received in the response=2C one of
   which is the reflexive address.  The response MUST be ignored if the
   receivedSocketAddress does not match the socket address to which the
   matching transaction ID was sent=2C as per ICE 7.1.3.2.  Upon receipt
   of a valid response the client also adds the now-verified address to
   the Transmit Whitelist=2C a list of socket addresses to which sending
   of media is now permissible. =20

[BA] Does it really make sense to add the verified address of a responding
client to the transmit whitelist=2C even if the nominated flag is not
set in the response?

 		 	   		  =

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<html>
<head>
<style><!--
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{
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body.hmmessage
{
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</head>
<body class=3D'hmmessage'>
Review of draft-kaufman-rtcweb-traversal<br><br>Overall=2C a very good docu=
ment. <br><br>Comments below:<br><br>Section 2.2.&nbsp=3B Hampers Innovatio=
n<br><br>&nbsp=3B&nbsp=3B One of the benefits of ICE is that it allows loca=
l implementation<br>&nbsp=3B&nbsp=3B flexibility in the way candidates are =
gathered=2C offered and<br>&nbsp=3B&nbsp=3B prioritized.&nbsp=3B However=2C=
 once ICE is baked into the browser=2C it is no<br>&nbsp=3B&nbsp=3B longer =
possible for that innovation to take place - or at least=2C it<br>&nbsp=3B&=
nbsp=3B leaves the hands of the voice application providers.&nbsp=3B To dat=
e=2C there<br>&nbsp=3B&nbsp=3B has been variability in this aspect of imple=
mentation=2C with different<br>&nbsp=3B&nbsp=3B providers tuning it to twea=
k their needs and deployments.<br><br>[BA] It might be pointed out this fle=
xibility has already proven useful<br>to enable support for HTTP failover a=
s well as in implementations of<br>ICE supporting IPV6.&nbsp=3B Given the w=
ide variety of IPv6 deployments in <br>progress=2C consideration experiment=
ation and tweaking has been required <br>to handle the wide variety of cond=
itions that can be experienced<br>(e.g. IPv6 tunnels of various flavors=2C =
network devices or services <br>with partial or no IPv6 support=2C&nbsp=3B =
missing IPv6 routes=2C NAT 64=2C etc.). <br><br>2.3.&nbsp=3B Unneccesary Co=
st in some Cases<br><br>&nbsp=3B&nbsp=3B There is a broad array of use case=
s for VoIP.&nbsp=3B It is used for<br>&nbsp=3B&nbsp=3B everything from cons=
umer Internet services (like Skype) to small<br>&nbsp=3B&nbsp=3B business p=
hone systems.&nbsp=3B Though clearly global consumer Internet<br>&nbsp=3B&n=
bsp=3B services require the kind of traversal technology provided by full<b=
r>&nbsp=3B&nbsp=3B ICE=2C it is not needed in other cases.&nbsp=3B One such=
 use case is=2C in fact=2C<br>&nbsp=3B&nbsp=3B enterprise telephony=2C wher=
e users make calls within the confines of<br>&nbsp=3B&nbsp=3B their corpora=
te network=2C and remote access is supported through VPN.<br>&nbsp=3B&nbsp=
=3B Today=2C VoIP endpoints in these environments do not generally use ICE.=
<br><br>[BA] ICE has in fact proven quite popular in enterprise deployments=
 <br>involving remote users (e.g. mobile devices)=2C so I'd reword the<br>l=
ast sentence to say "For example=2C in an enterprise deployment where<br>us=
ers make calls primarily within a corporate network without deployed <br>NA=
Ts=2C VoIP endpoints will not require ICE." <br><br>3.&nbsp=3B Proposed Mod=
el<br><br>&nbsp=3B&nbsp=3B For security purposes=2C the browser will refuse=
 to send=2C or accept=2C<br>&nbsp=3B&nbsp=3B media to or from a peer to whi=
ch a STUN transaction has not completed<br>&nbsp=3B&nbsp=3B successfully.&n=
bsp=3B This ensures that the browser cannot be used as a DoS<br>&nbsp=3B&nb=
sp=3B tool to launch a voice hammer attack.<br><br>[BA] This condition is n=
ecessary but not sufficient.&nbsp=3B For example=2C it<br>would not make se=
nse for the browser to authorize media to be sent to<br>an endpoint which c=
ompleted a STUN transaction without ICE extensions.<br>If this were allowed=
=2C then it would be possible to DoS a STUN server. <br><br>&nbsp=3B&nbsp=
=3B With this model=2C there is now a great deal of flexibility in how NAT<=
br>&nbsp=3B&nbsp=3B traversal can be done.&nbsp=3B Some of the models which=
 can now be supported<br>&nbsp=3B&nbsp=3B are:<br><br>&nbsp=3B&nbsp=3B ICE =
in Javascript:&nbsp=3B A full ICE implementation is possible in<br>&nbsp=3B=
&nbsp=3B&nbsp=3B&nbsp=3B&nbsp=3B Javascript itself.&nbsp=3B Because the imp=
lementation resides in<br>&nbsp=3B&nbsp=3B&nbsp=3B&nbsp=3B&nbsp=3B Javasrip=
t=2C it is trivially changed at any time.<br><br>&nbsp=3B&nbsp=3B Server-Ba=
sed ICE:&nbsp=3B A full ICE implementation can execute in the<br>&nbsp=3B&n=
bsp=3B&nbsp=3B&nbsp=3B&nbsp=3B server=2C using remote-control commands to i=
nform the browser to<br>&nbsp=3B&nbsp=3B&nbsp=3B&nbsp=3B&nbsp=3B send STUN =
transactions=2C and passing the results from the browser<br>&nbsp=3B&nbsp=
=3B&nbsp=3B&nbsp=3B&nbsp=3B back to the server.&nbsp=3B In essence - MGCP f=
or ICE.<br><br>&nbsp=3B&nbsp=3B STUN-Only:&nbsp=3B For deployments where th=
e peer is always publically<br>&nbsp=3B&nbsp=3B&nbsp=3B&nbsp=3B&nbsp=3B rea=
chable from clients - such as enterprises or PSTN termination<br>&nbsp=3B&n=
bsp=3B&nbsp=3B&nbsp=3B&nbsp=3B services - the Javascript can do a single ST=
UN transaction to<br>&nbsp=3B&nbsp=3B&nbsp=3B&nbsp=3B&nbsp=3B create a perm=
ission in the browser=2C and then proceed to send<br>&nbsp=3B&nbsp=3B&nbsp=
=3B&nbsp=3B&nbsp=3B media.<br><br>&nbsp=3B&nbsp=3B Non-ICE:&nbsp=3B Protoco=
ls similar to ICE=2C but not otherwise compliant=2C can<br>&nbsp=3B&nbsp=3B=
&nbsp=3B&nbsp=3B&nbsp=3B also be implemented.&nbsp=3B Negotiation of which =
NAT traversal mechanism<br>&nbsp=3B&nbsp=3B&nbsp=3B&nbsp=3B&nbsp=3B is need=
ed=2C is done by the application outside of the browser.<br><br>[BA] There =
are some interoperability implications to these<br>approaches that need to =
be explored.&nbsp=3B For example=2C the ICE<br>Javascript approach should e=
nable interoperability<br>between different browsers attached to the same s=
ervice.&nbsp=3B However=2C<br>it is not clear to me that this will necessar=
ily work for <br>inter-service communication (e.g. the clients could have <=
br>incompatible ICE Javascript libraries). <br><br>4.4.&nbsp=3B Receipt of =
Response<br><br>&nbsp=3B&nbsp=3B Upon receipt of a valid STUN transaction r=
esponse from the responding<br>&nbsp=3B&nbsp=3B client=2C the initiating cl=
ient MUST call a callback function that<br>&nbsp=3B&nbsp=3B delivers the at=
tribute/value pairs received in the response=2C one of<br>&nbsp=3B&nbsp=3B =
which is the reflexive address.&nbsp=3B The response MUST be ignored if the=
<br>&nbsp=3B&nbsp=3B receivedSocketAddress does not match the socket addres=
s to which the<br>&nbsp=3B&nbsp=3B matching transaction ID was sent=2C as p=
er ICE 7.1.3.2.&nbsp=3B Upon receipt<br>&nbsp=3B&nbsp=3B of a valid respons=
e the client also adds the now-verified address to<br>&nbsp=3B&nbsp=3B the =
Transmit Whitelist=2C a list of socket addresses to which sending<br>&nbsp=
=3B&nbsp=3B of media is now permissible. &nbsp=3B<br><br>[BA] Does it reall=
y make sense to add the verified address of a responding<br>client to the t=
ransmit whitelist=2C even if the nominated flag is not<br>set in the respon=
se?<br><br> 		 	   		  </body>
</html>=

--_d0bdc94c-8c8c-4c19-9c95-08a276807331_--

From matthew.kaufman@skype.net  Sat Jun 11 13:50:38 2011
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On Jun 11, 2011, at 1:43 PM, Bernard Aboba <bernard_aboba@hotmail.com> wrot=
e:

>=20
> [BA] This condition is necessary but not sufficient.  For example, it
> would not make sense for the browser to authorize media to be sent to
> an endpoint which completed a STUN transaction without ICE extensions.
> If this were allowed, then it would be possible to DoS a STUN server.=20
>=20

This is only true of STUN servers that don't require credentials.

If those really exist then yes, requiring at least one reasonable extension=
 makes sense.
>=20
>=20
> [BA] Does it really make sense to add the verified address of a respondin=
g
> client to the transmit whitelist, even if the nominated flag is not
> set in the response?

Quite possibly yes, especially for mobility cases, but see above that sone =
extension or flag might indeed be needed before adding.

Thanks for the comments, will review more thoroughly when the plane I'm sit=
ting on lands.

Matthew Kaufman
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Bernard,

at last week's call, the question was raised on whether it's actually 
possible to implement ICE in javascript according to the described model 
- the Javascript event loop has certain problems with precise timing.

The upshoot seemed to be that we'd like to see a demonstration that it's 
possible to create an ICE-interoperable implementation using Javascript 
as described before attempting to go further down this path.

Some other comments below.

On 06/11/2011 10:43 PM, Bernard Aboba wrote:
> Review of draft-kaufman-rtcweb-traversal
>
> Overall, a very good document.
>
> Comments below:
>
> Section 2.2.  Hampers Innovation
>
>    One of the benefits of ICE is that it allows local implementation
>    flexibility in the way candidates are gathered, offered and
>    prioritized.  However, once ICE is baked into the browser, it is no
>    longer possible for that innovation to take place - or at least, it
>    leaves the hands of the voice application providers.  To date, there
>    has been variability in this aspect of implementation, with different
>    providers tuning it to tweak their needs and deployments.
>
> [BA] It might be pointed out this flexibility has already proven useful
> to enable support for HTTP failover as well as in implementations of
> ICE supporting IPV6.  Given the wide variety of IPv6 deployments in
> progress, consideration experimentation and tweaking has been required
> to handle the wide variety of conditions that can be experienced
> (e.g. IPv6 tunnels of various flavors, network devices or services
> with partial or no IPv6 support,  missing IPv6 routes, NAT 64, etc.).

This is good information, and it seems reasonable that it should lead to 
a rework of ICE based on this experience. Is it written up anywhere you 
can share?

>    ICE in Javascript:  A full ICE implementation is possible in
>       Javascript itself.  Because the implementation resides in
>       Javasript, it is trivially changed at any time.
>
>    Server-Based ICE:  A full ICE implementation can execute in the
>       server, using remote-control commands to inform the browser to
>       send STUN transactions, and passing the results from the browser
>       back to the server.  In essence - MGCP for ICE.
>
>    STUN-Only:  For deployments where the peer is always publically
>       reachable from clients - such as enterprises or PSTN termination
>       services - the Javascript can do a single STUN transaction to
>       create a permission in the browser, and then proceed to send
>       media.
>
>    Non-ICE:  Protocols similar to ICE, but not otherwise compliant, can
>       also be implemented.  Negotiation of which NAT traversal mechanism
>       is needed, is done by the application outside of the browser.
>
> [BA] There are some interoperability implications to these
> approaches that need to be explored.  For example, the ICE
> Javascript approach should enable interoperability
> between different browsers attached to the same service.  However,
> it is not clear to me that this will necessarily work for
> inter-service communication (e.g. the clients could have
> incompatible ICE Javascript libraries).

If both javascript libraries implement ICE, they should by definition be 
interoperable - I am more worried about the case where one application 
supports ICE and another application supports some ICE-like mechanism.



--------------090607000706010609080805
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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 text="#000000" bgcolor="#ffffff">
    Bernard,<br>
    <br>
    at last week's call, the question was raised on whether it's
    actually possible to implement ICE in javascript according to the
    described model - the Javascript event loop has certain problems
    with precise timing.<br>
    <br>
    The upshoot seemed to be that we'd like to see a demonstration that
    it's possible to create an ICE-interoperable implementation using
    Javascript as described before attempting to go further down this
    path.<br>
    <br>
    Some other comments below.<br>
    <br>
    On 06/11/2011 10:43 PM, Bernard Aboba wrote:
    <blockquote cite="mid:BLU152-w64EE1689B126AA09DBD5593670@phx.gbl"
      type="cite">
      <style><!--
.hmmessage P
{
margin:0px;
padding:0px
}
body.hmmessage
{
font-size: 10pt;
font-family:Tahoma
}
--></style>
      Review of draft-kaufman-rtcweb-traversal<br>
      <br>
      Overall, a very good document. <br>
      <br>
      Comments below:<br>
      <br>
      Section 2.2.&nbsp; Hampers Innovation<br>
      <br>
      &nbsp;&nbsp; One of the benefits of ICE is that it allows local
      implementation<br>
      &nbsp;&nbsp; flexibility in the way candidates are gathered, offered and<br>
      &nbsp;&nbsp; prioritized.&nbsp; However, once ICE is baked into the browser, it
      is no<br>
      &nbsp;&nbsp; longer possible for that innovation to take place - or at
      least, it<br>
      &nbsp;&nbsp; leaves the hands of the voice application providers.&nbsp; To date,
      there<br>
      &nbsp;&nbsp; has been variability in this aspect of implementation, with
      different<br>
      &nbsp;&nbsp; providers tuning it to tweak their needs and deployments.<br>
      <br>
      [BA] It might be pointed out this flexibility has already proven
      useful<br>
      to enable support for HTTP failover as well as in implementations
      of<br>
      ICE supporting IPV6.&nbsp; Given the wide variety of IPv6 deployments
      in <br>
      progress, consideration experimentation and tweaking has been
      required <br>
      to handle the wide variety of conditions that can be experienced<br>
      (e.g. IPv6 tunnels of various flavors, network devices or services
      <br>
      with partial or no IPv6 support,&nbsp; missing IPv6 routes, NAT 64,
      etc.). <br>
    </blockquote>
    <br>
    This is good information, and it seems reasonable that it should
    lead to a rework of ICE based on this experience. Is it written up
    anywhere you can share?<br>
    <br>
    <blockquote cite="mid:BLU152-w64EE1689B126AA09DBD5593670@phx.gbl"
      type="cite">&nbsp;&nbsp; ICE in Javascript:&nbsp; A full ICE implementation is
      possible in<br>
      &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Javascript itself.&nbsp; Because the implementation resides in<br>
      &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Javasript, it is trivially changed at any time.<br>
      <br>
      &nbsp;&nbsp; Server-Based ICE:&nbsp; A full ICE implementation can execute in the<br>
      &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; server, using remote-control commands to inform the browser
      to<br>
      &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; send STUN transactions, and passing the results from the
      browser<br>
      &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; back to the server.&nbsp; In essence - MGCP for ICE.<br>
      <br>
      &nbsp;&nbsp; STUN-Only:&nbsp; For deployments where the peer is always publically<br>
      &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; reachable from clients - such as enterprises or PSTN
      termination<br>
      &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; services - the Javascript can do a single STUN transaction
      to<br>
      &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; create a permission in the browser, and then proceed to send<br>
      &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; media.<br>
      <br>
      &nbsp;&nbsp; Non-ICE:&nbsp; Protocols similar to ICE, but not otherwise
      compliant, can<br>
      &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; also be implemented.&nbsp; Negotiation of which NAT traversal
      mechanism<br>
      &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; is needed, is done by the application outside of the
      browser.<br>
      <br>
      [BA] There are some interoperability implications to these<br>
      approaches that need to be explored.&nbsp; For example, the ICE<br>
      Javascript approach should enable interoperability<br>
      between different browsers attached to the same service.&nbsp; However,<br>
      it is not clear to me that this will necessarily work for <br>
      inter-service communication (e.g. the clients could have <br>
      incompatible ICE Javascript libraries). <br>
    </blockquote>
    <br>
    If both javascript libraries implement ICE, they should by
    definition be interoperable - I am more worried about the case where
    one application supports ICE and another application supports some
    ICE-like mechanism.<br>
    <br>
    <br>
  </body>
</html>

--------------090607000706010609080805--

From bernard_aboba@hotmail.com  Sat Jun 11 14:40:22 2011
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Review of draft-rescorla-rtcweb-security

Overall=2C a reasonable -00 that needs expanded coverage in a number of are=
as before it
can be considered for adoption as a WG document.=20

Comments below:=20

   The Real-Time Communications on the Web (RTC-Web) working group is
   tasked with standardizing protocols for real-time communications
   between Web browsers.  The two major use cases for RTC-Web technology
   are real-time audio and/or video calls and direct data transfer.

[BA] Today=2C Web-based realtime communications technology is widely
deployed (albeit proprietary)=2C and one of the major use cases is
Web conferencing (e.g. WebEx).  So this is probably deserving
of consideration although this may not be convenient given the
discussion that follows.=20


   More subtly=2C if the
   exposed APIs allow the server to instruct the browser to send
   arbitrary content=2C then they can be used to bypass firewalls or mount
   denial of service attacks.  Any successful system will need to be
   resistant to this and other attacks.

[BA] This would include things like denial of service attacks on
public service answering points (PSAPs).=20

   Conventionally=2C we refer to either WEB ATTACKERS=2C who are able to
   induce you to visit their sites but do not control the network=2C and
   NETWORK ATTACKERS=2C who are able to control your network.  Network
   attackers correspond to the [RFC3552] "Internet Threat Model".  In
   general=2C it is desirable to build a system which is secure against
   both kinds of attackers=2C but realistically many sites do not run
   HTTPS [RFC2818] and so our ability to defend against network
   attackers is necessarily somewhat limited.  Most of the rest of this
   section is devoted to web attackers=2C with the assumption that
   protection against network attackers is provided by running HTTPS.

[BA] I'd suggest that the issue of "privacy leakage" also be mentioned=2C e=
ven
though this is a much nastier problem that is largely outside the scope of
the RTCWEB WG:
http://www.w3.org/2011/track-privacy/slides/wills.pdf

The issue of privacy may also be relevant in other contexts such as
the potential creation of a Javascript API to return the interface addresse=
s (e.g.
for use by ICE).  My take is that the document should take a position on th=
is
(e.g. that such an extension would be OK or not).=20

3.3.  Bypassing SOP: CORS=2C WebSockets=2C and consent to communicate

   While SOP serves an important security function=2C it also makes it
   inconvenient to write certain classes of applications.  In
   particular=2C mash-ups=2C in which a script from origin A uses resources
   from origin B=2C can only be achieved via a certain amount of hackery.
   The W3C Cross-Origin Resource Sharing (CORS) spec [CORS] is a
   response to this demand. =20

[BA] It probably should be noted that there are other ways around
the same ORIGIN policy that are widely used today.  These
include Javascript libraries as well as general-purpose=20
BOSH connection managers.

4.1.  Access to Local Devices

   Arguably=2C origin is not fine-grained enough.  Consider the situation
   where Alice visits a site and authorizes it to make a single call.
   If consent is expressed solely in terms of origin=2C then at any future
   visit to that site (including one induced via mash-up or ad network)=2C
   the site can bug Alice's computer.  While in principle Alice could
   grant and then revoke the privilege=2C in practice privileges
   accumulate=3B if we are concerned about this attack=2C something else is
   needed.  There are a number of potential countermeasures to this sort
   of issue.

[BA] Not only can they bug Alice's computer=2C but the site could also
do other things like making a bogus emergency call.=20

4.2.2.  Masking

   Once consent is verified=2C there still is some concern about
   misinterpretation attacks as described by Huang et al.[huang-w2sp].
   As long as communication is limited to UDP=2C then this risk is
   probably limited=2C thus masking is not required for UDP. =20

[BA] Are you saying that the browser should be able to send arbitrary
UDP traffic once "consent" is granted? =20

4.2.3.  Backward Compatibility

   A requirement to use ICE limits compatibility with legacy non-ICE
   clients.  It seems unsafe to completely remove the requirement for
   some check=2C but it might be possible to merely require a one-sided
   check where the legacy client was a STUN responder.  It's unclear
   whether that is in fact simpler than doing ICE-Lite.

[BA] This section needs to be expanded to provide more in depth
discussion of backward compatibility issues and potential approaches=2C
which might include:

a. RTCP-based approaches (e.g. no ICE). =20
b. STUN without ICE extensions

Since each of these approaches has weaknesses=2C it would be useful to
make clear what attacks can be launched against them and what
mitigating measures might be available=2C along with a discussion
of their effectiveness and an overall recommendation.

4.3.  Communications Security

   Technology for providing this
   service (for instance=2C DTLS [RFC4347] and DTLS-SRTP [RFC5763]) is
   well understood.  However=2C we must examine this technology to the
   RTC-Web context=2C where the threat model is somewhat different.

[BA] As suggested in draft-johnston-rtcweb-media-privacy=2C the threat
model is not the only potential issue in the RTCWEB context -- there
is also the problem of dependency on (non-standardized) signaling.=20





 		 	   		  =

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Review of draft-rescorla-rtcweb-security<br><br>Overall=2C a reasonable -00=
 that needs expanded coverage in a number of areas before it<br>can be cons=
idered for adoption as a WG document. <br><br>Comments below: <br><br>&nbsp=
=3B&nbsp=3B The Real-Time Communications on the Web (RTC-Web) working group=
 is<br>&nbsp=3B&nbsp=3B tasked with standardizing protocols for real-time c=
ommunications<br>&nbsp=3B&nbsp=3B between Web browsers.&nbsp=3B The two maj=
or use cases for RTC-Web technology<br>&nbsp=3B&nbsp=3B are real-time audio=
 and/or video calls and direct data transfer.<br><br>[BA] Today=2C Web-base=
d realtime communications technology is widely<br>deployed (albeit propriet=
ary)=2C and one of the major use cases is<br>Web conferencing (e.g. WebEx).=
&nbsp=3B So this is probably deserving<br>of consideration although this ma=
y not be convenient given the<br>discussion that follows. <br><br><br>&nbsp=
=3B&nbsp=3B More subtly=2C if the<br>&nbsp=3B&nbsp=3B exposed APIs allow th=
e server to instruct the browser to send<br>&nbsp=3B&nbsp=3B arbitrary cont=
ent=2C then they can be used to bypass firewalls or mount<br>&nbsp=3B&nbsp=
=3B denial of service attacks.&nbsp=3B Any successful system will need to b=
e<br>&nbsp=3B&nbsp=3B resistant to this and other attacks.<br><br>[BA] This=
 would include things like denial of service attacks on<br>public service a=
nswering points (PSAPs). <br><br>&nbsp=3B&nbsp=3B Conventionally=2C we refe=
r to either WEB ATTACKERS=2C who are able to<br>&nbsp=3B&nbsp=3B induce you=
 to visit their sites but do not control the network=2C and<br>&nbsp=3B&nbs=
p=3B NETWORK ATTACKERS=2C who are able to control your network.&nbsp=3B Net=
work<br>&nbsp=3B&nbsp=3B attackers correspond to the [RFC3552] "Internet Th=
reat Model".&nbsp=3B In<br>&nbsp=3B&nbsp=3B general=2C it is desirable to b=
uild a system which is secure against<br>&nbsp=3B&nbsp=3B both kinds of att=
ackers=2C but realistically many sites do not run<br>&nbsp=3B&nbsp=3B HTTPS=
 [RFC2818] and so our ability to defend against network<br>&nbsp=3B&nbsp=3B=
 attackers is necessarily somewhat limited.&nbsp=3B Most of the rest of thi=
s<br>&nbsp=3B&nbsp=3B section is devoted to web attackers=2C with the assum=
ption that<br>&nbsp=3B&nbsp=3B protection against network attackers is prov=
ided by running HTTPS.<br><br>[BA] I'd suggest that the issue of "privacy l=
eakage" also be mentioned=2C even<br>though this is a much nastier problem =
that is largely outside the scope of<br>the RTCWEB WG:<br>http://www.w3.org=
/2011/track-privacy/slides/wills.pdf<br><br>The issue of privacy may also b=
e relevant in other contexts such as<br>the potential creation of a Javascr=
ipt API to return the interface addresses (e.g.<br>for use by ICE).&nbsp=3B=
 My take is that the document should take a position on this<br>(e.g. that =
such an extension would be OK or not). <br><br>3.3.&nbsp=3B Bypassing SOP: =
CORS=2C WebSockets=2C and consent to communicate<br><br>&nbsp=3B&nbsp=3B Wh=
ile SOP serves an important security function=2C it also makes it<br>&nbsp=
=3B&nbsp=3B inconvenient to write certain classes of applications.&nbsp=3B =
In<br>&nbsp=3B&nbsp=3B particular=2C mash-ups=2C in which a script from ori=
gin A uses resources<br>&nbsp=3B&nbsp=3B from origin B=2C can only be achie=
ved via a certain amount of hackery.<br>&nbsp=3B&nbsp=3B The W3C Cross-Orig=
in Resource Sharing (CORS) spec [CORS] is a<br>&nbsp=3B&nbsp=3B response to=
 this demand.&nbsp=3B <br><br>[BA] It probably should be noted that there a=
re other ways around<br>the same ORIGIN policy that are widely used today.&=
nbsp=3B These<br>include Javascript libraries as well as general-purpose <b=
r>BOSH connection managers.<br><br>4.1.&nbsp=3B Access to Local Devices<br>=
<br>&nbsp=3B&nbsp=3B Arguably=2C origin is not fine-grained enough.&nbsp=3B=
 Consider the situation<br>&nbsp=3B&nbsp=3B where Alice visits a site and a=
uthorizes it to make a single call.<br>&nbsp=3B&nbsp=3B If consent is expre=
ssed solely in terms of origin=2C then at any future<br>&nbsp=3B&nbsp=3B vi=
sit to that site (including one induced via mash-up or ad network)=2C<br>&n=
bsp=3B&nbsp=3B the site can bug Alice's computer.&nbsp=3B While in principl=
e Alice could<br>&nbsp=3B&nbsp=3B grant and then revoke the privilege=2C in=
 practice privileges<br>&nbsp=3B&nbsp=3B accumulate=3B if we are concerned =
about this attack=2C something else is<br>&nbsp=3B&nbsp=3B needed.&nbsp=3B =
There are a number of potential countermeasures to this sort<br>&nbsp=3B&nb=
sp=3B of issue.<br><br>[BA] Not only can they bug Alice's computer=2C but t=
he site could also<br>do other things like making a bogus emergency call. <=
br><br>4.2.2.&nbsp=3B Masking<br><br>&nbsp=3B&nbsp=3B Once consent is verif=
ied=2C there still is some concern about<br>&nbsp=3B&nbsp=3B misinterpretat=
ion attacks as described by Huang et al.[huang-w2sp].<br>&nbsp=3B&nbsp=3B A=
s long as communication is limited to UDP=2C then this risk is<br>&nbsp=3B&=
nbsp=3B probably limited=2C thus masking is not required for UDP.&nbsp=3B <=
br><br>[BA] Are you saying that the browser should be able to send arbitrar=
y<br>UDP traffic once "consent" is granted?&nbsp=3B <br><br>4.2.3.&nbsp=3B =
Backward Compatibility<br><br>&nbsp=3B&nbsp=3B A requirement to use ICE lim=
its compatibility with legacy non-ICE<br>&nbsp=3B&nbsp=3B clients.&nbsp=3B =
It seems unsafe to completely remove the requirement for<br>&nbsp=3B&nbsp=
=3B some check=2C but it might be possible to merely require a one-sided<br=
>&nbsp=3B&nbsp=3B check where the legacy client was a STUN responder.&nbsp=
=3B It's unclear<br>&nbsp=3B&nbsp=3B whether that is in fact simpler than d=
oing ICE-Lite.<br><br>[BA] This section needs to be expanded to provide mor=
e in depth<br>discussion of backward compatibility issues and potential app=
roaches=2C<br>which might include:<br><br>a. RTCP-based approaches (e.g. no=
 ICE).&nbsp=3B <br>b. STUN without ICE extensions<br><br>Since each of thes=
e approaches has weaknesses=2C it would be useful to<br>make clear what att=
acks can be launched against them and what<br>mitigating measures might be =
available=2C along with a discussion<br>of their effectiveness and an overa=
ll recommendation.<br><br>4.3.&nbsp=3B Communications Security<br><br>&nbsp=
=3B&nbsp=3B Technology for providing this<br>&nbsp=3B&nbsp=3B service (for =
instance=2C DTLS [RFC4347] and DTLS-SRTP [RFC5763]) is<br>&nbsp=3B&nbsp=3B =
well understood.&nbsp=3B However=2C we must examine this technology to the<=
br>&nbsp=3B&nbsp=3B RTC-Web context=2C where the threat model is somewhat d=
ifferent.<br><br>[BA] As suggested in draft-johnston-rtcweb-media-privacy=
=2C the threat<br>model is not the only potential issue in the RTCWEB conte=
xt -- there<br>is also the problem of dependency on (non-standardized) sign=
aling. <br><br><br><br><br><br> 		 	   		  </body>
</html>=

--_ae8a1e60-1663-4a6e-9328-1e3370732de4_--

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These kind of "performance" arguments tend to fall victim to improvements i=
n technology=2C so even if they can be demonstrated to be true at a given p=
oint in time=2C the argument may prove invalid years or even months down th=
e road.  =20

In this particular case=2C there have been some major improvements in Javas=
cript performance in recent browser releases=2C and it is quite possible th=
at those improvements may continue. =20

So even if such a demonstration were to "fail" on current releases=2C  the =
target is not today's browser (which after all=2C doesn't support RTCWEB at=
 all)=2C but rather future releases that do (and may also incorporate Javas=
cript performance improvements).=20

To my mind=2C the real question isn't performance=2C it is interoperability=
.   The "monolithic" approach to ICE brings with it the potential for inter=
operability problems between different browsers operating on the same servi=
ce.  Mathew's approach of Javascript implementation in ICE addresses that i=
ssue=2C but I'm concerned about the implications for cross-service interope=
rability. =20

Date: Sat=2C 11 Jun 2011 23:23:04 +0200
From: harald@alvestrand.no
To: bernard_aboba@hotmail.com
CC: rtcweb@ietf.org
Subject: Re: [rtcweb] Review of draft-kaufman-rtcweb-traversal



 =20


   =20
 =20
 =20
    Bernard=2C

   =20

    at last week's call=2C the question was raised on whether it's
    actually possible to implement ICE in javascript according to the
    described model - the Javascript event loop has certain problems
    with precise timing.

   =20

    The upshoot seemed to be that we'd like to see a demonstration that
    it's possible to create an ICE-interoperable implementation using
    Javascript as described before attempting to go further down this
    path.

   =20

    Some other comments below.

   =20

    On 06/11/2011 10:43 PM=2C Bernard Aboba wrote:
   =20
     =20
      Review of draft-kaufman-rtcweb-traversal

     =20

      Overall=2C a very good document.=20

     =20

      Comments below:

     =20

      Section 2.2.  Hampers Innovation

     =20

         One of the benefits of ICE is that it allows local
      implementation

         flexibility in the way candidates are gathered=2C offered and

         prioritized.  However=2C once ICE is baked into the browser=2C it
      is no

         longer possible for that innovation to take place - or at
      least=2C it

         leaves the hands of the voice application providers.  To date=2C
      there

         has been variability in this aspect of implementation=2C with
      different

         providers tuning it to tweak their needs and deployments.

     =20

      [BA] It might be pointed out this flexibility has already proven
      useful

      to enable support for HTTP failover as well as in implementations
      of

      ICE supporting IPV6.  Given the wide variety of IPv6 deployments
      in=20

      progress=2C consideration experimentation and tweaking has been
      required=20

      to handle the wide variety of conditions that can be experienced

      (e.g. IPv6 tunnels of various flavors=2C network devices or services
     =20

      with partial or no IPv6 support=2C  missing IPv6 routes=2C NAT 64=2C
      etc.).=20

   =20
   =20

    This is good information=2C and it seems reasonable that it should
    lead to a rework of ICE based on this experience. Is it written up
    anywhere you can share?

   =20

       ICE in Javascript:  A full ICE implementation is
      possible in

            Javascript itself.  Because the implementation resides in

            Javasript=2C it is trivially changed at any time.

     =20

         Server-Based ICE:  A full ICE implementation can execute in the

            server=2C using remote-control commands to inform the browser
      to

            send STUN transactions=2C and passing the results from the
      browser

            back to the server.  In essence - MGCP for ICE.

     =20

         STUN-Only:  For deployments where the peer is always publically

            reachable from clients - such as enterprises or PSTN
      termination

            services - the Javascript can do a single STUN transaction
      to

            create a permission in the browser=2C and then proceed to send

            media.

     =20

         Non-ICE:  Protocols similar to ICE=2C but not otherwise
      compliant=2C can

            also be implemented.  Negotiation of which NAT traversal
      mechanism

            is needed=2C is done by the application outside of the
      browser.

     =20

      [BA] There are some interoperability implications to these

      approaches that need to be explored.  For example=2C the ICE

      Javascript approach should enable interoperability

      between different browsers attached to the same service.  However=2C

      it is not clear to me that this will necessarily work for=20

      inter-service communication (e.g. the clients could have=20

      incompatible ICE Javascript libraries).=20

   =20
   =20

    If both javascript libraries implement ICE=2C they should by
    definition be interoperable - I am more worried about the case where
    one application supports ICE and another application supports some
    ICE-like mechanism.

   =20

   =20
 		 	   		  =

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<body class=3D'hmmessage'>
These kind of "performance" arguments tend to fall victim to improvements i=
n technology=2C so even if they can be demonstrated to be true at a given p=
oint in time=2C the argument may prove invalid years or even months down th=
e road.&nbsp=3B&nbsp=3B <br><br>In this particular case=2C there have been =
some major improvements in Javascript performance in recent browser release=
s=2C and it is quite possible that those improvements may continue.&nbsp=3B=
 <br><br>So even if such a demonstration were to "fail" on current releases=
=2C&nbsp=3B the target is not today's browser (which after all=2C doesn't s=
upport RTCWEB at all)=2C but rather future releases that do (and may also i=
ncorporate Javascript performance improvements). <br><br>To my mind=2C the =
real question isn't performance=2C it is interoperability.&nbsp=3B&nbsp=3B =
The "monolithic" approach to ICE brings with it the potential for interoper=
ability problems between different browsers operating on the same service.&=
nbsp=3B Mathew's approach of Javascript implementation in ICE addresses tha=
t issue=2C but I'm concerned about the implications for cross-service inter=
operability.&nbsp=3B <br><br><hr id=3D"stopSpelling">Date: Sat=2C 11 Jun 20=
11 23:23:04 +0200<br>From: harald@alvestrand.no<br>To: bernard_aboba@hotmai=
l.com<br>CC: rtcweb@ietf.org<br>Subject: Re: [rtcweb] Review of draft-kaufm=
an-rtcweb-traversal<br><br>

 =20
<meta http-equiv=3D"Content-Type" content=3D"text/html=3B charset=3Dunicode=
">
<meta name=3D"Generator" content=3D"Microsoft SafeHTML">
   =20
 =20
 =20
    Bernard=2C<br>
    <br>
    at last week's call=2C the question was raised on whether it's
    actually possible to implement ICE in javascript according to the
    described model - the Javascript event loop has certain problems
    with precise timing.<br>
    <br>
    The upshoot seemed to be that we'd like to see a demonstration that
    it's possible to create an ICE-interoperable implementation using
    Javascript as described before attempting to go further down this
    path.<br>
    <br>
    Some other comments below.<br>
    <br>
    On 06/11/2011 10:43 PM=2C Bernard Aboba wrote:
    <blockquote cite=3D"mid:BLU152-w64EE1689B126AA09DBD5593670@phx.gbl">
      <style>
.ExternalClass .ecxhmmessage P
{padding:0px=3B}
.ExternalClass body.ecxhmmessage
{font-size:10pt=3Bfont-family:Tahoma=3B}

</style>
      Review of draft-kaufman-rtcweb-traversal<br>
      <br>
      Overall=2C a very good document. <br>
      <br>
      Comments below:<br>
      <br>
      Section 2.2.&nbsp=3B Hampers Innovation<br>
      <br>
      &nbsp=3B&nbsp=3B One of the benefits of ICE is that it allows local
      implementation<br>
      &nbsp=3B&nbsp=3B flexibility in the way candidates are gathered=2C of=
fered and<br>
      &nbsp=3B&nbsp=3B prioritized.&nbsp=3B However=2C once ICE is baked in=
to the browser=2C it
      is no<br>
      &nbsp=3B&nbsp=3B longer possible for that innovation to take place - =
or at
      least=2C it<br>
      &nbsp=3B&nbsp=3B leaves the hands of the voice application providers.=
&nbsp=3B To date=2C
      there<br>
      &nbsp=3B&nbsp=3B has been variability in this aspect of implementatio=
n=2C with
      different<br>
      &nbsp=3B&nbsp=3B providers tuning it to tweak their needs and deploym=
ents.<br>
      <br>
      [BA] It might be pointed out this flexibility has already proven
      useful<br>
      to enable support for HTTP failover as well as in implementations
      of<br>
      ICE supporting IPV6.&nbsp=3B Given the wide variety of IPv6 deploymen=
ts
      in <br>
      progress=2C consideration experimentation and tweaking has been
      required <br>
      to handle the wide variety of conditions that can be experienced<br>
      (e.g. IPv6 tunnels of various flavors=2C network devices or services
      <br>
      with partial or no IPv6 support=2C&nbsp=3B missing IPv6 routes=2C NAT=
 64=2C
      etc.). <br>
    </blockquote>
    <br>
    This is good information=2C and it seems reasonable that it should
    lead to a rework of ICE based on this experience. Is it written up
    anywhere you can share?<br>
    <br>
    <blockquote cite=3D"mid:BLU152-w64EE1689B126AA09DBD5593670@phx.gbl">&nb=
sp=3B&nbsp=3B ICE in Javascript:&nbsp=3B A full ICE implementation is
      possible in<br>
      &nbsp=3B&nbsp=3B&nbsp=3B&nbsp=3B&nbsp=3B Javascript itself.&nbsp=3B B=
ecause the implementation resides in<br>
      &nbsp=3B&nbsp=3B&nbsp=3B&nbsp=3B&nbsp=3B Javasript=2C it is trivially=
 changed at any time.<br>
      <br>
      &nbsp=3B&nbsp=3B Server-Based ICE:&nbsp=3B A full ICE implementation =
can execute in the<br>
      &nbsp=3B&nbsp=3B&nbsp=3B&nbsp=3B&nbsp=3B server=2C using remote-contr=
ol commands to inform the browser
      to<br>
      &nbsp=3B&nbsp=3B&nbsp=3B&nbsp=3B&nbsp=3B send STUN transactions=2C an=
d passing the results from the
      browser<br>
      &nbsp=3B&nbsp=3B&nbsp=3B&nbsp=3B&nbsp=3B back to the server.&nbsp=3B =
In essence - MGCP for ICE.<br>
      <br>
      &nbsp=3B&nbsp=3B STUN-Only:&nbsp=3B For deployments where the peer is=
 always publically<br>
      &nbsp=3B&nbsp=3B&nbsp=3B&nbsp=3B&nbsp=3B reachable from clients - suc=
h as enterprises or PSTN
      termination<br>
      &nbsp=3B&nbsp=3B&nbsp=3B&nbsp=3B&nbsp=3B services - the Javascript ca=
n do a single STUN transaction
      to<br>
      &nbsp=3B&nbsp=3B&nbsp=3B&nbsp=3B&nbsp=3B create a permission in the b=
rowser=2C and then proceed to send<br>
      &nbsp=3B&nbsp=3B&nbsp=3B&nbsp=3B&nbsp=3B media.<br>
      <br>
      &nbsp=3B&nbsp=3B Non-ICE:&nbsp=3B Protocols similar to ICE=2C but not=
 otherwise
      compliant=2C can<br>
      &nbsp=3B&nbsp=3B&nbsp=3B&nbsp=3B&nbsp=3B also be implemented.&nbsp=3B=
 Negotiation of which NAT traversal
      mechanism<br>
      &nbsp=3B&nbsp=3B&nbsp=3B&nbsp=3B&nbsp=3B is needed=2C is done by the =
application outside of the
      browser.<br>
      <br>
      [BA] There are some interoperability implications to these<br>
      approaches that need to be explored.&nbsp=3B For example=2C the ICE<b=
r>
      Javascript approach should enable interoperability<br>
      between different browsers attached to the same service.&nbsp=3B Howe=
ver=2C<br>
      it is not clear to me that this will necessarily work for <br>
      inter-service communication (e.g. the clients could have <br>
      incompatible ICE Javascript libraries). <br>
    </blockquote>
    <br>
    If both javascript libraries implement ICE=2C they should by
    definition be interoperable - I am more worried about the case where
    one application supports ICE and another application supports some
    ICE-like mechanism.<br>
    <br>
    <br> 		 	   		  </body>
</html>=

--_51b0a978-7c3e-4e7e-9951-e8bdb41cb409_--

From dzonatas@gmail.com  Sat Jun 11 16:14:08 2011
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Date: Sat, 11 Jun 2011 16:12:50 -0700
From: Dzonatas Sol <dzonatas@gmail.com>
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+1 "a demonstration that it's possible to create an ICE-interoperable 
implementation using Javascript"

Especially if the media is intentionally legal size document (or legal 
size video...).

On 06/11/2011 02:23 PM, Harald Alvestrand wrote:
> Bernard,
>
> at last week's call, the question was raised on whether it's actually 
> possible to implement ICE in javascript according to the described 
> model - the Javascript event loop has certain problems with precise 
> timing.
>
> The upshoot seemed to be that we'd like to see a demonstration that 
> it's possible to create an ICE-interoperable implementation using 
> Javascript as described before attempting to go further down this path.
>
> Some other comments below.
>
> On 06/11/2011 10:43 PM, Bernard Aboba wrote:
>> Review of draft-kaufman-rtcweb-traversal
>>
>> Overall, a very good document.
>>
>> Comments below:
>>
>> Section 2.2.  Hampers Innovation
>>
>>    One of the benefits of ICE is that it allows local implementation
>>    flexibility in the way candidates are gathered, offered and
>>    prioritized.  However, once ICE is baked into the browser, it is no
>>    longer possible for that innovation to take place - or at least, it
>>    leaves the hands of the voice application providers.  To date, there
>>    has been variability in this aspect of implementation, with different
>>    providers tuning it to tweak their needs and deployments.
>>
>> [BA] It might be pointed out this flexibility has already proven useful
>> to enable support for HTTP failover as well as in implementations of
>> ICE supporting IPV6.  Given the wide variety of IPv6 deployments in
>> progress, consideration experimentation and tweaking has been required
>> to handle the wide variety of conditions that can be experienced
>> (e.g. IPv6 tunnels of various flavors, network devices or services
>> with partial or no IPv6 support,  missing IPv6 routes, NAT 64, etc.).
>
> This is good information, and it seems reasonable that it should lead 
> to a rework of ICE based on this experience. Is it written up anywhere 
> you can share?
>
>>    ICE in Javascript:  A full ICE implementation is possible in
>>       Javascript itself.  Because the implementation resides in
>>       Javasript, it is trivially changed at any time.
>>
>>    Server-Based ICE:  A full ICE implementation can execute in the
>>       server, using remote-control commands to inform the browser to
>>       send STUN transactions, and passing the results from the browser
>>       back to the server.  In essence - MGCP for ICE.
>>
>>    STUN-Only:  For deployments where the peer is always publically
>>       reachable from clients - such as enterprises or PSTN termination
>>       services - the Javascript can do a single STUN transaction to
>>       create a permission in the browser, and then proceed to send
>>       media.
>>
>>    Non-ICE:  Protocols similar to ICE, but not otherwise compliant, can
>>       also be implemented.  Negotiation of which NAT traversal mechanism
>>       is needed, is done by the application outside of the browser.
>>
>> [BA] There are some interoperability implications to these
>> approaches that need to be explored.  For example, the ICE
>> Javascript approach should enable interoperability
>> between different browsers attached to the same service.  However,
>> it is not clear to me that this will necessarily work for
>> inter-service communication (e.g. the clients could have
>> incompatible ICE Javascript libraries).
>
> If both javascript libraries implement ICE, they should by definition 
> be interoperable - I am more worried about the case where one 
> application supports ICE and another application supports some 
> ICE-like mechanism.
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>    


-- 
--- https://twitter.com/Dzonatas_Sol ---
Web Development, Software Engineering, Virtual Reality, Consultant


From harald@alvestrand.no  Sat Jun 11 22:55:47 2011
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On 06/12/2011 12:09 AM, Bernard Aboba wrote:
> These kind of "performance" arguments tend to fall victim to 
> improvements in technology, so even if they can be demonstrated to be 
> true at a given point in time, the argument may prove invalid years or 
> even months down the road.
>
> In this particular case, there have been some major improvements in 
> Javascript performance in recent browser releases, and it is quite 
> possible that those improvements may continue.
Actually my worry is more the javascript model than absolute 
performance. My tests with current browsers indicate that the timer 
resolutions for timed events are on the order of magnitude of 10-50 
msec, depending on browser, and the single-threaded model of the 
Javascript event queue means that response to timers depends on nothing 
else happening in that Javascript context. While I'm not a Javascript 
expert, I haven't had any reassurance that we can get around this from 
Javascript experts I've asked.
>
> So even if such a demonstration were to "fail" on current releases,  
> the target is not today's browser (which after all, doesn't support 
> RTCWEB at all), but rather future releases that do (and may also 
> incorporate Javascript performance improvements).
We might be taking on a bit much if we, in addition to specifying RTCWEB 
protocols, also take on rewriting the Javascript runtime model.
If all it takes is performance tweaking, I might be more optimistic; a 
demo running ICE at half the speed/timing precision needed would 
probably convince me it could be done.
>
> To my mind, the real question isn't performance, it is 
> interoperability.   The "monolithic" approach to ICE brings with it 
> the potential for interoperability problems between different browsers 
> operating on the same service.  Mathew's approach of Javascript 
> implementation in ICE addresses that issue, but I'm concerned about 
> the implications for cross-service interoperability.
One of my implementor colleagues claims (in another context) that a main 
difference between an embedded-in-browser implementation and a 
Javascript one is that the browser one has less than 10 significant 
implementations that have to interoperate, while the Javascript one will 
have an essentially unbounded number of implementations ("millions of 
web pages"). His conclusion was that for the things we need to get right 
in order to have interoperability (for the case we were discussing, 
which was not ICE), it's better to have them embedded in the browser 
than to depend on Javascript.
>
> ------------------------------------------------------------------------
> Date: Sat, 11 Jun 2011 23:23:04 +0200
> From: harald@alvestrand.no
> To: bernard_aboba@hotmail.com
> CC: rtcweb@ietf.org
> Subject: Re: [rtcweb] Review of draft-kaufman-rtcweb-traversal
>
> Bernard,
>
> at last week's call, the question was raised on whether it's actually 
> possible to implement ICE in javascript according to the described 
> model - the Javascript event loop has certain problems with precise 
> timing.
>
> The upshoot seemed to be that we'd like to see a demonstration that 
> it's possible to create an ICE-interoperable implementation using 
> Javascript as described before attempting to go further down this path.
>
> Some other comments below.
>
> On 06/11/2011 10:43 PM, Bernard Aboba wrote:
>
>     Review of draft-kaufman-rtcweb-traversal
>
>     Overall, a very good document.
>
>     Comments below:
>
>     Section 2.2.  Hampers Innovation
>
>        One of the benefits of ICE is that it allows local implementation
>        flexibility in the way candidates are gathered, offered and
>        prioritized.  However, once ICE is baked into the browser, it is no
>        longer possible for that innovation to take place - or at least, it
>        leaves the hands of the voice application providers.  To date,
>     there
>        has been variability in this aspect of implementation, with
>     different
>        providers tuning it to tweak their needs and deployments.
>
>     [BA] It might be pointed out this flexibility has already proven
>     useful
>     to enable support for HTTP failover as well as in implementations of
>     ICE supporting IPV6.  Given the wide variety of IPv6 deployments in
>     progress, consideration experimentation and tweaking has been
>     required
>     to handle the wide variety of conditions that can be experienced
>     (e.g. IPv6 tunnels of various flavors, network devices or services
>     with partial or no IPv6 support,  missing IPv6 routes, NAT 64, etc.).
>
>
> This is good information, and it seems reasonable that it should lead 
> to a rework of ICE based on this experience. Is it written up anywhere 
> you can share?
>
>        ICE in Javascript:  A full ICE implementation is possible in
>           Javascript itself.  Because the implementation resides in
>           Javasript, it is trivially changed at any time.
>
>        Server-Based ICE:  A full ICE implementation can execute in the
>           server, using remote-control commands to inform the browser to
>           send STUN transactions, and passing the results from the browser
>           back to the server.  In essence - MGCP for ICE.
>
>        STUN-Only:  For deployments where the peer is always publically
>           reachable from clients - such as enterprises or PSTN termination
>           services - the Javascript can do a single STUN transaction to
>           create a permission in the browser, and then proceed to send
>           media.
>
>        Non-ICE:  Protocols similar to ICE, but not otherwise
>     compliant, can
>           also be implemented.  Negotiation of which NAT traversal
>     mechanism
>           is needed, is done by the application outside of the browser.
>
>     [BA] There are some interoperability implications to these
>     approaches that need to be explored.  For example, the ICE
>     Javascript approach should enable interoperability
>     between different browsers attached to the same service.  However,
>     it is not clear to me that this will necessarily work for
>     inter-service communication (e.g. the clients could have
>     incompatible ICE Javascript libraries).
>
>
> If both javascript libraries implement ICE, they should by definition 
> be interoperable - I am more worried about the case where one 
> application supports ICE and another application supports some 
> ICE-like mechanism.
>
>


--------------080106080206060403010608
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">
  </head>
  <body text="#000000" bgcolor="#ffffff">
    On 06/12/2011 12:09 AM, Bernard Aboba wrote:
    <blockquote cite="mid:BLU152-w6357991C9692D84FF2916393670@phx.gbl"
      type="cite">
      <style><!--
.hmmessage P
{
margin:0px;
padding:0px
}
body.hmmessage
{
font-size: 10pt;
font-family:Tahoma
}
--></style>
      These kind of "performance" arguments tend to fall victim to
      improvements in technology, so even if they can be demonstrated to
      be true at a given point in time, the argument may prove invalid
      years or even months down the road.&nbsp;&nbsp; <br>
      <br>
      In this particular case, there have been some major improvements
      in Javascript performance in recent browser releases, and it is
      quite possible that those improvements may continue.&nbsp; <br>
    </blockquote>
    Actually my worry is more the javascript model than absolute
    performance. My tests with current browsers indicate that the timer
    resolutions for timed events are on the order of magnitude of 10-50
    msec, depending on browser, and the single-threaded model of the
    Javascript event queue means that response to timers depends on
    nothing else happening in that Javascript context. While I'm not a
    Javascript expert, I haven't had any reassurance that we can get
    around this from Javascript experts I've asked.<br>
    <blockquote cite="mid:BLU152-w6357991C9692D84FF2916393670@phx.gbl"
      type="cite"><br>
      So even if such a demonstration were to "fail" on current
      releases,&nbsp; the target is not today's browser (which after all,
      doesn't support RTCWEB at all), but rather future releases that do
      (and may also incorporate Javascript performance improvements). <br>
    </blockquote>
    We might be taking on a bit much if we, in addition to specifying
    RTCWEB protocols, also take on rewriting the Javascript runtime
    model.<br>
    If all it takes is performance tweaking, I might be more optimistic;
    a demo running ICE at half the speed/timing precision needed would
    probably convince me it could be done.<br>
    <blockquote cite="mid:BLU152-w6357991C9692D84FF2916393670@phx.gbl"
      type="cite"><br>
      To my mind, the real question isn't performance, it is
      interoperability.&nbsp;&nbsp; The "monolithic" approach to ICE brings with
      it the potential for interoperability problems between different
      browsers operating on the same service.&nbsp; Mathew's approach of
      Javascript implementation in ICE addresses that issue, but I'm
      concerned about the implications for cross-service
      interoperability.&nbsp; <br>
    </blockquote>
    One of my implementor colleagues claims (in another context) that a
    main difference between an embedded-in-browser implementation and a
    Javascript one is that the browser one has less than 10 significant
    implementations that have to interoperate, while the Javascript one
    will have an essentially unbounded number of implementations
    ("millions of web pages"). His conclusion was that for the things we
    need to get right in order to have interoperability (for the case we
    were discussing, which was not ICE), it's better to have them
    embedded in the browser than to depend on Javascript.<br>
    <blockquote cite="mid:BLU152-w6357991C9692D84FF2916393670@phx.gbl"
      type="cite"><br>
      <hr id="stopSpelling">Date: Sat, 11 Jun 2011 23:23:04 +0200<br>
      From: <a class="moz-txt-link-abbreviated" href="mailto:harald@alvestrand.no">harald@alvestrand.no</a><br>
      To: <a class="moz-txt-link-abbreviated" href="mailto:bernard_aboba@hotmail.com">bernard_aboba@hotmail.com</a><br>
      CC: <a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
      Subject: Re: [rtcweb] Review of draft-kaufman-rtcweb-traversal<br>
      <br>
      <meta http-equiv="Content-Type" content="text/html;
        charset=ISO-8859-1">
      <meta name="Generator" content="Microsoft SafeHTML">
      Bernard,<br>
      <br>
      at last week's call, the question was raised on whether it's
      actually possible to implement ICE in javascript according to the
      described model - the Javascript event loop has certain problems
      with precise timing.<br>
      <br>
      The upshoot seemed to be that we'd like to see a demonstration
      that it's possible to create an ICE-interoperable implementation
      using Javascript as described before attempting to go further down
      this path.<br>
      <br>
      Some other comments below.<br>
      <br>
      On 06/11/2011 10:43 PM, Bernard Aboba wrote:
      <blockquote cite="mid:BLU152-w64EE1689B126AA09DBD5593670@phx.gbl">
        <style>
.ExternalClass .ecxhmmessage P
{padding:0px;}
.ExternalClass body.ecxhmmessage
{font-size:10pt;font-family:Tahoma;}

</style> Review of draft-kaufman-rtcweb-traversal<br>
        <br>
        Overall, a very good document. <br>
        <br>
        Comments below:<br>
        <br>
        Section 2.2.&nbsp; Hampers Innovation<br>
        <br>
        &nbsp;&nbsp; One of the benefits of ICE is that it allows local
        implementation<br>
        &nbsp;&nbsp; flexibility in the way candidates are gathered, offered and<br>
        &nbsp;&nbsp; prioritized.&nbsp; However, once ICE is baked into the browser, it
        is no<br>
        &nbsp;&nbsp; longer possible for that innovation to take place - or at
        least, it<br>
        &nbsp;&nbsp; leaves the hands of the voice application providers.&nbsp; To
        date, there<br>
        &nbsp;&nbsp; has been variability in this aspect of implementation, with
        different<br>
        &nbsp;&nbsp; providers tuning it to tweak their needs and deployments.<br>
        <br>
        [BA] It might be pointed out this flexibility has already proven
        useful<br>
        to enable support for HTTP failover as well as in
        implementations of<br>
        ICE supporting IPV6.&nbsp; Given the wide variety of IPv6 deployments
        in <br>
        progress, consideration experimentation and tweaking has been
        required <br>
        to handle the wide variety of conditions that can be experienced<br>
        (e.g. IPv6 tunnels of various flavors, network devices or
        services <br>
        with partial or no IPv6 support,&nbsp; missing IPv6 routes, NAT 64,
        etc.). <br>
      </blockquote>
      <br>
      This is good information, and it seems reasonable that it should
      lead to a rework of ICE based on this experience. Is it written up
      anywhere you can share?<br>
      <br>
      <blockquote cite="mid:BLU152-w64EE1689B126AA09DBD5593670@phx.gbl">&nbsp;&nbsp;
        ICE in Javascript:&nbsp; A full ICE implementation is possible in<br>
        &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Javascript itself.&nbsp; Because the implementation resides in<br>
        &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Javasript, it is trivially changed at any time.<br>
        <br>
        &nbsp;&nbsp; Server-Based ICE:&nbsp; A full ICE implementation can execute in
        the<br>
        &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; server, using remote-control commands to inform the
        browser to<br>
        &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; send STUN transactions, and passing the results from the
        browser<br>
        &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; back to the server.&nbsp; In essence - MGCP for ICE.<br>
        <br>
        &nbsp;&nbsp; STUN-Only:&nbsp; For deployments where the peer is always
        publically<br>
        &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; reachable from clients - such as enterprises or PSTN
        termination<br>
        &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; services - the Javascript can do a single STUN transaction
        to<br>
        &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; create a permission in the browser, and then proceed to
        send<br>
        &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; media.<br>
        <br>
        &nbsp;&nbsp; Non-ICE:&nbsp; Protocols similar to ICE, but not otherwise
        compliant, can<br>
        &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; also be implemented.&nbsp; Negotiation of which NAT traversal
        mechanism<br>
        &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; is needed, is done by the application outside of the
        browser.<br>
        <br>
        [BA] There are some interoperability implications to these<br>
        approaches that need to be explored.&nbsp; For example, the ICE<br>
        Javascript approach should enable interoperability<br>
        between different browsers attached to the same service.&nbsp;
        However,<br>
        it is not clear to me that this will necessarily work for <br>
        inter-service communication (e.g. the clients could have <br>
        incompatible ICE Javascript libraries). <br>
      </blockquote>
      <br>
      If both javascript libraries implement ICE, they should by
      definition be interoperable - I am more worried about the case
      where one application supports ICE and another application
      supports some ICE-like mechanism.<br>
      <br>
      <br>
    </blockquote>
    <br>
  </body>
</html>

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From: Matthew Kaufman <matthew.kaufman@skype.net>
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On Jun 12, 2011, at 7:55 AM, Harald Alvestrand wrote:

> On 06/12/2011 12:09 AM, Bernard Aboba wrote:
>>=20
>> These kind of "performance" arguments tend to fall victim to =
improvements in technology, so even if they can be demonstrated to be =
true at a given point in time, the argument may prove invalid years or =
even months down the road.  =20
>>=20
>> In this particular case, there have been some major improvements in =
Javascript performance in recent browser releases, and it is quite =
possible that those improvements may continue. =20
> Actually my worry is more the javascript model than absolute =
performance. My tests with current browsers indicate that the timer =
resolutions for timed events are on the order of magnitude of 10-50 =
msec, depending on browser, and the single-threaded model of the =
Javascript event queue means that response to timers depends on nothing =
else happening in that Javascript context. While I'm not a Javascript =
expert, I haven't had any reassurance that we can get around this from =
Javascript experts I've asked.

The resolution of waking up for a timer on WinNT is 30 msec and only 15 =
msec on WinXP and ICE works just fine there.

I am not at all convinced that for our use cases we need anything better =
than ~100 msec, so your 50 msec is already twice as good.

>>=20
>> So even if such a demonstration were to "fail" on current releases,  =
the target is not today's browser (which after all, doesn't support =
RTCWEB at all), but rather future releases that do (and may also =
incorporate Javascript performance improvements).=20
> We might be taking on a bit much if we, in addition to specifying =
RTCWEB protocols, also take on rewriting the Javascript runtime model.
> If all it takes is performance tweaking, I might be more optimistic; a =
demo running ICE at half the speed/timing precision needed would =
probably convince me it could be done.

See above.

>>=20
>> To my mind, the real question isn't performance, it is =
interoperability.   The "monolithic" approach to ICE brings with it the =
potential for interoperability problems between different browsers =
operating on the same service.  Mathew's approach of Javascript =
implementation in ICE addresses that issue, but I'm concerned about the =
implications for cross-service interoperability. =20
> One of my implementor colleagues claims (in another context) that a =
main difference between an embedded-in-browser implementation and a =
Javascript one is that the browser one has less than 10 significant =
implementations that have to interoperate, while the Javascript one will =
have an essentially unbounded number of implementations ("millions of =
web pages"). His conclusion was that for the things we need to get right =
in order to have interoperability (for the case we were discussing, =
which was not ICE), it's better to have them embedded in the browser =
than to depend on Javascript.

This is actually the only good argument I've heard... that if the =
"right" implementation is baked into browsers then it will "just work". =
On the other hand, we have numerous examples of things that are baked =
into browsers that would be much more useful/flexible if they weren't... =
or at the very least offered both options. (And the future flexibility =
requirements around IPv6 are certainly a hint of why we might need this, =
in addition to service providers with non-standard needs.)

What about having a full ICE implementation *and* the ability to run it =
"manually" under Javascript control?

(The example that keeps coming to mind is the difference between =
progressive playback of HTTP video (as is in Flash Player and most HTML5 =
video tag implementation) vs. adaptive HTTP streaming (which, in Flash =
Player was architected as low-level ActionScript primitives plus =
reference code, and which has already done several things that weren't =
originally imagined when the APIs were first added))

Matthew Kaufman=

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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 Jun 12, 2011, at 7:55 AM, Harald Alvestrand =
wrote:</div><br class=3D"Apple-interchange-newline"><blockquote =
type=3D"cite">
<div text=3D"#000000" bgcolor=3D"#ffffff">
    On 06/12/2011 12:09 AM, Bernard Aboba wrote:
    <blockquote cite=3D"mid:BLU152-w6357991C9692D84FF2916393670@phx.gbl" =
type=3D"cite">
      <style><!--
.hmmessage P
{
margin:0px;
padding:0px
}
body.hmmessage
{
font-size: 10pt;
font-family:Tahoma
}
--></style>
      These kind of "performance" arguments tend to fall victim to
      improvements in technology, so even if they can be demonstrated to
      be true at a given point in time, the argument may prove invalid
      years or even months down the road.&nbsp;&nbsp; <br>
      <br>
      In this particular case, there have been some major improvements
      in Javascript performance in recent browser releases, and it is
      quite possible that those improvements may continue.&nbsp; <br>
    </blockquote>
    Actually my worry is more the javascript model than absolute
    performance. My tests with current browsers indicate that the timer
    resolutions for timed events are on the order of magnitude of 10-50
    msec, depending on browser, and the single-threaded model of the
    Javascript event queue means that response to timers depends on
    nothing else happening in that Javascript context. While I'm not a
    Javascript expert, I haven't had any reassurance that we can get
    around this from Javascript experts I've =
asked.<br></div></blockquote><div><br></div>The resolution of waking up =
for a timer on WinNT is 30 msec and only 15 msec on WinXP and ICE works =
just fine there.</div><div><br></div><div>I am not at all convinced that =
for our use cases we need anything better than ~100 msec, so your 50 =
msec is already twice as good.</div><div><br><blockquote =
type=3D"cite"><div text=3D"#000000" bgcolor=3D"#ffffff">
    <blockquote cite=3D"mid:BLU152-w6357991C9692D84FF2916393670@phx.gbl" =
type=3D"cite"><br>
      So even if such a demonstration were to "fail" on current
      releases,&nbsp; the target is not today's browser (which after =
all,
      doesn't support RTCWEB at all), but rather future releases that do
      (and may also incorporate Javascript performance improvements). =
<br>
    </blockquote>
    We might be taking on a bit much if we, in addition to specifying
    RTCWEB protocols, also take on rewriting the Javascript runtime
    model.<br>
    If all it takes is performance tweaking, I might be more optimistic;
    a demo running ICE at half the speed/timing precision needed would
    probably convince me it could be =
done.<br></div></blockquote><div><br></div>See =
above.</div><div><br><blockquote type=3D"cite"><div text=3D"#000000" =
bgcolor=3D"#ffffff">
    <blockquote cite=3D"mid:BLU152-w6357991C9692D84FF2916393670@phx.gbl" =
type=3D"cite"><br>
      To my mind, the real question isn't performance, it is
      interoperability.&nbsp;&nbsp; The "monolithic" approach to ICE =
brings with
      it the potential for interoperability problems between different
      browsers operating on the same service.&nbsp; Mathew's approach of
      Javascript implementation in ICE addresses that issue, but I'm
      concerned about the implications for cross-service
      interoperability.&nbsp; <br>
    </blockquote>
    One of my implementor colleagues claims (in another context) that a
    main difference between an embedded-in-browser implementation and a
    Javascript one is that the browser one has less than 10 significant
    implementations that have to interoperate, while the Javascript one
    will have an essentially unbounded number of implementations
    ("millions of web pages"). His conclusion was that for the things we
    need to get right in order to have interoperability (for the case we
    were discussing, which was not ICE), it's better to have them
    embedded in the browser than to depend on =
Javascript.<br></div></blockquote><div><br></div><div>This is actually =
the only good argument I've heard... that if the "right" implementation =
is baked into browsers then it will "just work". On the other hand, we =
have numerous examples of things that are baked into browsers that would =
be much more useful/flexible if they weren't... or at the very least =
offered both options. (And the future flexibility requirements around =
IPv6 are certainly a hint of why we might need this, in addition to =
service providers with non-standard =
needs.)</div><div><br></div><div>What about having a full ICE =
implementation *and* the ability to run it "manually" under Javascript =
control?</div><div><br></div><div>(The example that keeps coming to mind =
is the difference between progressive playback of HTTP video (as is in =
Flash Player and most HTML5 video tag implementation) vs. adaptive HTTP =
streaming (which, in Flash Player was architected as low-level =
ActionScript primitives plus reference code, and which has already done =
several things that weren't originally imagined when the APIs were first =
added))</div><div><br></div><div>Matthew =
Kaufman</div></div></body></html>=

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From tim@phonefromhere.com  Sun Jun 12 02:36:44 2011
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To: Matthew Kaufman <matthew.kaufman@skype.net>
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On 12 Jun 2011, at 10:05, Matthew Kaufman wrote:

>=20
> On Jun 12, 2011, at 7:55 AM, Harald Alvestrand wrote:
>=20
>> On 06/12/2011 12:09 AM, Bernard Aboba wrote:
>>>=20
>>>=20
>=20
>>> To my mind, the real question isn't performance, it is =
interoperability.   The "monolithic" approach to ICE brings with it the =
potential for interoperability problems between different browsers =
operating on the same service.  Mathew's approach of Javascript =
implementation in ICE addresses that issue, but I'm concerned about the =
implications for cross-service interoperability. =20
>> One of my implementor colleagues claims (in another context) that a =
main difference between an embedded-in-browser implementation and a =
Javascript one is that the browser one has less than 10 significant =
implementations that have to interoperate, while the Javascript one will =
have an essentially unbounded number of implementations ("millions of =
web pages"). His conclusion was that for the things we need to get right =
in order to have interoperability (for the case we were discussing, =
which was not ICE), it's better to have them embedded in the browser =
than to depend on Javascript.
>=20
> This is actually the only good argument I've heard... that if the =
"right" implementation is baked into browsers then it will "just work". =
On the other hand, we have numerous examples of things that are baked =
into browsers that would be much more useful/flexible if they weren't... =
or at the very least offered both options. (And the future flexibility =
requirements around IPv6 are certainly a hint of why we might need this, =
in addition to service providers with non-standard needs.)

Maybe I've missed something, but to my mind there is no need for the =
javascript ICE's to interoperate with each other. All the users of =
XYZ-webconference service will load _exactly_the_same_ javascript - so =
for them to talk peer-to-peer the XYZ-ICE-lite implementation only has =
to interoperate with itself.

Are there important use-cases where we expect to have a visitor on =
XYZ.com web-page calling to ABC.com ? If we do then a baked in ICE=20
may be needed.

Tim.=

From harald@alvestrand.no  Sun Jun 12 02:58:05 2011
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On 06/12/2011 11:05 AM, Matthew Kaufman wrote:
>
> On Jun 12, 2011, at 7:55 AM, Harald Alvestrand wrote:
>
>> On 06/12/2011 12:09 AM, Bernard Aboba wrote:
>>> These kind of "performance" arguments tend to fall victim to 
>>> improvements in technology, so even if they can be demonstrated to 
>>> be true at a given point in time, the argument may prove invalid 
>>> years or even months down the road.
>>>
>>> In this particular case, there have been some major improvements in 
>>> Javascript performance in recent browser releases, and it is quite 
>>> possible that those improvements may continue.
>> Actually my worry is more the javascript model than absolute 
>> performance. My tests with current browsers indicate that the timer 
>> resolutions for timed events are on the order of magnitude of 10-50 
>> msec, depending on browser, and the single-threaded model of the 
>> Javascript event queue means that response to timers depends on 
>> nothing else happening in that Javascript context. While I'm not a 
>> Javascript expert, I haven't had any reassurance that we can get 
>> around this from Javascript experts I've asked.
>
> The resolution of waking up for a timer on WinNT is 30 msec and only 
> 15 msec on WinXP and ICE works just fine there.
>
> I am not at all convinced that for our use cases we need anything 
> better than ~100 msec, so your 50 msec is already twice as good.
The shortest timer in RFC 5245 (as far as I can see) is Ta (section 
16.1), which has a lower bound of 20 msec; however, this is a lower 
bound on transmission intervals, so it's possible that an implementation 
that fires only every 100 msec would be perfectly interoperable with an 
application that fires every 20 msec.

Testing will show....

>
>>>
>>> So even if such a demonstration were to "fail" on current releases,  
>>> the target is not today's browser (which after all, doesn't support 
>>> RTCWEB at all), but rather future releases that do (and may also 
>>> incorporate Javascript performance improvements).
>> We might be taking on a bit much if we, in addition to specifying 
>> RTCWEB protocols, also take on rewriting the Javascript runtime model.
>> If all it takes is performance tweaking, I might be more optimistic; 
>> a demo running ICE at half the speed/timing precision needed would 
>> probably convince me it could be done.
>
> See above.
>
>>>
>>> To my mind, the real question isn't performance, it is 
>>> interoperability.   The "monolithic" approach to ICE brings with it 
>>> the potential for interoperability problems between different 
>>> browsers operating on the same service.  Mathew's approach of 
>>> Javascript implementation in ICE addresses that issue, but I'm 
>>> concerned about the implications for cross-service interoperability.
>> One of my implementor colleagues claims (in another context) that a 
>> main difference between an embedded-in-browser implementation and a 
>> Javascript one is that the browser one has less than 10 significant 
>> implementations that have to interoperate, while the Javascript one 
>> will have an essentially unbounded number of implementations 
>> ("millions of web pages"). His conclusion was that for the things we 
>> need to get right in order to have interoperability (for the case we 
>> were discussing, which was not ICE), it's better to have them 
>> embedded in the browser than to depend on Javascript.
>
> This is actually the only good argument I've heard... that if the 
> "right" implementation is baked into browsers then it will "just 
> work". On the other hand, we have numerous examples of things that are 
> baked into browsers that would be much more useful/flexible if they 
> weren't... or at the very least offered both options. (And the future 
> flexibility requirements around IPv6 are certainly a hint of why we 
> might need this, in addition to service providers with non-standard 
> needs.)
>
> What about having a full ICE implementation *and* the ability to run 
> it "manually" under Javascript control?
I'd be happy to have a spec that allows us to run an "under the covers" 
fully compatible ICE implementation in the "normal case" (like what's 
currently implemented in Google's libjingle/WebRTC code), and have an 
extension (or another API spec) that allows "manual" driving of the STUN 
machinery, which we can agree on once it's proved viable. I'd be very 
unhappy at this point to have only a spec that required a 
partly-Javascript ICE implementation.

But that's my preferences.
>
> (The example that keeps coming to mind is the difference between 
> progressive playback of HTTP video (as is in Flash Player and most 
> HTML5 video tag implementation) vs. adaptive HTTP streaming (which, in 
> Flash Player was architected as low-level ActionScript primitives plus 
> reference code, and which has already done several things that weren't 
> originally imagined when the APIs were first added))
>
> Matthew Kaufman


--------------020800090504010509030308
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
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    On 06/12/2011 11:05 AM, Matthew Kaufman wrote:
    <blockquote
      cite="mid:6DA32DD7-6BB4-445C-B119-292246E24B0A@skype.net"
      type="cite"><br>
      <div>
        <div>On Jun 12, 2011, at 7:55 AM, Harald Alvestrand wrote:</div>
        <br class="Apple-interchange-newline">
        <blockquote type="cite">
          <div text="#000000" bgcolor="#ffffff"> On 06/12/2011 12:09 AM,
            Bernard Aboba wrote:
            <blockquote
              cite="mid:BLU152-w6357991C9692D84FF2916393670@phx.gbl"
              type="cite">
              <style><!--
.hmmessage P
{
margin:0px;
padding:0px
}
body.hmmessage
{
font-size: 10pt;
font-family:Tahoma
}
--></style> These kind of "performance" arguments tend to fall victim to
              improvements in technology, so even if they can be
              demonstrated to be true at a given point in time, the
              argument may prove invalid years or even months down the
              road.&nbsp;&nbsp; <br>
              <br>
              In this particular case, there have been some major
              improvements in Javascript performance in recent browser
              releases, and it is quite possible that those improvements
              may continue.&nbsp; <br>
            </blockquote>
            Actually my worry is more the javascript model than absolute
            performance. My tests with current browsers indicate that
            the timer resolutions for timed events are on the order of
            magnitude of 10-50 msec, depending on browser, and the
            single-threaded model of the Javascript event queue means
            that response to timers depends on nothing else happening in
            that Javascript context. While I'm not a Javascript expert,
            I haven't had any reassurance that we can get around this
            from Javascript experts I've asked.<br>
          </div>
        </blockquote>
        <div><br>
        </div>
        The resolution of waking up for a timer on WinNT is 30 msec and
        only 15 msec on WinXP and ICE works just fine there.</div>
      <div><br>
      </div>
      <div>I am not at all convinced that for our use cases we need
        anything better than ~100 msec, so your 50 msec is already twice
        as good.</div>
    </blockquote>
    The shortest timer in RFC 5245 (as far as I can see) is Ta (section
    16.1), which has a lower bound of 20 msec; however, this is a lower
    bound on transmission intervals, so it's possible that an
    implementation that fires only every 100 msec would be perfectly
    interoperable with an application that fires every 20 msec.<br>
    <br>
    Testing will show....<br>
    <br>
    <blockquote
      cite="mid:6DA32DD7-6BB4-445C-B119-292246E24B0A@skype.net"
      type="cite">
      <div><br>
        <blockquote type="cite">
          <div text="#000000" bgcolor="#ffffff">
            <blockquote
              cite="mid:BLU152-w6357991C9692D84FF2916393670@phx.gbl"
              type="cite"><br>
              So even if such a demonstration were to "fail" on current
              releases,&nbsp; the target is not today's browser (which after
              all, doesn't support RTCWEB at all), but rather future
              releases that do (and may also incorporate Javascript
              performance improvements). <br>
            </blockquote>
            We might be taking on a bit much if we, in addition to
            specifying RTCWEB protocols, also take on rewriting the
            Javascript runtime model.<br>
            If all it takes is performance tweaking, I might be more
            optimistic; a demo running ICE at half the speed/timing
            precision needed would probably convince me it could be
            done.<br>
          </div>
        </blockquote>
        <div><br>
        </div>
        See above.</div>
      <div><br>
        <blockquote type="cite">
          <div text="#000000" bgcolor="#ffffff">
            <blockquote
              cite="mid:BLU152-w6357991C9692D84FF2916393670@phx.gbl"
              type="cite"><br>
              To my mind, the real question isn't performance, it is
              interoperability.&nbsp;&nbsp; The "monolithic" approach to ICE
              brings with it the potential for interoperability problems
              between different browsers operating on the same service.&nbsp;
              Mathew's approach of Javascript implementation in ICE
              addresses that issue, but I'm concerned about the
              implications for cross-service interoperability.&nbsp; <br>
            </blockquote>
            One of my implementor colleagues claims (in another context)
            that a main difference between an embedded-in-browser
            implementation and a Javascript one is that the browser one
            has less than 10 significant implementations that have to
            interoperate, while the Javascript one will have an
            essentially unbounded number of implementations ("millions
            of web pages"). His conclusion was that for the things we
            need to get right in order to have interoperability (for the
            case we were discussing, which was not ICE), it's better to
            have them embedded in the browser than to depend on
            Javascript.<br>
          </div>
        </blockquote>
        <div><br>
        </div>
        <div>This is actually the only good argument I've heard... that
          if the "right" implementation is baked into browsers then it
          will "just work". On the other hand, we have numerous examples
          of things that are baked into browsers that would be much more
          useful/flexible if they weren't... or at the very least
          offered both options. (And the future flexibility requirements
          around IPv6 are certainly a hint of why we might need this, in
          addition to service providers with non-standard needs.)</div>
        <div><br>
        </div>
        <div>What about having a full ICE implementation *and* the
          ability to run it "manually" under Javascript control?</div>
      </div>
    </blockquote>
    I'd be happy to have a spec that allows us to run an "under the
    covers" fully compatible ICE implementation in the "normal case"
    (like what's currently implemented in Google's libjingle/WebRTC
    code), and have an extension (or another API spec) that allows
    "manual" driving of the STUN machinery, which we can agree on once
    it's proved viable. I'd be very unhappy at this point to have only a
    spec that required a partly-Javascript ICE implementation.<br>
    <br>
    But that's my preferences.<br>
    <blockquote
      cite="mid:6DA32DD7-6BB4-445C-B119-292246E24B0A@skype.net"
      type="cite">
      <div>
        <div><br>
        </div>
        <div>(The example that keeps coming to mind is the difference
          between progressive playback of HTTP video (as is in Flash
          Player and most HTML5 video tag implementation) vs. adaptive
          HTTP streaming (which, in Flash Player was architected as
          low-level ActionScript primitives plus reference code, and
          which has already done several things that weren't originally
          imagined when the APIs were first added))</div>
        <div><br>
        </div>
        <div>Matthew Kaufman</div>
      </div>
    </blockquote>
    <br>
  </body>
</html>

--------------020800090504010509030308--

From emil@sip-communicator.org  Sun Jun 12 03:02:49 2011
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Cc: Bernard Aboba <bernard_aboba@hotmail.com>, rtcweb@ietf.org
Subject: Re: [rtcweb] Review of draft-kaufman-rtcweb-traversal
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=D0=9D=D0=B0 12.06.11 11:36, Tim Panton =D0=BD=D0=B0=D0=BF=D0=B8=D1=81=D0=
=B0:
> Maybe I've missed something, but to my mind there is no need for the
> javascript ICE's to interoperate with each other. All the users of
> XYZ-webconference service will load _exactly_the_same_ javascript -
> so for them to talk peer-to-peer the XYZ-ICE-lite implementation only
> has to interoperate with itself.
>=20
> Are there important use-cases where we expect to have a visitor on
> XYZ.com web-page calling to ABC.com ? If we do then a baked in ICE=20
> may be needed.

Unless we are planning on promoting a skype-like model where the entire
world is stuck with the same provider and where PSTN is the only way to
have some sort of interop ... then I'd say that inter-domain calls are
the only use case that we have to consider.

The Gmail web client is a great example for how things could work. A
bunch of clients out there, connected to random XMPP services, can talk
to users connected on Gmail and this is great for both Gmail users and
those who'd prefer to stick with those other clients.

Emil


From matthew.kaufman@skype.net  Sun Jun 12 03:25:05 2011
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On Jun 12, 2011, at 11:36 AM, Tim Panton wrote:

>=20
> On 12 Jun 2011, at 10:05, Matthew Kaufman wrote:
>=20
>>=20
>> On Jun 12, 2011, at 7:55 AM, Harald Alvestrand wrote:
>>=20
>>> On 06/12/2011 12:09 AM, Bernard Aboba wrote:
>>>>=20
>>>>=20
>>=20
>>>> To my mind, the real question isn't performance, it is =
interoperability.   The "monolithic" approach to ICE brings with it the =
potential for interoperability problems between different browsers =
operating on the same service.  Mathew's approach of Javascript =
implementation in ICE addresses that issue, but I'm concerned about the =
implications for cross-service interoperability. =20
>>> One of my implementor colleagues claims (in another context) that a =
main difference between an embedded-in-browser implementation and a =
Javascript one is that the browser one has less than 10 significant =
implementations that have to interoperate, while the Javascript one will =
have an essentially unbounded number of implementations ("millions of =
web pages"). His conclusion was that for the things we need to get right =
in order to have interoperability (for the case we were discussing, =
which was not ICE), it's better to have them embedded in the browser =
than to depend on Javascript.
>>=20
>> This is actually the only good argument I've heard... that if the =
"right" implementation is baked into browsers then it will "just work". =
On the other hand, we have numerous examples of things that are baked =
into browsers that would be much more useful/flexible if they weren't... =
or at the very least offered both options. (And the future flexibility =
requirements around IPv6 are certainly a hint of why we might need this, =
in addition to service providers with non-standard needs.)
>=20
> Maybe I've missed something, but to my mind there is no need for the =
javascript ICE's to interoperate with each other. All the users of =
XYZ-webconference service will load _exactly_the_same_ javascript - so =
for them to talk peer-to-peer the XYZ-ICE-lite implementation only has =
to interoperate with itself.

Definitely true.

>=20
> Are there important use-cases where we expect to have a visitor on =
XYZ.com web-page calling to ABC.com ? If we do then a baked in ICE=20
> may be needed.
>=20

Not the case. One can build a perfectly interoperable ICE using =
Javascript, or a combination of Javascript calls and server-side ICE =
logic.

But when XYZ.com is calling itself it might be able to either save some =
of the steps, or do an even better job of NAT/firewall traversal by =
using more techniques than were baked in.

Matthew Kaufman


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Cc: Bernard Aboba <bernard_aboba@hotmail.com>, rtcweb@ietf.org
Subject: Re: [rtcweb] Review of draft-kaufman-rtcweb-traversal
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=D0=9D=D0=B0 12.06.11 11:57, Harald Alvestrand =D0=BD=D0=B0=D0=BF=D0=B8=D1=
=81=D0=B0:
> I'd be happy to have a spec that allows us to run an "under the covers"=

> fully compatible ICE implementation in the "normal case" (like what's
> currently implemented in Google's libjingle/WebRTC code), and have an
> extension (or another API spec) that allows "manual" driving of the STU=
N
> machinery, which we can agree on once it's proved viable. I'd be very
> unhappy at this point to have only a spec that required a
> partly-Javascript ICE implementation.

+1

Besides, unless I've missed something, most of the arguments against an
in-browser ICE implementation, like being able to use port-prediction,
IPv6 tweaks, etc. are about the way an application would harvest candidat=
es.

If we agree on that then wouldn't it make sense to make that part
extensible by applications (i.e. javascript), and have the rest handled
by the browser? In other words, the browser could come with a set of
default harvesters (i.e. host, stun, turn) and allow applications to
provide additional ones.

Connectivity checks, which are arguably where 90% of the ICE complexity
is can be handled by the browser which would also worry timing and such.
We've followed a similar model in ice4j.org and it has proven to be
rather practical.

Emil


From tim@phonefromhere.com  Sun Jun 12 04:21:57 2011
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On 12 Jun 2011, at 11:02, Emil Ivov wrote:

> =D0=9D=D0=B0 12.06.11 11:36, Tim Panton =D0=BD=D0=B0=D0=BF=D0=B8=D1=81=D0=
=B0:
>> Maybe I've missed something, but to my mind there is no need for the
>> javascript ICE's to interoperate with each other. All the users of
>> XYZ-webconference service will load _exactly_the_same_ javascript -
>> so for them to talk peer-to-peer the XYZ-ICE-lite implementation only
>> has to interoperate with itself.
>>=20
>> Are there important use-cases where we expect to have a visitor on
>> XYZ.com web-page calling to ABC.com ? If we do then a baked in ICE=20
>> may be needed.
>=20
> Unless we are planning on promoting a skype-like model where the =
entire
> world is stuck with the same provider and where PSTN is the only way =
to
> have some sort of interop ... then I'd say that inter-domain calls are
> the only use case that we have to consider.
>=20
Ok, we have explicit non-requirements that cross web site signalling=20
and auth are compatible (We push these issues up to the web service), =
but we propose=20
to mandate that the ICE is identical ? I'm not clear why.

Lets say in my hypothetical case of  an XYZ.com page viewer talking to =
an ABC.com viewer=20
both XYZ and ABC decide to support gtalk. Their web pages will have to =
include an gtalk=20
compatible auth component to get the user's gtalk identity. It will also =
have to include a gtalk
compatible signalling layer. But we are balking at loading a suitable =
ICE implementation?

I still don't see a case where the site supplying an ICE implementation =
is a problem, given that
it already has to supply the signalling and auth code.

Tim.=

From ekr@rtfm.com  Sun Jun 12 07:44:23 2011
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On Sun, Jun 12, 2011 at 2:05 AM, Matthew Kaufman
<matthew.kaufman@skype.net> wrote:
>
> On Jun 12, 2011, at 7:55 AM, Harald Alvestrand wrote:
>
> On 06/12/2011 12:09 AM, Bernard Aboba wrote:
>
> These kind of "performance" arguments tend to fall victim to improvements in
> technology, so even if they can be demonstrated to be true at a given point
> in time, the argument may prove invalid years or even months down the
> road.
>
> In this particular case, there have been some major improvements in
> Javascript performance in recent browser releases, and it is quite possible
> that those improvements may continue.
>
> Actually my worry is more the javascript model than absolute performance. My
> tests with current browsers indicate that the timer resolutions for timed
> events are on the order of magnitude of 10-50 msec, depending on browser,
> and the single-threaded model of the Javascript event queue means that
> response to timers depends on nothing else happening in that Javascript
> context. While I'm not a Javascript expert, I haven't had any reassurance
> that we can get around this from Javascript experts I've asked.
>
> The resolution of waking up for a timer on WinNT is 30 msec and only 15 msec
> on WinXP and ICE works just fine there.
> I am not at all convinced that for our use cases we need anything better
> than ~100 msec, so your 50 msec is already twice as good.

Remember that JS is single-threaded, so other parts of the app can easily
starve you for much longer than this if they are carelessly written. That's
why I'd like to see a demonstration in a site that is actually doing substantial
other JS action.

-Ekr

From stefan.lk.hakansson@ericsson.com  Sun Jun 12 07:52:33 2011
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From: =?iso-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>
To: Eric Rescorla <ekr@rtfm.com>, Matthew Kaufman <matthew.kaufman@skype.net>
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Cc: Bernard Aboba <bernard_aboba@hotmail.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
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=20

>> Actually my worry is more the javascript model than absolute=20
>performance. My
>> tests with current browsers indicate that the timer=20
>resolutions for timed
>> events are on the order of magnitude of 10-50 msec,=20
>depending on browser,
>> and the single-threaded model of the Javascript event queue=20
>means that
>> response to timers depends on nothing else happening in that=20
>Javascript
>> context. While I'm not a Javascript expert, I haven't had=20
>any reassurance
>> that we can get around this from Javascript experts I've asked.
>>
>> The resolution of waking up for a timer on WinNT is 30 msec=20
>and only 15 msec
>> on WinXP and ICE works just fine there.
>> I am not at all convinced that for our use cases we need=20
>anything better
>> than ~100 msec, so your 50 msec is already twice as good.
>
>Remember that JS is single-threaded, so other parts of the app=20
>can easily
>starve you for much longer than this if they are carelessly=20
>written. That's
>why I'd like to see a demonstration in a site that is actually=20
>doing substantial
>other JS action.
And on a device at the lower end of the performance spectrum.
--Stefan
>
>-Ekr
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb
>=

From harald@alvestrand.no  Sun Jun 12 08:45:41 2011
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On 06/12/2011 11:36 AM, Tim Panton wrote:
> On 12 Jun 2011, at 10:05, Matthew Kaufman wrote:
>
>> On Jun 12, 2011, at 7:55 AM, Harald Alvestrand wrote:
>>
>>> On 06/12/2011 12:09 AM, Bernard Aboba wrote:
>>>>
>>>> To my mind, the real question isn't performance, it is interoperability.   The "monolithic" approach to ICE brings with it the potential for interoperability problems between different browsers operating on the same service.  Mathew's approach of Javascript implementation in ICE addresses that issue, but I'm concerned about the implications for cross-service interoperability.
>>> One of my implementor colleagues claims (in another context) that a main difference between an embedded-in-browser implementation and a Javascript one is that the browser one has less than 10 significant implementations that have to interoperate, while the Javascript one will have an essentially unbounded number of implementations ("millions of web pages"). His conclusion was that for the things we need to get right in order to have interoperability (for the case we were discussing, which was not ICE), it's better to have them embedded in the browser than to depend on Javascript.
>> This is actually the only good argument I've heard... that if the "right" implementation is baked into browsers then it will "just work". On the other hand, we have numerous examples of things that are baked into browsers that would be much more useful/flexible if they weren't... or at the very least offered both options. (And the future flexibility requirements around IPv6 are certainly a hint of why we might need this, in addition to service providers with non-standard needs.)
> Maybe I've missed something, but to my mind there is no need for the javascript ICE's to interoperate with each other. All the users of XYZ-webconference service will load _exactly_the_same_ javascript - so for them to talk peer-to-peer the XYZ-ICE-lite implementation only has to interoperate with itself.
>
> Are there important use-cases where we expect to have a visitor on XYZ.com web-page calling to ABC.com ? If we do then a baked in ICE
> may be needed.
We've spent a number of messages touting scenarios where one application 
running in one browser would use a webserver to set up a media path to a 
different application running in another browser, possibly via gateways 
in the signalling path.

Basically all but the first scenario from Jonathan Rosenberg's 
presentation in Prague fall into this category (the one I copied my 
illustrations from in the presentation from last Tuesday).

Hm... in draft-holmberg-rtcweb-ucreqs, only the last scenario (telephony 
interoperation) describes this need, and that scenario could also be 
handled with a media transcoder. We may need to add some scenarios to 
that document if we want to make sure we capture the other scenarios in 
the requirements.

                         Harald


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Magnus wrote:
 >> David Singer skrev 2011-04-18 20:10:
 >>
 >> If I am doing a point-to-point call, and I have the connectivity
 >> established (i.e. IP addresses and the like), why should I trouble at
 >> all with CNAME?  And, indeed, are there privacy concerns with it
 >> (well, there are with using <userid>@<host>, I rather suspect).  Can
 >> the CNAME problem be handled when aggregation happens?
 >
 >Because you don't know when a peer is a single user, or a conference
 >bridge. In the later case of you will need the cname. Also for point to
 >point you might need it if you have multiple sources that should or
 >should not be synchronized. The point of the new RFC is that the cname
 >can be set session specific, by drawing a random number that is being
 >used. Thus you can avoid the downsides of CNAME from privacy point of
 >view, and the generation aspects.

(Working on catching up with traffic on this list... you may see some 
more answers to old messages)

The bridge aspect is a strong example.

Another random obvious example: even a single user might be sending 
multiple A/V pairs, which need to be synced to each other, but do not 
necessarily need to be synced to each other, or multiple 
non-synchronized audio streams.

-- 
Randell Jesup
randell-ietf@jesup.org



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On 6/12/2011 10:52 AM, Stefan Håkansson LK wrote:
>
>> Remember that JS is single-threaded, so other parts of the app
>> can easily
>> starve you for much longer than this if they are carelessly
>> written. That's
>> why I'd like to see a demonstration in a site that is actually
>> doing substantial
>> other JS action.
> And on a device at the lower end of the performance spectrum.
> --Stefan

Yes!  Please remember that while desktops have unheard-of performance 
compared to 10 or even 5 years ago, an ever-increasing use-case for 
browsers and applications are smartphones and tablets that have an order 
of magnitude (or more) less performance (even with JITs) compared to 
desktops.  They will get better in terms of performance, but even if 
they get fast (dual/quad core, higher clock rates, etc), that speed 
comes at a price - for mobile devices it's more an issue of energy 
expended once a certain amount of performance has been obtained.  So 
even if the mobile device *can* run a pure-JS implementation fast 
enough, it may expend several times more energy in order to do so.

Now, in some cases that doesn't matter - it's a one-time-ish operation.  
But in others it matters a lot (media), and there's a gray area 
inbetween (moderately-frequent or rare-but-expensive operations).  It's 
something that needs to be considered before a decision is made.  It 
doesn't mean JS is the wrong choice, just that raw performance 
(especially on a desktop) isn't the (only) bar to judge against.

The other recently-mentioned point is very important - lack of threading 
in JS may cause RT or semi-RT operations needed for "call" setup or 
in-"call" things to be seriously delayed.  We should verify this and 
options for dealing with it with some JS experts.  (Brendan, are you 
here? ;-)

(Note: still catching up and haven't read the proposal itself yet; just 
the last few messages discussing it.)

-- 
Randell Jesup
randell-ietf@jesup.org


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On 06/12/2011 02:57 AM, Harald Alvestrand wrote:
> On 06/12/2011 11:05 AM, Matthew Kaufman wrote:
>>
>> On Jun 12, 2011, at 7:55 AM, Harald Alvestrand wrote:
>>
>>> On 06/12/2011 12:09 AM, Bernard Aboba wrote:
>>>> These kind of "performance" arguments tend to fall victim to 
>>>> improvements in technology, so even if they can be demonstrated to 
>>>> be true at a given point in time, the argument may prove invalid 
>>>> years or even months down the road.
>>>>
>>>> In this particular case, there have been some major improvements in 
>>>> Javascript performance in recent browser releases, and it is quite 
>>>> possible that those improvements may continue.
>>> Actually my worry is more the javascript model than absolute 
>>> performance. My tests with current browsers indicate that the timer 
>>> resolutions for timed events are on the order of magnitude of 10-50 
>>> msec, depending on browser, and the single-threaded model of the 
>>> Javascript event queue means that response to timers depends on 
>>> nothing else happening in that Javascript context. While I'm not a 
>>> Javascript expert, I haven't had any reassurance that we can get 
>>> around this from Javascript experts I've asked.
>>
>> The resolution of waking up for a timer on WinNT is 30 msec and only 
>> 15 msec on WinXP and ICE works just fine there.
>>
>> I am not at all convinced that for our use cases we need anything 
>> better than ~100 msec, so your 50 msec is already twice as good.
> The shortest timer in RFC 5245 (as far as I can see) is Ta (section 
> 16.1), which has a lower bound of 20 msec; however, this is a lower 
> bound on transmission intervals, so it's possible that an 
> implementation that fires only every 100 msec would be perfectly 
> interoperable with an application that fires every 20 msec.
>
> Testing will show....

If there is some simple way to add something like the "Expires:" header 
then this may help settle the case of what is "real" per second across 
the page (legal size or 16:9). I can think of arguments against that and 
possible expansion yet synchronization simplicity and reimplementation 
avoidance is the possibility I had in mind to avoid further cost. 
What-if there are multiple script engines (or multiple browsers) on one 
page, so speed of one single javascript instance alone is not the best 
measurement (even in smaller msec).

One oversimplified example, let's say there are multiple tapes with date 
stamp per frame, and they are all played randomly. In this example, the 
system aligns the frames (per second) despite no guarantee each source 
starts with the same datestamp. Now scale that to colosseum size with 
added info for GPS, focal length, and direction where everybody records 
or recorded some part of any sports event.

Another example is in skeletal animation of multiple dancers. There are 
options that help automate alignment of each playback of dance moves 
among couples or several groups of people across each viewer. The 
significance is is the possible synchronization of less than per "real" 
second. Complexity varies in ability to decentralize from simulcast, yet 
we can assume some kind of tether exists, physical or tangible (in 
traversal). Some may get the gist of this for other use-cases.

-- 
--- https://twitter.com/Dzonatas_Sol ---
Web Development, Software Engineering, Virtual Reality, Consultant


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Emil Ivov said:

  Besides=2C unless I've missed something=2C most of the arguments against =
an
  in-browser ICE implementation=2C like being able to use port-prediction=
=2C
  IPv6 tweaks=2C etc. are about the way an application would harvest candid=
ates.

[BA] Many of the choices arise in formulating (and responding) to an offer.
>From RFC 5245 Section 4:

   In order to send the initial offer in an offer/answer exchange=2C an
   agent must (1) gather candidates=2C (2) prioritize them=2C (3) eliminate
   redundant candidates=2C (4) choose default candidates=2C and then (5)
   formulate and send the SDP offer.

Emil Ivov also said:

    If we agree on that then wouldn't it make sense to make that part
    extensible by applications (i.e. javascript)=2C and have the rest handl=
ed
    by the browser? In other words=2C the browser could come with a set of
    default harvesters (i.e. host=2C stun=2C turn) and allow applications t=
o
    provide additional ones.

   Connectivity checks=2C which are arguably where 90% of the ICE complexit=
y
   is can be handled by the browser which would also worry timing and such.
   We've followed a similar model in ice4j.org and it has proven to be
   rather practical.

[BA] I'd agree that the time-critical portions can be handled by the
browser while still preserving much of the flexibility that Mathew's
draft argues for.   So it isn't just a choice between "all in the browser"
or "all but STUN/TURN in Javascript".

Harald said:

   One of my implementor colleagues claims (in another context) that a main
   difference between an embedded-in-browser implementation and a javascrip=
t
   one is that the browser one has less than 10 significant implementations
   that have to interoperate

[BA] Based on the recent SIPIt summaries=2C it seems that we still have a w=
ay to go before demonstrating interop between 10 ICE implementations:

https://www.sipit.net/SIPit23_Summary
https://www.sipit.net/SIPit24_Summary
https://www.sipit.net/SIPit26_Summary
https://www.sipit.net/SIPit27_Summary

To understand the challenges=2C it might be useful to go over the ICE inter=
op learnings at IETF 81.=20

Harald also said:

    Hm... in draft-holmberg-rtcweb-ucreqs=2C only the last scenario (teleph=
ony=20
interoperation)=20
    describes this need=2C and that scenario could also be=20
handled with a media transcoder.=20
    We may need to add some scenarios to=20
that document if we want to make sure we capture=20
    the other scenarios in=20
the requirements.

[BA] That sounds like a good idea.=20
 		 	   		  =

--_94323b45-75ed-4026-93f2-a8f8b73ef671_
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<body class=3D'hmmessage'>
Emil Ivov said:<br><br>&nbsp=3B Besides=2C unless I've missed something=2C =
most of the arguments against an<br>&nbsp=3B in-browser ICE implementation=
=2C like being able to use port-prediction=2C<br>&nbsp=3B IPv6 tweaks=2C et=
c. are about the way an application would harvest candidates.<br><br>[BA] M=
any of the choices arise in formulating (and responding) to an offer.<br>Fr=
om RFC 5245 Section 4:<br><br>&nbsp=3B&nbsp=3B In order to send the initial=
 offer in an offer/answer exchange=2C an<br>&nbsp=3B&nbsp=3B agent must (1)=
 gather candidates=2C (2) prioritize them=2C (3) eliminate<br>&nbsp=3B&nbsp=
=3B redundant candidates=2C (4) choose default candidates=2C and then (5)<b=
r>&nbsp=3B&nbsp=3B formulate and send the SDP offer.<br><br>Emil Ivov also =
said:<br><br>&nbsp=3B&nbsp=3B&nbsp=3B If we agree on that then wouldn't it =
make sense to make that part<br>&nbsp=3B&nbsp=3B&nbsp=3B extensible by appl=
ications (i.e. javascript)=2C and have the rest handled<br>&nbsp=3B&nbsp=3B=
&nbsp=3B by the browser? In other words=2C the browser could come with a se=
t of<br>&nbsp=3B&nbsp=3B&nbsp=3B default harvesters (i.e. host=2C stun=2C t=
urn) and allow applications to<br>&nbsp=3B&nbsp=3B&nbsp=3B provide addition=
al ones.<br><br>&nbsp=3B&nbsp=3B Connectivity checks=2C which are arguably =
where 90% of the ICE complexity<br>&nbsp=3B&nbsp=3B is can be handled by th=
e browser which would also worry timing and such.<br>&nbsp=3B&nbsp=3B We've=
 followed a similar model in ice4j.org and it has proven to be<br>&nbsp=3B&=
nbsp=3B rather practical.<br><br>[BA] I'd agree that the time-critical port=
ions can be handled by the<br>browser while still preserving much of the fl=
exibility that Mathew's<br>draft argues for.&nbsp=3B&nbsp=3B So it isn't ju=
st a choice between "all in the browser"<br>or "all but STUN/TURN in Javasc=
ript".<br><br>Harald said:<br><br>&nbsp=3B&nbsp=3B One of my implementor co=
lleagues claims (in another context) that a main<br>&nbsp=3B&nbsp=3B differ=
ence between an embedded-in-browser implementation and a javascript<br>&nbs=
p=3B&nbsp=3B one is that the browser one has less than 10 significant imple=
mentations<br>&nbsp=3B&nbsp=3B that have to interoperate<br><br>[BA] Based =
on the recent SIPIt summaries=2C it seems that we still have a way to go be=
fore demonstrating interop between 10 ICE implementations:<br><br>https://w=
ww.sipit.net/SIPit23_Summary<br>https://www.sipit.net/SIPit24_Summary<br>ht=
tps://www.sipit.net/SIPit26_Summary<br>https://www.sipit.net/SIPit27_Summar=
y<br><br>To understand the challenges=2C it might be useful to go over the =
ICE interop learnings at IETF 81. <br><br>Harald also said:<br><br><tt>&nbs=
p=3B&nbsp=3B&nbsp=3B Hm... in draft-holmberg-rtcweb-ucreqs=2C only the last=
 scenario (telephony=20
</tt><tt>interoperation) <br>&nbsp=3B&nbsp=3B&nbsp=3B describes this need=
=2C and that scenario could also be=20
</tt><tt>handled with a media transcoder. <br>&nbsp=3B&nbsp=3B&nbsp=3B We m=
ay need to add some scenarios to=20
</tt><tt>that document if we want to make sure we capture <br>&nbsp=3B&nbsp=
=3B&nbsp=3B the other scenarios in=20
</tt><tt>the requirements.
</tt><pre style=3D"margin: 0em=3B"><br></pre>[BA] That sounds like a good i=
dea. <br> 		 	   		  </body>
</html>=

--_94323b45-75ed-4026-93f2-a8f8b73ef671_--

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Emil said:=20

   Connectivity checks=2C which are arguably where 90% of the ICE complexit=
y
   is can be handled by the browser which would also worry timing and such.
   We've followed a similar model in ice4j.org and it has proven to be
   rather practical.

[BA] The issue here is not just performance/flexibility=2C there are archit=
ectural implications as well.  If we assume that it necessary to allow vari=
ation in signaling=2C then the signaling and connectivity checking function=
s of ICE cannot be combined into a single "monolithic" implementation in th=
e browser.   For example=2C  the actual SDP negotiation may be handled with=
in a Javascript library supporting a given signaling approach (e.g. Strophe=
 for XMPP).   However=2C to enable the building of an offer (and response)=
=2C the ICE candidates need to be gathered=2C prioritized=2C etc.  This imp=
lies the creation of APIs to support this.  After the (implementation depen=
dent) offer/answer is completed=2C then the connectivity checking portion c=
an be handled natively in the browser=2C but you have to have some API supp=
ort to enable getting to that point.
 		 	   		  =

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<body class=3D'hmmessage'>
Emil said: <br><br>&nbsp=3B&nbsp=3B Connectivity checks=2C which are arguab=
ly where 90% of the ICE complexity<br>&nbsp=3B&nbsp=3B is can be handled by=
 the browser which would also worry timing and such.<br>&nbsp=3B&nbsp=3B We=
've followed a similar model in ice4j.org and it has proven to be<br>&nbsp=
=3B&nbsp=3B rather practical.<br><br>[BA] The issue here is not just perfor=
mance/flexibility=2C there are architectural implications as well.&nbsp=3B =
If we assume that it necessary to allow variation in signaling=2C then the =
signaling and connectivity checking functions of ICE cannot be combined int=
o a single "monolithic" implementation in the browser.&nbsp=3B&nbsp=3B For =
example=2C&nbsp=3B the actual SDP negotiation may be handled within a Javas=
cript library supporting a given signaling approach (e.g. Strophe for XMPP)=
.&nbsp=3B&nbsp=3B However=2C to enable the building of an offer (and respon=
se)=2C the ICE candidates need to be gathered=2C prioritized=2C etc.&nbsp=
=3B This implies the creation of APIs to support this.&nbsp=3B After the (i=
mplementation dependent) offer/answer is completed=2C then the connectivity=
 checking portion can be handled natively in the browser=2C but you have to=
 have some API support to enable getting to that point.<br> 		 	   		  </bo=
dy>
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From emil@sip-communicator.org  Sun Jun 12 15:16:19 2011
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Subject: Re: [rtcweb] Review of draft-kaufman-rtcweb-traversal
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=D0=9D=D0=B0 12.06.11 20:55, Bernard Aboba =D0=BD=D0=B0=D0=BF=D0=B8=D1=81=
=D0=B0:
> Emil said:
>=20
>    Connectivity checks, which are arguably where 90% of the ICE complex=
ity
>    is can be handled by the browser which would also worry timing and s=
uch.
>    We've followed a similar model in ice4j.org and it has proven to be
>    rather practical.
>=20
> [BA] The issue here is not just performance/flexibility, there are
> architectural implications as well.  If we assume that it necessary to
> allow variation in signaling, then the signaling and connectivity
> checking functions of ICE cannot be combined into a single "monolithic"=

> implementation in the browser.=20

When you say signalling, do you mean the exchange of candidates between
peers? If so, and provided we are not to pick a signalling protocol
here, then of course, this should most definitely be outside the ICE
impl and left to the javascript, just as it would be the case with a
regular application and an ICE stack.

>  For example,  the actual SDP
> negotiation may be handled within a Javascript library supporting a
> given signaling approach (e.g. Strophe for XMPP).

Right.

>  However, to enable
> the building of an offer (and response), the ICE candidates need to be
> gathered, prioritized, etc.  This implies the creation of APIs to
> support this.=20

Right.

> After the (implementation dependent) offer/answer is
> completed, then the connectivity checking portion can be handled
> natively in the browser, but you have to have some API support to enabl=
e
> getting to that point.

Yes. Basically the browser would act as the ICE stack and the javascript
as the application.


Emil


From harald@alvestrand.no  Mon Jun 13 01:00:05 2011
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On 06/12/11 20:55, Bernard Aboba wrote:
> Emil said:
>
>    Connectivity checks, which are arguably where 90% of the ICE complexity
>    is can be handled by the browser which would also worry timing and 
> such.
>    We've followed a similar model in ice4j.org and it has proven to be
>    rather practical.
>
> [BA] The issue here is not just performance/flexibility, there are 
> architectural implications as well.  If we assume that it necessary to 
> allow variation in signaling, then the signaling and connectivity 
> checking functions of ICE cannot be combined into a single 
> "monolithic" implementation in the browser.   For example,  the actual 
> SDP negotiation may be handled within a Javascript library supporting 
> a given signaling approach (e.g. Strophe for XMPP).
SDP or mechanisms of similar expressive power, yes.
>   However, to enable the building of an offer (and response), the ICE 
> candidates need to be gathered, prioritized, etc.  This implies the 
> creation of APIs to support this.  After the (implementation 
> dependent) offer/answer is completed, then the connectivity checking 
> portion can be handled natively in the browser, but you have to have 
> some API support to enable getting to that point.
I think we agree on that point (that you have to have some API support 
to enable getting to that point). In the WHATWG proposal, that interface 
consists of SDP blobs thrown across the JS/browser interface; in the 
Google RTCWeb implementation, that interface consists of JSON blobs 
thrown in the same fashion. It's possible to write a JS app that just 
passes these blobs to a backend - or one can have a JS app that parses 
the blobs and does more with them before sending them off to wherever 
they need to go - but it's kind of obvious that these blobs need to 
contain the ICE candidates that the other end should learn about.



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    On 06/12/11 20:55, Bernard Aboba wrote:
    <blockquote cite="mid:BLU152-w23BCE3555BBE0867EDEC2F93660@phx.gbl"
      type="cite">
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      Emil said: <br>
      <br>
      &nbsp;&nbsp; Connectivity checks, which are arguably where 90% of the ICE
      complexity<br>
      &nbsp;&nbsp; is can be handled by the browser which would also worry timing
      and such.<br>
      &nbsp;&nbsp; We've followed a similar model in ice4j.org and it has proven
      to be<br>
      &nbsp;&nbsp; rather practical.<br>
      <br>
      [BA] The issue here is not just performance/flexibility, there are
      architectural implications as well.&nbsp; If we assume that it
      necessary to allow variation in signaling, then the signaling and
      connectivity checking functions of ICE cannot be combined into a
      single "monolithic" implementation in the browser.&nbsp;&nbsp; For example,&nbsp;
      the actual SDP negotiation may be handled within a Javascript
      library supporting a given signaling approach (e.g. Strophe for
      XMPP). <br>
    </blockquote>
    SDP or mechanisms of similar expressive power, yes.<br>
    <blockquote cite="mid:BLU152-w23BCE3555BBE0867EDEC2F93660@phx.gbl"
      type="cite">&nbsp; However, to enable the building of an offer (and
      response), the ICE candidates need to be gathered, prioritized,
      etc.&nbsp; This implies the creation of APIs to support this.&nbsp; After
      the (implementation dependent) offer/answer is completed, then the
      connectivity checking portion can be handled natively in the
      browser, but you have to have some API support to enable getting
      to that point.<br>
    </blockquote>
    I think we agree on that point (that you have to have some API
    support to enable getting to that point). In the WHATWG proposal,
    that interface consists of SDP blobs thrown across the JS/browser
    interface; in the Google RTCWeb implementation, that interface
    consists of JSON blobs thrown in the same fashion. It's possible to
    write a JS app that just passes these blobs to a backend - or one
    can have a JS app that parses the blobs and does more with them
    before sending them off to wherever they need to go - but it's kind
    of obvious that these blobs need to contain the ICE candidates that
    the other end should learn about.<br>
    <br>
    <br>
  </body>
</html>

--------------050101070301070909050305--

From harald@alvestrand.no  Mon Jun 13 05:14:18 2011
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I thought I'd try to capture my understanding of the state of play wrt 
RTP multiplexing as of the end of Wednesday's meeting.

As I understand it, we have two possibilities for guidelines, and the 
difference between them can be decided on whether it's technically 
feasible or not to put audio and video into the same RTP session.

(for the sake of clarity of argument, I'm claiming that only two types 
of media exist: audio and video, and I'm ignoring the question of 
additional sessions for FEC. It's simpler to see what we come out with 
that way.)

-------------------------------------------------

1) draft-perkins-rtcweb-rtp-usage, as clarified in discussion:

Each RTCWEB user has two RTP sessions active. All outgoing and incoming 
audio channels go on one RTP session; all outgoing and incoming RTP 
sessions go on the other RTP session.

Audio and video from a single user is correlated based on the CNAME; the 
SSRC for streams with the same CNAME can be different in the two RTP 
sessions.

Each RTP session, together with its RTCP data, is carried over a single 
pair of ports (5-tuple, for those who prefer that terminology) between 
each pair of participants. The RTP sessions are bidirectional.

(Note: in draft-holmberg's scenario 4.3, multiparty without central 
server, there are multiple participants; I don't know if this would have 
only one pair of RTP sessions or one per pair.)

2) A not-yet-written-up alternative:

Each RTCWEB user has one RTP session active. Audio and video both use 
the same RTP session. Streams from the same user all have the same 
CNAME; the audio and video stream [must have different SSRCs|can have 
the same SSRC|must have the same SSRC]. <not sure what to write here - 
last alternative doesn't make sense if a single user sends multiple 
streams of the same PT>

All data between two participants is carried over a single 5-tuple.

-------------------------------------------
I recognize that this is a gross simplification that glosses over 
significant detail, but ... can we agree that this is a roughly correct 
description of the alternatives we were talking about?

                 Harald



From csp@csperkins.org  Mon Jun 13 05:29:31 2011
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On 13 Jun 2011, at 13:14, Harald Alvestrand wrote:
> I thought I'd try to capture my understanding of the state of play wrt =
RTP multiplexing as of the end of Wednesday's meeting.
>=20
> As I understand it, we have two possibilities for guidelines, and the =
difference between them can be decided on whether it's technically =
feasible or not to put audio and video into the same RTP session.
>=20
> (for the sake of clarity of argument, I'm claiming that only two types =
of media exist: audio and video, and I'm ignoring the question of =
additional sessions for FEC. It's simpler to see what we come out with =
that way.)
>=20
> -------------------------------------------------
>=20
> 1) draft-perkins-rtcweb-rtp-usage, as clarified in discussion:
>=20
> Each RTCWEB user has two RTP sessions active. All outgoing and =
incoming audio channels go on one RTP session; all outgoing and incoming =
RTP sessions go on the other RTP session.
>=20
> Audio and video from a single user is correlated based on the CNAME; =
the SSRC for streams with the same CNAME can be different in the two RTP =
sessions.
>=20
> Each RTP session, together with its RTCP data, is carried over a =
single pair of ports (5-tuple, for those who prefer that terminology) =
between each pair of participants. The RTP sessions are bidirectional.
>=20
> (Note: in draft-holmberg's scenario 4.3, multiparty without central =
server, there are multiple participants; I don't know if this would have =
only one pair of RTP sessions or one per pair.)
>=20
> 2) A not-yet-written-up alternative:
>=20
> Each RTCWEB user has one RTP session active. Audio and video both use =
the same RTP session. Streams from the same user all have the same =
CNAME; the audio and video stream [must have different SSRCs|can have =
the same SSRC|must have the same SSRC]. <not sure what to write here - =
last alternative doesn't make sense if a single user sends multiple =
streams of the same PT>
>=20
> All data between two participants is carried over a single 5-tuple.
>=20
> -------------------------------------------
> I recognize that this is a gross simplification that glosses over =
significant detail, but ... can we agree that this is a roughly correct =
description of the alternatives we were talking about?


RTP requires some lower-layer demultiplexing point to identify sessions. =
A third alternative is to introduce some shim-layer between RTP and UDP =
that can provide this demultiplexing point. This has the advantage that =
it also only uses a single 5-tuple, but unlike option 2, doesn't break =
the RTP protocol.

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




From harald@alvestrand.no  Mon Jun 13 05:34:26 2011
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On 06/13/11 14:27, Colin Perkins wrote:
> On 13 Jun 2011, at 13:14, Harald Alvestrand wrote:
>> I thought I'd try to capture my understanding of the state of play wrt RTP multiplexing as of the end of Wednesday's meeting.
>>
>> As I understand it, we have two possibilities for guidelines, and the difference between them can be decided on whether it's technically feasible or not to put audio and video into the same RTP session.
>>
>> (for the sake of clarity of argument, I'm claiming that only two types of media exist: audio and video, and I'm ignoring the question of additional sessions for FEC. It's simpler to see what we come out with that way.)
>>
>> -------------------------------------------------
>>
>> 1) draft-perkins-rtcweb-rtp-usage, as clarified in discussion:
>>
>> Each RTCWEB user has two RTP sessions active. All outgoing and incoming audio channels go on one RTP session; all outgoing and incoming RTP sessions go on the other RTP session.
>>
>> Audio and video from a single user is correlated based on the CNAME; the SSRC for streams with the same CNAME can be different in the two RTP sessions.
>>
>> Each RTP session, together with its RTCP data, is carried over a single pair of ports (5-tuple, for those who prefer that terminology) between each pair of participants. The RTP sessions are bidirectional.
>>
>> (Note: in draft-holmberg's scenario 4.3, multiparty without central server, there are multiple participants; I don't know if this would have only one pair of RTP sessions or one per pair.)
>>
>> 2) A not-yet-written-up alternative:
>>
>> Each RTCWEB user has one RTP session active. Audio and video both use the same RTP session. Streams from the same user all have the same CNAME; the audio and video stream [must have different SSRCs|can have the same SSRC|must have the same SSRC].<not sure what to write here - last alternative doesn't make sense if a single user sends multiple streams of the same PT>
>>
>> All data between two participants is carried over a single 5-tuple.
>>
>> -------------------------------------------
>> I recognize that this is a gross simplification that glosses over significant detail, but ... can we agree that this is a roughly correct description of the alternatives we were talking about?
>
> RTP requires some lower-layer demultiplexing point to identify sessions. A third alternative is to introduce some shim-layer between RTP and UDP that can provide this demultiplexing point. This has the advantage that it also only uses a single 5-tuple, but unlike option 2, doesn't break the RTP protocol.
>
Yes - the disadvantage is of course that it kills all hope of 
interoperability with any other RTP implementation; people may or may 
not think of that as "breaking the RTP protocol".

An interesting property of RTP is that if a third party monitors two 
packet streams on different ports, I don't think he can tell whether 
they belong to the same RTP session or not - the tying of transport 
5-tuples to RTP sessions is n-to-n, and must be done using singalling.

              Harald


From csp@csperkins.org  Mon Jun 13 05:37:43 2011
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On 13 Jun 2011, at 13:34, Harald Alvestrand wrote:
> On 06/13/11 14:27, Colin Perkins wrote:
>> On 13 Jun 2011, at 13:14, Harald Alvestrand wrote:
>>> I thought I'd try to capture my understanding of the state of play =
wrt RTP multiplexing as of the end of Wednesday's meeting.
>>>=20
>>> As I understand it, we have two possibilities for guidelines, and =
the difference between them can be decided on whether it's technically =
feasible or not to put audio and video into the same RTP session.
>>>=20
>>> (for the sake of clarity of argument, I'm claiming that only two =
types of media exist: audio and video, and I'm ignoring the question of =
additional sessions for FEC. It's simpler to see what we come out with =
that way.)
>>>=20
>>> -------------------------------------------------
>>>=20
>>> 1) draft-perkins-rtcweb-rtp-usage, as clarified in discussion:
>>>=20
>>> Each RTCWEB user has two RTP sessions active. All outgoing and =
incoming audio channels go on one RTP session; all outgoing and incoming =
RTP sessions go on the other RTP session.
>>>=20
>>> Audio and video from a single user is correlated based on the CNAME; =
the SSRC for streams with the same CNAME can be different in the two RTP =
sessions.
>>>=20
>>> Each RTP session, together with its RTCP data, is carried over a =
single pair of ports (5-tuple, for those who prefer that terminology) =
between each pair of participants. The RTP sessions are bidirectional.
>>>=20
>>> (Note: in draft-holmberg's scenario 4.3, multiparty without central =
server, there are multiple participants; I don't know if this would have =
only one pair of RTP sessions or one per pair.)
>>>=20
>>> 2) A not-yet-written-up alternative:
>>>=20
>>> Each RTCWEB user has one RTP session active. Audio and video both =
use the same RTP session. Streams from the same user all have the same =
CNAME; the audio and video stream [must have different SSRCs|can have =
the same SSRC|must have the same SSRC].<not sure what to write here - =
last alternative doesn't make sense if a single user sends multiple =
streams of the same PT>
>>>=20
>>> All data between two participants is carried over a single 5-tuple.
>>>=20
>>> -------------------------------------------
>>> I recognize that this is a gross simplification that glosses over =
significant detail, but ... can we agree that this is a roughly correct =
description of the alternatives we were talking about?
>>=20
>> RTP requires some lower-layer demultiplexing point to identify =
sessions. A third alternative is to introduce some shim-layer between =
RTP and UDP that can provide this demultiplexing point. This has the =
advantage that it also only uses a single 5-tuple, but unlike option 2, =
doesn't break the RTP protocol.
>>=20
> Yes - the disadvantage is of course that it kills all hope of =
interoperability with any other RTP implementation; people may or may =
not think of that as "breaking the RTP protocol".

Option 2 cannot directly interoperate with standard RTP either. They =
both need some form of gateway to separate the sessions.=20

Colin


> An interesting property of RTP is that if a third party monitors two =
packet streams on different ports, I don't think he can tell whether =
they belong to the same RTP session or not - the tying of transport =
5-tuples to RTP sessions is n-to-n, and must be done using singalling.
>=20
>             Harald
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb



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




From salvatore.loreto@ericsson.com  Mon Jun 13 07:45:01 2011
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From: Salvatore Loreto <salvatore.loreto@ericsson.com>
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Subject: Re: [rtcweb] How do we implement SIP event subscription in RTCweb?
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On 6/11/11 1:29 AM, Justin Uberti wrote:
>
>
> On Fri, Jun 10, 2011 at 3:01 PM, Igor Faynberg 
> <igor.faynberg@alcatel-lucent.com 
> <mailto:igor.faynberg@alcatel-lucent.com>> wrote:
>
>     A question.
>
>     When SIP is implemented in an application within the present
>     RTCweb architecture, how exactly could we use WebSockets for
>     SUBSCRIBE/NOTIFY?
>
>     It appears to me that the only way to achieve that is to open a
>     persistent connection.  Am I right?
>
>
> Yes, that is correct. This is how asynchronous web applications work 
> today.
>
>     If so, is it not a problem, given the potential duration of such
>     connections and potential tie-up of network resources?
>
>
> An unused TCP/WebSockets connection consumes very few resources.

more there is also the intention in the HyBi wg to work on a MUX 
extension to the WebSocket protocol



-- 

Salvatore Loreto
www.sloreto.com


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    On 6/11/11 1:29 AM, Justin Uberti wrote:
    <blockquote
cite="mid:BANLkTimeBoAbC1GndiSFeuM4Yabpf7iBXen1OwGQAr05er1f+Q@mail.gmail.com"
      type="cite"><br>
      <br>
      <div class="gmail_quote">On Fri, Jun 10, 2011 at 3:01 PM, Igor
        Faynberg <span dir="ltr">&lt;<a moz-do-not-send="true"
            href="mailto:igor.faynberg@alcatel-lucent.com">igor.faynberg@alcatel-lucent.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;">
          A question.<br>
          <br>
          When SIP is implemented in an application within the present
          RTCweb architecture, how exactly could we use WebSockets for
          SUBSCRIBE/NOTIFY?<br>
          <br>
          It appears to me that the only way to achieve that is to open
          a persistent connection. Â Am I right? <br>
        </blockquote>
        <div><br>
        </div>
        <div>Yes, that is correct. This is how asynchronous web
          applications work today.</div>
        <div>
          Â </div>
        <blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt
          0.8ex; border-left: 1px solid rgb(204, 204, 204);
          padding-left: 1ex;">
          If so, is it not a problem, given the potential duration of
          such connections and potential tie-up of network resources?<br>
        </blockquote>
        <div><br>
        </div>
        <div>An unused TCP/WebSockets connection consumes very few
          resources.</div>
      </div>
    </blockquote>
    <br>
    more there is also the intention in the HyBi wg to work on a MUX
    extension to the WebSocket protocol<br>
    <br>
    <br>
    <br>
    -- <br>
    <pre class="moz-signature" cols="72">Salvatore Loreto
<a class="moz-txt-link-abbreviated" href="http://www.sloreto.com">www.sloreto.com</a></pre>
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Subject: Re: [rtcweb] RTP multiplexing - my understanding of status
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> On 13 Jun 2011, at 13:34, Harald Alvestrand wrote:
>> On 06/13/11 14:27, Colin Perkins wrote:
>>> On 13 Jun 2011, at 13:14, Harald Alvestrand wrote:
>>>> Each RTCWEB user has two RTP sessions active. All outgoing and incoming
>>>> audio channels go on one RTP session; all outgoing and incoming RTP
>>>> sessions go on the other RTP session.

s/sessions go/video goes/, I assume?

>>>> 2) A not-yet-written-up alternative:
>>>>
>>>> Each RTCWEB user has one RTP session active. Audio and video both use the

Allow me to make a suggestion for how this might work below...

>>> RTP requires some lower-layer demultiplexing point to identify sessions. A
>>> third alternative is to introduce some shim-layer between RTP and UDP that
>>> can provide this demultiplexing point. This has the advantage that it also
>>> only uses a single 5-tuple, but unlike option 2, doesn't break the RTP
>>> protocol.

So, I looked at some of RFCs mentioned in the rtp-usage-01 draft.

RFC 2733 (FEC) says, "For unicast, different ports or different SSRC may 
be used. Each of these approaches has drawbacks and benefits which 
depend on the application." This flexibility was apparently removed 8 
years later in RFC 5109, though I suspect there are a number of people 
in this WG who still believe the second sentence from RFC 2733 is true.

RFC 4588 (retransmission) also says, "These two streams may be 
multiplexed either by sending them in two different sessions, i.e., 
session-multiplexing, or in the same session using different SSRC 
values, i.e., SSRC-multiplexing."

>> Yes - the disadvantage is of course that it kills all hope of interoperability
>> with any other RTP implementation; people may or may not think of that as
>> "breaking the RTP protocol".

The problem, as I see it, is that ports are rare and expensive, and SSRC 
values are plentiful and cheap. With NATs, many endpoints (and 
potentially pairs of hosts) must share the same port namespace, and 
firewalls/routers/etc. consume resources for each one. In contrast, 
there are 32-bits of available SSRC space per CNAME, but the number of 
actual streams needed is usually quite small and self-limiting (since 
there are normally significant processing requirements at the endpoints 
associated with each one).

I propose taking the most significant, say, 8 bits of the SSRC value and 
using them as a "pseudo-session" identifier. Then the association of 
mutliple streams for purposes of scalability, retransmission, and FEC 
can be done simply by clearing the top 8 bits, with no signaling required.

This can be thought of as a variation of the "shim-layer" approach (what 
has been meant by that has never been clear to me, but my assumption has 
always been that it would encapsulate the RTP packets with yet another 
header), but it has some advantages:

1) No additional bytes must be transmitted. IP+UDP+RTP is already 40 
bytes, or 16 kbps when sending 50 packets per second (typical for audio, 
where this is a significant fraction of the total bandwidth).

2) Interoperability with existing RTP implementations works for the 
cases where only a single session is needed (e.g., simple audio, or even 
audio with some amount of FEC, as the packet sizes are typically much 
smaller than an MTU).

3) As the change and semantics are relatively simple, eventually 
upgrading existing RTP deployments should not be hard.

For interoperability in more complex cases, you can add the same 
translation tool a shim-layer would normally need, and it can be just as 
simple and generic.

So, what does this break?

RFC 3550 Section 5.2 lists 5 points. The first 3 are eliminated by using 
different SSRCs. Point 4 is, "An RTP mixer would not be able to combine 
interleaved streams of incompatible media into one stream," which is 
still true, but it would be able to combine them into a small number of 
streams, the same as if they had been transmitted in different sessions, 
based on media type (which it would have to know to be able recombine 
them at all), so I'm not sure what the actual drawback is here. Point 5 
says SSRC-multiplexing precludes 3 things:

A) "...the use of different network paths or network resource 
allocations if appropriate..." Resource allocation already requires deep 
packet inspection for many parts of the internet, and the inspection 
required here is not particularly onerous (masking off a fixed-size 
field at a fixed offset that is always transmitted in the clear, even 
for SRTP).

B) "...reception of a subset of the media, if desired, for example just 
audio if video would exceed the available bandwidth..." This is true, 
but I expect rtcweb to be mostly point-to-point, and that these 
decisions should be made during call setup, so this doesn't really apply 
(I assume it was meant for multicast environments where one might wish 
to subscribe to a subset of the available streams).

C) "...receiver implementations that use separate processes for the 
different media..." It would not be hard to write packet-filter rules to 
distribute the separate session data to multiple processes if this was 
really desired, but at least at the endpoints, is there really much 
expectation that audio and video data which must be synchronized to a 
single, common clock can be processed in separate processes?

What else am I missing?

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On 06/13/11 17:28, Timothy B. Terriberry wrote:
>> On 13 Jun 2011, at 13:34, Harald Alvestrand wrote:
>>> On 06/13/11 14:27, Colin Perkins wrote:
>>>> On 13 Jun 2011, at 13:14, Harald Alvestrand wrote:
>>>>> Each RTCWEB user has two RTP sessions active. All outgoing and 
>>>>> incoming
>>>>> audio channels go on one RTP session; all outgoing and incoming RTP
>>>>> sessions go on the other RTP session.
>
> s/sessions go/video goes/, I assume?
Yes, sorry about that. Silly of me, since it's the whole point.

I'll hold off on commenting on the rest of the message.


From randell1@jesup.org  Mon Jun 13 08:51:55 2011
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On 6/13/2011 8:37 AM, Colin Perkins wrote:
> On 13 Jun 2011, at 13:34, Harald Alvestrand wrote:
>> On 06/13/11 14:27, Colin Perkins wrote:
>>> On 13 Jun 2011, at 13:14, Harald Alvestrand wrote:
>>>> I thought I'd try to capture my understanding of the state of play wrt RTP multiplexing as of the end of Wednesday's meeting.
>>>>
>>>> As I understand it, we have two possibilities for guidelines, and the difference between them can be decided on whether it's technically feasible or not to put audio and video into the same RTP session.
>>>>
>>>> (for the sake of clarity of argument, I'm claiming that only two types of media exist: audio and video, and I'm ignoring the question of additional sessions for FEC. It's simpler to see what we come out with that way.)
>>>>
>>>> -------------------------------------------------
>>>>
>>>> 1) draft-perkins-rtcweb-rtp-usage, as clarified in discussion:
>>>>
>>>> Each RTCWEB user has two RTP sessions active. All outgoing and incoming audio channels go on one RTP session; all outgoing and incoming RTP sessions go on the other RTP session.
>>>>
>>>> Audio and video from a single user is correlated based on the CNAME; the SSRC for streams with the same CNAME can be different in the two RTP sessions.
>>>>
>>>> Each RTP session, together with its RTCP data, is carried over a single pair of ports (5-tuple, for those who prefer that terminology) between each pair of participants. The RTP sessions are bidirectional.
>>>>
>>>> (Note: in draft-holmberg's scenario 4.3, multiparty without central server, there are multiple participants; I don't know if this would have only one pair of RTP sessions or one per pair.)
>>>>
>>>> 2) A not-yet-written-up alternative:
>>>>
>>>> Each RTCWEB user has one RTP session active. Audio and video both use the same RTP session. Streams from the same user all have the same CNAME; the audio and video stream [must have different SSRCs|can have the same SSRC|must have the same SSRC].<not sure what to write here - last alternative doesn't make sense if a single user sends multiple streams of the same PT>
>>>>
>>>> All data between two participants is carried over a single 5-tuple.
>>>>
>>>> -------------------------------------------
>>>> I recognize that this is a gross simplification that glosses over significant detail, but ... can we agree that this is a roughly correct description of the alternatives we were talking about?
>>> RTP requires some lower-layer demultiplexing point to identify sessions. A third alternative is to introduce some shim-layer between RTP and UDP that can provide this demultiplexing point. This has the advantage that it also only uses a single 5-tuple, but unlike option 2, doesn't break the RTP protocol.

I assume the Matthew Kaufman here is the same one who presented the 
paper on RTMFP for Adobe at IETF 77.  :-)  From the email address, I 
take it you work for Skype now.

While that protocol per se may be out-of-bounds (proprietary, patents), 
some of the problems they were trying to solve aren't entirely different 
than what we're looking at here.  In particular, there may be advantages 
in setup time and complexity to reducing the number of "5-tuples" 
required and multiplexing data.  For example, if you need to poke holes 
in a firewall using UPnP it can be painful and slow (to the point where 
we at WorldGate pre-poked holes to use if a call came in, and we had to 
keep 2 or 3 sets of holes open to handle call-waiting cases, etc).  Even 
ignoring that and just considering ICE, the number of messages to open N 
ports is a lot higher than 1 or a pair of ports, and it may also 
increase the chance of a single-packet loss interfering and slowing down 
convergence (or pushing ICE to select a sub-optimal connection method).  
Also, what are the odds of ICE selecting different methods for the 
different streams?

As mentioned, if this is done as a shim/multiplex layer it simplifies 
interop with existing voice gateways.  Side note: there is a draft for 
(just) multiplexing RTP and RTCP which I had weighed in on and 
supported; I forget where it stands currently.

Another goal from RTMFP that is worth considering is consolidated 
congestion control - doing congestion control and bandwidth 
measurement/adaptation across all the streams exchanged with another 
endpoint together.  This is simplified by using a single or smaller 
number of "5-tuples", though it isn't gated on that.

Thought should be given to some of the other goals of RTMFP and if 
they're in-scope or not.  (IP addr changes, etc).

>> Yes - the disadvantage is of course that it kills all hope of interoperability with any other RTP implementation; people may or may not think of that as "breaking the RTP protocol".
> Option 2 cannot directly interoperate with standard RTP either. They both need some form of gateway to separate the sessions.
>
> Colin


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On 6/13/2011 11:28 AM, Timothy B. Terriberry wrote:
>> On 13 Jun 2011, at 13:34, Harald Alvestrand wrote:
>>> On 06/13/11 14:27, Colin Perkins wrote:
>>>> On 13 Jun 2011, at 13:14, Harald Alvestrand wrote:
>>>>> Each RTCWEB user has two RTP sessions active. All outgoing and 
>>>>> incoming
>>>>> audio channels go on one RTP session; all outgoing and incoming RTP
>>>>> sessions go on the other RTP session.
>
> s/sessions go/video goes/, I assume?
>
>>>>> 2) A not-yet-written-up alternative:
>>>>>
>>>>> Each RTCWEB user has one RTP session active. Audio and video both 
>>>>> use the
>
> Allow me to make a suggestion for how this might work below...
>
>>>> RTP requires some lower-layer demultiplexing point to identify 
>>>> sessions. A
>>>> third alternative is to introduce some shim-layer between RTP and 
>>>> UDP that
>>>> can provide this demultiplexing point. This has the advantage that 
>>>> it also
>>>> only uses a single 5-tuple, but unlike option 2, doesn't break the RTP
>>>> protocol.
>
> So, I looked at some of RFCs mentioned in the rtp-usage-01 draft.
>
> RFC 2733 (FEC) says, "For unicast, different ports or different SSRC 
> may be used. Each of these approaches has drawbacks and benefits which 
> depend on the application." This flexibility was apparently removed 8 
> years later in RFC 5109, though I suspect there are a number of people 
> in this WG who still believe the second sentence from RFC 2733 is true.
>
> RFC 4588 (retransmission) also says, "These two streams may be 
> multiplexed either by sending them in two different sessions, i.e., 
> session-multiplexing, or in the same session using different SSRC 
> values, i.e., SSRC-multiplexing."
>
>>> Yes - the disadvantage is of course that it kills all hope of 
>>> interoperability
>>> with any other RTP implementation; people may or may not think of 
>>> that as
>>> "breaking the RTP protocol".
>
> The problem, as I see it, is that ports are rare and expensive, and 
> SSRC values are plentiful and cheap. With NATs, many endpoints (and 
> potentially pairs of hosts) must share the same port namespace, and 
> firewalls/routers/etc. consume resources for each one. In contrast, 
> there are 32-bits of available SSRC space per CNAME, but the number of 
> actual streams needed is usually quite small and self-limiting (since 
> there are normally significant processing requirements at the 
> endpoints associated with each one).
>
> I propose taking the most significant, say, 8 bits of the SSRC value 
> and using them as a "pseudo-session" identifier. Then the association 
> of mutliple streams for purposes of scalability, retransmission, and 
> FEC can be done simply by clearing the top 8 bits, with no signaling 
> required.

I can understand the temptation to re-purpose the SSRC in this manner.

> This can be thought of as a variation of the "shim-layer" approach 
> (what has been meant by that has never been clear to me, but my 
> assumption has always been that it would encapsulate the RTP packets 
> with yet another header), but it has some advantages:

I would assume the "shim layer" suggested is a multiplexor byte or more 
between RTP/RTCP and UDP.

>
> 1) No additional bytes must be transmitted. IP+UDP+RTP is already 40 
> bytes, or 16 kbps when sending 50 packets per second (typical for 
> audio, where this is a significant fraction of the total bandwidth).

Agreed.

> 2) Interoperability with existing RTP implementations works for the 
> cases where only a single session is needed (e.g., simple audio, or 
> even audio with some amount of FEC, as the packet sizes are typically 
> much smaller than an MTU).

In my experience, most video implementations (which can fill packets) 
avoid using every possible byte in an ethernet MTU because this can 
cause problems with things like PPPOE/etc (which is a form of shim layer 
itself).  It does mean you don't need to strip the shim layer outgoing, 
and you don't need to fragment (in the rare case of full-MTU incoming).  
Note however this may cause issues with interop requiring SSRC 
translation, since the interop device may change SSRCs to any value at 
will, and SSRC translation gets fun when you have to deal with RTCP and 
all the variants like XR report blocks.  Perhaps I'm overplaying the 
issues here; Magnus & Colin are probably more familiar with the issues.

> 3) As the change and semantics are relatively simple, eventually 
> upgrading existing RTP deployments should not be hard.

You mean "RTP v3"?  Perhaps I mis-understand your comment.  Technically 
it wouldn't be hard, but in practice I think you'd have to wait a long 
time even if there was agreement that this was the "next step" for RTP - 
you'd need pretty much for all existing equipment to be end-of-lifed and 
retired.  There's little or no incentive for them to switch, let alone 
upgrade existing devices and servers.

> For interoperability in more complex cases, you can add the same 
> translation tool a shim-layer would normally need, and it can be just 
> as simple and generic.
>
> So, what does this break?
>
> RFC 3550 Section 5.2 lists 5 points. The first 3 are eliminated by 
> using different SSRCs. Point 4 is, "An RTP mixer would not be able to 
> combine interleaved streams of incompatible media into one stream," 
> which is still true, but it would be able to combine them into a small 
> number of streams, the same as if they had been transmitted in 
> different sessions, based on media type (which it would have to know 
> to be able recombine them at all), so I'm not sure what the actual 
> drawback is here. Point 5 says SSRC-multiplexing precludes 3 things:
>
> A) "...the use of different network paths or network resource 
> allocations if appropriate..." Resource allocation already requires 
> deep packet inspection for many parts of the internet, and the 
> inspection required here is not particularly onerous (masking off a 
> fixed-size field at a fixed offset that is always transmitted in the 
> clear, even for SRTP).

An example of this is resource-reservations (in particular upstream 
bandwidth) done on HFC cable access networks.  I know such things used 
to be done circa 4-5 years ago, or at least were being played with by 
the cable companies.  In theory different media (audio vs video) could 
be treated differently by a resource-reservation if they're on separate 
ports; I'm not sure that's ever actually done however.

> B) "...reception of a subset of the media, if desired, for example 
> just audio if video would exceed the available bandwidth..." This is 
> true, but I expect rtcweb to be mostly point-to-point, and that these 
> decisions should be made during call setup, so this doesn't really 
> apply (I assume it was meant for multicast environments where one 
> might wish to subscribe to a subset of the available streams).

This is a fairly bogus issue at least for the rtcweb use-cases I know 
of, in that you need the sender to throttle if you want to save 
available bandwidth, though in theory a midbox or conference server 
could do so for you.  These decisions may be made in-call instead of at 
setup, but they still require coordination with the sender.

> C) "...receiver implementations that use separate processes for the 
> different media..." It would not be hard to write packet-filter rules 
> to distribute the separate session data to multiple processes if this 
> was really desired, but at least at the endpoints, is there really 
> much expectation that audio and video data which must be synchronized 
> to a single, common clock can be processed in separate processes?

Well, RTP is designed to not only allow separate processes to handle 
different media types (streams) but also to allow different devices (IP 
addresses) for each stream in a session.  And yes, such devices really 
do exist.  In the context of rtcweb, however, I can't think of a 
use-case for this ability.  (Magnus, Colin?)

-- 
Randell Jesup
randell-ietf@jesup.org


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On 13 Jun 2011, at 16:28, Timothy B. Terriberry wrote:
>> On 13 Jun 2011, at 13:34, Harald Alvestrand wrote:
>>> On 06/13/11 14:27, Colin Perkins wrote:
>>>> On 13 Jun 2011, at 13:14, Harald Alvestrand wrote:
>>>>> Each RTCWEB user has two RTP sessions active. All outgoing and =
incoming audio channels go on one RTP session; all outgoing and incoming =
RTP sessions go on the other RTP session.
>=20
> s/sessions go/video goes/, I assume?
>=20
>>>>> 2) A not-yet-written-up alternative:
>>>>>=20
>>>>> Each RTCWEB user has one RTP session active. Audio and video both =
use the
>=20
> Allow me to make a suggestion for how this might work below...
>=20
>>>> RTP requires some lower-layer demultiplexing point to identify =
sessions. A third alternative is to introduce some shim-layer between =
RTP and UDP that can provide this demultiplexing point. This has the =
advantage that it also only uses a single 5-tuple, but unlike option 2, =
doesn't break the RTPprotocol.
>=20
> So, I looked at some of RFCs mentioned in the rtp-usage-01 draft.
>=20
> RFC 2733 (FEC) says, "For unicast, different ports or different SSRC =
may be used. Each of these approaches has drawbacks and benefits which =
depend on the application." This flexibility was apparently removed 8 =
years later in RFC 5109, though I suspect there are a number of people =
in this WG who still believe the second sentence from RFC 2733 is true.

I'm sure there are. Unfortunately, the SSRC multiplexing mechanism used =
in RFC 2733 breaks when a participant tries to send several streams in =
the session, which is why RFC 5109 mandates a separate RTP session for =
the FEC.=20

> RFC 4588 (retransmission) also says, "These two streams may be =
multiplexed either by sending them in two different sessions, i.e., =
session-multiplexing, or in the same session using different SSRC =
values, i.e., SSRC-multiplexing."

Again, as Section 5.3 of RFC 4588 explains, there are difficulties when =
using SSRC multiplexing when a participant tries to send several streams =
in a session.

In both cases, SSRC multiplexing appears to work, until you consider the =
more complex scenarios in which RTP is used.

>>> Yes - the disadvantage is of course that it kills all hope of =
interoperability with any other RTP implementation; people may or may =
not think of that as "breaking the RTP protocol".
>=20
> The problem, as I see it, is that ports are rare and expensive, and =
SSRC values are plentiful and cheap. With NATs, many endpoints (and =
potentially pairs of hosts) must share the same port namespace, and =
firewalls/routers/etc. consume resources for each one.

So add a shim layer to demultiplex, sitting between the UDP header and =
the RTP header. Don't break RTP.

> In contrast, there are 32-bits of available SSRC space per CNAME, but =
the number of actual streams needed is usually quite small and =
self-limiting (since there are normally significant processing =
requirements at the endpoints associated with each one).
>=20
> I propose taking the most significant, say, 8 bits of the SSRC value =
and using them as a "pseudo-session" identifier. Then the association of =
mutliple streams for purposes of scalability, retransmission, and FEC =
can be done simply by clearing the top 8 bits, with no signaling =
required.
>=20
> This can be thought of as a variation of the "shim-layer" approach =
(what has been meant by that has never been clear to me, but my =
assumption has always been that it would encapsulate the RTP packets =
with yet another header), but it has some advantages:

The shim could be a simple as a single byte to identify the RTP session =
prepended onto the packets (there might be advantages in using something =
more complex like RTP in DCCP in UDP encapsulation, but this also has =
some significant costs).

> 1) No additional bytes must be transmitted. IP+UDP+RTP is already 40 =
bytes, or 16 kbps when sending 50 packets per second (typical for audio, =
where this is a significant fraction of the total bandwidth).
>=20
> 2) Interoperability with existing RTP implementations works for the =
cases where only a single session is needed (e.g., simple audio, or even =
audio with some amount of FEC, as the packet sizes are typically much =
smaller than an MTU).

It's not clear that this interoperates, since the other endpoint might =
pick one of the reserved SSRC values. You'd need to translate the SSRC =
at the gateway, which is non-trival once you start including RTCP =
packets in the translation.

> 3) As the change and semantics are relatively simple, eventually =
upgrading existing RTP deployments should not be hard.

The semantics are complex, especially once you start to think about the =
interactions with the RTCP timing rules. It would be much simpler to add =
a demultiplexing shim.

> For interoperability in more complex cases, you can add the same =
translation tool a shim-layer would normally need, and it can be just as =
simple and generic.

It's much more complex, since you may need to rewrite SSRC values in =
RTCP and deal with the differences in RTCP timing and participant =
timeout.

> So, what does this break?
>=20
> RFC 3550 Section 5.2 lists 5 points. The first 3 are eliminated by =
using different SSRCs. Point 4 is, "An RTP mixer would not be able to =
combine interleaved streams of incompatible media into one stream," =
which is still true, but it would be able to combine them into a small =
number of streams, the same as if they had been transmitted in different =
sessions, based on media type (which it would have to know to be able =
recombine them at all), so I'm not sure what the actual drawback is =
here. Point 5 says SSRC-multiplexing precludes 3 things:
>=20
> A) "...the use of different network paths or network resource =
allocations if appropriate..." Resource allocation already requires deep =
packet inspection for many parts of the internet, and the inspection =
required here is not particularly onerous (masking off a fixed-size =
field at a fixed offset that is always transmitted in the clear, even =
for SRTP).

Demultiplexing based on a shim is also no more onerous.

> B) "...reception of a subset of the media, if desired, for example =
just audio if video would exceed the available bandwidth..." This is =
true, but I expect rtcweb to be mostly point-to-point, and that these =
decisions should be made during call setup, so this doesn't really apply =
(I assume it was meant for multicast environments where one might wish =
to subscribe to a subset of the available streams).

It's also useful in unicast environments.=20

> C) "...receiver implementations that use separate processes for the =
different media..." It would not be hard to write packet-filter rules to =
distribute the separate session data to multiple processes if this was =
really desired, but at least at the endpoints, is there really much =
expectation that audio and video data which must be synchronized to a =
single, common clock can be processed in separate processes?

There have been implementations that work that way, but a more common =
issue might be where audio and video are rendered by different devices.

> What else am I missing?

You also break the RTCP timing rules and participant timeouts, the SSRC =
allocation, and the SSRC collision detection and resolution algorithm. =
All of these make it incompatible with other RTP implementations.=20

And all to avoid a one byte shim layer...

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




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Subject: Re: [rtcweb] RTP multiplexing - my understanding of status
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On 13 Jun 2011, at 18:07, Randell Jesup wrote:
> On 6/13/2011 11:28 AM, Timothy B. Terriberry wrote:
>>> On 13 Jun 2011, at 13:34, Harald Alvestrand wrote:
>>>> On 06/13/11 14:27, Colin Perkins wrote:
>>>>> On 13 Jun 2011, at 13:14, Harald Alvestrand wrote:
>>>>>> Each RTCWEB user has two RTP sessions active. All outgoing and =
incoming audio channels go on one RTP session; all outgoing and incoming =
RTP sessions go on the other RTP session.
>>=20
>> s/sessions go/video goes/, I assume?
>>=20
>>>>>> 2) A not-yet-written-up alternative:
>>>>>>=20
>>>>>> Each RTCWEB user has one RTP session active. Audio and video both =
use the
>>=20
>> Allow me to make a suggestion for how this might work below...
>>=20
>>>>> RTP requires some lower-layer demultiplexing point to identify =
sessions. A third alternative is to introduce some shim-layer between =
RTP and UDP that can provide this demultiplexing point. This has the =
advantage that it also only uses a single 5-tuple, but unlike option 2, =
doesn't break the RTP protocol.
>>=20
>> So, I looked at some of RFCs mentioned in the rtp-usage-01 draft.
>>=20
>> RFC 2733 (FEC) says, "For unicast, different ports or different SSRC =
may be used. Each of these approaches has drawbacks and benefits which =
depend on the application." This flexibility was apparently removed 8 =
years later in RFC 5109, though I suspect there are a number of people =
in this WG who still believe the second sentence from RFC 2733 is true.
>>=20
>> RFC 4588 (retransmission) also says, "These two streams may be =
multiplexed either by sending them in two different sessions, i.e., =
session-multiplexing, or in the same session using different SSRC =
values, i.e., SSRC-multiplexing."
>>=20
>>>> Yes - the disadvantage is of course that it kills all hope of =
interoperability with any other RTP implementation; people may or may =
not think of that as "breaking the RTP protocol".
>>=20
>> The problem, as I see it, is that ports are rare and expensive, and =
SSRC values are plentiful and cheap. With NATs, many endpoints (and =
potentially pairs of hosts) must share the same port namespace, and =
firewalls/routers/etc. consume resources for each one. In contrast, =
there are 32-bits of available SSRC space per CNAME, but the number of =
actual streams needed is usually quite small and self-limiting (since =
there are normally significant processing requirements at the endpoints =
associated with each one).
>>=20
>> I propose taking the most significant, say, 8 bits of the SSRC value =
and using them as a "pseudo-session" identifier. Then the association of =
mutliple streams for purposes of scalability, retransmission, and FEC =
can be done simply by clearing the top 8 bits, with no signaling =
required.
>=20
> I can understand the temptation to re-purpose the SSRC in this manner.
>=20
>> This can be thought of as a variation of the "shim-layer" approach =
(what has been meant by that has never been clear to me, but my =
assumption has always been that it would encapsulate the RTP packets =
with yet another header), but it has some advantages:
>=20
> I would assume the "shim layer" suggested is a multiplexor byte or =
more between RTP/RTCP and UDP.
>=20
>>=20
>> 1) No additional bytes must be transmitted. IP+UDP+RTP is already 40 =
bytes, or 16 kbps when sending 50 packets per second (typical for audio, =
where this is a significant fraction of the total bandwidth).
>=20
> Agreed.
>=20
>> 2) Interoperability with existing RTP implementations works for the =
cases where only a single session is needed (e.g., simple audio, or even =
audio with some amount of FEC, as the packet sizes are typically much =
smaller than an MTU).
>=20
> In my experience, most video implementations (which can fill packets) =
avoid using every possible byte in an ethernet MTU because this can =
cause problems with things like PPPOE/etc (which is a form of shim layer =
itself).  It does mean you don't need to strip the shim layer outgoing, =
and you don't need to fragment (in the rare case of full-MTU incoming).  =
Note however this may cause issues with interop requiring SSRC =
translation, since the interop device may change SSRCs to any value at =
will, and SSRC translation gets fun when you have to deal with RTCP and =
all the variants like XR report blocks.  Perhaps I'm overplaying the =
issues here; Magnus & Colin are probably more familiar with the issues.

No, the translation will get very difficult once you consider all the =
RTCP packet types. The trivial case can be made to work, but the =
complexity rapidly spirals up as you consider all the other ways in =
which RTP is used.

>> 3) As the change and semantics are relatively simple, eventually =
upgrading existing RTP deployments should not be hard.
>=20
> You mean "RTP v3"?  Perhaps I mis-understand your comment.  =
Technically it wouldn't be hard, but in practice I think you'd have to =
wait a long time even if there was agreement that this was the "next =
step" for RTP - you'd need pretty much for all existing equipment to be =
end-of-lifed and retired.  There's little or no incentive for them to =
switch, let alone upgrade existing devices and servers.
>=20
>> For interoperability in more complex cases, you can add the same =
translation tool a shim-layer would normally need, and it can be just as =
simple and generic.
>>=20
>> So, what does this break?
>>=20
>> RFC 3550 Section 5.2 lists 5 points. The first 3 are eliminated by =
using different SSRCs. Point 4 is, "An RTP mixer would not be able to =
combine interleaved streams of incompatible media into one stream," =
which is still true, but it would be able to combine them into a small =
number of streams, the same as if they had been transmitted in different =
sessions, based on media type (which it would have to know to be able =
recombine them at all), so I'm not sure what the actual drawback is =
here. Point 5 says SSRC-multiplexing precludes 3 things:
>>=20
>> A) "...the use of different network paths or network resource =
allocations if appropriate..." Resource allocation already requires deep =
packet inspection for many parts of the internet, and the inspection =
required here is not particularly onerous (masking off a fixed-size =
field at a fixed offset that is always transmitted in the clear, even =
for SRTP).
>=20
> An example of this is resource-reservations (in particular upstream =
bandwidth) done on HFC cable access networks.  I know such things used =
to be done circa 4-5 years ago, or at least were being played with by =
the cable companies.  In theory different media (audio vs video) could =
be treated differently by a resource-reservation if they're on separate =
ports; I'm not sure that's ever actually done however.
>=20
>> B) "...reception of a subset of the media, if desired, for example =
just audio if video would exceed the available bandwidth..." This is =
true, but I expect rtcweb to be mostly point-to-point, and that these =
decisions should be made during call setup, so this doesn't really apply =
(I assume it was meant for multicast environments where one might wish =
to subscribe to a subset of the available streams).
>=20
> This is a fairly bogus issue at least for the rtcweb use-cases I know =
of, in that you need the sender to throttle if you want to save =
available bandwidth, though in theory a midbox or conference server =
could do so for you.  These decisions may be made in-call instead of at =
setup, but they still require coordination with the sender.
>=20
>> C) "...receiver implementations that use separate processes for the =
different media..." It would not be hard to write packet-filter rules to =
distribute the separate session data to multiple processes if this was =
really desired, but at least at the endpoints, is there really much =
expectation that audio and video data which must be synchronized to a =
single, common clock can be processed in separate processes?
>=20
> Well, RTP is designed to not only allow separate processes to handle =
different media types (streams) but also to allow different devices (IP =
addresses) for each stream in a session.  And yes, such devices really =
do exist.  In the context of rtcweb, however, I can't think of a =
use-case for this ability.  (Magnus, Colin?)


We don't have a use case, or we don't have a use case _yet_? While we =
may not want to support more advanced scenarios today, we don't want to =
make decisions which preclude them if we can avoid it (and, in this =
case, we can easily avoid it by either using separate UDP ports, or by =
adding a shim layer between UDP and RTP, and leaving the SSRC semantics =
alone).

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




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Cc: Bernard Aboba <bernard_aboba@hotmail.com>, rtcweb@ietf.org
Subject: Re: [rtcweb] Review of draft-kaufman-rtcweb-traversal
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On 6/11/2011 10:55 PM, Harald Alvestrand wrote:
> One of my implementor colleagues claims (in another context) that a 
> main difference between an embedded-in-browser implementation and a 
> Javascript one is that the browser one has less than 10 significant 
> implementations that have to interoperate, while the Javascript one 
> will have an essentially unbounded number of implementations 
> ("millions of web pages"). His conclusion was that for the things we 
> need to get right in order to have interoperability (for the case we 
> were discussing, which was not ICE), it's better to have them embedded 
> in the browser than to depend on Javascript.

Agreed on this point.  The level for this API is probably pretty high to 
reduce the matrix for testing an interop.

--Chris

From blizzard@mozilla.com  Mon Jun 13 10:45:05 2011
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On 6/12/2011 2:05 AM, Matthew Kaufman wrote:
> (The example that keeps coming to mind is the difference between 
> progressive playback of HTTP video (as is in Flash Player and most 
> HTML5 video tag implementation) vs. adaptive HTTP streaming (which, in 
> Flash Player was architected as low-level ActionScript primitives plus 
> reference code, and which has already done several things that weren't 
> originally imagined when the APIs were first added))

For this case you should know that we're interested in the web browser 
to do both - have a "just works" mode for people that want adaptive 
bandwidth and a low-level API for people who want to control it 
directly.  But keep in mind that the interop matrix there is small - 
we're still talking about a few products and the site has control of its 
entire stack.  Getting services talking to each other with a lower level 
API is very different from a single site.

--Chris

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On Jun 13, 2011, at 11:27 AM, Colin Perkins wrote:

>>=20
>> Well, RTP is designed to not only allow separate processes to handle =
different media types (streams) but also to allow different devices (IP =
addresses) for each stream in a session.  And yes, such devices really =
do exist.  In the context of rtcweb, however, I can't think of a =
use-case for this ability.  (Magnus, Colin?)
>=20
>=20
> We don't have a use case, or we don't have a use case _yet_? While we =
may not want to support more advanced scenarios today, we don't want to =
make decisions which preclude them if we can avoid it (and, in this =
case, we can easily avoid it by either using separate UDP ports, or by =
adding a shim layer between UDP and RTP, and leaving the SSRC semantics =
alone).

One possible use case is I want the video displayed in my browser but =
the audio going to a high quality speaker phone on my desk. People often =
do this with soft phones today so it is easy to imagine them wanting to =
do with a browser and phone. So I can impinge a use case, however... I'm =
not seeing that this use case changes much. Assuming that SSRC multiplex =
is signaled and both ends negotiate using it, I'm not understanding =
Colin's & Magnus draft well enough to see what breaks that people care =
about.=20

Cullen <in my individual contributor role>




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Subject: Re: [rtcweb] How do we implement SIP event subscription in RTCweb?
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That part I thought was easy: The application will send REGISTER.

I was worried about keeping a persistent connection, but Justin (to whom 
many thanks for answering my query right away!) assures me that this is 
not a problem, and so far no one said anything otherwise.

Igor

Avasarala, Ranjit wrote:
> Hi
>
> Any idea how the initial SIP signaling would take place between an RTC web application and SIP server?
>
> Regards
> Ranjit
>
> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of Igor Faynberg
> Sent: Saturday, June 11, 2011 3:31 AM
> To: rtcweb@ietf.org
> Subject: [rtcweb] How do we implement SIP event subscription in RTCweb?
>
> A question.
>
> When SIP is implemented in an application within the present RTCweb 
> architecture, how exactly could we use WebSockets for SUBSCRIBE/NOTIFY?
>
> It appears to me that the only way to achieve that is to open a 
> persistent connection.  Am I right? 
>
> If so, is it not a problem, given the potential duration of such 
> connections and potential tie-up of network resources?
>
> If not, how else could that be done?
>
> With many thanks,
>
> Igor
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>   

From blizzard@mozilla.com  Mon Jun 13 11:09:02 2011
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On 6/13/2011 11:05 AM, Igor Faynberg wrote:
> I was worried about keeping a persistent connection, but Justin (to 
> whom many thanks for answering my query right away!) assures me that 
> this is not a problem, and so far no one said anything otherwise. 

The larger issue is that we need a way to do connections & applications 
outside of the single page context.  How do you signal from a service 
when someone might not have a web page open?  This is a problem that's 
outside of the scope of both the IETF and W3C groups, but it's a problem 
in any case.

--Chris

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On Jun 11, 2011, at 3:40 PM, Bernard Aboba wrote:

> 4.2.2.  Masking
>=20
>    Once consent is verified, there still is some concern about
>    misinterpretation attacks as described by Huang et al.[huang-w2sp].
>    As long as communication is limited to UDP, then this risk is
>    probably limited, thus masking is not required for UDP. =20
>=20
> [BA] Are you saying that the browser should be able to send arbitrary
> UDP traffic once "consent" is granted? =20

I could be very confused about this but .... I think it is OK with UDP =
(after consent is given on the 5 tuple) or even with TLS over TCP. I'm =
not sure how long the consent should last. The only place I know of =
where this is a problem is when there are middle boxes that intercept =
traffic that is not addressed to them. I don't think this happens to the =
existing UDP based protocols. Not sure if I have this right but I hope =
some discussions ensues....
=20



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On 13 Jun 2011, at 18:53, Cullen Jennings wrote:
> On Jun 13, 2011, at 11:27 AM, Colin Perkins wrote:
>>> Well, RTP is designed to not only allow separate processes to handle =
different media types (streams) but also to allow different devices (IP =
addresses) for each stream in a session.  And yes, such devices really =
do exist.  In the context of rtcweb, however, I can't think of a =
use-case for this ability.  (Magnus, Colin?)
>>=20
>> We don't have a use case, or we don't have a use case _yet_? While we =
may not want to support more advanced scenarios today, we don't want to =
make decisions which preclude them if we can avoid it (and, in this =
case, we can easily avoid it by either using separate UDP ports, or by =
adding a shim layer between UDP and RTP, and leaving the SSRC semantics =
alone).
>=20
> One possible use case is I want the video displayed in my browser but =
the audio going to a high quality speaker phone on my desk. People often =
do this with soft phones today so it is easy to imagine them wanting to =
do with a browser and phone. So I can impinge a use case, however... I'm =
not seeing that this use case changes much. Assuming that SSRC multiplex =
is signaled and both ends negotiate using it, I'm not understanding =
Colin's & Magnus draft well enough to see what breaks that people care =
about.=20


The problem is in signalling the SSRC multiplexing, partitioning the =
SSRC space between the different RTP sessions, and updating the RTP =
end-points and middleboxes to cope with the changed SSRC semantics and =
RTCP timing rules. All of this is possible, but the result isn't =
compatible with RTP as it stands now, and is not easy to gateway into a =
standard RTP session.=20

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




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On 06/13/2011 10:07 AM, Randell Jesup wrote:
> On 6/13/2011 11:28 AM, Timothy B. Terriberry wrote:
>
>> C) "...receiver implementations that use separate processes for the 
>> different media..." It would not be hard to write packet-filter rules 
>> to distribute the separate session data to multiple processes if this 
>> was really desired, but at least at the endpoints, is there really 
>> much expectation that audio and video data which must be synchronized 
>> to a single, common clock can be processed in separate processes?
>
> Well, RTP is designed to not only allow separate processes to handle 
> different media types (streams) but also to allow different devices 
> (IP addresses) for each stream in a session.  And yes, such devices 
> really do exist.  In the context of rtcweb, however, I can't think of 
> a use-case for this ability.  (Magnus, Colin?)
>


The basic common clock is most available in kernel mode, so of course 
use-cases are harder to delineate that way unless there are ties into 
that mode (into user mode). Exploitation of that clock can lead to 
over-usage of resources, which then lead to known depletion of energy or 
cause resource imbalances.

That's why my previous message hinted about the colosseum, yet think 
where every seat has their own clock however set (i.e. for now, 
cellphone-cams or D8 media). Start with one solution as if the Olympics 
would setup some kind of heartbeat server for all media that stems 
(ports) from such an event. It's not an odd use-case unless the idea of 
software development doesn't belong in any kind of Olympic theme (which 
seems anti-Olympian and only supports privatization in such idea). I 
have distributed processes in mind that go over collections of assets or 
posted content and render the scene. The harder part is how-to layout 
the words for specific use-case that does not favor any particular 
development or group other than to say "on the Olympic scale."

The Olympics prove how resourceful people are in reality, and the rings 
represent such elementary thinking.


-- 
--- https://twitter.com/Dzonatas_Sol ---
Web Development, Software Engineering, Virtual Reality, Consultant


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On 06/13/2011 10:45 AM, Christopher Blizzard wrote:
> On 6/12/2011 2:05 AM, Matthew Kaufman wrote:
>> (The example that keeps coming to mind is the difference between 
>> progressive playback of HTTP video (as is in Flash Player and most 
>> HTML5 video tag implementation) vs. adaptive HTTP streaming (which, 
>> in Flash Player was architected as low-level ActionScript primitives 
>> plus reference code, and which has already done several things that 
>> weren't originally imagined when the APIs were first added))
>
> For this case you should know that we're interested in the web browser 
> to do both - have a "just works" mode for people that want adaptive 
> bandwidth and a low-level API for people who want to control it 
> directly.  But keep in mind that the interop matrix there is small - 
> we're still talking about a few products and the site has control of 
> its entire stack.  Getting services talking to each other with a lower 
> level API is very different from a single site.
>
>

Yes and people try to build browser in a browser in a browser etc 
turtles on turtles as they sort out the layers. With hypervisors 
available, the concept of service stacks as an entire separate system on 
the same hardware is known, and browsers can use those instead of 
endless built-in extensions (or major re-implementations). I wouldn't be 
surprised to see hypervisors that designate one guest system per port, 
per protocol (API), per URI scheme, and/or per URN.


-- 
--- https://twitter.com/Dzonatas_Sol ---
Web Development, Software Engineering, Virtual Reality, Consultant


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On 06/13/11 20:09, Christopher Blizzard wrote:
> On 6/13/2011 11:05 AM, Igor Faynberg wrote:
>> I was worried about keeping a persistent connection, but Justin (to 
>> whom many thanks for answering my query right away!) assures me that 
>> this is not a problem, and so far no one said anything otherwise. 
>
> The larger issue is that we need a way to do connections & 
> applications outside of the single page context.  How do you signal 
> from a service when someone might not have a web page open?  This is a 
> problem that's outside of the scope of both the IETF and W3C groups, 
> but it's a problem in any case.
Isn't this the "Web Application / Background Page" problem - that is, 
having stuff run in the browser when no pages are open?
I believe there are W3C groups hard at work on that; we should punt it 
to them if we can.

                Harald

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We should, and YOU are the best person to do that, since you are already 
there (and with a lot of clout).

But... what can we do in the IETF?

Igotr

Harald Alvestrand wrote:
> On 06/13/11 20:09, Christopher Blizzard wrote:
>> On 6/13/2011 11:05 AM, Igor Faynberg wrote:
>>> I was worried about keeping a persistent connection, but Justin (to 
>>> whom many thanks for answering my query right away!) assures me that 
>>> this is not a problem, and so far no one said anything otherwise. 
>>
>> The larger issue is that we need a way to do connections & 
>> applications outside of the single page context.  How do you signal 
>> from a service when someone might not have a web page open?  This is 
>> a problem that's outside of the scope of both the IETF and W3C 
>> groups, but it's a problem in any case.
> Isn't this the "Web Application / Background Page" problem - that is, 
> having stuff run in the browser when no pages are open?
> I believe there are W3C groups hard at work on that; we should punt it 
> to them if we can.
>
>                Harald

From Hannes.Tschofenig@gmx.net  Tue Jun 14 07:33:37 2011
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Hi Igor,

 I do not quite understand where you are heading with your question.

 Your question can be interpreted in two ways, namely:

 a) How do I implement SIP event subscription alike concepts in a browser?
 (without using SIP itself but with browser technologies)

 b) I want to provide interworking with an existing SIP implementation that uses the SIP event package: how do I do it?

 In any case, you need to provide more details about what you want to accomplish in order for someone to provide you a good answer.

 For example, you may want to provide some information about
 * what data would you like to subscribe to,
 * where is the data,
 * who is supposed to receive it? 

 Ciao
 Hannes

 PS: Your remark that maintaining connectivity alive for asynchronous communication is correct but what do you want to do if that's your desired functionality?
 This aspect is independent of SIP, XMPP and the Web.

----- UrsprÃ¼ngliche Nachricht -----
Von: Igor Faynberg
Gesendet: 11.06.11 01:01 Uhr
An: rtcweb@ietf.org
Betreff: [rtcweb] How do we implement SIP event subscription in RTCweb?

 A question. When SIP is implemented in an application within the present RTCweb architecture, how exactly could we use WebSockets for SUBSCRIBE/NOTIFY? It appears to me that the only way to achieve that is to open a persistent connection. Am I right? If so, is it not a problem, given the potential duration of such connections and potential tie-up of network resources? If not, how else could that be done? With many thanks, Igor _______________________________________________ rtcweb mailing list rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb

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<span style=3D'font-family:Verdana'><span style=3D'font-size:12px'>Hi Igor,=
<br />=20
<br />=20
I do not quite understand where you are heading with your question.<br />=
=20
<br />=20
Your question can be interpreted in two ways, namely:<br />=20
<br />=20
a) How do I implement SIP event subscription alike concepts in a browser?<b=
r />=20
(without using SIP itself but with browser technologies)<br />=20
<br />=20
b) I want to provide interworking with an existing SIP implementation that =
uses the SIP event package: how do I do it?<br />=20
<br />=20
In any case, you need to provide more details about what you want to accomp=
lish in order for someone to provide you a good answer.<br />=20
<br />=20
For example, you may want to provide some information about<br />=20
* what data would you like to subscribe to,<br />=20
* where is the data,<br />=20
* who is supposed to receive it? &nbsp;<br />=20
<br />=20
Ciao<br />=20
Hannes<br />=20
<br />=20
PS: Your remark that maintaining connectivity alive for asynchronous commun=
ication is correct but what do you want to do if that's your desired functi=
onality?<br />=20
This aspect is independent of SIP, XMPP and the Web.<br />=20
<br />=20
<p style=3D"margin:0px; padding:0px;" >=20
	=C2=A0</p>=20
<blockquote style=3D"border-left: 1px solid #CCC; padding-left: 5px; margin=
-left: 5px; margin-bottom: 0px; margin-top: 0px; margin-right: 0px;" type=
=3D"cite">=20
	<p style=3D"margin:0px; padding:0px;" >=20
		<span style=3D"font-family:Verdana"><span style=3D"font-size:12px">----- =
Urspr=C3=BCngliche Nachricht -----</span></span></p>=20
	<p style=3D"margin:0px; padding:0px;" >=20
		<span style=3D"font-family:Verdana"><span style=3D"font-size:12px">Von: I=
gor Faynberg</span></span></p>=20
	<p style=3D"margin:0px; padding:0px;" >=20
		<span style=3D"font-family:Verdana"><span style=3D"font-size:12px">Gesend=
et: 11.06.11 01:01 Uhr</span></span></p>=20
	<p style=3D"margin:0px; padding:0px;" >=20
		<span style=3D"font-family:Verdana"><span style=3D"font-size:12px">An: rt=
cweb@ietf.org</span></span></p>=20
	<p style=3D"margin:0px; padding:0px;" >=20
		<span style=3D"font-family:Verdana"><span style=3D"font-size:12px">Betref=
f: [rtcweb] How do we implement SIP event subscription in RTCweb?</span></s=
pan></p>=20
	<br />=20
	<div>=20
		<div>=20
			<pre style=3D"white-space: pre-wrap; word-wrap: break-word; font-size:11=
;pre">=20
A question.=20

When SIP is implemented in an application within the present RTCweb=20
architecture, how exactly could we use WebSockets for SUBSCRIBE/NOTIFY?=20

It appears to me that the only way to achieve that is to open a=20
persistent connection.  Am I right?=20

If so, is it not a problem, given the potential duration of such=20
connections and potential tie-up of network resources?=20

If not, how else could that be done?=20

With many thanks,=20

Igor=20
_______________________________________________=20
rtcweb mailing list=20
rtcweb@ietf.org=20
https://www.ietf.org/mailman/listinfo/rtcweb</pre>=20
		</div>=20
	</div>=20
</blockquote>=20
<p style=3D"margin:0px; padding:0px;" >=20
	=C2=A0</p>=20
</span></span>

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Subject: Re: [rtcweb] How do we implement SIP event subscription in RTCweb?
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--------------070102060704080703000108
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Hannes,

Thanks for following up! I actually think that my question has already 
been answered, but to reply to  your query, I think that of  the two 
options you mentioned, the closest one to my question  is

...
b) I want to provide interworking with an existing SIP implementation 
that uses the SIP event package: how do I do it?
...

More precisely, the problem statement was: I need to have SIP on my 
Internet appliance. Now that I have only the browser to rely on, how do 
I implement it?

  I assume that I have 1) JavaScript to program in and 2) the WebSockets 
interface.  All is well until asynchronous processing comes in.  To 
implement SUBSCRIBE/NOTIFY, I would need to be able to receive the 
NOTIFY messages (i.e., catch interrupts) whenever they come.   I thought 
that the only way to achieve that was to keep an open TCP connection... 
forever.  (I was under the impression that this was not considered a 
good practice--both in terms of resending "keep alive" and holding 
resources (such as NAT).

The responses that I got so far confirmed that my assumption about 
keeping a TCP connection open  was right, but that the practice of 
keeping a TCP connection open has no longer been considered wasteful and 
is, in fact, implemented.  There was also a message about the necessity 
of being tied to the page, but this appears to me as something that can 
be meaningfully worked around.

I guess more clarifications and implementation reports will come my 
way.  I do think it is critical that SIP be implementable in the 
browser-only environment.  I understand the reasons for the decision NOT 
to have SIP (or, for that matter ICE) as part of the browser, but that 
implied that SIP can be implemented in the application.

Igor

On 6/14/2011 10:33 AM, Hannes Tschofenig wrote:
> Hi Igor,
>
> I do not quite understand where you are heading with your question.
>
> Your question can be interpreted in two ways, namely:
>
> a) How do I implement SIP event subscription alike concepts in a browser?
> (without using SIP itself but with browser technologies)
>
> b) I want to provide interworking with an existing SIP implementation 
> that uses the SIP event package: how do I do it?
>
> In any case, you need to provide more details about what you want to 
> accomplish in order for someone to provide you a good answer.
>
> For example, you may want to provide some information about
> * what data would you like to subscribe to,
> * where is the data,
> * who is supposed to receive it?
>
> Ciao
> Hannes
>
> PS: Your remark that maintaining connectivity alive for asynchronous 
> communication is correct but what do you want to do if that's your 
> desired functionality?
> This aspect is independent of SIP, XMPP and the Web.
>
>> ----- UrsprÃ¼ngliche Nachricht -----
>>
>> Von: Igor Faynberg
>>
>> Gesendet: 11.06.11 01:01 Uhr
>>
>> An: rtcweb@ietf.org
>>
>> Betreff: [rtcweb] How do we implement SIP event subscription in RTCweb?
>>
>>
>>
>> A question.
>>
>> When SIP is implemented in an application within the present RTCweb
>> architecture, how exactly could we use WebSockets for SUBSCRIBE/NOTIFY?
>>
>> It appears to me that the only way to achieve that is to open a
>> persistent connection.  Am I right?
>>
>> If so, is it not a problem, given the potential duration of such
>> connections and potential tie-up of network resources?
>>
>> If not, how else could that be done?
>>
>> With many thanks,
>>
>> Igor
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>

--------------070102060704080703000108
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: 8bit

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
  <head>
    <meta content="text/html; charset=UTF-8" http-equiv="Content-Type">
  </head>
  <body text="#000000" bgcolor="#ffffff">
    Hannes,<br>
    <br>
    Thanks for following up! I actually think that my question has
    already been answered, but to reply toÂ  your query, I think that ofÂ 
    the two options you mentioned, the closest one to my questionÂ  is<br>
    <br>
    ...<br>
    <span style="font-family: Verdana;"><span style="font-size: 12px;">b)
        I want to provide interworking with an existing SIP
        implementation that uses the SIP event package: how do I do it?</span></span><br>
    ...<br>
    <br>
    More precisely, the problem statement was: I need to have SIP on my
    Internet appliance. Now that I have only the browser to rely on, how
    do I implement it?Â Â  <br>
    <br>
    Â I assume that I have 1) JavaScript to program in and 2) the
    WebSockets interface.Â  All is well until asynchronous processing
    comes in.Â  To implement SUBSCRIBE/NOTIFY, I would need to be able to
    receive the NOTIFY messages (i.e., catch interrupts) whenever they
    come.Â Â  I thought that the only way to achieve that was to keep an
    open TCP connection... forever.Â  (I was under the impression that
    this was not considered a good practice--both in terms of resending
    "keep alive" and holding resources (such as NAT).<br>
    <br>
    The responses that I got so far confirmed that my assumption about
    keeping a TCP connection openÂ  was right, but that the practice of
    keeping a TCP connection open has no longer been considered wasteful
    and is, in fact, implemented.Â  There was also a message about the
    necessity of being tied to the page, but this appears to me as
    something that can be meaningfully worked around.<br>
    <br>
    I guess more clarifications and implementation reports will come my
    way.Â  I do think it is critical that SIP be implementable in the
    browser-only environment.Â  I understand the reasons for the decision
    NOT to have SIP (or, for that matter ICE) as part of the browser,
    but that implied that SIP can be implemented in the application.<br>
    <br>
    Igor<br>
    <br>
    On 6/14/2011 10:33 AM, Hannes Tschofenig wrote:
    <blockquote cite="mid:20110614143331.43770@gmx.net" type="cite"><span
        style="font-family: Verdana;"><span style="font-size: 12px;">Hi
          Igor,<br>
          <br>
          I do not quite understand where you are heading with your
          question.<br>
          <br>
          Your question can be interpreted in two ways, namely:<br>
          <br>
          a) How do I implement SIP event subscription alike concepts in
          a browser?<br>
          (without using SIP itself but with browser technologies)<br>
          <br>
          b) I want to provide interworking with an existing SIP
          implementation that uses the SIP event package: how do I do
          it?<br>
          <br>
          In any case, you need to provide more details about what you
          want to accomplish in order for someone to provide you a good
          answer.<br>
          <br>
          For example, you may want to provide some information about<br>
          * what data would you like to subscribe to,<br>
          * where is the data,<br>
          * who is supposed to receive it? Â <br>
          <br>
          Ciao<br>
          Hannes<br>
          <br>
          PS: Your remark that maintaining connectivity alive for
          asynchronous communication is correct but what do you want to
          do if that's your desired functionality?<br>
          This aspect is independent of SIP, XMPP and the Web.<br>
          <br>
          <p style="margin: 0px; padding: 0px;"> Â </p>
          <blockquote style="border-left: 1px solid rgb(204, 204, 204);
            padding-left: 5px; margin: 0px 0px 0px 5px;" type="cite">
            <p style="margin: 0px; padding: 0px;"> <span
                style="font-family: Verdana;"><span style="font-size:
                  12px;">----- UrsprÃ¼ngliche Nachricht -----</span></span></p>
            <p style="margin: 0px; padding: 0px;"> <span
                style="font-family: Verdana;"><span style="font-size:
                  12px;">Von: Igor Faynberg</span></span></p>
            <p style="margin: 0px; padding: 0px;"> <span
                style="font-family: Verdana;"><span style="font-size:
                  12px;">Gesendet: 11.06.11 01:01 Uhr</span></span></p>
            <p style="margin: 0px; padding: 0px;"> <span
                style="font-family: Verdana;"><span style="font-size:
                  12px;">An: <a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a></span></span></p>
            <p style="margin: 0px; padding: 0px;"> <span
                style="font-family: Verdana;"><span style="font-size:
                  12px;">Betreff: [rtcweb] How do we implement SIP event
                  subscription in RTCweb?</span></span></p>
            <br>
            <div>
              <div>
                <pre style="white-space: pre-wrap; word-wrap: break-word; font-size: 11px;"> 
A question. 

When SIP is implemented in an application within the present RTCweb 
architecture, how exactly could we use WebSockets for SUBSCRIBE/NOTIFY? 

It appears to me that the only way to achieve that is to open a 
persistent connection.  Am I right? 

If so, is it not a problem, given the potential duration of such 
connections and potential tie-up of network resources? 

If not, how else could that be done? 

With many thanks, 

Igor 
_______________________________________________ 
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>
              </div>
            </div>
          </blockquote>
          <p style="margin: 0px; padding: 0px;"> Â </p>
        </span></span>
    </blockquote>
  </body>
</html>

--------------070102060704080703000108--

From mthornbu@adobe.com  Tue Jun 14 18:08:30 2011
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To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Tue, 14 Jun 2011 18:08:24 -0700
Thread-Topic: Re: [rtcweb] RTP multiplexing - my understanding of status
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(apologies for breaking the "in-reply-to" threadiness; i've been reading th=
e list through the archive web page)

On 6/13/2011 11:49:34 -0400, Randell Jesup wrote:
> I assume the Matthew Kaufman here is the same one who presented the paper=
 on RTMFP for Adobe at IETF 77. :-) From the email address, I take it you w=
ork for Skype now.

> While that protocol per se may be out-of-bounds (proprietary, patents), s=
ome of the problems they were trying to solve aren't entirely different tha=
n what we're looking at here.

except for "interoperation with legacy systems" and "no plug-ins", the prob=
lems we set out to solve with RTMFP were almost exactly the same: secure, r=
eal-time, congestion controlled, safe P2P communication of voice, video, an=
d data in the browser, with NAT traversal.

> In particular, there may be advantages in setup time and complexity to re=
ducing the number of "5-tuples" required and multiplexing data. For example=
, if you need to poke holes in a firewall using UPnP it can be painful and =
slow (to the point where we at WorldGate pre-poked holes to use if a call c=
ame in, and we had to keep 2 or 3 sets of holes open to handle call-waiting=
 cases, etc). Even ignoring that and just considering ICE, the number of me=
ssages to open N ports is a lot higher than 1 or a pair of ports, and it ma=
y also increase the chance of a single-packet loss interfering and slowing =
down convergence (or pushing ICE to select a sub-optimal connection method)=
. Also, what are the odds of ICE selecting different methods for the differ=
ent streams?

RTMFP takes multiplexing over UDP further: communications with multiple dis=
tinct endpoints can (if desired) share the same UDP port on a host, which c=
an make more efficient use of NAT resources and traversal-helpers (the "Red=
irectors" and "Forwarders" from Matthew's presentation at IETF 77 [1, pp19-=
30]).

> As mentioned, if this is done as a shim/multiplex layer it simplifies int=
erop with existing voice gateways. Side note: there is a draft for (just) m=
ultiplexing RTP and RTCP which I had weighed in on and supported; I forget =
where it stands currently.

> Another goal from RTMFP that is worth considering is consolidated congest=
ion control - doing congestion control and bandwidth measurement/adaptation=
 across all the streams exchanged with another endpoint together. This is s=
implified by using a single or smaller number of "5-tuples", though it isn'=
t gated on that.

> Thought should be given to some of the other goals of RTMFP and if they'r=
e in-scope or not. (IP addr changes, etc).

some other goals of RTMFP to which thought should be given:
  * all traffic is congestion controlled all the time (i know RTCWeb is alr=
eady thinking about this, but we think it's critical so i'm mentioning it a=
gain)
     * consolidated congestion control for prioritization of traffic (for e=
xample, control/signaling over media, audio over video & data, real-time da=
ta over bulk data)
     * multi-endpoint cooperative congestion control for real-time vs not-r=
eal-time concurrent communications [1, p34]
  * full or partial reliability for data (and audio & video media)
  * ordered or un-ordered (ASAP) receipt of messages
  * fragmentation/reassembly of large (>504 bytes) data messages
  * flow association to provide unambiguous channel grouping of distinct pa=
rallel communications between a single pair of endpoints, still sharing a s=
ingle security association, UDP port, congestion control and prioritization=
 domain
  * perform endpoint "white listing" based on ephemeral cryptographic ident=
ity instead of IP address/port number, while still using end-to-end encrypt=
ion keying

Michael Thornburgh

[1] http://www.ietf.org/proceedings/10mar/slides/tsvarea-1.pdf


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At the interim meeting discussion, the point was brought up that the use 
cases document at the moment doesn't contain any example where 
interoperability between clients driven by different Web applications 
could be an issue.

Rather than leaving this for later, I'll try to suggest some text below, 
in a form that should be easy to insert into draft-holmberg.

Comments on the below? Too little? Too much? Wrong? Just right?

                 Harald

----------------------------------------------------------------------

4.X. Use-case: Simple video communication service with inter-operator 
calling

4.X.1.  Description

Two users have logged into two different web applications, provided by 
different
service providers.

The service providers are interconnected by some means, but exchange no 
more information about the users than what can be carried using SIP. 
(Note: More profiling of what this means may be needed.)

Each web service publishes information about user login status for users 
that have a relationship with the other user; how this is established is 
out of scope.

The same functionality as in the "Simple Video Communication Service" is 
available.
The same issues with connectivity apply.

4.X.2 Derived requirements

(This describes requirements adding to the existing requirements)

FX1 The browser MUST be able to initiate and accept a media session 
using a common, standard protocol where the data needed for 
establishment can be carried in SIP.

FX2 The browser MUST support a baseline audio and video codec

FX3 There SHOULD be a mapping of the minimum needed data for setting up 
connections into SIP, so that the restriction to SIP-carriable data can 
be verified.

(There might be more here)



From stefan.lk.hakansson@ericsson.com  Thu Jun 16 06:26:27 2011
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Subject: Re: [rtcweb] Use case addition: Interoperable calling
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I think this additional use case makes a lot of sense. Some comments in 
line.
On 2011-06-16 14:35, Harald Alvestrand wrote:
> At the interim meeting discussion, the point was brought up that the use
> cases document at the moment doesn't contain any example where
> interoperability between clients driven by different Web applications
> could be an issue.
>
> Rather than leaving this for later, I'll try to suggest some text below,
> in a form that should be easy to insert into draft-holmberg.
>
> Comments on the below? Too little? Too much? Wrong? Just right?
>
>                   Harald
>
> ----------------------------------------------------------------------
>
> 4.X. Use-case: Simple video communication service with inter-operator
> calling
>
> 4.X.1.  Description
>
> Two users have logged into two different web applications, provided by
> different
> service providers.
>
> The service providers are interconnected by some means, but exchange no
> more information about the users than what can be carried using SIP.
> (Note: More profiling of what this means may be needed.)
>
> Each web service publishes information about user login status for users
> that have a relationship with the other user; how this is established is
> out of scope.
>
> The same functionality as in the "Simple Video Communication Service" is
> available.
> The same issues with connectivity apply.
>
> 4.X.2 Derived requirements
>
> (This describes requirements adding to the existing requirements)
>
> FX1 The browser MUST be able to initiate and accept a media session
> using a common, standard protocol where the data needed for
> establishment can be carried in SIP.
Do you by "common, standard protocol" mean http,  WS or something else?
>
> FX2 The browser MUST support a baseline audio and video codec
Related to the current F18. Video is added, and then I guess it is up 
for discussion if the telephony codec is sufficient as baseline audio.
>
> FX3 There SHOULD be a mapping of the minimum needed data for setting up
> connections into SIP, so that the restriction to SIP-carriable data can
> be verified.
Do you mean a mapping document the WG produces? Or do you mean that the 
browser should have some "SIP-validator" function implemented?

>
> (There might be more here)
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From harald@alvestrand.no  Thu Jun 16 06:53:21 2011
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On 06/16/11 15:26, Stefan H=E5kansson LK wrote:
> I think this additional use case makes a lot of sense. Some comments=20
> in line.
> On 2011-06-16 14:35, Harald Alvestrand wrote:
>> At the interim meeting discussion, the point was brought up that the u=
se
>> cases document at the moment doesn't contain any example where
>> interoperability between clients driven by different Web applications
>> could be an issue.
>>
>> Rather than leaving this for later, I'll try to suggest some text belo=
w,
>> in a form that should be easy to insert into draft-holmberg.
>>
>> Comments on the below? Too little? Too much? Wrong? Just right?
>>
>>                   Harald
>>
>> ----------------------------------------------------------------------=

>>
>> 4.X. Use-case: Simple video communication service with inter-operator
>> calling
>>
>> 4.X.1.  Description
>>
>> Two users have logged into two different web applications, provided by=

>> different
>> service providers.
>>
>> The service providers are interconnected by some means, but exchange n=
o
>> more information about the users than what can be carried using SIP.
>> (Note: More profiling of what this means may be needed.)
>>
>> Each web service publishes information about user login status for use=
rs
>> that have a relationship with the other user; how this is established =
is
>> out of scope.
>>
>> The same functionality as in the "Simple Video Communication Service" =
is
>> available.
>> The same issues with connectivity apply.
>>
>> 4.X.2 Derived requirements
>>
>> (This describes requirements adding to the existing requirements)
>>
>> FX1 The browser MUST be able to initiate and accept a media session
>> using a common, standard protocol where the data needed for
>> establishment can be carried in SIP.
> Do you by "common, standard protocol" mean http,  WS or something else?=

I was actually thinking "common" as in "both parties implement it", and=20
thinking of the browser-to-browser connection, not the=20
browser-to-webserver connection.

ICE + RTP is the combination I was thinking about, but naming protocols=20
in requirements isn't nice if we can avoid it.
>>
>> FX2 The browser MUST support a baseline audio and video codec
> Related to the current F18. Video is added, and then I guess it is up=20
> for discussion if the telephony codec is sufficient as baseline audio.
Good argument to have. I think U-Law falls way short of a good baseline, =

but it's not obvious that this falls out of any present usecase.
>>
>> FX3 There SHOULD be a mapping of the minimum needed data for setting u=
p
>> connections into SIP, so that the restriction to SIP-carriable data ca=
n
>> be verified.
> Do you mean a mapping document the WG produces? Or do you mean that=20
> the browser should have some "SIP-validator" function implemented?

A mapping document that the WG produces would satisfy me as a way to=20
show that it can be mapped.
Code is even better, of course.




From stefan.lk.hakansson@ericsson.com  Thu Jun 16 07:09:30 2011
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On 2011-06-16 15:53, Harald Alvestrand wrote:
> On 06/16/11 15:26, Stefan Håkansson LK wrote:
>> I think this additional use case makes a lot of sense. Some comments
>> in line.
>> On 2011-06-16 14:35, Harald Alvestrand wrote:
>>> At the interim meeting discussion, the point was brought up that the use
>>> cases document at the moment doesn't contain any example where
>>> interoperability between clients driven by different Web applications
>>> could be an issue.
>>>
>>> Rather than leaving this for later, I'll try to suggest some text below,
>>> in a form that should be easy to insert into draft-holmberg.
>>>
>>> Comments on the below? Too little? Too much? Wrong? Just right?
>>>
>>>                    Harald
>>>
>>> ----------------------------------------------------------------------
>>>
>>> 4.X. Use-case: Simple video communication service with inter-operator
>>> calling
>>>
>>> 4.X.1.  Description
>>>
>>> Two users have logged into two different web applications, provided by
>>> different
>>> service providers.
>>>
>>> The service providers are interconnected by some means, but exchange no
>>> more information about the users than what can be carried using SIP.
>>> (Note: More profiling of what this means may be needed.)
>>>
>>> Each web service publishes information about user login status for users
>>> that have a relationship with the other user; how this is established is
>>> out of scope.
>>>
>>> The same functionality as in the "Simple Video Communication Service" is
>>> available.
>>> The same issues with connectivity apply.
>>>
>>> 4.X.2 Derived requirements
>>>
>>> (This describes requirements adding to the existing requirements)
>>>
>>> FX1 The browser MUST be able to initiate and accept a media session
>>> using a common, standard protocol where the data needed for
>>> establishment can be carried in SIP.
>> Do you by "common, standard protocol" mean http,  WS or something else?
> I was actually thinking "common" as in "both parties implement it", and
> thinking of the browser-to-browser connection, not the
> browser-to-webserver connection.
>
> ICE + RTP is the combination I was thinking about, but naming protocols
> in requirements isn't nice if we can avoid it.
Aha, then I get it. The new part is "data needed for establishment can 
be carried in SIP"; the rest is already covered (F2 - F4).

Maybe it should be slightly re-phrased; when I read it "using a common, 
standard protocol" confuses, perhaps those words could just be removed?

If there is a need to state that the media session should use "a common, 
standard protocol" it is much earlier, this is relevant already to the 
first use case.
>>>
>>> FX2 The browser MUST support a baseline audio and video codec
>> Related to the current F18. Video is added, and then I guess it is up
>> for discussion if the telephony codec is sufficient as baseline audio.
> Good argument to have. I think U-Law falls way short of a good baseline,
> but it's not obvious that this falls out of any present usecase.
>>>
>>> FX3 There SHOULD be a mapping of the minimum needed data for setting up
>>> connections into SIP, so that the restriction to SIP-carriable data can
>>> be verified.
>> Do you mean a mapping document the WG produces? Or do you mean that
>> the browser should have some "SIP-validator" function implemented?
>
> A mapping document that the WG produces would satisfy me as a way to
> show that it can be mapped.
> Code is even better, of course.
>
>
>


From emcho@sip-communicator.org  Thu Jun 16 08:40:26 2011
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On Thu, Jun 16, 2011 at 2:35 PM, Harald Alvestrand <harald@alvestrand.no>wrote:

>
> Each web service publishes information about user login status for users
> that have a relationship with the other user; how this is established is out
> of scope.
>

I am not sure if I am understanding this part correctly. To me it sounds
like some sort of presence exchanged between users and, if so, I was
wondering if it was really necessary for inter-operator communication.

Emil

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<br><br><div class=3D"gmail_quote">On Thu, Jun 16, 2011 at 2:35 PM, 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;"=
>

<br>
Each web service publishes information about user login status for users th=
at have a relationship with the other user; how this is established is out =
of scope.<br></blockquote><div><br>I am not sure if I am understanding this=
 part correctly. To me it sounds like some sort of presence exchanged betwe=
en users and, if so, I was wondering if it was really necessary for inter-o=
perator communication.<br>

<br>Emil <br></div></div>

--20cf307f37fc655a3e04a5d6161e--

From harald@alvestrand.no  Thu Jun 16 09:20:47 2011
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From: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] Use case addition: Interoperable calling
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On 06/16/11 17:40, Emil Ivov wrote:
>
>
> On Thu, Jun 16, 2011 at 2:35 PM, Harald Alvestrand 
> <harald@alvestrand.no <mailto:harald@alvestrand.no>> wrote:
>
>
>     Each web service publishes information about user login status for
>     users that have a relationship with the other user; how this is
>     established is out of scope.
>
>
> I am not sure if I am understanding this part correctly. To me it 
> sounds like some sort of presence exchanged between users and, if so, 
> I was wondering if it was really necessary for inter-operator 
> communication.
I don't know - all chat networks and web-based calling networks use 
network presence pretty heavily; the phone network's able to live 
without it, assuming that "everyone's there all the time".

The only reason for mentioning it here is that it's out of our scope to 
standardize it - if we don't mention it, some people might believe it's 
in scope, others that it's out of scope.



--------------090707030000060005080407
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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 06/16/11 17:40, Emil Ivov wrote:
    <blockquote
cite="mid:BANLkTin60Ao=5BrcN=6vUGF8Be1A3-uCBFm=5n0QzONS+eVAJg@mail.gmail.com"
      type="cite"><br>
      <br>
      <div class="gmail_quote">On Thu, Jun 16, 2011 at 2:35 PM, 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;">
          <br>
          Each web service publishes information about user login status
          for users that have a relationship with the other user; how
          this is established is out of scope.<br>
        </blockquote>
        <div><br>
          I am not sure if I am understanding this part correctly. To me
          it sounds like some sort of presence exchanged between users
          and, if so, I was wondering if it was really necessary for
          inter-operator communication.<br>
        </div>
      </div>
    </blockquote>
    I don't know - all chat networks and web-based calling networks use
    network presence pretty heavily; the phone network's able to live
    without it, assuming that "everyone's there all the time".<br>
    <br>
    The only reason for mentioning it here is that it's out of our scope
    to standardize it - if we don't mention it, some people might
    believe it's in scope, others that it's out of scope.<br>
    <br>
    <br>
  </body>
</html>

--------------090707030000060005080407--

From tedh@ipinfusion.com  Thu Jun 16 21:30:55 2011
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From: Ted Hardie <hardie@ipinfusion.com>
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--20cf307f3ac6b886e704a5e0d983
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Hi Folks,

In order to allow folks time to submit draft-ietf-rtcweb versions in time
for the upcoming 00 deadline, the chairs would like to first outline our
theory of the document splits to meet our chartered milestones (to refresh
you self on them: http://datatracker.ietf.org/wg/rtcweb/charter/) in the
August time frame.

We'd like to have one document which describes the use cases.
We'd like to have one document that gives a system overview and outlines the
overall model.
We'd like to have one document that describes the privacy and security
model.
We'd like to have one document that  describes the connectivity model (for
NAT traversal etc.).

Later documents will be: signaling and negotiation methods, media
transports, datagram transport for non-media data, and one or more documents
on media processing and codecs.

Note that this set of "documents we'd like" is missing one that says
"requirements".  Discussion among the chairs has come to consensus that we
have at least three different kinds of requirements being discussed and the
the use of the term "requirement" is hindering both our internal discussion
and the balance of work between us and the W3C at the moment.  So, we'd like
to have the focus be on what the parts of the system *do* within the context
of a system overview, rather than on a discussion of requirements or, even
worse, a meta discussion of what the word "requirements" means in particular
contexts.

If folks our okay with this document set, the chairs will then appoint
editors from among those who have contributed text so far.  This may mean
that the initial -00 versions have a strong resemblance to the individual
submissions they'd already presented; it also may not.  We'd like to remind
everyone that taking on the role of document editor in an IETF context means
accepting the responsibility to make the document reflect the consensus of
the group.  If a WG -00 is missing items or needs revision to meet the needs
of the group, in other words, that's normal--send text.

Ted, Magnus, and Cullen

--20cf307f3ac6b886e704a5e0d983
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Hi Folks,<br>
<br>
In order to allow folks time to submit draft-ietf-rtcweb versions in=20
time for the upcoming 00 deadline, the chairs would like to first=20
outline our theory of the document splits to meet our chartered=20
milestones (to refresh you self on them: <a href=3D"http://datatracker.ietf=
.org/wg/rtcweb/charter/" target=3D"_blank">http://datatracker.ietf.org/wg/r=
tcweb/charter/</a>) in the August time frame.<br>
<br>
We&#39;d like to have one document which describes the use cases.<br>
We&#39;d like to have one document that gives a system overview and outline=
s the overall model.<br>
We&#39;d like to have one document that describes the privacy and security =
model.<br>
We&#39;d like to have one document that =A0describes the connectivity model=
 (for NAT traversal etc.).<br>
<br>
Later documents will be: signaling and negotiation methods, media=20
transports, datagram transport for non-media data, and one or more=20
documents on media processing and codecs.<br>
<br>
Note that this set of &quot;documents we&#39;d like&quot; is missing one th=
at says=20
&quot;requirements&quot;. =A0Discussion among the chairs has come to consen=
sus that=20
we have at least three different kinds of requirements being discussed=20
and the the use of the term &quot;requirement&quot; is hindering both our i=
nternal
 discussion and the balance of work between us and the W3C at the=20
moment. =A0So, we&#39;d like to have the focus be on what the parts of the=
=20
system *do* within the context of a system overview, rather than on a=20
discussion of requirements or, even worse, a meta discussion of what the
 word &quot;requirements&quot; means in particular contexts.<br>
<br>
If folks our okay with this document set, the chairs will then appoint=20
editors from among those who have contributed text so far. =A0This may=20
mean that the initial -00 versions have a strong resemblance to the=20
individual submissions they&#39;d already presented; it also may not. =A0We=
&#39;d=20
like to remind everyone that taking on the role of document editor in an
 IETF context means accepting the responsibility to make the document=20
reflect the consensus of the group. =A0If a WG -00 is missing items or=20
needs revision to meet the needs of the group, in other words, that&#39;s=
=20
normal--send text.<br>
<br>
Ted, Magnus, and Cullen<br>

--20cf307f3ac6b886e704a5e0d983--

From dhc@dcrocker.net  Thu Jun 16 22:21:25 2011
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On 6/16/2011 9:30 PM, Ted Hardie wrote:
> We'd like to have one document which describes the use cases.
> We'd like to have one document that gives a system overview and outlines the
> overall model.
> We'd like to have one document that describes the privacy and security model.
> We'd like to have one document that  describes the connectivity model (for NAT
> traversal etc.).
>
> Later documents will be: signaling and negotiation methods, media transports,
> datagram transport for non-media data, and one or more documents on media
> processing and codecs.


If someone wants to implement the simplest, core capability that is useful 
within this context of service, how many docs are they going to have to read?

Which ones? How long before they will be written?

For classic Web, I think it is still just 2.  Same for email.

My question is motivated by the usual concern about barriers to adoption that 
can stymie new services.

d/
-- 

   Dave Crocker
   Brandenburg InternetWorking
   bbiw.net

From christer.holmberg@ericsson.com  Fri Jun 17 00:16:04 2011
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Ted Hardie <hardie@ipinfusion.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Fri, 17 Jun 2011 09:15:56 +0200
Thread-Topic: [rtcweb] Call for comment on document split
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Hi,

I think the proposed document split looks very good.

Regards,

Christer

________________________________
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Ted Hardie
Sent: 17. kes=E4kuuta 2011 7:31
To: rtcweb@ietf.org
Subject: [rtcweb] Call for comment on document split

Hi Folks,

In order to allow folks time to submit draft-ietf-rtcweb versions in time f=
or the upcoming 00 deadline, the chairs would like to first outline our the=
ory of the document splits to meet our chartered milestones (to refresh you=
 self on them: http://datatracker.ietf.org/wg/rtcweb/charter/) in the Augus=
t time frame.

We'd like to have one document which describes the use cases.
We'd like to have one document that gives a system overview and outlines th=
e overall model.
We'd like to have one document that describes the privacy and security mode=
l.
We'd like to have one document that  describes the connectivity model (for =
NAT traversal etc.).

Later documents will be: signaling and negotiation methods, media transport=
s, datagram transport for non-media data, and one or more documents on medi=
a processing and codecs.

Note that this set of "documents we'd like" is missing one that says "requi=
rements".  Discussion among the chairs has come to consensus that we have a=
t least three different kinds of requirements being discussed and the the u=
se of the term "requirement" is hindering both our internal discussion and =
the balance of work between us and the W3C at the moment.  So, we'd like to=
 have the focus be on what the parts of the system *do* within the context =
of a system overview, rather than on a discussion of requirements or, even =
worse, a meta discussion of what the word "requirements" means in particula=
r contexts.

If folks our okay with this document set, the chairs will then appoint edit=
ors from among those who have contributed text so far.  This may mean that =
the initial -00 versions have a strong resemblance to the individual submis=
sions they'd already presented; it also may not.  We'd like to remind every=
one that taking on the role of document editor in an IETF context means acc=
epting the responsibility to make the document reflect the consensus of the=
 group.  If a WG -00 is missing items or needs revision to meet the needs o=
f the group, in other words, that's normal--send text.

Ted, Magnus, and Cullen

--_000_7F2072F1E0DE894DA4B517B93C6A0585194E3E71AEESESSCMS0356e_
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<DIV dir=3Dltr align=3Dleft><SPAN class=3D961351507-17062011><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>Hi,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D961351507-17062011><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D961351507-17062011><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>I think the proposed document split looks very=20
good.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D961351507-17062011><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D961351507-17062011><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>Regards,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D961351507-17062011><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D961351507-17062011><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>Christer</FONT></SPAN></DIV><BR>
<BLOCKQUOTE=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px soli=
d; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> rtcweb-bounces@ietf.org=20
  [mailto:rtcweb-bounces@ietf.org] <B>On Behalf Of </B>Ted=20
  Hardie<BR><B>Sent:</B> 17. kes=E4kuuta 2011 7:31<BR><B>To:</B>=20
  rtcweb@ietf.org<BR><B>Subject:</B> [rtcweb] Call for comment on document=
=20
  split<BR></FONT><BR></DIV>
  <DIV></DIV>Hi Folks,<BR><BR>In order to allow folks time to submit=20
  draft-ietf-rtcweb versions in time for the upcoming 00 deadline, the chai=
rs=20
  would like to first outline our theory of the document splits to meet our=
=20
  chartered milestones (to refresh you self on them: <A=20
  href=3D"http://datatracker.ietf.org/wg/rtcweb/charter/"=20
  target=3D_blank>http://datatracker.ietf.org/wg/rtcweb/charter/</A>) in th=
e=20
  August time frame.<BR><BR>We'd like to have one document which describes =
the=20
  use cases.<BR>We'd like to have one document that gives a system overview=
 and=20
  outlines the overall model.<BR>We'd like to have one document that descri=
bes=20
  the privacy and security model.<BR>We'd like to have one document that=20
  &nbsp;describes the connectivity model (for NAT traversal etc.).<BR><BR>L=
ater=20
  documents will be: signaling and negotiation methods, media transports,=20
  datagram transport for non-media data, and one or more documents on media=
=20
  processing and codecs.<BR><BR>Note that this set of "documents we'd like"=
 is=20
  missing one that says "requirements". &nbsp;Discussion among the chairs h=
as=20
  come to consensus that we have at least three different kinds of requirem=
ents=20
  being discussed and the the use of the term "requirement" is hindering bo=
th=20
  our internal discussion and the balance of work between us and the W3C at=
 the=20
  moment. &nbsp;So, we'd like to have the focus be on what the parts of the=
=20
  system *do* within the context of a system overview, rather than on a=20
  discussion of requirements or, even worse, a meta discussion of what the =
word=20
  "requirements" means in particular contexts.<BR><BR>If folks our okay wit=
h=20
  this document set, the chairs will then appoint editors from among those =
who=20
  have contributed text so far. &nbsp;This may mean that the initial -00=20
  versions have a strong resemblance to the individual submissions they'd=20
  already presented; it also may not. &nbsp;We'd like to remind everyone th=
at=20
  taking on the role of document editor in an IETF context means accepting =
the=20
  responsibility to make the document reflect the consensus of the group.=20
  &nbsp;If a WG -00 is missing items or needs revision to meet the needs of=
 the=20
  group, in other words, that's normal--send text.<BR><BR>Ted, Magnus, and=
=20
  Cullen<BR></BLOCKQUOTE></BODY></HTML>

--_000_7F2072F1E0DE894DA4B517B93C6A0585194E3E71AEESESSCMS0356e_--

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On 2011-06-17 07:21, Dave CROCKER wrote:
> 
> 
> On 6/16/2011 9:30 PM, Ted Hardie wrote:
>> We'd like to have one document which describes the use cases.
>> We'd like to have one document that gives a system overview and outlines the
>> overall model.
>> We'd like to have one document that describes the privacy and security model.
>> We'd like to have one document that  describes the connectivity model (for NAT
>> traversal etc.).
>>
>> Later documents will be: signaling and negotiation methods, media transports,
>> datagram transport for non-media data, and one or more documents on media
>> processing and codecs.
> 
> 
> If someone wants to implement the simplest, core capability that is useful 
> within this context of service, how many docs are they going to have to read?

That will likely be the full set of standards track documents, and the
overview to know how they fit together

> 
> Which ones? How long before they will be written?

The point of getting them written is exactly why we have this split. We
try to distribute the editing load over a number of people and people
expertise to get high quality documents.

> 
> For classic Web, I think it is still just 2.  Same for email.
> 
> My question is motivated by the usual concern about barriers to adoption that 
> can stymie new services.

Well, RTCWEB is primarily umbrella standardization. Something we rarely
do in IETF. This is actually about gluing together some 20-30 or more
specifications into a service.

Thus I think comparing it with basic HTTP service is not that relevant,
which by the way still requires you to read a bit more than 2 specs.
Even if I assume you have a working TCP/IP with DNS system to build on,
you still need the HTTP, the URI, the MIME spec and likely a few small
pieces more before you even can fetch an referenced object.


Cheers

Magnus Westerlund

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Subject: [rtcweb] WG adoption for Use Case document
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WG,

We would like to adopt a WG document with the purpose for describing use
cases for the RTCWEB work. This document would be targeted as
publication as an Inforamtional RFC eventually.

The candidate document for this is:
Web Real-Time Communication Use-cases and Requirements
https://datatracker.ietf.org/doc/draft-holmberg-rtcweb-ucreqs/

Based on our interim last week it was clear that there are some use
cases that clearly needs development. However, the WG chairs believe
that future development of the document can be done as WG item.

WG participants please indicate if you agree in adopting this document
as a basis, or if not, what the short comings you would like to see
addressed prior to adoption. Once more, the document will be continued
to be developed under the WG control and this is far from any final
version.

Please provide your view no later than Friday the 24th.

Best Regards

Magnus Westerlund
WG chair

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From randell1@jesup.org  Fri Jun 17 04:32:03 2011
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On 6/17/2011 4:00 AM, Magnus Westerlund wrote:
> WG,
>
> We would like to adopt a WG document with the purpose for describing use
> cases for the RTCWEB work. This document would be targeted as
> publication as an Inforamtional RFC eventually.
>
> The candidate document for this is:
> Web Real-Time Communication Use-cases and Requirements
> https://datatracker.ietf.org/doc/draft-holmberg-rtcweb-ucreqs/

Agreed

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I think the document is a good starting point, although, I personally would
prefer that the mapping would should the use cases that were used to derive
the requirements versus the requirements that are derived from the use case=
.
 Otherise, you have to   search through the use cases to see the one from
which the requirement was derived. Or perhaps having the mapping both
directions.

Also, I would suggest some reformatting so that new use cases can be easily
added - i.e., it doesn't look like the list is a result of the
<list>...</list> xml formatting. If that's not the case, I still don't like
this dashy list style.

Regards,
Mary.

On Fri, Jun 17, 2011 at 3:00 AM, Magnus Westerlund <
magnus.westerlund@ericsson.com> wrote:

> WG,
>
> We would like to adopt a WG document with the purpose for describing use
> cases for the RTCWEB work. This document would be targeted as
> publication as an Inforamtional RFC eventually.
>
> The candidate document for this is:
> Web Real-Time Communication Use-cases and Requirements
> https://datatracker.ietf.org/doc/draft-holmberg-rtcweb-ucreqs/
>
> Based on our interim last week it was clear that there are some use
> cases that clearly needs development. However, the WG chairs believe
> that future development of the document can be done as WG item.
>
> WG participants please indicate if you agree in adopting this document
> as a basis, or if not, what the short comings you would like to see
> addressed prior to adoption. Once more, the document will be continued
> to be developed under the WG control and this is far from any final
> version.
>
> Please provide your view no later than Friday the 24th.
>
> Best Regards
>
> Magnus Westerlund
> WG chair
>
> ----------------------------------------------------------------------
> 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
>

--bcaec53f8f4f3409ce04a5e71e15
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

I think the document is a good starting point, although, I personally would=
 prefer that the mapping would should the use cases that were used to deriv=
e the requirements versus the requirements that are derived from the use ca=
se. =A0Otherise, you have to =A0 search through the use cases to see the on=
e from which the requirement was derived. Or perhaps having the mapping bot=
h directions. =A0<div>
<br></div><div>Also, I would suggest some reformatting so that new use case=
s can be easily added - i.e., it doesn&#39;t look like the list is a result=
 of the &lt;list&gt;...&lt;/list&gt; xml formatting. If that&#39;s not the =
case, I still don&#39;t like this dashy list style.=A0</div>
<div><br></div><div>Regards,</div><div>Mary.=A0<br><br><div class=3D"gmail_=
quote">On Fri, Jun 17, 2011 at 3:00 AM, Magnus Westerlund <span dir=3D"ltr"=
>&lt;<a href=3D"mailto:magnus.westerlund@ericsson.com">magnus.westerlund@er=
icsson.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;">WG,<br>
<br>
We would like to adopt a WG document with the purpose for describing use<br=
>
cases for the RTCWEB work. This document would be targeted as<br>
publication as an Inforamtional RFC eventually.<br>
<br>
The candidate document for this is:<br>
Web Real-Time Communication Use-cases and Requirements<br>
<a href=3D"https://datatracker.ietf.org/doc/draft-holmberg-rtcweb-ucreqs/" =
target=3D"_blank">https://datatracker.ietf.org/doc/draft-holmberg-rtcweb-uc=
reqs/</a><br>
<br>
Based on our interim last week it was clear that there are some use<br>
cases that clearly needs development. However, the WG chairs believe<br>
that future development of the document can be done as WG item.<br>
<br>
WG participants please indicate if you agree in adopting this document<br>
as a basis, or if not, what the short comings you would like to see<br>
addressed prior to adoption. Once more, the document will be continued<br>
to be developed under the WG control and this is far from any final<br>
version.<br>
<br>
Please provide your view no later than Friday the 24th.<br>
<br>
Best Regards<br>
<br>
Magnus Westerlund<br>
WG chair<br>
<br>
----------------------------------------------------------------------<br>
Multimedia Technologies, Ericsson Research EAB/TVM<br>
----------------------------------------------------------------------<br>
Ericsson AB =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Phone =A0+46 10 7148287<br>
F=E4r=F6gatan 6 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Mobile +46 73 0949079<br>
SE-164 80 Stockholm, Sweden| mailto: <a href=3D"mailto:magnus.westerlund@er=
icsson.com">magnus.westerlund@ericsson.com</a><br>
----------------------------------------------------------------------<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></div>

--bcaec53f8f4f3409ce04a5e71e15--

From christer.holmberg@ericsson.com  Fri Jun 17 05:27:38 2011
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Mary Barnes <mary.ietf.barnes@gmail.com>, Magnus Westerlund <magnus.westerlund@ericsson.com>
Date: Fri, 17 Jun 2011 14:26:15 +0200
Thread-Topic: [rtcweb] WG adoption for Use Case document
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Hi Mary,
=09
>I think the document is a good starting point, although, I personally woul=
d prefer that the mapping would should the use cases that were used to deri=
ve the requirements versus the requirements=20
>that are derived from the use case.=20

We can fix that.

>Otherise, you have to   search through the use cases to see the one from w=
hich the requirement was derived. Or perhaps having the mapping both direct=
ions.  =20
>
>Also, I would suggest some reformatting so that new use cases can be easil=
y added - i.e., it doesn't look like the list is a result of the <list>...<=
/list> xml formatting. If that's not the=20
>case, I still don't like this dashy list style.=20

We can fix that too.

Thanks!

Regards,

Christer



=09
=09
	On Fri, Jun 17, 2011 at 3:00 AM, Magnus Westerlund <magnus.westerlund@eric=
sson.com> wrote:
=09

		WG,
	=09
		We would like to adopt a WG document with the purpose for describing use
		cases for the RTCWEB work. This document would be targeted as
		publication as an Inforamtional RFC eventually.
	=09
		The candidate document for this is:
		Web Real-Time Communication Use-cases and Requirements
		https://datatracker.ietf.org/doc/draft-holmberg-rtcweb-ucreqs/
	=09
		Based on our interim last week it was clear that there are some use
		cases that clearly needs development. However, the WG chairs believe
		that future development of the document can be done as WG item.
	=09
		WG participants please indicate if you agree in adopting this document
		as a basis, or if not, what the short comings you would like to see
		addressed prior to adoption. Once more, the document will be continued
		to be developed under the WG control and this is far from any final
		version.
	=09
		Please provide your view no later than Friday the 24th.
	=09
		Best Regards
	=09
		Magnus Westerlund
		WG chair
	=09
		----------------------------------------------------------------------
		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
		----------------------------------------------------------------------
	=09
		_______________________________________________
		rtcweb mailing list
		rtcweb@ietf.org
		https://www.ietf.org/mailman/listinfo/rtcweb
	=09



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Agreed.=20

Anil

On Jun 17, 2011, at 1:00 AM, Magnus Westerlund <magnus.westerlund@ericsson.c=
om> wrote:

> WG,
>=20
> We would like to adopt a WG document with the purpose for describing use
> cases for the RTCWEB work. This document would be targeted as
> publication as an Inforamtional RFC eventually.
>=20
> The candidate document for this is:
> Web Real-Time Communication Use-cases and Requirements
> https://datatracker.ietf.org/doc/draft-holmberg-rtcweb-ucreqs/
>=20
> Based on our interim last week it was clear that there are some use
> cases that clearly needs development. However, the WG chairs believe
> that future development of the document can be done as WG item.
>=20
> WG participants please indicate if you agree in adopting this document
> as a basis, or if not, what the short comings you would like to see
> addressed prior to adoption. Once more, the document will be continued
> to be developed under the WG control and this is far from any final
> version.
>=20
> Please provide your view no later than Friday the 24th.
>=20
> Best Regards
>=20
> Magnus Westerlund
> WG chair
>=20
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=C3=A4r=C3=B6gatan 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

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On Jun 16, 2011, at 11:21 PM, Dave CROCKER wrote:

>=20
>=20
> If someone wants to implement the simplest, core capability that is =
useful within this context of service, how many docs are they going to =
have to read?

Just as a random guess, probably around 20 to 50. This is a bit more =
complicated that email. There's not consensus on what the set would be =
yet but I would guess that in the end we will end up with a list of docs =
a browser would need to implement (beyond what they already implement) =
that is roughly along the lines of:  the docs on RTP, SRTP and various =
keying approach, various audio and video codecs, various offer answer =
schemes, the whole ICE family of docs including STUN and TURN families, =
a bunch of docs on SDP and various offer answer schemes.=20=

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On Jun 16, 2011, at 10:30 PM, Ted Hardie wrote:

> Hi Folks,
>=20
> In order to allow folks time to submit draft-ietf-rtcweb versions in =
time for the upcoming 00 deadline, the chairs would like to first =
outline our theory of the document splits to meet our chartered =
milestones (to refresh you self on them: =
http://datatracker.ietf.org/wg/rtcweb/charter/) in the August time =
frame.
>=20
> We'd like to have one document which describes the use cases.
> We'd like to have one document that gives a system overview and =
outlines the overall model.
> We'd like to have one document that describes the privacy and security =
model.
> We'd like to have one document that  describes the connectivity model =
(for NAT traversal etc.).
>=20
> Later documents will be: signaling and negotiation methods, media =
transports, datagram transport for non-media data, and one or more =
documents on media processing and codecs.

With my individual contributor hat on ...=20

I don't see the reason to wait for "later" on these other docs.  I think =
some of them are as ready now as the connectivity model stuff is. Let's =
get moving with this stuff.


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This is a multi-part message in MIME format.
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Ted,

I think the proposed split is well in line with the Internet 
Architecture principle, "Modularity is always good."

It is well thought through.

Igor

On 6/17/2011 12:30 AM, Ted Hardie wrote:
> Hi Folks,
>
> In order to allow folks time to submit draft-ietf-rtcweb versions in 
> time for the upcoming 00 deadline, the chairs would like to first 
> outline our theory of the document splits to meet our chartered 
> milestones (to refresh you self on them: 
> http://datatracker.ietf.org/wg/rtcweb/charter/) in the August time frame.
>
> We'd like to have one document which describes the use cases.
> We'd like to have one document that gives a system overview and 
> outlines the overall model.
> We'd like to have one document that describes the privacy and security 
> model.
> We'd like to have one document that  describes the connectivity model 
> (for NAT traversal etc.).
>
> Later documents will be: signaling and negotiation methods, media 
> transports, datagram transport for non-media data, and one or more 
> documents on media processing and codecs.
>
> Note that this set of "documents we'd like" is missing one that says 
> "requirements".  Discussion among the chairs has come to consensus 
> that we have at least three different kinds of requirements being 
> discussed and the the use of the term "requirement" is hindering both 
> our internal discussion and the balance of work between us and the W3C 
> at the moment.  So, we'd like to have the focus be on what the parts 
> of the system *do* within the context of a system overview, rather 
> than on a discussion of requirements or, even worse, a meta discussion 
> of what the word "requirements" means in particular contexts.
>
> If folks our okay with this document set, the chairs will then appoint 
> editors from among those who have contributed text so far.  This may 
> mean that the initial -00 versions have a strong resemblance to the 
> individual submissions they'd already presented; it also may not. 
>  We'd like to remind everyone that taking on the role of document 
> editor in an IETF context means accepting the responsibility to make 
> the document reflect the consensus of the group.  If a WG -00 is 
> missing items or needs revision to meet the needs of the group, in 
> other words, that's normal--send text.
>
> Ted, Magnus, and Cullen
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

--------------080905050800000306010108
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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">
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  <body text="#000000" bgcolor="#ffffff">
    Ted,<br>
    <br>
    I think the proposed split is well in line with the Internet
    Architecture principle, "Modularity is always good."<br>
    <br>
    It is well thought through.<br>
    <br>
    Igor<br>
    <br>
    On 6/17/2011 12:30 AM, Ted Hardie wrote:
    <blockquote
      cite="mid:BANLkTinu402NoPovU6nDWAKKUBKfbJyk3Q@mail.gmail.com"
      type="cite">Hi Folks,<br>
      <br>
      In order to allow folks time to submit draft-ietf-rtcweb versions
      in time for the upcoming 00 deadline, the chairs would like to
      first outline our theory of the document splits to meet our
      chartered milestones (to refresh you self on them: <a
        moz-do-not-send="true"
        href="http://datatracker.ietf.org/wg/rtcweb/charter/"
        target="_blank">http://datatracker.ietf.org/wg/rtcweb/charter/</a>)
      in the August time frame.<br>
      <br>
      We'd like to have one document which describes the use cases.<br>
      We'd like to have one document that gives a system overview and
      outlines the overall model.<br>
      We'd like to have one document that describes the privacy and
      security model.<br>
      We'd like to have one document that &nbsp;describes the connectivity
      model (for NAT traversal etc.).<br>
      <br>
      Later documents will be: signaling and negotiation methods, media
      transports, datagram transport for non-media data, and one or more
      documents on media processing and codecs.<br>
      <br>
      Note that this set of "documents we'd like" is missing one that
      says "requirements". &nbsp;Discussion among the chairs has come to
      consensus that we have at least three different kinds of
      requirements being discussed and the the use of the term
      "requirement" is hindering both our internal discussion and the
      balance of work between us and the W3C at the moment. &nbsp;So, we'd
      like to have the focus be on what the parts of the system *do*
      within the context of a system overview, rather than on a
      discussion of requirements or, even worse, a meta discussion of
      what the word "requirements" means in particular contexts.<br>
      <br>
      If folks our okay with this document set, the chairs will then
      appoint editors from among those who have contributed text so far.
      &nbsp;This may mean that the initial -00 versions have a strong
      resemblance to the individual submissions they'd already
      presented; it also may not. &nbsp;We'd like to remind everyone that
      taking on the role of document editor in an IETF context means
      accepting the responsibility to make the document reflect the
      consensus of the group. &nbsp;If a WG -00 is missing items or needs
      revision to meet the needs of the group, in other words, that's
      normal--send text.<br>
      <br>
      Ted, Magnus, and Cullen<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>

--------------080905050800000306010108--

From igor.faynberg@alcatel-lucent.com  Fri Jun 17 10:24:40 2011
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Subject: Re: [rtcweb] Call for comment on document split
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I share the concern, but yet no matter how humble the beginning was 
(e.g., SIP), things ended up with far too many specifications for anyone 
to grasp (e.g., ...SIP).  don't think we will get away with two 
specifications no matter what (but I am a registered pessimist).I

This is why I think that modularity thought through in the beginning 
(vs. ad-hoc additions) will ensure order in the future.  In addition, 
with the controlled break-up, we could know in advance whether (and how) 
change in one document would affect others.

It is true though that a short Informational describing the structure of 
the project and references to other documents would be helpful.

Igor

On 6/17/2011 1:21 AM, Dave CROCKER wrote:
>
>
> On 6/16/2011 9:30 PM, Ted Hardie wrote:
>> We'd like to have one document which describes the use cases.
>> We'd like to have one document that gives a system overview and 
>> outlines the
>> overall model.
>> We'd like to have one document that describes the privacy and 
>> security model.
>> We'd like to have one document that  describes the connectivity model 
>> (for NAT
>> traversal etc.).
>>
>> Later documents will be: signaling and negotiation methods, media 
>> transports,
>> datagram transport for non-media data, and one or more documents on 
>> media
>> processing and codecs.
>
>
> If someone wants to implement the simplest, core capability that is 
> useful within this context of service, how many docs are they going to 
> have to read?
>
> Which ones? How long before they will be written?
>
> For classic Web, I think it is still just 2.  Same for email.
>
> My question is motivated by the usual concern about barriers to 
> adoption that can stymie new services.
>
> d/

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On Jun 12, 2011, at 3:57 AM, Harald Alvestrand wrote:

>> The resolution of waking up for a timer on WinNT is 30 msec and only =
15 msec on WinXP and ICE works just fine there.
>>=20
>> I am not at all convinced that for our use cases we need anything =
better than ~100 msec, so your 50 msec is already twice as good.
> The shortest timer in RFC 5245 (as far as I can see) is Ta (section =
16.1), which has a lower bound of 20 msec; however, this is a lower =
bound on transmission intervals, so it's possible that an implementation =
that fires only every 100 msec would be perfectly interoperable with an =
application that fires every 20 msec.
>=20
> Testing will show....

One complication is that there is ICE in theory and ICE in practice. =
Some of the times in the theory are bit longer than some implementations =
use in practice. This theory numbers are larger due to congestion =
control concerns at IETF. Some of the practice numbers are a bit lower =
to reduce the latency to have the call set up.=20

Testing will tell ... :-)




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From: Cullen Jennings <fluffy@cisco.com>
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Subject: [rtcweb] use case:
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If people agree with the following use cases, I was hoping they might =
get added to draft-holmberg-rtcweb-ucreqs.


Video Size Change.

Alice and Bob are in a video call in their browsers and have negotiate a =
high resolution video. Bob decides to change the size of the windows his =
browser is displaying video to a small size.  Bob's browser should be =
able to regenerate the video codec parameters with Alice's browser to =
change the resolution of video Alice sends to match the smaller size.=20


Call Fedex

Alice uses her web browser with a  service something like Skype to be =
able to phone PSTN numbers. Alice calls 1-800-gofedex. Alice should be =
able to hear the initial prompts from the fedex IVR and when the IVR =
says press 1, there should be a way for Alice to navigate the IVR.=20



NIC Change

Alice is on her notebook computer that is plugged in to 1G ethernet and =
has 802.11 wireless interface. Alice is in a call talking with Bob and =
decides to unplug here notebook computer and walk down to a different =
room. The call should continue as Alice unplugs from the fat network =
interface and moves to a slower one. The media codecs may end up =
renegotiating to select a different bandwidth consumption on the =
wireless network.=20



QoS Marking=20

Alice's browser is on a computer behind a common residential router that =
supports prioritization of traffic . The browser should be able to take =
advantage of the capabilities of the  router to prioritize voice and =
video appropriately.=20



Cullen <in my individual contributor role>





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Subject: Re: [rtcweb] Use case addition: Interoperable calling
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On Thu, Jun 16, 2011 at 2:35 PM, Harald Alvestrand <harald@alvestrand.no>
wrote:


Each web service publishes information about user login status for users
that have a relationship with the other user; how this is established is out
of scope.


I am not sure if I am understanding this part correctly. To me it sounds
like some sort of presence exchanged between users and, if so, I was
wondering if it was really necessary for inter-operator communication.

I don't know - all chat networks and web-based calling networks use network
presence pretty heavily; the phone network's able to live without it,
assuming that "everyone's there all the time".

Mobile networks have their own presence.  HLR/VLR or HSS in IMS terms.


The only reason for mentioning it here is that it's out of our scope to
standardize it - if we don't mention it, some people might believe it's in
scope, others that it's out of scope.

So if this is out of scope and its seems it is , then this entire WEB-RTC
exercise seems to perpetuate the kind of silos we seen in Instant Messaging
and other forms of RTC.

Pardon me for sounding overly cynical..

 






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class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'><o:p>&nbsp;</o:p></p><div><p =
class=3DMsoNormal>On Thu, Jun 16, 2011 at 2:35 PM, Harald Alvestrand =
&lt;<a href=3D"mailto:harald@alvestrand.no">harald@alvestrand.no</a>&gt; =
wrote:<o:p></o:p></p><p class=3DMsoNormal><br>Each web service publishes =
information about user login status for users that have a relationship =
with the other user; how this is established is out of =
scope.<o:p></o:p></p><div><p class=3DMsoNormal><br>I am not sure if I am =
understanding this part correctly. To me it sounds like some sort of =
presence exchanged between users and, if so, I was wondering if it was =
really necessary for inter-operator =
communication.<o:p></o:p></p></div></div><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'>I don't know - all chat networks and =
web-based calling networks use network presence pretty heavily; the =
phone network's able to live without it, assuming that &quot;everyone's =
there all the time&quot;.<span =
style=3D'color:#1F497D'><o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Mobile networks have their own presence.&nbsp; HLR/VLR or HSS in IMS =
terms.<o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'><br>The only reason for mentioning it =
here is that it's out of our scope to standardize it - if we don't =
mention it, some people might believe it's in scope, others that it's =
out of scope.<span style=3D'color:#1F497D'><o:p></o:p></span></p><p =
class=3DMsoNormal style=3D'margin-bottom:12.0pt'><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>So if this is out of scope and its seems it is , then this entire =
WEB-RTC exercise seems to perpetuate the kind of silos we seen in =
Instant Messaging and other forms of RTC.<o:p></o:p></span></p><p =
class=3DMsoNormal style=3D'margin-bottom:12.0pt'><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Pardon me for sounding overly cynical..<o:p></o:p></span></p><p =
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style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'><br><br><o:p></o:p></p></div></body></html=
>
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From randell1@jesup.org  Fri Jun 17 11:52:23 2011
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Subject: Re: [rtcweb] Use case addition: Interoperable calling
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On 6/16/2011 12:20 PM, Harald Alvestrand wrote:
> On 06/16/11 17:40, Emil Ivov wrote:
>> On Thu, Jun 16, 2011 at 2:35 PM, Harald Alvestrand 
>> <harald@alvestrand.no <mailto:harald@alvestrand.no>> wrote:
>>
>>
>>     Each web service publishes information about user login status
>>     for users that have a relationship with the other user; how this
>>     is established is out of scope.
>>

I don't even think that's necessary - the user could indicate he wants 
to talk to Joe_OHara@facebook.com, without having to know if Joe is 
currently on and registered with Facebook or not.  Also, a web service 
may implement the equivalent to network answering machines (forward to 
voice/videomail on not registered or no answer, or forward to another 
user, etc).

>>
>> I am not sure if I am understanding this part correctly. To me it 
>> sounds like some sort of presence exchanged between users and, if so, 
>> I was wondering if it was really necessary for inter-operator 
>> communication.
> I don't know - all chat networks and web-based calling networks use 
> network presence pretty heavily; the phone network's able to live 
> without it, assuming that "everyone's there all the time".

Well, the PSTN network doesn't really assume that.   It knows where a 
number will get routed (which provider), but the device in question may 
not be reachable (especially for cellphones and VoIP phones).

> The only reason for mentioning it here is that it's out of our scope 
> to standardize it - if we don't mention it, some people might believe 
> it's in scope, others that it's out of scope.

Sure.

-- 
Randell Jesup
randell-ietf@jesup.org

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On Fri, Jun 17, 2011 at 11:09 AM, Cullen Jennings <fluffy@cisco.com> wrote:

>
> Video Size Change.
>
> Alice and Bob are in a video call in their browsers and have negotiate a
> high resolution video. Bob decides to change the size of the windows his
> browser is displaying video to a small size.  Bob's browser should be able
> to regenerate the video codec parameters with Alice's browser to change the
> resolution of video Alice sends to match the smaller size.
>
>
> Changing compression due to a user change in window size is usually not
done since it has a bad long term effect on video quality. In almost all the
major video calling systems, video resolution is a function of current
bandwidth (which changes as the rate control system detects differences)
and/or constrained by the physical device, but not a function of a user
dragging the window size. At an equivalent bitrate, it is better to compress
a larger resolution and display it smaller (higher PSNR), then compress a
smaller resolution if you are within the codec's effective bitrate for that
larger resolution.

Aron Rosenberg
LifeSize, A Division of Logitech

--0015174ff1f4dcda1f04a5ed8743
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div class=3D"gmail_quote">On Fri, Jun 17, 2011 at 11:09 AM, Cullen Jenning=
s <span dir=3D"ltr">&lt;<a href=3D"mailto:fluffy@cisco.com">fluffy@cisco.co=
m</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margi=
n:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">

<br>Video Size Change.<br>
<br>
Alice and Bob are in a video call in their browsers and have negotiate a hi=
gh resolution video. Bob decides to change the size of the windows his brow=
ser is displaying video to a small size. =A0Bob&#39;s browser should be abl=
e to regenerate the video codec parameters with Alice&#39;s browser to chan=
ge the resolution of video Alice sends to match the smaller size.<br>


<br>
<br></blockquote><div>Changing compression due to a user change in window s=
ize is usually not done since it has a bad long term effect on video qualit=
y. In almost all the major video calling systems, video resolution is a fun=
ction of current bandwidth (which changes as the rate control system detect=
s differences) and/or constrained by the physical device, but not a functio=
n of a user dragging the window size. At an=A0equivalent=A0bitrate, it is b=
etter to compress a larger resolution and display it smaller (higher PSNR),=
 then compress a smaller resolution if you are within the codec&#39;s effec=
tive bitrate for that larger resolution.</div>

<div><br></div><div>Aron Rosenberg</div><div>LifeSize, A Division of Logite=
ch</div></div><br>

--0015174ff1f4dcda1f04a5ed8743--

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From: Kundan Singh <kundansingh_99@yahoo.com>
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=0A=0AFrom a software developer's point of view, I feel the following will =
be most productive:=0A=0A1. One or two complete document that are necessary=
 and sufficient to implement RTC-Web/WebRTC.=A0=0AEverything else that is n=
ot covered by this document must be an existing popular standard with exist=
ing popular libraries/APIs.=0A=0A=0A2. Create the documents based on actual=
 implementation, rather than the reverse. =0A=0APerhaps include pseudo-code=
 in Internet drafts to avoid ambiguity.=0A=0AIf interaction with external s=
tandards are needed, pseudo-code could show how it is done based on interfa=
ces of existing popular libraries/APIs for that external standard.=0A=0AHav=
ing to read and implement many documents to do one thing=A0 is a recipe for=
 disaster, e.g., (non) interoperability matrix, frustration, ...=0A=0AIf we=
 manage to create tens of documents that cover all corner cases, all featur=
es, all authors, all existing extensions, but scare away actual developers,=
 then we haven't learned from our mistakes...=0A=0AJust a thought.=0A=0A--=
=0AKundan Singh=0A=0A=0A=0A----- Original Message -----=0A> From: Igor Fayn=
berg <igor.faynberg@alcatel-lucent.com>=0A> To: rtcweb@ietf.org=0A> Cc: =0A=
> Sent: Friday, June 17, 2011 10:24 AM=0A> Subject: Re: [rtcweb] Call for c=
omment on document split=0A> =0A> I share the concern, but yet no matter ho=
w humble the beginning was (e.g., SIP), =0A> things ended up with far too m=
any specifications for anyone to grasp (e.g., =0A> ...SIP).=A0 don't think =
we will get away with two specifications no matter =0A> what (but I am a re=
gistered pessimist).I=0A> =0A> This is why I think that modularity thought =
through in the beginning (vs. ad-hoc =0A> additions) will ensure order in t=
he future.=A0 In addition, with the controlled =0A> break-up, we could know=
 in advance whether (and how) change in one document =0A> would affect othe=
rs.=0A> =0A> It is true though that a short Informational describing the st=
ructure of the =0A> project and references to other documents would be help=
ful.=0A> =0A> Igor=0A> =0A> On 6/17/2011 1:21 AM, Dave CROCKER wrote:=0A>> =
=0A>> =0A>>  On 6/16/2011 9:30 PM, Ted Hardie wrote:=0A>>>  We'd like to ha=
ve one document which describes the use cases.=0A>>>  We'd like to have one=
 document that gives a system overview and =0A> outlines the=0A>>>  overall=
 model.=0A>>>  We'd like to have one document that describes the privacy an=
d =0A> security model.=0A>>>  We'd like to have one document that=A0 descri=
bes the connectivity =0A> model (for NAT=0A>>>  traversal etc.).=0A>>> =0A>=
>>  Later documents will be: signaling and negotiation methods, media =0A> =
transports,=0A>>>  datagram transport for non-media data, and one or more d=
ocuments on =0A> media=0A>>>  processing and codecs.=0A>> =0A>> =0A>>  If s=
omeone wants to implement the simplest, core capability that is useful =0A>=
 within this context of service, how many docs are they going to have to re=
ad?=0A>> =0A>>  Which ones? How long before they will be written?=0A>> =0A>=
>  For classic Web, I think it is still just 2.=A0 Same for email.=0A>> =0A=
>>  My question is motivated by the usual concern about barriers to adoptio=
n =0A> that can stymie new services.=0A>> =0A>>  d/=0A> ___________________=
____________________________=0A> rtcweb mailing list=0A> rtcweb@ietf.org=0A=
> https://www.ietf.org/mailman/listinfo/rtcweb=0A>

From petithug@acm.org  Fri Jun 17 13:11:53 2011
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

As a software developer myself, I believe that this approach is wrong.  The goal
of the IETF is to make the Internet work better, not to make the developers life
better.

Implementers should contribute running code to help the IETF make the Internet
work better, by having multiple implementations of the same protocol and find
any interoperability, security and extensibility issue as soon as possible.
Documenting one implementation as protocol creates monoculture, and everybody
should know better by now than encouraging this.

And BTW, I do not think that "scaring away" developers is a bad thing.

On 06/17/2011 12:47 PM, Kundan Singh wrote:
> 
> 
>>From a software developer's point of view, I feel the following will be most productive:
> 
> 1. One or two complete document that are necessary and sufficient to implement RTC-Web/WebRTC. 
> Everything else that is not covered by this document must be an existing popular standard with existing popular libraries/APIs.
> 
> 
> 2. Create the documents based on actual implementation, rather than the reverse. 
> 
> Perhaps include pseudo-code in Internet drafts to avoid ambiguity.
> 
> If interaction with external standards are needed, pseudo-code could show how it is done based on interfaces of existing popular libraries/APIs for that external standard.
> 
> Having to read and implement many documents to do one thing  is a recipe for disaster, e.g., (non) interoperability matrix, frustration, ...
> 
> If we manage to create tens of documents that cover all corner cases, all features, all authors, all existing extensions, but scare away actual developers, then we haven't learned from our mistakes...
> 
> Just a thought.
> 
> --
> Kundan Singh
> 
> 
> 
> ----- Original Message -----
>> From: Igor Faynberg <igor.faynberg@alcatel-lucent.com>
>> To: rtcweb@ietf.org
>> Cc: 
>> Sent: Friday, June 17, 2011 10:24 AM
>> Subject: Re: [rtcweb] Call for comment on document split
>>
>> I share the concern, but yet no matter how humble the beginning was (e.g., SIP), 
>> things ended up with far too many specifications for anyone to grasp (e.g., 
>> ...SIP).  don't think we will get away with two specifications no matter 
>> what (but I am a registered pessimist).I
>>
>> This is why I think that modularity thought through in the beginning (vs. ad-hoc 
>> additions) will ensure order in the future.  In addition, with the controlled 
>> break-up, we could know in advance whether (and how) change in one document 
>> would affect others.
>>
>> It is true though that a short Informational describing the structure of the 
>> project and references to other documents would be helpful.
>>
>> Igor
>>
>> On 6/17/2011 1:21 AM, Dave CROCKER wrote:
>>>
>>>
>>>  On 6/16/2011 9:30 PM, Ted Hardie wrote:
>>>>  We'd like to have one document which describes the use cases.
>>>>  We'd like to have one document that gives a system overview and 
>> outlines the
>>>>  overall model.
>>>>  We'd like to have one document that describes the privacy and 
>> security model.
>>>>  We'd like to have one document that  describes the connectivity 
>> model (for NAT
>>>>  traversal etc.).
>>>>
>>>>  Later documents will be: signaling and negotiation methods, media 
>> transports,
>>>>  datagram transport for non-media data, and one or more documents on 
>> media
>>>>  processing and codecs.
>>>
>>>
>>>  If someone wants to implement the simplest, core capability that is useful 
>> within this context of service, how many docs are they going to have to read?
>>>
>>>  Which ones? How long before they will be written?
>>>
>>>  For classic Web, I think it is still just 2.  Same for email.
>>>
>>>  My question is motivated by the usual concern about barriers to adoption 
>> that can stymie new services.
>>>
>>>  d/
>> _______________________________________________
>> 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 Jun 20 01:20:52 2011
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I agree with adopting the document as a WG starting point.

Naturally, I expect multiple use cases to be added (and perhaps even 
some use cases marked up with "RTCWEB can't do this") before we're finished.

             Harald


On 06/17/11 10:00, Magnus Westerlund wrote:
> WG,
>
> We would like to adopt a WG document with the purpose for describing use
> cases for the RTCWEB work. This document would be targeted as
> publication as an Inforamtional RFC eventually.
>
> The candidate document for this is:
> Web Real-Time Communication Use-cases and Requirements
> https://datatracker.ietf.org/doc/draft-holmberg-rtcweb-ucreqs/
>
> Based on our interim last week it was clear that there are some use
> cases that clearly needs development. However, the WG chairs believe
> that future development of the document can be done as WG item.
>
> WG participants please indicate if you agree in adopting this document
> as a basis, or if not, what the short comings you would like to see
> addressed prior to adoption. Once more, the document will be continued
> to be developed under the WG control and this is far from any final
> version.
>
> Please provide your view no later than Friday the 24th.
>
> Best Regards
>
> Magnus Westerlund
> WG chair
>
> ----------------------------------------------------------------------
> 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
>


From harald@alvestrand.no  Mon Jun 20 01:30:35 2011
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Subject: [rtcweb] Changing video size (Re:  use case:)
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On 06/17/11 21:38, Aron Rosenberg wrote:
> On Fri, Jun 17, 2011 at 11:09 AM, Cullen Jennings <fluffy@cisco.com 
> <mailto:fluffy@cisco.com>> wrote:
>
>
>     Video Size Change.
>
>     Alice and Bob are in a video call in their browsers and have
>     negotiate a high resolution video. Bob decides to change the size
>     of the windows his browser is displaying video to a small size.
>      Bob's browser should be able to regenerate the video codec
>     parameters with Alice's browser to change the resolution of video
>     Alice sends to match the smaller size.
>
>
> Changing compression due to a user change in window size is usually 
> not done since it has a bad long term effect on video quality. In 
> almost all the major video calling systems, video resolution is a 
> function of current bandwidth (which changes as the rate control 
> system detects differences) and/or constrained by the physical device, 
> but not a function of a user dragging the window size. At 
> an equivalent bitrate, it is better to compress a larger resolution 
> and display it smaller (higher PSNR), then compress a smaller 
> resolution if you are within the codec's effective bitrate for that 
> larger resolution.
Interesting info, but I don't know if it's true in all cases.

If the overall constraint on a conferencing system is bandwidth, the 
system might find the saving in average bandwidth saving of encoding 
from a base image of 140x200 for "small" windows rather than "HD" 
(720x1024?) worth considering, even though the sending of a complete 
I-frame to reinitialize the video stream might take some effort.

I like having the use case of "change recipient video window size, and 
as a result change sender bandwidth usage"; we might argue about the 
exact phrasing of the use case.

                    Harald





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    On 06/17/11 21:38, Aron Rosenberg wrote:
    <blockquote
      cite="mid:BANLkTi=g13whA3PCXKPW5Q7a2PzEDY3ESg@mail.gmail.com"
      type="cite">
      <div class="gmail_quote">On Fri, Jun 17, 2011 at 11:09 AM, Cullen
        Jennings <span dir="ltr">&lt;<a moz-do-not-send="true"
            href="mailto:fluffy@cisco.com">fluffy@cisco.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;">
          <br>
          Video Size Change.<br>
          <br>
          Alice and Bob are in a video call in their browsers and have
          negotiate a high resolution video. Bob decides to change the
          size of the windows his browser is displaying video to a small
          size. &nbsp;Bob's browser should be able to regenerate the video
          codec parameters with Alice's browser to change the resolution
          of video Alice sends to match the smaller size.<br>
          <br>
          <br>
        </blockquote>
        <div>Changing compression due to a user change in window size is
          usually not done since it has a bad long term effect on video
          quality. In almost all the major video calling systems, video
          resolution is a function of current bandwidth (which changes
          as the rate control system detects differences) and/or
          constrained by the physical device, but not a function of a
          user dragging the window size. At an&nbsp;equivalent&nbsp;bitrate, it is
          better to compress a larger resolution and display it smaller
          (higher PSNR), then compress a smaller resolution if you are
          within the codec's effective bitrate for that larger
          resolution.<br>
        </div>
      </div>
    </blockquote>
    Interesting info, but I don't know if it's true in all cases.<br>
    <br>
    If the overall constraint on a conferencing system is bandwidth, the
    system might find the saving in average bandwidth saving of encoding
    from a base image of 140x200 for "small" windows rather than "HD"
    (720x1024?) worth considering, even though the sending of a complete
    I-frame to reinitialize the video stream might take some effort.<br>
    <br>
    I like having the use case of "change recipient video window size,
    and as a result change sender bandwidth usage"; we might argue about
    the exact phrasing of the use case.<br>
    <br>
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Harald<br>
    <br>
    <br>
    <br>
    <br>
  </body>
</html>

--------------080402000208040607040408--

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Subject: Re: [rtcweb] WG adoption for Use Case document
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+1

On 6/20/11 11:20 AM, Harald Alvestrand wrote:
> I agree with adopting the document as a WG starting point.
>
> Naturally, I expect multiple use cases to be added (and perhaps even
> some use cases marked up with "RTCWEB can't do this") before we're finished.
>
>               Harald
>
>
> On 06/17/11 10:00, Magnus Westerlund wrote:
>> WG,
>>
>> We would like to adopt a WG document with the purpose for describing use
>> cases for the RTCWEB work. This document would be targeted as
>> publication as an Inforamtional RFC eventually.
>>
>> The candidate document for this is:
>> Web Real-Time Communication Use-cases and Requirements
>> https://datatracker.ietf.org/doc/draft-holmberg-rtcweb-ucreqs/
>>
>> Based on our interim last week it was clear that there are some use
>> cases that clearly needs development. However, the WG chairs believe
>> that future development of the document can be done as WG item.
>>
>> WG participants please indicate if you agree in adopting this document
>> as a basis, or if not, what the short comings you would like to see
>> addressed prior to adoption. Once more, the document will be continued
>> to be developed under the WG control and this is far from any final
>> version.
>>
>> Please provide your view no later than Friday the 24th.
>>
>> Best Regards
>>
>> Magnus Westerlund
>> WG chair
>>
>> ----------------------------------------------------------------------
>> 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
>>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From stewe@stewe.org  Mon Jun 20 06:52:56 2011
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Applications requiring (or making desirable) adequate response of the video
codec to changes of rendering window sizes are a perfect scenario for the
use of spatial scalability.  One avoids the waste of sending bits for a high
resolution picture that cannot be rendered at high resolution, but has
freedom to start sending high resolution, without the delay involved in
sending an I frame, at any time.

Scalable video is also a useful tool to combat losses and to adapt
seamlessly to changes in the channel bandwidth (to combat congestion, for
example).

Stephan
  

From:  Harald Alvestrand <harald@alvestrand.no>
Date:  Mon, 20 Jun 2011 10:30:34 +0200
To:  Aron Rosenberg <arosenberg@logitech.com>
Cc:  <rtcweb@ietf.org>
Subject:  [rtcweb] Changing video size (Re:  use case:)

    
 On 06/17/11 21:38, Aron Rosenberg wrote:
>  
> On Fri, Jun 17, 2011 at 11:09 AM, Cullen Jennings <fluffy@cisco.com> wrote:
>  
>>  
>>  Video Size Change.
>>  
>>  Alice and Bob are in a video call in their browsers and have negotiate a
>> high resolution video. Bob decides to change the size of the windows his
>> browser is displaying video to a small size.  Bob's browser should be able to
>> regenerate the video codec parameters with Alice's browser to change the
>> resolution of video Alice sends to match the smaller size.
>>  
>>  
>>  
>  
> Changing compression due to a user change in window size is usually not done
> since it has a bad long term effect on video quality. In almost all the major
> video calling systems, video resolution is a function of current bandwidth
> (which changes as the rate control system detects differences) and/or
> constrained by the physical device, but not a function of a user dragging the
> window size. At an equivalent bitrate, it is better to compress a larger
> resolution and display it smaller (higher PSNR), then compress a smaller
> resolution if you are within the codec's effective bitrate for that larger
> resolution.
>  
>  
>  
 Interesting info, but I don't know if it's true in all cases.
 
 If the overall constraint on a conferencing system is bandwidth, the system
might find the saving in average bandwidth saving of encoding from a base
image of 140x200 for "small" windows rather than "HD" (720x1024?) worth
considering, even though the sending of a complete I-frame to reinitialize
the video stream might take some effort.
 
 I like having the use case of "change recipient video window size, and as a
result change sender bandwidth usage"; we might argue about the exact
phrasing of the use case.
 
                    Harald
 
 
 
 
 
_______________________________________________ rtcweb mailing list
rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb


--B_3391397569_1056160
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<html><head></head><body style=3D"word-wrap: break-word; -webkit-nbsp-mode: s=
pace; -webkit-line-break: after-white-space; color: rgb(0, 0, 0); font-size:=
 14px; font-family: Calibri, sans-serif; "><div>Applications requiring (or m=
aking desirable) adequate response of the video codec to changes of renderin=
g window sizes are a perfect scenario for the use of spatial scalability. &n=
bsp;One avoids the waste of sending bits for a high resolution picture that =
cannot be rendered at high resolution, but has freedom to start sending high=
 resolution, without the delay involved in sending an I frame, at any time. =
&nbsp;</div><div><br></div><div>Scalable video is also a useful tool to comb=
at losses and to adapt seamlessly to changes in the channel bandwidth (to co=
mbat congestion, for example).</div><div><br></div><div>Stephan</div><div>&n=
bsp;&nbsp;</div><div><br></div><span id=3D"OLK_SRC_BODY_SECTION"><div style=3D"f=
ont-family:Calibri; font-size:11pt; text-align:left; color:black; BORDER-BOT=
TOM: medium none; BORDER-LEFT: medium none; PADDING-BOTTOM: 0in; PADDING-LEF=
T: 0in; PADDING-RIGHT: 0in; BORDER-TOP: #b5c4df 1pt solid; BORDER-RIGHT: med=
ium none; PADDING-TOP: 3pt"><span style=3D"font-weight:bold">From: </span> Har=
ald Alvestrand &lt;<a href=3D"mailto:harald@alvestrand.no">harald@alvestrand.n=
o</a>&gt;<br><span style=3D"font-weight:bold">Date: </span> Mon, 20 Jun 2011 1=
0:30:34 +0200<br><span style=3D"font-weight:bold">To: </span> Aron Rosenberg &=
lt;<a href=3D"mailto:arosenberg@logitech.com">arosenberg@logitech.com</a>&gt;<=
br><span style=3D"font-weight:bold">Cc: </span> &lt;<a href=3D"mailto:rtcweb@iet=
f.org">rtcweb@ietf.org</a>&gt;<br><span style=3D"font-weight:bold">Subject: </=
span> [rtcweb] Changing video size (Re:  use case:)<br></div><div><br></div>=
<div>
  
    
  
  <div bgcolor=3D"#ffffff" text=3D"#000000">
    On 06/17/11 21:38, Aron Rosenberg wrote:
    <blockquote cite=3D"mid:BANLkTi=3Dg13whA3PCXKPW5Q7a2PzEDY3ESg@mail.gmail.co=
m" type=3D"cite">
      <div class=3D"gmail_quote">On Fri, Jun 17, 2011 at 11:09 AM, Cullen
        Jennings <span dir=3D"ltr">&lt;<a moz-do-not-send=3D"true" href=3D"mailto=
:fluffy@cisco.com">fluffy@cisco.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;">
          <br>
          Video Size Change.<br>
          <br>
          Alice and Bob are in a video call in their browsers and have
          negotiate a high resolution video. Bob decides to change the
          size of the windows his browser is displaying video to a small
          size. &nbsp;Bob's browser should be able to regenerate the video
          codec parameters with Alice's browser to change the resolution
          of video Alice sends to match the smaller size.<br>
          <br>
          <br>
        </blockquote>
        <div>Changing compression due to a user change in window size is
          usually not done since it has a bad long term effect on video
          quality. In almost all the major video calling systems, video
          resolution is a function of current bandwidth (which changes
          as the rate control system detects differences) and/or
          constrained by the physical device, but not a function of a
          user dragging the window size. At an&nbsp;equivalent&nbsp;bitrate=
, it is
          better to compress a larger resolution and display it smaller
          (higher PSNR), then compress a smaller resolution if you are
          within the codec's effective bitrate for that larger
          resolution.<br>
        </div>
      </div>
    </blockquote>
    Interesting info, but I don't know if it's true in all cases.<br>
    <br>
    If the overall constraint on a conferencing system is bandwidth, the
    system might find the saving in average bandwidth saving of encoding
    from a base image of 140x200 for "small" windows rather than "HD"
    (720x1024?) worth considering, even though the sending of a complete
    I-frame to reinitialize the video stream might take some effort.<br>
    <br>
    I like having the use case of "change recipient video window size,
    and as a result change sender bandwidth usage"; we might argue about
    the exact phrasing of the use case.<br>
    <br>
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Harald<br>
    <br>
    <br>
    <br>
    <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>

--B_3391397569_1056160--



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Subject: Re: [rtcweb] Changing video size (Re:  use case:)
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On 6/20/2011 4:30 AM, Harald Alvestrand wrote:
> On 06/17/11 21:38, Aron Rosenberg wrote:
>> On Fri, Jun 17, 2011 at 11:09 AM, Cullen Jennings <fluffy@cisco.com 
>> <mailto:fluffy@cisco.com>> wrote:
>>
>>
>>     Video Size Change.
>>
>>     Alice and Bob are in a video call in their browsers and have
>>     negotiate a high resolution video. Bob decides to change the size
>>     of the windows his browser is displaying video to a small size.
>>      Bob's browser should be able to regenerate the video codec
>>     parameters with Alice's browser to change the resolution of video
>>     Alice sends to match the smaller size.
>>
>>
>> Changing compression due to a user change in window size is usually 
>> not done since it has a bad long term effect on video quality. In 
>> almost all the major video calling systems, video resolution is a 
>> function of current bandwidth (which changes as the rate control 
>> system detects differences) and/or constrained by the physical 
>> device, but not a function of a user dragging the window size. At 
>> an equivalent bitrate, it is better to compress a larger resolution 
>> and display it smaller (higher PSNR), then compress a smaller 
>> resolution if you are within the codec's effective bitrate for that 
>> larger resolution.

Don't use "most current systems" as a guide.  Just because they don't do 
it doesn't mean it's a bad idea (it might be, but more proof than their 
not doing it is needed).

I would assert that in general, it's better to scale before encoding 
than to scale after decoding; and one scale operation is generally 
better than two (at camera->encoder, and decoder->display), given a 
constant bitrate.

> Interesting info, but I don't know if it's true in all cases.
>
> If the overall constraint on a conferencing system is bandwidth, the 
> system might find the saving in average bandwidth saving of encoding 
> from a base image of 140x200 for "small" windows rather than "HD" 
> (720x1024?) worth considering, even though the sending of a complete 
> I-frame to reinitialize the video stream might take some effort.
>
> I like having the use case of "change recipient video window size, and 
> as a result change sender bandwidth usage"; we might argue about the 
> exact phrasing of the use case.
>

The more general issue here is one of communication: events such as 
window resize are of interest to the sender, and may cause the sender to 
change what they're sending.    Events could include bandwidth changes, 
decode resolution changes, decode framerate changes  (probably not 
common, but perhaps for managing battery use), exposure (decode partly- 
or fully-obscured, or perhaps the user has zoomed in on one portion of 
the video), etc.  How the encoder reacts to that is largely up to the 
encoder/app (and input from the decoder app on what it wants); 
transmission of the information to allow decision-making is what I'm 
concerned with.

Some of these can/should be spoken to in use-cases.

-- 
Randell Jesup
randell-ietf@jesup.org


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From: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>
To: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Mon, 20 Jun 2011 19:11:02 +0200
Thread-Topic: [rtcweb] WG adoption for Use Case document
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I also agree with adopting this document and will contribute to the develop=
ment of the use cases.

Regards
Andy

> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Harald Alvestrand
> Sent: 20 June 2011 09:21
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] WG adoption for Use Case document
>=20
> I agree with adopting the document as a WG starting point.
>=20
> Naturally, I expect multiple use cases to be added (and perhaps even
> some use cases marked up with "RTCWEB can't do this") before we're
> finished.
>=20
>              Harald
>=20
>=20
> On 06/17/11 10:00, Magnus Westerlund wrote:
> > WG,
> >
> > We would like to adopt a WG document with the purpose for describing
> use
> > cases for the RTCWEB work. This document would be targeted as
> > publication as an Inforamtional RFC eventually.
> >
> > The candidate document for this is:
> > Web Real-Time Communication Use-cases and Requirements
> > https://datatracker.ietf.org/doc/draft-holmberg-rtcweb-ucreqs/
> >
> > Based on our interim last week it was clear that there are some use
> > cases that clearly needs development. However, the WG chairs believe
> > that future development of the document can be done as WG item.
> >
> > WG participants please indicate if you agree in adopting this
> document
> > as a basis, or if not, what the short comings you would like to see
> > addressed prior to adoption. Once more, the document will be
> continued
> > to be developed under the WG control and this is far from any final
> > version.
> >
> > Please provide your view no later than Friday the 24th.
> >
> > Best Regards
> >
> > Magnus Westerlund
> > WG chair
> >
> > ---------------------------------------------------------------------
> -
> > 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
> >
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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Date: Tue, 21 Jun 2011 09:45:43 +0200
From: Varun Singh <vsingh.ietf@gmail.com>
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Subject: Re: [rtcweb] Changing video size (Re: use case:)
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In case of RTP, RTCP carries network level reception statistics and video/voice metrics (RTCP XR etc). However, there is a no standardized congestion control for RTP (http://tools.ietf.org/html/draft-gharai-avtcore-rtp-tfrc-00 uses packet loss and delay not some of the other cues mentioned in the mail).

Unless we identify a congestion control, I would suggest that the use-case capture that we need an extensible mechanism to carry this type of information and then the sender implements its own congestion control (best to its ability).

-- 
Varun Singh

On Monday, June 20, 2011 at 4:11 PM, Randell Jesup wrote:

> On 6/20/2011 4:30 AM, Harald Alvestrand wrote:
> > On 06/17/11 21:38, Aron Rosenberg wrote:
> > > On Fri, Jun 17, 2011 at 11:09 AM, Cullen Jennings <fluffy@cisco.com (mailto:fluffy@cisco.com) 
> > > <mailto:fluffy@cisco.com>> wrote:
> > > 
> > > 
> > >  Video Size Change.
> > > 
> > >  Alice and Bob are in a video call in their browsers and have
> > >  negotiate a high resolution video. Bob decides to change the size
> > >  of the windows his browser is displaying video to a small size.
> > >  Bob's browser should be able to regenerate the video codec
> > >  parameters with Alice's browser to change the resolution of video
> > >  Alice sends to match the smaller size.
> > > 
> > > 
> > > Changing compression due to a user change in window size is usually 
> > > not done since it has a bad long term effect on video quality. In 
> > > almost all the major video calling systems, video resolution is a 
> > > function of current bandwidth (which changes as the rate control 
> > > system detects differences) and/or constrained by the physical 
> > > device, but not a function of a user dragging the window size. At 
> > > an equivalent bitrate, it is better to compress a larger resolution 
> > > and display it smaller (higher PSNR), then compress a smaller 
> > > resolution if you are within the codec's effective bitrate for that 
> > > larger resolution.
> 
> Don't use "most current systems" as a guide. Just because they don't do 
> it doesn't mean it's a bad idea (it might be, but more proof than their 
> not doing it is needed).
> 
> I would assert that in general, it's better to scale before encoding 
> than to scale after decoding; and one scale operation is generally 
> better than two (at camera->encoder, and decoder->display), given a 
> constant bitrate.
> 
> > Interesting info, but I don't know if it's true in all cases.
> > 
> > If the overall constraint on a conferencing system is bandwidth, the 
> > system might find the saving in average bandwidth saving of encoding 
> > from a base image of 140x200 for "small" windows rather than "HD" 
> > (720x1024?) worth considering, even though the sending of a complete 
> > I-frame to reinitialize the video stream might take some effort.
> > 
> > I like having the use case of "change recipient video window size, and 
> > as a result change sender bandwidth usage"; we might argue about the 
> > exact phrasing of the use case.
> 
> The more general issue here is one of communication: events such as 
> window resize are of interest to the sender, and may cause the sender to 
> change what they're sending. Events could include bandwidth changes, 
> decode resolution changes, decode framerate changes (probably not 
> common, but perhaps for managing battery use), exposure (decode partly- 
> or fully-obscured, or perhaps the user has zoomed in on one portion of 
> the video), etc. How the encoder reacts to that is largely up to the 
> encoder/app (and input from the decoder app on what it wants); 
> transmission of the information to allow decision-making is what I'm 
> concerned with.
> 
> Some of these can/should be spoken to in use-cases.
> 
> -- 
> Randell Jesup
> randell-ietf@jesup.org (mailto:randell-ietf@jesup.org)
> 
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org (mailto:rtcweb@ietf.org)
> https://www.ietf.org/mailman/listinfo/rtcweb



From stefan.lk.hakansson@ericsson.com  Tue Jun 21 02:48:57 2011
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Subject: Re: [rtcweb] Changing video size (Re:  use case:)
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I just want to highlight that changing video display window size is 
captured already in the very first use case ("Simple Video Communication 
Service") in 
http://datatracker.ietf.org/doc/draft-holmberg-rtcweb-ucreqs/?include_text=1. 


In the preparation of the document we also discussed putting a 
requirement on that the receiving end should explicitly signal the size 
or the required codec parameters to the sender but concluded at that 
time that more discussion is needed.

My personal view is that this kind of functionality should be there. 
After all the suitable codec settings depend a lot on e.g. the display 
size. But I'm not sure that a requirement is needed. There must be a 
signaling channel of some kind available to set up streams (to negotiate 
codecs and codec settings). I think it can be re-used for this purpose.

Setting it up this way there could initially be simpler but standard 
compliant implementations that does not support this, with support added 
later - with the web application being unaffected.

//Stefan
On 2011-06-20 16:11, Randell Jesup wrote:
> On 6/20/2011 4:30 AM, Harald Alvestrand wrote:
>> On 06/17/11 21:38, Aron Rosenberg wrote:
>>> On Fri, Jun 17, 2011 at 11:09 AM, Cullen Jennings<fluffy@cisco.com
>>> <mailto:fluffy@cisco.com>>  wrote:
>>>
>>>
>>>      Video Size Change.
>>>
>>>      Alice and Bob are in a video call in their browsers and have
>>>      negotiate a high resolution video. Bob decides to change the size
>>>      of the windows his browser is displaying video to a small size.
>>>       Bob's browser should be able to regenerate the video codec
>>>      parameters with Alice's browser to change the resolution of video
>>>      Alice sends to match the smaller size.
>>>
>>>
>>> Changing compression due to a user change in window size is usually
>>> not done since it has a bad long term effect on video quality. In
>>> almost all the major video calling systems, video resolution is a
>>> function of current bandwidth (which changes as the rate control
>>> system detects differences) and/or constrained by the physical
>>> device, but not a function of a user dragging the window size. At
>>> an equivalent bitrate, it is better to compress a larger resolution
>>> and display it smaller (higher PSNR), then compress a smaller
>>> resolution if you are within the codec's effective bitrate for that
>>> larger resolution.
>
> Don't use "most current systems" as a guide.  Just because they don't do
> it doesn't mean it's a bad idea (it might be, but more proof than their
> not doing it is needed).
>
> I would assert that in general, it's better to scale before encoding
> than to scale after decoding; and one scale operation is generally
> better than two (at camera->encoder, and decoder->display), given a
> constant bitrate.
>
>> Interesting info, but I don't know if it's true in all cases.
>>
>> If the overall constraint on a conferencing system is bandwidth, the
>> system might find the saving in average bandwidth saving of encoding
>> from a base image of 140x200 for "small" windows rather than "HD"
>> (720x1024?) worth considering, even though the sending of a complete
>> I-frame to reinitialize the video stream might take some effort.
>>
>> I like having the use case of "change recipient video window size, and
>> as a result change sender bandwidth usage"; we might argue about the
>> exact phrasing of the use case.
>>
>
> The more general issue here is one of communication: events such as
> window resize are of interest to the sender, and may cause the sender to
> change what they're sending.    Events could include bandwidth changes,
> decode resolution changes, decode framerate changes  (probably not
> common, but perhaps for managing battery use), exposure (decode partly-
> or fully-obscured, or perhaps the user has zoomed in on one portion of
> the video), etc.  How the encoder reacts to that is largely up to the
> encoder/app (and input from the decoder app on what it wants);
> transmission of the information to allow decision-making is what I'm
> concerned with.
>
> Some of these can/should be spoken to in use-cases.
>


From christer.holmberg@ericsson.com  Tue Jun 21 05:39:43 2011
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Tue, 21 Jun 2011 14:39:38 +0200
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Hi,

It's nice to see the amount of use-cases people have submitted :)


Now, since many of the use-cases are related to each other, and in order to=
 get a better and easy-to-read structure of the document, I would like to s=
uggest to devide the use-cases into different "use-case groups".

Below is a proposed grouping, and examples of use-cases which would be list=
ed in each group. I have only used the existing use-cases, and the ones sug=
gested by Cullen, but I am aware that more use-cases have been proposed :)


Telephony use-cases:

    - Telephony terminal use-case (draft)
    - Call fedex use-case (Cullen)


Video conferenceing use-cases:

    - With central server (draft)
    - Without central server (draft)
    - Hockey (draft)
    - Screen re-size (Cullen)


Embedded voice communicatoin use-case:

   - Multiparty on-line game with voice communication (draft)


Bandwidth/QoS/mobility use-cases:

   - NIC Change (Cullen)
   - QoS Marking (Cullen)



Later, we can of course modify/add/remove groups, but I think the grouping =
above would be a good start, and I think all the use-cases so far proposed =
would fit into one of those groups.

Comments?

Regards,

Christer


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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
<meta name=3D"Generator" content=3D"Microsoft Exchange Server">
<!-- converted from rtf -->
<style><!-- .EmailQuote { margin-left: 1pt; padding-left: 4pt; border-left:=
 #800000 2px solid; } --></style>
</head>
<body>
<font face=3D"Arial, sans-serif" size=3D"3">
<div>&nbsp;</div>
<div><font size=3D"2">Hi,</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">It's nice to see the amount of use-cases people have =
submitted :)</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">Now, since many of the use-cases are related to each =
other, and in order to get a better and easy-to-read structure of the docum=
ent, I would like to suggest to devide the use-cases into different &quot;u=
se-case groups&quot;.</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">Below is a proposed grouping, and examples of use-cas=
es which would be listed in each group. I have only used the existing use-c=
ases, and the ones suggested by Cullen, but I am aware that more use-cases =
have been proposed :)</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">Telephony use-cases:</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">&nbsp;&nbsp;&nbsp; - Telephony terminal use-case (dra=
ft)</font></div>
<div><font size=3D"2">&nbsp;&nbsp;&nbsp; - Call fedex use-case (Cullen)</fo=
nt></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">Video conferenceing use-cases:</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">&nbsp;&nbsp;&nbsp; - With central server (draft)</fon=
t></div>
<div><font size=3D"2">&nbsp;&nbsp;&nbsp; - Without central server (draft)</=
font></div>
<div><font size=3D"2">&nbsp;&nbsp;&nbsp; - Hockey (draft)</font></div>
<div><font size=3D"2">&nbsp;&nbsp;&nbsp; - Screen re-size (Cullen)</font></=
div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">Embedded voice communicatoin use-case:</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">&nbsp;&nbsp; - Multiparty on-line game with voice com=
munication (draft)</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">Bandwidth/QoS/mobility use-cases:</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">&nbsp;&nbsp; - NIC Change (Cullen)</font></div>
<div><font size=3D"2">&nbsp;&nbsp; - QoS Marking (Cullen)</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">Later, we can of course modify/add/remove groups, but=
 I think the grouping above would be a good start, and I think all the use-=
cases so far proposed would fit into one of those groups.</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">Comments?</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">Regards,</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">Christer</font></div>
<div><font size=3D"2">&nbsp;</font></div>
</font>
</body>
</html>

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From fd@w3.org  Tue Jun 21 06:57:51 2011
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Subject: [rtcweb] W3C Web Real-Time Communications face-to-face on 23 July 2011 in Quebec City, Canada
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Dear participants in the IETF RTCWEB group,

The W3C Web Real-Time Communications working group will meet on Saturday 23 July 2011 afternoon, starting at or shortly after 1PM, in Quebec City, Canada, co-located with the IETF Meeting. Exact time range, agenda and location will be announced once known.

The choice of the date and place is meant to favor synergies between the W3C WebRTC and the IETF RTCWEB groups. Participants in the IETF RTCWEB group are welcome to attend this W3C face-to-face as meeting guests.

To facilitate logistics purpose, I have set up a registration form, available at:
  http://www.w3.org/2002/09/wbs/47318/webrtc-f2f-quebec/

Anyone can fill out the form.
People willing to attend the F2F need to register before 15 July 2011 (the sooner, the better).

Please let me know if you have questions.

Cheers,
Francois Daoust,
W3C Team Contact for the W3C WebRTC WG.

From magnus.westerlund@ericsson.com  Tue Jun 21 08:22:09 2011
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Subject: [rtcweb] WG adoption of Architecture and Overview of RTCWEB document
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WG,

We would like to adopt a document that describes the RTCWEB architecture
and provides an overview to the functionality that will be specified in
various documents targeted to specific pieces of the functionalities we
need for a complete solution.

There has been several documents that describes architecture, frame
works or overviews. None of them are complete when it comes to covering
the different aspects. The documents are:

https://datatracker.ietf.org/doc/draft-alvestrand-rtcweb-overview/
https://datatracker.ietf.org/doc/draft-rosenberg-rtcweb-framework/
https://datatracker.ietf.org/doc/draft-jennings-rtcweb-api/

The WG chairs' proposal is that we adopt one document and then add in
aspects that are needed from the other documents. In the process putting
together an good editor team for the WG document from the persons that
has contributed so far and want to continue.

The WG chairs propose that that the WG adopts:
https://datatracker.ietf.org/doc/draft-alvestrand-rtcweb-overview/
as a WG item to form the core of an Architecture and Overview document.

WG participants please provide either support for adoption or provide
feedback on any issues you see with this approach.

Please provide your view no later than Tuesday the 28th at 17.00 CEST.

Magnus Westerlund
WG Chair

----------------------------------------------------------------------
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 harald@alvestrand.no  Tue Jun 21 08:26:41 2011
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On 06/21/11 09:45, Varun Singh wrote:
> In case of RTP, RTCP carries network level reception statistics and video/voice metrics (RTCP XR etc). However, there is a no standardized congestion control for RTP (http://tools.ietf.org/html/draft-gharai-avtcore-rtp-tfrc-00 uses packet loss and delay not some of the other cues mentioned in the mail).
>
> Unless we identify a congestion control, I would suggest that the use-case capture that we need an extensible mechanism to carry this type of information and then the sender implements its own congestion control (best to its ability).
>
I think we need to separate the two requirements:

- response to network conditions (congestion control)
- response to recipient system conditions such as window size, CPU load, 
window being obscured, and so on.

Even though the possible actions to take might be the same, the two are 
fundamentally different; responses to end system events concern only the 
sender and the recipient, while responses to congestion events have the 
requirement that the behavior be globally sensible - that is, works well 
with the algorithms deployed by other users of the network.

                          Harald


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Subject: Re: [rtcweb] Use-case grouping
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Seems to me sensible to have groupings.

The use case I don't see a place for at the moment is the single 
point-to-point call; the proposed use case I don't see mentioned below 
is my suggestion for "interoperable calling".

So this might require more tuning.

On 06/21/11 14:39, Christer Holmberg wrote:
> Hi,
> It's nice to see the amount of use-cases people have submitted :)
> Now, since many of the use-cases are related to each other, and in 
> order to get a better and easy-to-read structure of the document, I 
> would like to suggest to devide the use-cases into different "use-case 
> groups".
> Below is a proposed grouping, and examples of use-cases which would be 
> listed in each group. I have only used the existing use-cases, and the 
> ones suggested by Cullen, but I am aware that more use-cases have been 
> proposed :)
> Telephony use-cases:
>     - Telephony terminal use-case (draft)
>     - Call fedex use-case (Cullen)
> Video conferenceing use-cases:
>     - With central server (draft)
>     - Without central server (draft)
>     - Hockey (draft)
>     - Screen re-size (Cullen)
> Embedded voice communicatoin use-case:
>    - Multiparty on-line game with voice communication (draft)
> Bandwidth/QoS/mobility use-cases:
>    - NIC Change (Cullen)
>    - QoS Marking (Cullen)
> Later, we can of course modify/add/remove groups, but I think the 
> grouping above would be a good start, and I think all the use-cases so 
> far proposed would fit into one of those groups.
> Comments?
> Regards,
> Christer
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


--------------070201030908020204060002
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    <meta content="text/html; charset=ISO-8859-1"
      http-equiv="Content-Type">
  </head>
  <body bgcolor="#ffffff" text="#000000">
    Seems to me sensible to have groupings.<br>
    <br>
    The use case I don't see a place for at the moment is the single
    point-to-point call; the proposed use case I don't see mentioned
    below is my suggestion for "interoperable calling".<br>
    <br>
    So this might require more tuning.<br>
    <br>
    On 06/21/11 14:39, Christer Holmberg wrote:
    <blockquote
cite="mid:7F2072F1E0DE894DA4B517B93C6A05851DB5336E53@ESESSCMS0356.eemea.ericsson.se"
      type="cite">
      <meta http-equiv="Content-Type" content="text/html;
        charset=ISO-8859-1">
      <meta name="Generator" content="Microsoft Exchange Server">
      <!-- converted from rtf -->
      <style><!-- .EmailQuote { margin-left: 1pt; padding-left: 4pt; border-left: #800000 2px solid; } --></style>
      <font face="Arial, sans-serif" size="3">
        <div>&nbsp;</div>
        <div><font size="2">Hi,</font></div>
        <div><font size="2">&nbsp;</font></div>
        <div><font size="2">It's nice to see the amount of use-cases
            people have submitted :)</font></div>
        <div><font size="2">&nbsp;</font></div>
        <div><font size="2">&nbsp;</font></div>
        <div><font size="2">Now, since many of the use-cases are related
            to each other, and in order to get a better and easy-to-read
            structure of the document, I would like to suggest to devide
            the use-cases into different "use-case groups".</font></div>
        <div><font size="2">&nbsp;</font></div>
        <div><font size="2">Below is a proposed grouping, and examples
            of use-cases which would be listed in each group. I have
            only used the existing use-cases, and the ones suggested by
            Cullen, but I am aware that more use-cases have been
            proposed :)</font></div>
        <div><font size="2">&nbsp;</font></div>
        <div><font size="2">&nbsp;</font></div>
        <div><font size="2">Telephony use-cases:</font></div>
        <div><font size="2">&nbsp;</font></div>
        <div><font size="2">&nbsp;&nbsp;&nbsp; - Telephony terminal use-case (draft)</font></div>
        <div><font size="2">&nbsp;&nbsp;&nbsp; - Call fedex use-case (Cullen)</font></div>
        <div><font size="2">&nbsp;</font></div>
        <div><font size="2">&nbsp;</font></div>
        <div><font size="2">Video conferenceing use-cases:</font></div>
        <div><font size="2">&nbsp;</font></div>
        <div><font size="2">&nbsp;&nbsp;&nbsp; - With central server (draft)</font></div>
        <div><font size="2">&nbsp;&nbsp;&nbsp; - Without central server (draft)</font></div>
        <div><font size="2">&nbsp;&nbsp;&nbsp; - Hockey (draft)</font></div>
        <div><font size="2">&nbsp;&nbsp;&nbsp; - Screen re-size (Cullen)</font></div>
        <div><font size="2">&nbsp;</font></div>
        <div><font size="2">&nbsp;</font></div>
        <div><font size="2">Embedded voice communicatoin use-case:</font></div>
        <div><font size="2">&nbsp;</font></div>
        <div><font size="2">&nbsp;&nbsp; - Multiparty on-line game with voice
            communication (draft)</font></div>
        <div><font size="2">&nbsp;</font></div>
        <div><font size="2">&nbsp;</font></div>
        <div><font size="2">Bandwidth/QoS/mobility use-cases:</font></div>
        <div><font size="2">&nbsp;</font></div>
        <div><font size="2">&nbsp;&nbsp; - NIC Change (Cullen)</font></div>
        <div><font size="2">&nbsp;&nbsp; - QoS Marking (Cullen)</font></div>
        <div><font size="2">&nbsp;</font></div>
        <div><font size="2">&nbsp;</font></div>
        <div><font size="2">&nbsp;</font></div>
        <div><font size="2">Later, we can of course modify/add/remove
            groups, but I think the grouping above would be a good
            start, and I think all the use-cases so far proposed would
            fit into one of those groups.</font></div>
        <div><font size="2">&nbsp;</font></div>
        <div><font size="2">Comments?</font></div>
        <div><font size="2">&nbsp;</font></div>
        <div><font size="2">Regards,</font></div>
        <div><font size="2">&nbsp;</font></div>
        <div><font size="2">Christer</font></div>
        <div><font size="2">&nbsp;</font></div>
      </font>
      <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>

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+1
On 2011-06-21 17:26, Harald Alvestrand wrote:
> On 06/21/11 09:45, Varun Singh wrote:
>> In case of RTP, RTCP carries network level reception statistics and video/voice metrics (RTCP XR etc). However, there is a no standardized congestion control for RTP (http://tools.ietf.org/html/draft-gharai-avtcore-rtp-tfrc-00 uses packet loss and delay not some of the other cues mentioned in the mail).
>>
>> Unless we identify a congestion control, I would suggest that the use-case capture that we need an extensible mechanism to carry this type of information and then the sender implements its own congestion control (best to its ability).
>>
> I think we need to separate the two requirements:
>
> - response to network conditions (congestion control)
> - response to recipient system conditions such as window size, CPU load,
> window being obscured, and so on.
>
> Even though the possible actions to take might be the same, the two are
> fundamentally different; responses to end system events concern only the
> sender and the recipient, while responses to congestion events have the
> requirement that the behavior be globally sensible - that is, works well
> with the algorithms deployed by other users of the network.
>
>                            Harald
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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Yes, I like this grouping.

On Jun 21, 2011, at 5:39 , Christer Holmberg wrote:

> =20
> Hi,
> =20
> It's nice to see the amount of use-cases people have submitted :)
> =20
> =20
> Now, since many of the use-cases are related to each other, and in =
order to get a better and easy-to-read structure of the document, I =
would like to suggest to devide the use-cases into different "use-case =
groups".
> =20
> Below is a proposed grouping, and examples of use-cases which would be =
listed in each group. I have only used the existing use-cases, and the =
ones suggested by Cullen, but I am aware that more use-cases have been =
proposed :)
> =20
> =20
> Telephony use-cases:
> =20
>     - Telephony terminal use-case (draft)
>     - Call fedex use-case (Cullen)
> =20
> =20
> Video conferenceing use-cases:
> =20
>     - With central server (draft)
>     - Without central server (draft)
>     - Hockey (draft)
>     - Screen re-size (Cullen)
> =20
> =20
> Embedded voice communicatoin use-case:
> =20
>    - Multiparty on-line game with voice communication (draft)
> =20
> =20
> Bandwidth/QoS/mobility use-cases:
> =20
>    - NIC Change (Cullen)
>    - QoS Marking (Cullen)
> =20
> =20
> =20
> Later, we can of course modify/add/remove groups, but I think the =
grouping above would be a good start, and I think all the use-cases so =
far proposed would fit into one of those groups.
> =20
> Comments?
> =20
> Regards,
> =20
> Christer
> =20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


Cullen Jennings
For corporate legal information go to:
http://www.cisco.com/web/about/doing_business/legal/cri/index.html



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Magnus' proposal of picking Harald's draft is fine with me.
Stephan


On 6.21.2011 08:22 , "Magnus Westerlund" <magnus.westerlund@ericsson.com>
wrote:

>WG,
>
>We would like to adopt a document that describes the RTCWEB architecture
>and provides an overview to the functionality that will be specified in
>various documents targeted to specific pieces of the functionalities we
>need for a complete solution.
>
>There has been several documents that describes architecture, frame
>works or overviews. None of them are complete when it comes to covering
>the different aspects. The documents are:
>
>https://datatracker.ietf.org/doc/draft-alvestrand-rtcweb-overview/
>https://datatracker.ietf.org/doc/draft-rosenberg-rtcweb-framework/
>https://datatracker.ietf.org/doc/draft-jennings-rtcweb-api/
>
>The WG chairs' proposal is that we adopt one document and then add in
>aspects that are needed from the other documents. In the process putting
>together an good editor team for the WG document from the persons that
>has contributed so far and want to continue.
>
>The WG chairs propose that that the WG adopts:
>https://datatracker.ietf.org/doc/draft-alvestrand-rtcweb-overview/
>as a WG item to form the core of an Architecture and Overview document.
>
>WG participants please provide either support for adoption or provide
>feedback on any issues you see with this approach.
>
>Please provide your view no later than Tuesday the 28th at 17.00 CEST.
>
>Magnus Westerlund
>WG Chair
>
>----------------------------------------------------------------------
>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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Subject: Re: [rtcweb] Changing video size (Re:  use case:)
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On Jun 20, 2011, at 1:30 , Harald Alvestrand wrote:

> On 06/17/11 21:38, Aron Rosenberg wrote:
>> On Fri, Jun 17, 2011 at 11:09 AM, Cullen Jennings <fluffy@cisco.com> =
wrote:
>>=20
>> Video Size Change.
>>=20
>> Alice and Bob are in a video call in their browsers and have =
negotiate a high resolution video. Bob decides to change the size of the =
windows his browser is displaying video to a small size.  Bob's browser =
should be able to regenerate the video codec parameters with Alice's =
browser to change the resolution of video Alice sends to match the =
smaller size.
>>=20
>>=20
>> Changing compression due to a user change in window size is usually =
not done since it has a bad long term effect on video quality. In almost =
all the major video calling systems, video resolution is a function of =
current bandwidth (which changes as the rate control system detects =
differences) and/or constrained by the physical device, but not a =
function of a user dragging the window size. At an equivalent bitrate, =
it is better to compress a larger resolution and display it smaller =
(higher PSNR), then compress a smaller resolution if you are within the =
codec's effective bitrate for that larger resolution.

Consider two different video flows using the same codec ...

Stream A is 640 x 480 at 1mbps which is this scaled to 320x240 and =
displayed

Stream B is 320 x 240 at 1mbps which is displayed at 320x240.=20

My experience has been with modern video codecs that stream B will look =
better than stream A. As well as looking better, it will typically also =
have a better PSNR. There's a bunch of reasons why which are probably =
not worth going into here but give it a try and you will see what I =
mean. The key point is both streams where 1mbps. If stream B was sent at =
256 kbps, then A would look better.=20





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On Jun 21, 2011, at 8:35 , Stefan H=E5kansson LK wrote:

> +1
> On 2011-06-21 17:26, Harald Alvestrand wrote:
>> On 06/21/11 09:45, Varun Singh wrote:
>>> In case of RTP, RTCP carries network level reception statistics and =
video/voice metrics (RTCP XR etc). However, there is a no standardized =
congestion control for RTP =
(http://tools.ietf.org/html/draft-gharai-avtcore-rtp-tfrc-00 uses packet =
loss and delay not some of the other cues mentioned in the mail).
>>>=20
>>> Unless we identify a congestion control, I would suggest that the =
use-case capture that we need an extensible mechanism to carry this type =
of information and then the sender implements its own congestion control =
(best to its ability).
>>>=20
>> I think we need to separate the two requirements:
>>=20
>> - response to network conditions (congestion control)
>> - response to recipient system conditions such as window size, CPU =
load,
>> window being obscured, and so on.
>>=20
>> Even though the possible actions to take might be the same, the two =
are
>> fundamentally different; responses to end system events concern only =
the
>> sender and the recipient, while responses to congestion events have =
the
>> requirement that the behavior be globally sensible - that is, works =
well
>> with the algorithms deployed by other users of the network.
>>=20
>>                           Harald

+1 It seems to me that what resolution video you send often is the =
largest size that is smaller than several constraints. The constraints =
include what you have bandwidth to transports, what the receiver is =
capable of displaying, and the desires of the two users involved. No one =
argues we should send 1080P video to a systems that is only capable of =
decoding 720P. I think what I was trying to get at in the use case is =
that these things depend on the desires of both the sending and =
receiving systems and they change dynamically during the session.=20




From stewe@stewe.org  Tue Jun 21 09:23:48 2011
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From: Stephan Wenger <stewe@stewe.org>
To: Cullen Jennings <fluffy@cisco.com>, Aron Rosenberg <arosenberg@logitech.com>
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What Cullen writes below matches my experience.  And, its not even limited
to "modern" codecs.  I would suggest that the observed behavior is true
for any video codec that has sub-pel motion compensation, i.e. any video
compression standard developed since H.263 (1998).
Stephan

On 6.21.2011 09:11 , "Cullen Jennings" <fluffy@cisco.com> wrote:


>[...]
>
>Consider two different video flows using the same codec ...
>
>Stream A is 640 x 480 at 1mbps which is this scaled to 320x240 and
>displayed
>
>Stream B is 320 x 240 at 1mbps which is displayed at 320x240.
>
>My experience has been with modern video codecs that stream B will look
>better than stream A. As well as looking better, it will typically also
>have a better PSNR. There's a bunch of reasons why which are probably not
>worth going into here but give it a try and you will see what I mean. The
>key point is both streams where 1mbps. If stream B was sent at 256 kbps,
>then A would look better.
>
>
>
>
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb



From christer.holmberg@ericsson.com  Tue Jun 21 10:11:41 2011
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Tue, 21 Jun 2011 19:10:44 +0200
Thread-Topic: [rtcweb] Use-case grouping
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Hi Harald,

We could add a "Point-to-point", or "Browser-to-browser", use-case group.

Regards,

Christer

________________________________
From: rtcweb-bounces@ietf.org [rtcweb-bounces@ietf.org] On Behalf Of Harald=
 Alvestrand [harald@alvestrand.no]
Sent: Tuesday, June 21, 2011 6:30 PM
To: rtcweb@ietf.org
Subject: Re: [rtcweb] Use-case grouping

Seems to me sensible to have groupings.

The use case I don't see a place for at the moment is the single point-to-p=
oint call; the proposed use case I don't see mentioned below is my suggesti=
on for "interoperable calling".

So this might require more tuning.

On 06/21/11 14:39, Christer Holmberg wrote:

Hi,

It's nice to see the amount of use-cases people have submitted :)


Now, since many of the use-cases are related to each other, and in order to=
 get a better and easy-to-read structure of the document, I would like to s=
uggest to devide the use-cases into different "use-case groups".

Below is a proposed grouping, and examples of use-cases which would be list=
ed in each group. I have only used the existing use-cases, and the ones sug=
gested by Cullen, but I am aware that more use-cases have been proposed :)


Telephony use-cases:

    - Telephony terminal use-case (draft)
    - Call fedex use-case (Cullen)


Video conferenceing use-cases:

    - With central server (draft)
    - Without central server (draft)
    - Hockey (draft)
    - Screen re-size (Cullen)


Embedded voice communicatoin use-case:

   - Multiparty on-line game with voice communication (draft)


Bandwidth/QoS/mobility use-cases:

   - NIC Change (Cullen)
   - QoS Marking (Cullen)



Later, we can of course modify/add/remove groups, but I think the grouping =
above would be a good start, and I think all the use-cases so far proposed =
would fit into one of those groups.

Comments?

Regards,

Christer



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rtcweb@ietf.org<mailto:rtcweb@ietf.org>
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From Bert.Greevenbosch@huawei.com  Tue Jun 21 20:20:00 2011
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Hi Cullen, all,

<cite>
Consider two different video flows using the same codec ...

Stream A is 640 x 480 at 1mbps which is this scaled to 320x240 and displayed

Stream B is 320 x 240 at 1mbps which is displayed at 320x240. 

My experience has been with modern video codecs that stream B will look better than stream A. As well as looking better, it will typically also have a better PSNR. There's a bunch of reasons why which are probably not worth going into here but give it a try and you will see what I mean. The key point is both streams where 1mbps. If stream B was sent at 256 kbps, then A would look better.
</cite>

Can I ask what you mean with "both streams where 1mbps"? Do you refer to the capacity of the channel (i.e. how much data the channel can transport), or to the actually codec bitrate (i.e. how much data is sent from the streaming server).

In the first case, assuming that the size of stream B is considerably less than the size of stream A, the quality of Stream B can be expected to be higher than the quality of Stream A, since a higher capacity of the channel results in less packet loss. 

In the second case, assuming both streams are coded at the same bitrate, the quality of Stream B can also be expected to be higher than the quality of Stream A, due to the availability of more bits per pixel (though similar packet loss).

However, if we reduce the capacity of the channel, it becomes more complicated. I think this highly depends on which codec is being used. For example, I am not sure whether  Stream B with 25% channel capacity would look better than Stream A (after scaling) with 100% channel capacity. I think this highly depends on the codec and scaling algorithm, and saturation of the channel.

To come to the question on whether negotiation of the video size is needed: I feel that it could be a useful feature. Even if the channel allows streaming the 640x480 version without congestion problems now, where is the guarantee that it will not happen later? Probably there will be multiple video streams (at least partially) using the same path. If congestion happens later, the codec bitrate of multiple streams may need to be reduced. How to know from which streams the bitrate needs to be reduced? It seems to be better to prevent this and use a reduced codec bitrate from the start for those streams that are known not to require a higher bitrate.

However, whether we can cater for this also depends on the actual technical solution for negotiation and which codec to use. If adding video size negotiation will not increase complexity too much, I see no reason not to do it. However, if it adds much complexity to the solution, we might need to consider whether it is worth it. Therefore, maybe it should not be a requirement, but something weaker; i.e. it could be phrased as a SHOULD rather than a MUST.

Best regards,
Bert


-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of Cullen Jennings
Sent: 22 June 2011 00:11
To: Aron Rosenberg
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Changing video size (Re: use case:)


On Jun 20, 2011, at 1:30 , Harald Alvestrand wrote:

> On 06/17/11 21:38, Aron Rosenberg wrote:
>> On Fri, Jun 17, 2011 at 11:09 AM, Cullen Jennings <fluffy@cisco.com> wrote:
>> 
>> Video Size Change.
>> 
>> Alice and Bob are in a video call in their browsers and have negotiate a high resolution video. Bob decides to change the size of the windows his browser is displaying video to a small size.  Bob's browser should be able to regenerate the video codec parameters with Alice's browser to change the resolution of video Alice sends to match the smaller size.
>> 
>> 
>> Changing compression due to a user change in window size is usually not done since it has a bad long term effect on video quality. In almost all the major video calling systems, video resolution is a function of current bandwidth (which changes as the rate control system detects differences) and/or constrained by the physical device, but not a function of a user dragging the window size. At an equivalent bitrate, it is better to compress a larger resolution and display it smaller (higher PSNR), then compress a smaller resolution if you are within the codec's effective bitrate for that larger resolution.

Consider two different video flows using the same codec ...

Stream A is 640 x 480 at 1mbps which is this scaled to 320x240 and displayed

Stream B is 320 x 240 at 1mbps which is displayed at 320x240. 

My experience has been with modern video codecs that stream B will look better than stream A. As well as looking better, it will typically also have a better PSNR. There's a bunch of reasons why which are probably not worth going into here but give it a try and you will see what I mean. The key point is both streams where 1mbps. If stream B was sent at 256 kbps, then A would look better. 




_______________________________________________
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rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

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I am fine with the proposal, but should the overview draft be informational=
, instead of standard track?

Thanks,
Kind Regards
Kepeng
-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Stephan Wenger
Sent: Wednesday, June 22, 2011 12:10 AM
To: Magnus Westerlund; rtcweb@ietf.org
Subject: Re: [rtcweb] WG adoption of Architecture and Overview of RTCWEB do=
cument

Magnus' proposal of picking Harald's draft is fine with me.
Stephan

On 6.21.2011 08:22 , "Magnus Westerlund" <magnus.westerlund@ericsson.com>
wrote:

>WG,
>
>We would like to adopt a document that describes the RTCWEB architecture
>and provides an overview to the functionality that will be specified in
>various documents targeted to specific pieces of the functionalities we
>need for a complete solution.
>
>There has been several documents that describes architecture, frame
>works or overviews. None of them are complete when it comes to covering
>the different aspects. The documents are:
>
>https://datatracker.ietf.org/doc/draft-alvestrand-rtcweb-overview/
>https://datatracker.ietf.org/doc/draft-rosenberg-rtcweb-framework/
>https://datatracker.ietf.org/doc/draft-jennings-rtcweb-api/
>
>The WG chairs' proposal is that we adopt one document and then add in
>aspects that are needed from the other documents. In the process putting
>together an good editor team for the WG document from the persons that
>has contributed so far and want to continue.
>
>The WG chairs propose that that the WG adopts:
>https://datatracker.ietf.org/doc/draft-alvestrand-rtcweb-overview/
>as a WG item to form the core of an Architecture and Overview document.
>
>WG participants please provide either support for adoption or provide
>feedback on any issues you see with this approach.
>
>Please provide your view no later than Tuesday the 28th at 17.00 CEST.
>
>Magnus Westerlund
>WG Chair
>
>----------------------------------------------------------------------
>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
>----------------------------------------------------------------------

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--0015177407d05d3b9404a644ff1f
Content-Type: text/plain; charset=UTF-8

Given the fact that this request for a video size change is an optimization
(either of resources or user experience), I think that SHOULD is probably
the strongest label we can put on it. Consider the case of a middlebox; if
it has no capacity to scale the stream, due to compute or codec
restrictions, there's not much it can do with this request.


> However, whether we can cater for this also depends on the actual technical
> solution for negotiation and which codec to use. If adding video size
> negotiation will not increase complexity too much, I see no reason not to do
> it. However, if it adds much complexity to the solution, we might need to
> consider whether it is worth it. Therefore, maybe it should not be a
> requirement, but something weaker; i.e. it could be phrased as a SHOULD
> rather than a MUST.
>
> Best regards,
> Bert
>
>
> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
> Of Cullen Jennings
> Sent: 22 June 2011 00:11
> To: Aron Rosenberg
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Changing video size (Re: use case:)
>
>
> On Jun 20, 2011, at 1:30 , Harald Alvestrand wrote:
>
> > On 06/17/11 21:38, Aron Rosenberg wrote:
> >> On Fri, Jun 17, 2011 at 11:09 AM, Cullen Jennings <fluffy@cisco.com>
> wrote:
> >>
> >> Video Size Change.
> >>
> >> Alice and Bob are in a video call in their browsers and have negotiate a
> high resolution video. Bob decides to change the size of the windows his
> browser is displaying video to a small size.  Bob's browser should be able
> to regenerate the video codec parameters with Alice's browser to change the
> resolution of video Alice sends to match the smaller size.
> >>
> >>
> >> Changing compression due to a user change in window size is usually not
> done since it has a bad long term effect on video quality. In almost all the
> major video calling systems, video resolution is a function of current
> bandwidth (which changes as the rate control system detects differences)
> and/or constrained by the physical device, but not a function of a user
> dragging the window size. At an equivalent bitrate, it is better to compress
> a larger resolution and display it smaller (higher PSNR), then compress a
> smaller resolution if you are within the codec's effective bitrate for that
> larger resolution.
>
> Consider two different video flows using the same codec ...
>
> Stream A is 640 x 480 at 1mbps which is this scaled to 320x240 and
> displayed
>
> Stream B is 320 x 240 at 1mbps which is displayed at 320x240.
>
> My experience has been with modern video codecs that stream B will look
> better than stream A. As well as looking better, it will typically also have
> a better PSNR. There's a bunch of reasons why which are probably not worth
> going into here but give it a try and you will see what I mean. The key
> point is both streams where 1mbps. If stream B was sent at 256 kbps, then A
> would look better.
>
>
>
>
> _______________________________________________
> 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
>

--0015177407d05d3b9404a644ff1f
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div class=3D"gmail_quote"><div>Given the fact that this request for a vide=
o size change is an optimization (either of resources or user experience), =
I think that SHOULD is probably the strongest label we can put on it. Consi=
der the case of a middlebox; if it has no capacity to scale the stream, due=
 to compute or codec restrictions, there&#39;s not much it can do with this=
 request.</div>

<div><br></div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex=
;border-left:1px #ccc solid;padding-left:1ex;">
<br>
However, whether we can cater for this also depends on the actual technical=
 solution for negotiation and which codec to use. If adding video size nego=
tiation will not increase complexity too much, I see no reason not to do it=
. However, if it adds much complexity to the solution, we might need to con=
sider whether it is worth it. Therefore, maybe it should not be a requireme=
nt, but something weaker; i.e. it could be phrased as a SHOULD rather than =
a MUST.<br>


<br>
Best regards,<br>
<font color=3D"#888888">Bert<br>
</font><div class=3D"im"><br>
<br>
-----Original Message-----<br>
From: <a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a=
> [mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.or=
g</a>] On Behalf Of Cullen Jennings<br>
Sent: 22 June 2011 00:11<br>
To: Aron Rosenberg<br>
Cc: <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
</div><div><div></div><div class=3D"h5">Subject: Re: [rtcweb] Changing vide=
o size (Re: use case:)<br>
<br>
<br>
On Jun 20, 2011, at 1:30 , Harald Alvestrand wrote:<br>
<br>
&gt; On 06/17/11 21:38, Aron Rosenberg wrote:<br>
&gt;&gt; On Fri, Jun 17, 2011 at 11:09 AM, Cullen Jennings &lt;<a href=3D"m=
ailto:fluffy@cisco.com">fluffy@cisco.com</a>&gt; wrote:<br>
&gt;&gt;<br>
&gt;&gt; Video Size Change.<br>
&gt;&gt;<br>
&gt;&gt; Alice and Bob are in a video call in their browsers and have negot=
iate a high resolution video. Bob decides to change the size of the windows=
 his browser is displaying video to a small size. =C2=A0Bob&#39;s browser s=
hould be able to regenerate the video codec parameters with Alice&#39;s bro=
wser to change the resolution of video Alice sends to match the smaller siz=
e.<br>


&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; Changing compression due to a user change in window size is usuall=
y not done since it has a bad long term effect on video quality. In almost =
all the major video calling systems, video resolution is a function of curr=
ent bandwidth (which changes as the rate control system detects differences=
) and/or constrained by the physical device, but not a function of a user d=
ragging the window size. At an equivalent bitrate, it is better to compress=
 a larger resolution and display it smaller (higher PSNR), then compress a =
smaller resolution if you are within the codec&#39;s effective bitrate for =
that larger resolution.<br>


<br>
Consider two different video flows using the same codec ...<br>
<br>
Stream A is 640 x 480 at 1mbps which is this scaled to 320x240 and displaye=
d<br>
<br>
Stream B is 320 x 240 at 1mbps which is displayed at 320x240.<br>
<br>
My experience has been with modern video codecs that stream B will look bet=
ter than stream A. As well as looking better, it will typically also have a=
 better PSNR. There&#39;s a bunch of reasons why which are probably not wor=
th going into here but give it a try and you will see what I mean. The key =
point is both streams where 1mbps. If stream B was sent at 256 kbps, then A=
 would look better.<br>


<br>
<br>
<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>
_______________________________________________<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>
</div></div></blockquote></div><br>

--0015177407d05d3b9404a644ff1f--

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Date: Wed, 22 Jun 2011 05:55:52 +0000
From: Bert Greevenbosch <Bert.Greevenbosch@huawei.com>
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Hi Justin, all,

Yes, that is a second issue. Even if the receiving end can signal its current display size, the streaming source might not be able to provide it. Therefore flexibility would be needed for the source on how to handle the display size information.

Best regards,
Bert


From: Justin Uberti [mailto:juberti@google.com]
Sent: 22 June 2011 11:59
To: Bert Greevenbosch
Cc: Cullen Jennings; Aron Rosenberg; rtcweb@ietf.org
Subject: Re: [rtcweb] Changing video size (Re: use case:)

Given the fact that this request for a video size change is an optimization (either of resources or user experience), I think that SHOULD is probably the strongest label we can put on it. Consider the case of a middlebox; if it has no capacity to scale the stream, due to compute or codec restrictions, there's not much it can do with this request.


However, whether we can cater for this also depends on the actual technical solution for negotiation and which codec to use. If adding video size negotiation will not increase complexity too much, I see no reason not to do it. However, if it adds much complexity to the solution, we might need to consider whether it is worth it. Therefore, maybe it should not be a requirement, but something weaker; i.e. it could be phrased as a SHOULD rather than a MUST.

Best regards,
Bert


-----Original Message-----
From: rtcweb-bounces@ietf.org<mailto:rtcweb-bounces@ietf.org> [mailto:rtcweb-bounces@ietf.org<mailto:rtcweb-bounces@ietf.org>] On Behalf Of Cullen Jennings
Sent: 22 June 2011 00:11
To: Aron Rosenberg
Cc: rtcweb@ietf.org<mailto:rtcweb@ietf.org>
Subject: Re: [rtcweb] Changing video size (Re: use case:)


On Jun 20, 2011, at 1:30 , Harald Alvestrand wrote:

> On 06/17/11 21:38, Aron Rosenberg wrote:
>> On Fri, Jun 17, 2011 at 11:09 AM, Cullen Jennings <fluffy@cisco.com<mailto:fluffy@cisco.com>> wrote:
>>
>> Video Size Change.
>>
>> Alice and Bob are in a video call in their browsers and have negotiate a high resolution video. Bob decides to change the size of the windows his browser is displaying video to a small size.  Bob's browser should be able to regenerate the video codec parameters with Alice's browser to change the resolution of video Alice sends to match the smaller size.
>>
>>
>> Changing compression due to a user change in window size is usually not done since it has a bad long term effect on video quality. In almost all the major video calling systems, video resolution is a function of current bandwidth (which changes as the rate control system detects differences) and/or constrained by the physical device, but not a function of a user dragging the window size. At an equivalent bitrate, it is better to compress a larger resolution and display it smaller (higher PSNR), then compress a smaller resolution if you are within the codec's effective bitrate for that larger resolution.

Consider two different video flows using the same codec ...

Stream A is 640 x 480 at 1mbps which is this scaled to 320x240 and displayed

Stream B is 320 x 240 at 1mbps which is displayed at 320x240.

My experience has been with modern video codecs that stream B will look better than stream A. As well as looking better, it will typically also have a better PSNR. There's a bunch of reasons why which are probably not worth going into here but give it a try and you will see what I mean. The key point is both streams where 1mbps. If stream B was sent at 256 kbps, then A would look better.




_______________________________________________
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


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From stefan.lk.hakansson@ericsson.com  Tue Jun 21 23:06:34 2011
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From: =?iso-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>
To: Bert Greevenbosch <Bert.Greevenbosch@huawei.com>, Justin Uberti <juberti@google.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Wed, 22 Jun 2011 08:06:28 +0200
Thread-Topic: [rtcweb] Changing video size (Re: use case:)
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Subject: Re: [rtcweb] Changing video size (Re: use case:)
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All,

this RFC could perhaps be useful for this purpose of signaling display size=
: http://www.rfc-editor.org/rfc/rfc6236.txt

Stefan

________________________________
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Bert Greevenbosch
Sent: den 22 juni 2011 07:56
To: Justin Uberti
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Changing video size (Re: use case:)

Hi Justin, all,

Yes, that is a second issue. Even if the receiving end can signal its curre=
nt display size, the streaming source might not be able to provide it. Ther=
efore flexibility would be needed for the source on how to handle the displ=
ay size information.

Best regards,
Bert


From: Justin Uberti [mailto:juberti@google.com]
Sent: 22 June 2011 11:59
To: Bert Greevenbosch
Cc: Cullen Jennings; Aron Rosenberg; rtcweb@ietf.org
Subject: Re: [rtcweb] Changing video size (Re: use case:)

Given the fact that this request for a video size change is an optimization=
 (either of resources or user experience), I think that SHOULD is probably =
the strongest label we can put on it. Consider the case of a middlebox; if =
it has no capacity to scale the stream, due to compute or codec restriction=
s, there's not much it can do with this request.


However, whether we can cater for this also depends on the actual technical=
 solution for negotiation and which codec to use. If adding video size nego=
tiation will not increase complexity too much, I see no reason not to do it=
. However, if it adds much complexity to the solution, we might need to con=
sider whether it is worth it. Therefore, maybe it should not be a requireme=
nt, but something weaker; i.e. it could be phrased as a SHOULD rather than =
a MUST.

Best regards,
Bert


-----Original Message-----
From: rtcweb-bounces@ietf.org<mailto:rtcweb-bounces@ietf.org> [mailto:rtcwe=
b-bounces@ietf.org<mailto:rtcweb-bounces@ietf.org>] On Behalf Of Cullen Jen=
nings
Sent: 22 June 2011 00:11
To: Aron Rosenberg
Cc: rtcweb@ietf.org<mailto:rtcweb@ietf.org>
Subject: Re: [rtcweb] Changing video size (Re: use case:)


On Jun 20, 2011, at 1:30 , Harald Alvestrand wrote:

> On 06/17/11 21:38, Aron Rosenberg wrote:
>> On Fri, Jun 17, 2011 at 11:09 AM, Cullen Jennings <fluffy@cisco.com<mail=
to:fluffy@cisco.com>> wrote:
>>
>> Video Size Change.
>>
>> Alice and Bob are in a video call in their browsers and have negotiate a=
 high resolution video. Bob decides to change the size of the windows his b=
rowser is displaying video to a small size.  Bob's browser should be able t=
o regenerate the video codec parameters with Alice's browser to change the =
resolution of video Alice sends to match the smaller size.
>>
>>
>> Changing compression due to a user change in window size is usually not =
done since it has a bad long term effect on video quality. In almost all th=
e major video calling systems, video resolution is a function of current ba=
ndwidth (which changes as the rate control system detects differences) and/=
or constrained by the physical device, but not a function of a user draggin=
g the window size. At an equivalent bitrate, it is better to compress a lar=
ger resolution and display it smaller (higher PSNR), then compress a smalle=
r resolution if you are within the codec's effective bitrate for that large=
r resolution.

Consider two different video flows using the same codec ...

Stream A is 640 x 480 at 1mbps which is this scaled to 320x240 and displaye=
d

Stream B is 320 x 240 at 1mbps which is displayed at 320x240.

My experience has been with modern video codecs that stream B will look bet=
ter than stream A. As well as looking better, it will typically also have a=
 better PSNR. There's a bunch of reasons why which are probably not worth g=
oing into here but give it a try and you will see what I mean. The key poin=
t is both streams where 1mbps. If stream B was sent at 256 kbps, then A wou=
ld look better.




_______________________________________________
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


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<BODY lang=3DEN-GB vLink=3Dpurple link=3Dblue>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D568580006-22062011><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>All,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D568580006-22062011><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D568580006-22062011><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>this RFC could perhaps be useful for this purpose =
of=20
signaling display size: <A=20
href=3D"http://www.rfc-editor.org/rfc/rfc6236.txt">http://www.rfc-editor.or=
g/rfc/rfc6236.txt</A></FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D568580006-22062011><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D568580006-22062011><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>Stefan</FONT></SPAN></DIV><BR>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px soli=
d; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> rtcweb-bounces@ietf.org=20
  [mailto:rtcweb-bounces@ietf.org] <B>On Behalf Of </B>Bert=20
  Greevenbosch<BR><B>Sent:</B> den 22 juni 2011 07:56<BR><B>To:</B> Justin=
=20
  Uberti<BR><B>Cc:</B> rtcweb@ietf.org<BR><B>Subject:</B> Re: [rtcweb] Chan=
ging=20
  video size (Re: use case:)<BR></FONT><BR></DIV>
  <DIV></DIV>
  <DIV class=3DSection1>
  <P class=3DMsoNormal><SPAN=20
  style=3D"FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: 'Calibri','sans-se=
rif'">Hi=20
  Justin, all,<o:p></o:p></SPAN></P>
  <P class=3DMsoNormal><SPAN=20
  style=3D"FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: 'Calibri','sans-se=
rif'"><o:p>&nbsp;</o:p></SPAN></P>
  <P class=3DMsoNormal><SPAN=20
  style=3D"FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: 'Calibri','sans-se=
rif'">Yes,=20
  that is a second issue. Even if the receiving end can signal its current=
=20
  display size, the streaming source might not be able to provide it. There=
fore=20
  flexibility would be needed for the source on how to handle the display s=
ize=20
  information.<o:p></o:p></SPAN></P>
  <P class=3DMsoNormal><SPAN=20
  style=3D"FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: 'Calibri','sans-se=
rif'"><o:p>&nbsp;</o:p></SPAN></P>
  <P class=3DMsoNormal><SPAN=20
  style=3D"FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: 'Calibri','sans-se=
rif'">Best=20
  regards,<o:p></o:p></SPAN></P>
  <P class=3DMsoNormal><SPAN=20
  style=3D"FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: 'Calibri','sans-se=
rif'">Bert<o:p></o:p></SPAN></P>
  <P class=3DMsoNormal><SPAN=20
  style=3D"FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: 'Calibri','sans-se=
rif'"><o:p>&nbsp;</o:p></SPAN></P>
  <P class=3DMsoNormal><SPAN=20
  style=3D"FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: 'Calibri','sans-se=
rif'"><o:p>&nbsp;</o:p></SPAN></P>
  <DIV=20
  style=3D"BORDER-RIGHT: medium none; PADDING-RIGHT: 0cm; BORDER-TOP: #b5c4=
df 1pt solid; PADDING-LEFT: 0cm; PADDING-BOTTOM: 0cm; BORDER-LEFT: medium n=
one; PADDING-TOP: 3pt; BORDER-BOTTOM: medium none">
  <P class=3DMsoNormal><B><SPAN lang=3DEN-US=20
  style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Tahoma','sans-serif'">From:</SPAN=
></B><SPAN=20
  lang=3DEN-US style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Tahoma','sans-serif'=
"> Justin=20
  Uberti [mailto:juberti@google.com] <BR><B>Sent:</B> 22 June 2011=20
  11:59<BR><B>To:</B> Bert Greevenbosch<BR><B>Cc:</B> Cullen Jennings; Aron=
=20
  Rosenberg; rtcweb@ietf.org<BR><B>Subject:</B> Re: [rtcweb] Changing video=
 size=20
  (Re: use case:)<o:p></o:p></SPAN></P></DIV>
  <P class=3DMsoNormal><o:p>&nbsp;</o:p></P>
  <DIV>
  <DIV>
  <P class=3DMsoNormal>Given the fact that this request for a video size ch=
ange is=20
  an optimization (either of resources or user experience), I think that SH=
OULD=20
  is probably the strongest label we can put on it. Consider the case of a=
=20
  middlebox; if it has no capacity to scale the stream, due to compute or c=
odec=20
  restrictions, there's not much it can do with this=20
  request.<o:p></o:p></P></DIV>
  <DIV>
  <P class=3DMsoNormal><o:p>&nbsp;</o:p></P></DIV>
  <BLOCKQUOTE=20
  style=3D"BORDER-RIGHT: medium none; PADDING-RIGHT: 0cm; BORDER-TOP: mediu=
m none; PADDING-LEFT: 6pt; PADDING-BOTTOM: 0cm; MARGIN-LEFT: 4.8pt; BORDER-=
LEFT: #cccccc 1pt solid; MARGIN-RIGHT: 0cm; PADDING-TOP: 0cm; BORDER-BOTTOM=
: medium none">
    <P class=3DMsoNormal><BR>However, whether we can cater for this also de=
pends=20
    on the actual technical solution for negotiation and which codec to use=
. If=20
    adding video size negotiation will not increase complexity too much, I =
see=20
    no reason not to do it. However, if it adds much complexity to the solu=
tion,=20
    we might need to consider whether it is worth it. Therefore, maybe it s=
hould=20
    not be a requirement, but something weaker; i.e. it could be phrased as=
 a=20
    SHOULD rather than a MUST.<BR><BR>Best regards,<BR><SPAN=20
    style=3D"COLOR: #888888">Bert</SPAN><o:p></o:p></P>
    <DIV>
    <P class=3DMsoNormal><BR><BR>-----Original Message-----<BR>From: <A=20
    href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</A> [ma=
ilto:<A=20
    href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</A>] On=
 Behalf=20
    Of Cullen Jennings<BR>Sent: 22 June 2011 00:11<BR>To: Aron Rosenberg<BR=
>Cc:=20
    <A href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</A><o:p></o:p></P></=
DIV>
    <DIV>
    <DIV>
    <P class=3DMsoNormal>Subject: Re: [rtcweb] Changing video size (Re: use=
=20
    case:)<BR><BR><BR>On Jun 20, 2011, at 1:30 , Harald Alvestrand=20
    wrote:<BR><BR>&gt; On 06/17/11 21:38, Aron Rosenberg wrote:<BR>&gt;&gt;=
 On=20
    Fri, Jun 17, 2011 at 11:09 AM, Cullen Jennings &lt;<A=20
    href=3D"mailto:fluffy@cisco.com">fluffy@cisco.com</A>&gt;=20
    wrote:<BR>&gt;&gt;<BR>&gt;&gt; Video Size Change.<BR>&gt;&gt;<BR>&gt;&g=
t;=20
    Alice and Bob are in a video call in their browsers and have negotiate =
a=20
    high resolution video. Bob decides to change the size of the windows hi=
s=20
    browser is displaying video to a small size. &nbsp;Bob's browser should=
 be=20
    able to regenerate the video codec parameters with Alice's browser to c=
hange=20
    the resolution of video Alice sends to match the smaller=20
    size.<BR>&gt;&gt;<BR>&gt;&gt;<BR>&gt;&gt; Changing compression due to a=
 user=20
    change in window size is usually not done since it has a bad long term=
=20
    effect on video quality. In almost all the major video calling systems,=
=20
    video resolution is a function of current bandwidth (which changes as t=
he=20
    rate control system detects differences) and/or constrained by the phys=
ical=20
    device, but not a function of a user dragging the window size. At an=20
    equivalent bitrate, it is better to compress a larger resolution and di=
splay=20
    it smaller (higher PSNR), then compress a smaller resolution if you are=
=20
    within the codec's effective bitrate for that larger=20
    resolution.<BR><BR>Consider two different video flows using the same co=
dec=20
    ...<BR><BR>Stream A is 640 x 480 at 1mbps which is this scaled to 320x2=
40=20
    and displayed<BR><BR>Stream B is 320 x 240 at 1mbps which is displayed =
at=20
    320x240.<BR><BR>My experience has been with modern video codecs that st=
ream=20
    B will look better than stream A. As well as looking better, it will=20
    typically also have a better PSNR. There's a bunch of reasons why which=
 are=20
    probably not worth going into here but give it a try and you will see w=
hat I=20
    mean. The key point is both streams where 1mbps. If stream B was sent a=
t 256=20
    kbps, then A would look=20
    better.<BR><BR><BR><BR><BR>____________________________________________=
___<BR>rtcweb=20
    mailing list<BR><A href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</A><=
BR><A=20
    href=3D"https://www.ietf.org/mailman/listinfo/rtcweb"=20
    target=3D_blank>https://www.ietf.org/mailman/listinfo/rtcweb</A><BR>___=
____________________________________________<BR>rtcweb=20
    mailing list<BR><A href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</A><=
BR><A=20
    href=3D"https://www.ietf.org/mailman/listinfo/rtcweb"=20
    target=3D_blank>https://www.ietf.org/mailman/listinfo/rtcweb</A><o:p></=
o:p></P></DIV></DIV></BLOCKQUOTE></DIV>
  <P class=3DMsoNormal><o:p>&nbsp;</o:p></P></DIV></BLOCKQUOTE></BODY></HTM=
L>

--_000_BBF498F2D030E84AB1179E24D1AC41D6147C4DA4CEESESSCMS0362e_--

From magnus.westerlund@ericsson.com  Wed Jun 22 00:15:09 2011
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Date: Wed, 22 Jun 2011 09:14:53 +0200
From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: Re: [rtcweb] WG adoption of Architecture and Overview of RTCWEB document
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On 2011-06-22 05:49, Likepeng wrote:
> I am fine with the proposal, but should the overview draft be
> informational, instead of standard track?

I think that is an open question. It depends on how this document is
written. If it in fact mandates which functionality block that must be
implemented it is a standards track. If it becomes written in a more
informal way it could be informational. The milestone says:

Sep 2011 Architecture and Security, Privacy, and Threat Model
document(s) to IESG as Informational

Milestones can be changed if the WG feels it has a preference for a
standards track document. From my perspective, this is a question we can
delay.

Cheers

Magnus Westerlund
WG chair

> 
> Thanks, Kind Regards Kepeng -----Original Message----- From:
> rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of
> Stephan Wenger Sent: Wednesday, June 22, 2011 12:10 AM To: Magnus
> Westerlund; rtcweb@ietf.org Subject: Re: [rtcweb] WG adoption of
> Architecture and Overview of RTCWEB document
> 
> Magnus' proposal of picking Harald's draft is fine with me. Stephan
> 
> On 6.21.2011 08:22 , "Magnus Westerlund"
> <magnus.westerlund@ericsson.com> wrote:
> 
>> WG,
>> 
>> We would like to adopt a document that describes the RTCWEB
>> architecture and provides an overview to the functionality that
>> will be specified in various documents targeted to specific pieces
>> of the functionalities we need for a complete solution.
>> 
>> There has been several documents that describes architecture,
>> frame works or overviews. None of them are complete when it comes
>> to covering the different aspects. The documents are:
>> 
>> https://datatracker.ietf.org/doc/draft-alvestrand-rtcweb-overview/ 
>> https://datatracker.ietf.org/doc/draft-rosenberg-rtcweb-framework/ 
>> https://datatracker.ietf.org/doc/draft-jennings-rtcweb-api/
>> 
>> The WG chairs' proposal is that we adopt one document and then add
>> in aspects that are needed from the other documents. In the process
>> putting together an good editor team for the WG document from the
>> persons that has contributed so far and want to continue.
>> 
>> The WG chairs propose that that the WG adopts: 
>> https://datatracker.ietf.org/doc/draft-alvestrand-rtcweb-overview/ 
>> as a WG item to form the core of an Architecture and Overview
>> document.
>> 
>> WG participants please provide either support for adoption or
>> provide feedback on any issues you see with this approach.
>> 
>> Please provide your view no later than Tuesday the 28th at 17.00
>> CEST.
>> 
>> Magnus Westerlund WG Chair
>> 
>> ----------------------------------------------------------------------
>>
>> 
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 
>> ----------------------------------------------------------------------
>
>> 
-- 

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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"Point-to-point" seems more flexible. Such a group would allow us to addres=
s use cases invovling devices (such as IP phones or mobile phones) which do=
n't have a browser.

Huilan=20

> -----Original Message-----
> From: rtcweb-bounces@ietf.org=20
> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Christer Holmberg
> Sent: Tuesday, June 21, 2011 1:11 PM
> To: Harald Alvestrand; rtcweb@ietf.org
> Subject: Re: [rtcweb] Use-case grouping
>=20
> Hi Harald,
>=20
> We could add a "Point-to-point", or "Browser-to-browser",=20
> use-case group.
>=20
> Regards,
>=20
> Christer
>=20
> ________________________________
> From: rtcweb-bounces@ietf.org [rtcweb-bounces@ietf.org] On=20
> Behalf Of Harald Alvestrand [harald@alvestrand.no]
> Sent: Tuesday, June 21, 2011 6:30 PM
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] Use-case grouping
>=20
> Seems to me sensible to have groupings.
>=20
> The use case I don't see a place for at the moment is the=20
> single point-to-point call; the proposed use case I don't see=20
> mentioned below is my suggestion for "interoperable calling".
>=20
> So this might require more tuning.
>=20
> On 06/21/11 14:39, Christer Holmberg wrote:
>=20
> Hi,
>=20
> It's nice to see the amount of use-cases people have submitted :)
>=20
>=20
> Now, since many of the use-cases are related to each other,=20
> and in order to get a better and easy-to-read structure of=20
> the document, I would like to suggest to devide the use-cases=20
> into different "use-case groups".
>=20
> Below is a proposed grouping, and examples of use-cases which=20
> would be listed in each group. I have only used the existing=20
> use-cases, and the ones suggested by Cullen, but I am aware=20
> that more use-cases have been proposed :)
>=20
>=20
> Telephony use-cases:
>=20
>     - Telephony terminal use-case (draft)
>     - Call fedex use-case (Cullen)
>=20
>=20
> Video conferenceing use-cases:
>=20
>     - With central server (draft)
>     - Without central server (draft)
>     - Hockey (draft)
>     - Screen re-size (Cullen)
>=20
>=20
> Embedded voice communicatoin use-case:
>=20
>    - Multiparty on-line game with voice communication (draft)
>=20
>=20
> Bandwidth/QoS/mobility use-cases:
>=20
>    - NIC Change (Cullen)
>    - QoS Marking (Cullen)
>=20
>=20
>=20
> Later, we can of course modify/add/remove groups, but I think=20
> the grouping above would be a good start, and I think all the=20
> use-cases so far proposed would fit into one of those groups.
>=20
> Comments?
>=20
> Regards,
>=20
> Christer
>=20
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org<mailto:rtcweb@ietf.org>
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> =

From csp@csperkins.org  Wed Jun 22 07:46:00 2011
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This seems reasonable to me.

Colin


On 21 Jun 2011, at 16:22, Magnus Westerlund wrote:
> WG,
>=20
> We would like to adopt a document that describes the RTCWEB =
architecture
> and provides an overview to the functionality that will be specified =
in
> various documents targeted to specific pieces of the functionalities =
we
> need for a complete solution.
>=20
> There has been several documents that describes architecture, frame
> works or overviews. None of them are complete when it comes to =
covering
> the different aspects. The documents are:
>=20
> https://datatracker.ietf.org/doc/draft-alvestrand-rtcweb-overview/
> https://datatracker.ietf.org/doc/draft-rosenberg-rtcweb-framework/
> https://datatracker.ietf.org/doc/draft-jennings-rtcweb-api/
>=20
> The WG chairs' proposal is that we adopt one document and then add in
> aspects that are needed from the other documents. In the process =
putting
> together an good editor team for the WG document from the persons that
> has contributed so far and want to continue.
>=20
> The WG chairs propose that that the WG adopts:
> https://datatracker.ietf.org/doc/draft-alvestrand-rtcweb-overview/
> as a WG item to form the core of an Architecture and Overview =
document.
>=20
> WG participants please provide either support for adoption or provide
> feedback on any issues you see with this approach.
>=20
> Please provide your view no later than Tuesday the 28th at 17.00 CEST.
>=20
> Magnus Westerlund
> WG Chair
>=20
> ----------------------------------------------------------------------
> 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
> ----------------------------------------------------------------------
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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




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Tentative times for meetings at IETF 81. 

Begin forwarded message:

> RTCWEB Session 1 (2 hours)
> Tuesday, Afternoon Session I 1300-1500
> Room Name: 205 ABC
> ----------------------------------------------
> RTCWEB Session 2 (2 hours)
> Thursday, Afternoon Session I 1300-1500
> Room Name: 205 ABC
> ----------------------------------------------
> 



From magnus.westerlund@ericsson.com  Sat Jun 25 07:51:19 2011
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Subject: Re: [rtcweb] WG adoption for Use Case document
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WG,

I have reviewed the input from the WG participants, and there has all
been in support for adopting this document as WG item. So hereby I
declare https://datatracker.ietf.org/doc/draft-holmberg-rtcweb-ucreqs/ a
WG document. At this point the WG chairs request no changes in the
document editor team.

In addition there has been input on how to improve the document which we
request the editors to take into account.

Editors, please submit the document as
draft-ietf-rtcweb-use-cases-and-requirements-00.txt.

Cheers

Magnus Westerlund
WG chair

On 2011-06-17 10:00, Magnus Westerlund wrote:
> WG,
> 
> We would like to adopt a WG document with the purpose for describing use
> cases for the RTCWEB work. This document would be targeted as
> publication as an Inforamtional RFC eventually.
> 
> The candidate document for this is:
> Web Real-Time Communication Use-cases and Requirements
> https://datatracker.ietf.org/doc/draft-holmberg-rtcweb-ucreqs/
> 
> Based on our interim last week it was clear that there are some use
> cases that clearly needs development. However, the WG chairs believe
> that future development of the document can be done as WG item.
> 
> WG participants please indicate if you agree in adopting this document
> as a basis, or if not, what the short comings you would like to see
> addressed prior to adoption. Once more, the document will be continued
> to be developed under the WG control and this is far from any final
> version.
> 
> Please provide your view no later than Friday the 24th.
> 
> Best Regards
> 
> Magnus Westerlund
> WG chair
> 
> ----------------------------------------------------------------------
> 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 magnus.westerlund@ericsson.com  Mon Jun 27 00:36:36 2011
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Subject: [rtcweb] Non-media data service consensus and requirements
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WG,

At the interim it was planned to have a bit discussion on the datagram
service for RTCWEB. The first question to try to resolve if there
is consensus for including some form of non real-time media (i.e. not
audio, video) service between peers. This is a bit tangled with the
actual requirements and use cases. But there was views both for it and
against it on the mailing list. So lets continue and try to come to a
conclusion on this discussion.

The use cases mentioned on the mailing list are:

- Dynamic meta data for Conference and other real-time services

- Gaming data with low latency requirements

Does anyone like to add additional use cases?

Based on my personal understanding this points to primarily have the
RTCWEB provide a unreliable datagram service. This clearly needs
additional requirements to be secure and safe to deploy, but more about
this below. I still like to ask the WG here a question.

Are you supporting the inclusion of a unreliable datagram service
directly between peers? Please provide your view and any additional
statements of motivation that you desire to provide.

Secondly, there is a question if there needs to have something that
provides reliable message (of arbitrary size) or byte stream oriented
data transport between the peers. I personally foresee that people will
build JS libraries for this on top of a unreliable datagram service. If
you desire reliable data service as part of the standardized solution
please provide motivation and use case and requirements.

I also want to take a stab on what I personally see as the requirements
that exist on unreliable datagram service in the context of RTCWEB.

- Unreliable data transmission
- Datagram oriented
   * Size limited by MTU
     - Path MTU discovery needed
   * Fragmentation by the application
- Low latency, i.e. Peer to Peer preferable
- Congestion Controlled, to be
   * Network friendly
   * Not become a Denial of Service tool
- Security
  * Confidentiality
  * Integrity Protected
  * Source Authenticated (at least bound to the signalling peer)
  * Ensure consent to receive data

Please debate the above. This is an attempt to ensure that we can
establish WG consensus on both data service and any requirements.

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
----------------------------------------------------------------------


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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 Use-cases and Requirements
	Author(s)       : Christer Holmberg
                          Stefan Hakansson
                          Goran AP Eriksson
	Filename        : draft-ietf-rtcweb-use-cases-and-requirements-00.txt
	Pages           : 15
	Date            : 2011-06-27

   This document describes web based real-time communication use-cases.
   Based on the use-cases, the document also derives requirements
   related to the browser, and the API used by web applications to
   request and control media stream services provided by the browser.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-rtcweb-use-cases-and-require=
ments-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-use-cases-and-requirem=
ents-00.txt

From emil@sip-communicator.org  Mon Jun 27 07:26:14 2011
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Non-media data service consensus and requirements
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=D0=9D=D0=B0 27.06.11 09:36, Magnus Westerlund =D0=BD=D0=B0=D0=BF=D0=B8=D1=
=81=D0=B0:
> WG,
>=20
> At the interim it was planned to have a bit discussion on the datagram
> service for RTCWEB. The first question to try to resolve if there
> is consensus for including some form of non real-time media (i.e. not
> audio, video) service between peers. This is a bit tangled with the
> actual requirements and use cases. But there was views both for it and
> against it on the mailing list. So lets continue and try to come to a
> conclusion on this discussion.
>=20
> The use cases mentioned on the mailing list are:
>=20
> - Dynamic meta data for Conference and other real-time services
>=20
> - Gaming data with low latency requirements
>=20
> Does anyone like to add additional use cases?

- File transfer? Skype, Google Talk, Windows Live, and pretty much any
existing service has it so I suppose it is reasonable to assume that
people using rtcweb would like to add the feature to their web applicatio=
ns.

- Transferring input events (e.g. pressed keys and mouse clicks) in
desktop sharing applicaitons?

> Based on my personal understanding this points to primarily have the
> RTCWEB provide a unreliable datagram service. This clearly needs
> additional requirements to be secure and safe to deploy, but more about=

> this below. I still like to ask the WG here a question.
>=20
> Are you supporting the inclusion of a unreliable datagram service
> directly between peers? Please provide your view and any additional
> statements of motivation that you desire to provide.

Yes. Requirements and code necessary to support a generic datagram
service would hardly differ from those we have for RTP and it would
definitely help a lot in the use cases described above.

> Secondly, there is a question if there needs to have something that
> provides reliable message (of arbitrary size) or byte stream oriented
> data transport between the peers. I personally foresee that people will=

> build JS libraries for this on top of a unreliable datagram service. If=

> you desire reliable data service as part of the standardized solution
> please provide motivation and use case and requirements.

I definitely think it would be a great idea for the IETF to specify a
datagram based reliable service similar to Google's pseudo TCP. Whether
or not this happens in RTCWEB is of course a different matter. As I
wrote above many applications are likely to need such a think and I am
not sure that the flexibility we'll get by leaving this to JS libs is
worth the interoperability nightmare it would imply.

Emil

>=20
> I also want to take a stab on what I personally see as the requirements=

> that exist on unreliable datagram service in the context of RTCWEB.
>=20
> - Unreliable data transmission
> - Datagram oriented
>    * Size limited by MTU
>      - Path MTU discovery needed
>    * Fragmentation by the application
> - Low latency, i.e. Peer to Peer preferable
> - Congestion Controlled, to be
>    * Network friendly
>    * Not become a Denial of Service tool
> - Security
>   * Confidentiality
>   * Integrity Protected
>   * Source Authenticated (at least bound to the signalling peer)
>   * Ensure consent to receive data
>=20
> Please debate the above. This is an attempt to ensure that we can
> establish WG consensus on both data service and any requirements.
>=20
> cheers
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=C3=A4r=C3=B6gatan 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
>=20

--=20
Emil Ivov, Ph.D.                       67000 Strasbourg,
Project Lead                           France
Jitsi
emcho@jitsi.org                        PHONE: +33.1.77.62.43.30
http://jitsi.org                       FAX:   +33.1.77.62.47.31


From tedh@ipinfusion.com  Mon Jun 27 09:22:19 2011
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Subject: [rtcweb] Document split is approved
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--bcaec501c58c4688ec04a6b3f458
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The document split presented seems to have rough consensus, so we will
consider it the plan of record for now.

As an aside, I seem to be having some trouble with email from the list to my
work email, so I will shift back to using ted.ietf@gmail.com for mailing
list traffic from now.

thanks,

Ted

--bcaec501c58c4688ec04a6b3f458
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The document split presented seems to have rough consensus, so we will cons=
ider it the plan of record for now.<div><br></div><div>As an aside, I seem =
to be having some trouble with email from the list to my work email, so I w=
ill shift back to using <a href=3D"mailto:ted.ietf@gmail.com">ted.ietf@gmai=
l.com</a> for mailing list traffic from now.</div>
<div><br></div><div>thanks,</div><div><br></div><div>Ted</div>

--bcaec501c58c4688ec04a6b3f458--

From bernard_aboba@hotmail.com  Mon Jun 27 15:35:18 2011
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I do not support an unreliable datagram service that can be used to send ar=
bitrary data. =20

For example=2C it seems dangerous for a Web browser under the control of an=
 attacker to be able to send RIP=2C SNMP or DNS packets to arbitrary destin=
ations.=20

For these transactional exchanges the overhead of ICE would be excessive an=
d so there will be a very strong temptation to cut corners.=20

Assuming that the goal is not to send arbitrary data=2C then we need to dig=
 into the transport requirements more.=20

For example=2C is the non-media data to be synchronized with media (e.g. re=
al-time text)?

Is there a session associated with the non-media data (e.g. XMPP or MSRP ex=
changes)?=20

Is there a reliability requirement?=20

Is it congestion-controlled?=20

How long-lived are the flows?=20




---------------------------------------------------
From: magnus.westerlund@ericsson.com
To: rtcweb@ietf.org
Date: Mon=2C 27 Jun 2011 09:36:30 +0200
Subject: [rtcweb] Non-media data service consensus and requirements

WG=2C
=20
At the interim it was planned to have a bit discussion on the datagram
service for RTCWEB. The first question to try to resolve if there
is consensus for including some form of non real-time media (i.e. not
audio=2C video) service between peers. This is a bit tangled with the
actual requirements and use cases. But there was views both for it and
against it on the mailing list. So lets continue and try to come to a
conclusion on this discussion.
=20
The use cases mentioned on the mailing list are:
=20
- Dynamic meta data for Conference and other real-time services
=20
- Gaming data with low latency requirements
=20
Does anyone like to add additional use cases?
=20
Based on my personal understanding this points to primarily have the
RTCWEB provide a unreliable datagram service. This clearly needs
additional requirements to be secure and safe to deploy=2C but more about
this below. I still like to ask the WG here a question.
=20
Are you supporting the inclusion of a unreliable datagram service
directly between peers? Please provide your view and any additional
statements of motivation that you desire to provide.
=20
Secondly=2C there is a question if there needs to have something that
provides reliable message (of arbitrary size) or byte stream oriented
data transport between the peers. I personally foresee that people will
build JS libraries for this on top of a unreliable datagram service. If
you desire reliable data service as part of the standardized solution
please provide motivation and use case and requirements.
=20
I also want to take a stab on what I personally see as the requirements
that exist on unreliable datagram service in the context of RTCWEB.
=20
- Unreliable data transmission
- Datagram oriented
   * Size limited by MTU
     - Path MTU discovery needed
   * Fragmentation by the application
- Low latency=2C i.e. Peer to Peer preferable
- Congestion Controlled=2C to be
   * Network friendly
   * Not become a Denial of Service tool
- Security
  * Confidentiality
  * Integrity Protected
  * Source Authenticated (at least bound to the signalling peer)
  * Ensure consent to receive data
=20
Please debate the above. This is an attempt to ensure that we can
establish WG consensus on both data service and any requirements.
=20
cheers
=20
Magnus Westerlund
=20
----------------------------------------------------------------------
Multimedia Technologies=2C Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
F=E4r=F6gatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm=2C Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------
 		 	   		  =

--_619c0170-1f0c-4026-ba0e-62b9c45533cd_
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<html>
<head>
<style><!--
.hmmessage P
{
margin:0px=3B
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}
body.hmmessage
{
font-size: 10pt=3B
font-family:Tahoma
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</head>
<body class=3D'hmmessage'><div dir=3D'ltr'>
I do not support an unreliable datagram service that can be used to send ar=
bitrary data.&nbsp=3B <br><br>For example=2C it seems dangerous for a Web b=
rowser under the control of an attacker to be able to send RIP=2C SNMP or D=
NS packets to arbitrary destinations. <br><br>For these transactional excha=
nges the overhead of ICE would be excessive and so there will be a very str=
ong temptation to cut corners. <br><br>Assuming that the goal is not to sen=
d arbitrary data=2C then we need to dig into the transport requirements mor=
e. <br><br>For example=2C is the non-media data to be synchronized with med=
ia (e.g. real-time text)?<br><br>Is there a session associated with the non=
-media data (e.g. XMPP or MSRP exchanges)? <br><br>Is there a reliability r=
equirement? <br><br>Is it congestion-controlled? <br><br>How long-lived are=
 the flows? <br><br><br><br><br>-------------------------------------------=
--------<br>From: magnus.westerlund@ericsson.com<br>To: rtcweb@ietf.org<br>=
Date: Mon=2C 27 Jun 2011 09:36:30 +0200<br>Subject: [rtcweb] Non-media data=
 service consensus and requirements<br><br><pre>WG=2C<br> <br>At the interi=
m it was planned to have a bit discussion on the datagram<br>service for RT=
CWEB. The first question to try to resolve if there<br>is consensus for inc=
luding some form of non real-time media (i.e. not<br>audio=2C video) servic=
e between peers. This is a bit tangled with the<br>actual requirements and =
use cases. But there was views both for it and<br>against it on the mailing=
 list. So lets continue and try to come to a<br>conclusion on this discussi=
on.<br> <br>The use cases mentioned on the mailing list are:<br> <br>- Dyna=
mic meta data for Conference and other real-time services<br> <br>- Gaming =
data with low latency requirements<br> <br>Does anyone like to add addition=
al use cases?<br> <br>Based on my personal understanding this points to pri=
marily have the<br>RTCWEB provide a unreliable datagram service. This clear=
ly needs<br>additional requirements to be secure and safe to deploy=2C but =
more about<br>this below. I still like to ask the WG here a question.<br> <=
br>Are you supporting the inclusion of a unreliable datagram service<br>dir=
ectly between peers? Please provide your view and any additional<br>stateme=
nts of motivation that you desire to provide.<br> <br>Secondly=2C there is =
a question if there needs to have something that<br>provides reliable messa=
ge (of arbitrary size) or byte stream oriented<br>data transport between th=
e peers. I personally foresee that people will<br>build JS libraries for th=
is on top of a unreliable datagram service. If<br>you desire reliable data =
service as part of the standardized solution<br>please provide motivation a=
nd use case and requirements.<br> <br>I also want to take a stab on what I =
personally see as the requirements<br>that exist on unreliable datagram ser=
vice in the context of RTCWEB.<br> <br>- Unreliable data transmission<br>- =
Datagram oriented<br>   * Size limited by MTU<br>     - Path MTU discovery =
needed<br>   * Fragmentation by the application<br>- Low latency=2C i.e. Pe=
er to Peer preferable<br>- Congestion Controlled=2C to be<br>   * Network f=
riendly<br>   * Not become a Denial of Service tool<br>- Security<br>  * Co=
nfidentiality<br>  * Integrity Protected<br>  * Source Authenticated (at le=
ast bound to the signalling peer)<br>  * Ensure consent to receive data<br>=
 <br>Please debate the above. This is an attempt to ensure that we can<br>e=
stablish WG consensus on both data service and any requirements.<br> <br>ch=
eers<br> <br>Magnus Westerlund<br> <br>------------------------------------=
----------------------------------<br>Multimedia Technologies=2C Ericsson R=
esearch EAB/TVM<br>--------------------------------------------------------=
--------------<br>Ericsson AB                | Phone  +46 10 7148287<br>F=
=E4r=F6gatan 6                | Mobile +46 73 0949079<br>SE-164 80 Stockhol=
m=2C Sweden| mailto: magnus.westerlund@ericsson.com<br>--------------------=
--------------------------------------------------</pre><br> 		 	   		  </d=
iv></body>
</html>=

--_619c0170-1f0c-4026-ba0e-62b9c45533cd_--

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=D0=9D=D0=B0 28.06.11 00:35, Bernard Aboba =D0=BD=D0=B0=D0=BF=D0=B8=D1=81=
=D0=B0:
> I do not support an unreliable datagram service that can be used to sen=
d
> arbitrary data.=20
>=20
> For example, it seems dangerous for a Web browser under the control of
> an attacker to be able to send RIP, SNMP or DNS packets to arbitrary
> destinations.
>=20
> For these transactional exchanges the overhead of ICE would be excessiv=
e
> and so there will be a very strong temptation to cut corners.

Well, if ICE is part of the browser we could condition sending such data
on the successful termination of ICE processing with the intended
destination. Same as with RTP. Wouldn't this work?

Emil

> Assuming that the goal is not to send arbitrary data, then we need to
> dig into the transport requirements more.
>=20
> For example, is the non-media data to be synchronized with media (e.g.
> real-time text)?
>=20
> Is there a session associated with the non-media data (e.g. XMPP or MSR=
P
> exchanges)?
>=20
> Is there a reliability requirement?
>=20
> Is it congestion-controlled?
>=20
> How long-lived are the flows?
>=20
>=20
>=20
>=20
> ---------------------------------------------------
> From: magnus.westerlund@ericsson.com
> To: rtcweb@ietf.org
> Date: Mon, 27 Jun 2011 09:36:30 +0200
> Subject: [rtcweb] Non-media data service consensus and requirements
>=20
> WG,
> =20
> At the interim it was planned to have a bit discussion on the datagram
> service for RTCWEB. The first question to try to resolve if there
> is consensus for including some form of non real-time media (i.e. not
> audio, video) service between peers. This is a bit tangled with the
> actual requirements and use cases. But there was views both for it and
> against it on the mailing list. So lets continue and try to come to a
> conclusion on this discussion.
> =20
> The use cases mentioned on the mailing list are:
> =20
> - Dynamic meta data for Conference and other real-time services
> =20
> - Gaming data with low latency requirements
> =20
> Does anyone like to add additional use cases?
> =20
> Based on my personal understanding this points to primarily have the
> RTCWEB provide a unreliable datagram service. This clearly needs
> additional requirements to be secure and safe to deploy, but more about=

> this below. I still like to ask the WG here a question.
> =20
> Are you supporting the inclusion of a unreliable datagram service
> directly between peers? Please provide your view and any additional
> statements of motivation that you desire to provide.
> =20
> Secondly, there is a question if there needs to have something that
> provides reliable message (of arbitrary size) or byte stream oriented
> data transport between the peers. I personally foresee that people will=

> build JS libraries for this on top of a unreliable datagram service. If=

> you desire reliable data service as part of the standardized solution
> please provide motivation and use case and requirements.
> =20
> I also want to take a stab on what I personally see as the requirements=

> that exist on unreliable datagram service in the context of RTCWEB.
> =20
> - Unreliable data transmission
> - Datagram oriented
>    * Size limited by MTU
>      - Path MTU discovery needed
>    * Fragmentation by the application
> - Low latency, i.e. Peer to Peer preferable
> - Congestion Controlled, to be
>    * Network friendly
>    * Not become a Denial of Service tool
> - Security
>   * Confidentiality
>   * Integrity Protected
>   * Source Authenticated (at least bound to the signalling peer)
>   * Ensure consent to receive data
> =20
> Please debate the above. This is an attempt to ensure that we can
> establish WG consensus on both data service and any requirements.
> =20
> cheers
> =20
> Magnus Westerlund
> =20
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=C3=A4r=C3=B6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=20
>=20
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

--=20
Emil Ivov, Ph.D.                       67000 Strasbourg,
Project Lead                           France
Jitsi
emcho@jitsi.org                        PHONE: +33.1.77.62.43.30
http://jitsi.org                       FAX:   +33.1.77.62.47.31


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On Jun 27, 2011, at 3:35 PM, Bernard Aboba wrote:

> I do not support an unreliable datagram service that can be used to =
send arbitrary data. =20

Mostly disagree. It should be possible to send arbitrary data from one =
browser to another.

>=20
> For example, it seems dangerous for a Web browser under the control of =
an attacker to be able to send RIP, SNMP or DNS packets to arbitrary =
destinations.=20
>=20

Agreed, so the encapsulation must prevent this.

> For these transactional exchanges the overhead of ICE would be =
excessive and so there will be a very strong temptation to cut corners.=20=

>=20

Disagree. Browser vendors will have a strong incentive to not cut =
corners.

> Assuming that the goal is not to send arbitrary data, then we need to =
dig into the transport requirements more.=20
>=20
> For example, is the non-media data to be synchronized with media (e.g. =
real-time text)?
>=20

Maybe.

> Is there a session associated with the non-media data (e.g. XMPP or =
MSRP exchanges)?=20
>=20

Maybe.

> Is there a reliability requirement?=20
>=20

Maybe.

> Is it congestion-controlled?=20
>=20

This must be enforced.

> How long-lived are the flows?=20
>=20

Totally depends on the application.

Matthew Kaufman


--Apple-Mail-63-194939573
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<html><head><base href=3D"x-msg://2449/"></head><body style=3D"word-wrap: =
break-word; -webkit-nbsp-mode: space; -webkit-line-break: =
after-white-space; "><br><div><div>On Jun 27, 2011, at 3:35 PM, 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-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">I do not support an unreliable datagram service that =
can be used to send arbitrary data.&nbsp;<span =
class=3D"Apple-converted-space">&nbsp;</span><br></div></div></span></bloc=
kquote><div><br></div>Mostly disagree. It should be possible to send =
arbitrary data from one browser to another.</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-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"><br>For example, it seems dangerous for a Web browser =
under the control of an attacker to be able to send RIP, SNMP or DNS =
packets to arbitrary destinations.<span =
class=3D"Apple-converted-space">&nbsp;</span><br><br></div></div></span></=
blockquote><div><br></div>Agreed, so the encapsulation must prevent =
this.</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-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">For these transactional exchanges the overhead of ICE =
would be excessive and so there will be a very strong temptation to cut =
corners.<span =
class=3D"Apple-converted-space">&nbsp;</span><br><br></div></div></span></=
blockquote><div><br></div>Disagree. Browser vendors will have a strong =
incentive to not cut corners.</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-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">Assuming that the goal is not to send arbitrary data, =
then we need to dig into the transport requirements more.<span =
class=3D"Apple-converted-space">&nbsp;</span><br><br>For example, is the =
non-media data to be synchronized with media (e.g. real-time =
text)?<br><br></div></div></span></blockquote><div><br></div>Maybe.</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-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">Is there a session associated with the non-media data =
(e.g. XMPP or MSRP exchanges)?<span =
class=3D"Apple-converted-space">&nbsp;</span><br><br></div></div></span></=
blockquote><div><br></div>Maybe.</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-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">Is there a reliability requirement?<span =
class=3D"Apple-converted-space">&nbsp;</span><br><br></div></div></span></=
blockquote><div><br></div>Maybe.</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-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">Is it congestion-controlled?<span =
class=3D"Apple-converted-space">&nbsp;</span><br><br></div></div></span></=
blockquote><div><br></div>This must be =
enforced.</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-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">How long-lived are the flows?<span =
class=3D"Apple-converted-space">&nbsp;</span><br><br></div></div></span></=
blockquote><div><br></div>Totally depends on the =
application.</div><div><br></div><div>Matthew =
Kaufman</div><div><br></div></body></html>=

--Apple-Mail-63-194939573--

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On 6/27/2011 3:43 PM, Emil Ivov wrote:
> Well, if ICE is part of the browser we could condition sending such data
> on the successful termination of ICE processing with the intended
> destination. Same as with RTP. Wouldn't this work?
>
> Emil
>

Yes, assuming that we have an established session I wouldn't support 
APIs that would allow sending of data until that session was 
established, and only to the other browser(s) in the session.

--Chris

From bernard_aboba@hotmail.com  Mon Jun 27 16:14:27 2011
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=20
> Well=2C if ICE is part of the browser we could condition sending such dat=
a
> on the successful termination of ICE processing with the intended
> destination. Same as with RTP. Wouldn't this work?

It depends on the exact requirements.=20

There has already been discussion of relaxing the ICE requirement so as to =
better enable backward compatibility with legacy implementations (e.g. usin=
g RTCP instead). =20
There is also a potential backward compatibility requirement for non-media =
data=2C which will be even more difficult=2C since ICE (or Jingle or SDP) i=
s not routinely used today (think online gaming)=2C and the volume is poten=
tially quite heavy so that gateways may represent a significant burden.

So before saying that ICE would "solve the problem"=2C it's important to un=
derstand what exactly the problem is.=20

Are we talking about sending arbitrary data-grams in transactional exchange=
s?   Or perhaps text encoded in RTP?  Or maybe data in a specified encoding=
 (e.g. not RTP or arbitrary datagrams) so as to prevent spoofing?

Are there multiple use cases here=2C each with slightly different requireme=
nts?  =20
 		 	   		  =

--_f00dabe5-1ecf-4a55-a0e0-a6e93dce161f_
Content-Type: text/html; charset="iso-8859-1"
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<html>
<head>
<style><!--
.hmmessage P
{
margin:0px=3B
padding:0px
}
body.hmmessage
{
font-size: 10pt=3B
font-family:Tahoma
}
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</head>
<body class=3D'hmmessage'><div dir=3D'ltr'>
&nbsp=3B<br><div>&gt=3B Well=2C if ICE is part of the browser we could cond=
ition sending such data<br>&gt=3B on the successful termination of ICE proc=
essing with the intended<br>&gt=3B destination. Same as with RTP. Wouldn't =
this work?<br><br>It depends on the exact requirements. <br><br>There has a=
lready been discussion of relaxing the ICE requirement so as to better enab=
le backward compatibility with legacy implementations (e.g. using RTCP inst=
ead).&nbsp=3B <br>There is also a potential backward compatibility requirem=
ent for non-media data=2C which will be even more difficult=2C since ICE (o=
r Jingle or SDP) is not routinely used today (think online gaming)=2C and t=
he volume is potentially quite heavy so that gateways may represent a signi=
ficant burden.<br><br>So before saying that ICE would "solve the problem"=
=2C it's important to understand what exactly the problem is. <br><br>Are w=
e talking about sending arbitrary data-grams in transactional exchanges?&nb=
sp=3B&nbsp=3B Or perhaps text encoded in RTP?&nbsp=3B Or maybe data in a sp=
ecified encoding (e.g. not RTP or arbitrary datagrams) so as to prevent spo=
ofing?<br><br>Are there multiple use cases here=2C each with slightly diffe=
rent requirements? &nbsp=3B <br></div> 		 	   		  </div></body>
</html>=

--_f00dabe5-1ecf-4a55-a0e0-a6e93dce161f_--

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On Jun 27, 2011, at 4:14 PM, Bernard Aboba wrote:

>=20
> Are there multiple use cases here, each with slightly different =
requirements?  =20

There's quite a few possible use cases... what we're looking for is the =
lowest common denominator use case so we can build solutions for the =
rest on top of that, rather than trying to build a dozen different =
special-purpose protocols into the browser.

Matthew Kaufman


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<html><head><base href=3D"x-msg://2469/"></head><body style=3D"word-wrap: =
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after-white-space; "><br><div><div>On Jun 27, 2011, at 4:14 PM, 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: =
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"><div dir=3D"ltr"><div><br>Are there multiple use cases here, each with =
slightly different requirements? &nbsp;<span =
class=3D"Apple-converted-space">&nbsp;</span><br></div></div></div></span>=
</blockquote><br></div><div>There's quite a few possible use cases... =
what we're looking for is the lowest common denominator use case so we =
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From emil@sip-communicator.org  Mon Jun 27 23:38:13 2011
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=D0=9D=D0=B0 28.06.11 01:14, Bernard Aboba =D0=BD=D0=B0=D0=BF=D0=B8=D1=81=
=D0=B0:
>> Well, if ICE is part of the browser we could condition sending such da=
ta
>> on the successful termination of ICE processing with the intended
>> destination. Same as with RTP. Wouldn't this work?
>=20
> It depends on the exact requirements.
>=20
> There has already been discussion of relaxing the ICE requirement so as=

> to better enable backward compatibility with legacy implementations

I supposed I missed the discussion on dropping ICE. However, I don't
quite understand the issue with legacy implementations. If they are not
using ICE then they are most probably relying on server-side relaying
techniques such as latching. If that's the case then rtcweb clients
would establish sessions with a server which would be the one required
to do ICE (or ICE lite) rather than talking directly to the remote party.=


This does indeed require the server to support ICE (lite) but is this
really an issue?

> (e.g. using RTCP instead).=20

Not sure I understand. Is someone using RTCP for NAT traversal and
connectivity establishment?

> There is also a potential backward compatibility requirement for
> non-media data, which will be even more difficult, since ICE (or Jingle=

> or SDP) is not routinely used today (think online gaming), and the
> volume is potentially quite heavy so that gateways may represent a
> significant burden.

Same as above. If they weren't using ICE before then they were either
relaying through a server that can now take care of ICE (lite), or they
were using a home-grown ICE variant which wouldn't be compatible anyway.
Am I missing something?

> So before saying that ICE would "solve the problem", it's important to
> understand what exactly the problem is.
>=20
> Are we talking about sending arbitrary data-grams in transactional
> exchanges?   Or perhaps text encoded in RTP?  Or maybe data in a
> specified encoding (e.g. not RTP or arbitrary datagrams) so as to
> prevent spoofing?
>=20
> Are there multiple use cases here, each with slightly different
> requirements?

Right. Can we then agree that most use cases would be similar to a
regular VoIP session without the limitation of putting RTP inside packets=
?

Emil


--=20
http://jitsi.org


From igor.faynberg@alcatel-lucent.com  Tue Jun 28 05:52:42 2011
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At the interim meeting,  there was a strong argument for NOT having ICE 
as part of the browser, and I remember that no one objected.  My reading 
is that it has been the decision.

Igor


On 6/27/2011 6:43 PM, Emil Ivov wrote:
> ...
>> For these transactional exchanges the overhead of ICE would be excessive
>> and so there will be a very strong temptation to cut corners.
> Well, if ICE is part of the browser we could condition sending such data
> on the successful termination of ICE processing with the intended
> destination. Same as with RTP. Wouldn't this work?
>
> Emil
>
>> Assuming that the goal is not to send arbitrary data, then we need to
>> dig into the transport requirements more.
>>
>> For example, is the non-media data to be synchronized with media (e.g.
>> real-time text)?
>>
>> Is there a session associated with the non-media data (e.g. XMPP or MSRP
>> exchanges)?
>>
>> Is there a reliability requirement?
>>
>> Is it congestion-controlled?
>>
>> How long-lived are the flows?
>>
>>
>>
>>
>> ---------------------------------------------------
>> From: magnus.westerlund@ericsson.com
>> To: rtcweb@ietf.org
>> Date: Mon, 27 Jun 2011 09:36:30 +0200
>> Subject: [rtcweb] Non-media data service consensus and requirements
>>
>> WG,
>>
>> At the interim it was planned to have a bit discussion on the datagram
>> service for RTCWEB. The first question to try to resolve if there
>> is consensus for including some form of non real-time media (i.e. not
>> audio, video) service between peers. This is a bit tangled with the
>> actual requirements and use cases. But there was views both for it and
>> against it on the mailing list. So lets continue and try to come to a
>> conclusion on this discussion.
>>
>> The use cases mentioned on the mailing list are:
>>
>> - Dynamic meta data for Conference and other real-time services
>>
>> - Gaming data with low latency requirements
>>
>> Does anyone like to add additional use cases?
>>
>> Based on my personal understanding this points to primarily have the
>> RTCWEB provide a unreliable datagram service. This clearly needs
>> additional requirements to be secure and safe to deploy, but more about
>> this below. I still like to ask the WG here a question.
>>
>> Are you supporting the inclusion of a unreliable datagram service
>> directly between peers? Please provide your view and any additional
>> statements of motivation that you desire to provide.
>>
>> Secondly, there is a question if there needs to have something that
>> provides reliable message (of arbitrary size) or byte stream oriented
>> data transport between the peers. I personally foresee that people will
>> build JS libraries for this on top of a unreliable datagram service. If
>> you desire reliable data service as part of the standardized solution
>> please provide motivation and use case and requirements.
>>
>> I also want to take a stab on what I personally see as the requirements
>> that exist on unreliable datagram service in the context of RTCWEB.
>>
>> - Unreliable data transmission
>> - Datagram oriented
>>     * Size limited by MTU
>>       - Path MTU discovery needed
>>     * Fragmentation by the application
>> - Low latency, i.e. Peer to Peer preferable
>> - Congestion Controlled, to be
>>     * Network friendly
>>     * Not become a Denial of Service tool
>> - Security
>>    * Confidentiality
>>    * Integrity Protected
>>    * Source Authenticated (at least bound to the signalling peer)
>>    * Ensure consent to receive data
>>
>> Please debate the above. This is an attempt to ensure that we can
>> establish WG consensus on both data service and any requirements.
>>
>> cheers
>>
>> Magnus Westerlund
>>
>> ----------------------------------------------------------------------
>> Multimedia Technologies, Ericsson Research EAB/TVM
>> ----------------------------------------------------------------------
>> Ericsson AB                | Phone  +46 10 7148287
>> FÃ¤rÃ¶gatan 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

From emil@sip-communicator.org  Tue Jun 28 06:28:39 2011
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=D0=9D=D0=B0 28.06.11 14:52, Igor Faynberg =D0=BD=D0=B0=D0=BF=D0=B8=D1=81=
=D0=B0:
> At the interim meeting,  there was a strong argument for NOT having ICE=
=20
> as part of the browser, and I remember that no one objected.  My readin=
g=20
> is that it has been the decision.

Well. I did attend the meeting but I remember no such decision or even
rough consensus. I actually remember people discussing the fact that
properly implementing ICE in javascript might be tricky due to the timer
granularity and the pacing requirements in 5245.

Have I missed something?

Emil

>=20
> Igor
>=20
>=20
> On 6/27/2011 6:43 PM, Emil Ivov wrote:
>> ...
>>> For these transactional exchanges the overhead of ICE would be excess=
ive
>>> and so there will be a very strong temptation to cut corners.
>> Well, if ICE is part of the browser we could condition sending such da=
ta
>> on the successful termination of ICE processing with the intended
>> destination. Same as with RTP. Wouldn't this work?
>>
>> Emil
>>
>>> Assuming that the goal is not to send arbitrary data, then we need to=

>>> dig into the transport requirements more.
>>>
>>> For example, is the non-media data to be synchronized with media (e.g=
=2E
>>> real-time text)?
>>>
>>> Is there a session associated with the non-media data (e.g. XMPP or M=
SRP
>>> exchanges)?
>>>
>>> Is there a reliability requirement?
>>>
>>> Is it congestion-controlled?
>>>
>>> How long-lived are the flows?
>>>
>>>
>>>
>>>
>>> ---------------------------------------------------
>>> From: magnus.westerlund@ericsson.com
>>> To: rtcweb@ietf.org
>>> Date: Mon, 27 Jun 2011 09:36:30 +0200
>>> Subject: [rtcweb] Non-media data service consensus and requirements
>>>
>>> WG,
>>>
>>> At the interim it was planned to have a bit discussion on the datagra=
m
>>> service for RTCWEB. The first question to try to resolve if there
>>> is consensus for including some form of non real-time media (i.e. not=

>>> audio, video) service between peers. This is a bit tangled with the
>>> actual requirements and use cases. But there was views both for it an=
d
>>> against it on the mailing list. So lets continue and try to come to a=

>>> conclusion on this discussion.
>>>
>>> The use cases mentioned on the mailing list are:
>>>
>>> - Dynamic meta data for Conference and other real-time services
>>>
>>> - Gaming data with low latency requirements
>>>
>>> Does anyone like to add additional use cases?
>>>
>>> Based on my personal understanding this points to primarily have the
>>> RTCWEB provide a unreliable datagram service. This clearly needs
>>> additional requirements to be secure and safe to deploy, but more abo=
ut
>>> this below. I still like to ask the WG here a question.
>>>
>>> Are you supporting the inclusion of a unreliable datagram service
>>> directly between peers? Please provide your view and any additional
>>> statements of motivation that you desire to provide.
>>>
>>> Secondly, there is a question if there needs to have something that
>>> provides reliable message (of arbitrary size) or byte stream oriented=

>>> data transport between the peers. I personally foresee that people wi=
ll
>>> build JS libraries for this on top of a unreliable datagram service. =
If
>>> you desire reliable data service as part of the standardized solution=

>>> please provide motivation and use case and requirements.
>>>
>>> I also want to take a stab on what I personally see as the requiremen=
ts
>>> that exist on unreliable datagram service in the context of RTCWEB.
>>>
>>> - Unreliable data transmission
>>> - Datagram oriented
>>>     * Size limited by MTU
>>>       - Path MTU discovery needed
>>>     * Fragmentation by the application
>>> - Low latency, i.e. Peer to Peer preferable
>>> - Congestion Controlled, to be
>>>     * Network friendly
>>>     * Not become a Denial of Service tool
>>> - Security
>>>    * Confidentiality
>>>    * Integrity Protected
>>>    * Source Authenticated (at least bound to the signalling peer)
>>>    * Ensure consent to receive data
>>>
>>> Please debate the above. This is an attempt to ensure that we can
>>> establish WG consensus on both data service and any requirements.
>>>
>>> cheers
>>>
>>> Magnus Westerlund
>>>
>>> ---------------------------------------------------------------------=
-
>>> Multimedia Technologies, Ericsson Research EAB/TVM
>>> ---------------------------------------------------------------------=
-
>>> Ericsson AB                | Phone  +46 10 7148287
>>> F=C3=A4r=C3=B6gatan 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
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

--=20
Emil Ivov, Ph.D.                       67000 Strasbourg,
Project Lead                           France
Jitsi
emcho@jitsi.org                        PHONE: +33.1.77.62.43.30
http://jitsi.org                       FAX:   +33.1.77.62.47.31


From magnus.westerlund@ericsson.com  Tue Jun 28 07:01:07 2011
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On 2011-06-28 14:52, Igor Faynberg wrote:
> At the interim meeting,  there was a strong argument for NOT having ICE 
> as part of the browser, and I remember that no one objected.  My reading 
> is that it has been the decision.

Yes, there was no such consensus decision in the interim meeting. We
haven't done a call on the list either at this point.

There was a clear question if one could implement ICE in a reasonable
way in a JS environment and meet the timing needs.

There has also been discussion on the mailing list after the Interim
that questions interoperability implications of the proposal.

So there are open questions. Please continue the various aspects so that
we can can come to a conclusion.

Cheers

Magnus Westerlund

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


From msimoni@gmail.com  Tue Jun 28 07:18:46 2011
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On Mon, Jun 27, 2011 at 9:36 AM, Magnus Westerlund
<magnus.westerlund@ericsson.com> wrote:
> At the interim it was planned to have a bit discussion on the datagram
> service for RTCWEB. The first question to try to resolve if there
> is consensus for including some form of non real-time media (i.e. not
> audio, video) service between peers. This is a bit tangled with the
> actual requirements and use cases. But there was views both for it and
> against it on the mailing list. So lets continue and try to come to a
> conclusion on this discussion.
>
> The use cases mentioned on the mailing list are:
>
> - Dynamic meta data for Conference and other real-time services
>
> - Gaming data with low latency requirements
>
> Does anyone like to add additional use cases?

Hello,

I don't understand the networking details, but I am sure that people
will want to transfer files between peers, and if only an unreliable
service is provided, people *will* implement it on top of the
unreliable service. Badly, probably.

Manuel Simoni

From magnus.westerlund@ericsson.com  Tue Jun 28 07:33:08 2011
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On 2011-06-28 16:18, Manuel Simoni wrote:
> On Mon, Jun 27, 2011 at 9:36 AM, Magnus Westerlund
> <magnus.westerlund@ericsson.com> wrote:
>> At the interim it was planned to have a bit discussion on the datagram
>> service for RTCWEB. The first question to try to resolve if there
>> is consensus for including some form of non real-time media (i.e. not
>> audio, video) service between peers. This is a bit tangled with the
>> actual requirements and use cases. But there was views both for it and
>> against it on the mailing list. So lets continue and try to come to a
>> conclusion on this discussion.
>>
>> The use cases mentioned on the mailing list are:
>>
>> - Dynamic meta data for Conference and other real-time services
>>
>> - Gaming data with low latency requirements
>>
>> Does anyone like to add additional use cases?
> 
> Hello,
> 
> I don't understand the networking details, but I am sure that people
> will want to transfer files between peers, and if only an unreliable
> service is provided, people *will* implement it on top of the
> unreliable service. Badly, probably.

Manual, as I asked lower down in the email:

> Secondly, there is a question if there needs to have something that
> provides reliable message (of arbitrary size) or byte stream oriented
> data transport between the peers. I personally foresee that people will
> build JS libraries for this on top of a unreliable datagram service. If
> you desire reliable data service as part of the standardized solution
> please provide motivation and use case and requirements.
> 

When it comes to reliable data I interpret you being a supporter for it.
However, can you elaborate on any usages you see for reliable object in
web-applications to try to see if there are any requirements we are
missing.

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
----------------------------------------------------------------------


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Here is our rough idea for an agenda. Let us know if you want to present =
a draft in one of the section or if you think we need change the =
section. This was based off our document split.=20

Thanks - Cullen, Magnus, and Ted=20


Draft Agenda - Version 1
         =20
         =20
          Agenda Bash
         =20
          Status report (Chairs) - 10 min
         =20
          Liaison Report from W3C WebRTC -(WebRTC Chairs) - 10 min
          The W3C will have just had a meeting the Saturday before
         =20
          Implementation Experience - 20 min
          - short little update on what people are implementing
          - discussion of things to help such as an implementors list
          - do we need an interoperability event later in year
          - expect to have 5 minute updates from 3 or more groups
         =20
          Use Cases ( 30 min )
          - we expect to have a WG draft by the meeting
         =20
          Architecture and System Overview ( 30 min)
          - expect to have have 1 draft here
         =20
          Security ( 20 min )
          - expect to have 1 draft here
         =20
         =20
         =20
          Start of Session two (Chairs) - 5 min
         =20
          Negotiation & signaling (25 min)
          - expect to have 1 draft and wide variety of opinions here
         =20
          NAT ( 20 min)
          - expect to have 1 draft here
         =20
          Media transport (30 min)
          - would like to to come to a decision on the architecture =
approach to number of UDP ports used for media and how, if any, =
multiplexing works
          - expect to have 2 drafts here
         =20
          Datagram transports (20 min)
          - expect to have 1 to 2 drafts here
         =20
          Media processing & codec (20 min)
          - expect to have at last 1 draft here
         =20
         =20=

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RTCWEB Interim Meeting Minutes
June 8, 2011
Chairs: Magnus, Cullen, Ted
Session recording url: Will be provided in later email to list.=20
Jabber: rtcweb@jabber.ietf.org
Materials: http://trac.tools.ietf.org/wg/rtcweb/trac/wiki

Agenda:

Architecture (discussion led by Harald)
Use Cases and Solution Requirements (discussion led by Christer)
Security (discussion led by EKR and Alan)
NAT Traversal (discussion led by Matthew)
RTP Usage (discussion led by Colin)
Non audio/video data (discussion led by Chairs)
Wrap Up

Note well reviewed
Harald reviewed slides at:
=
http://trac.tools.ietf.org/wg/rtcweb/trac/attachment/wiki/WikiStart/june-a=
rch-preso.pdf

Question raised: Harald mentions =93telephone events=94. What does that =
mean?
Cullen replied that this is DTMF. It is mostly used in PSTN gateway =
situations.
Andy Hutton: Is interoperability in the motherhood and apple pie slide =
set?
Harald's response: if we get browser-to-browser running in high quality, =
then direct interoperability may not be a requirement (achieving =
interoperability with a gateway, rather than =93cluttering up=94 the =
browser implementation).
Magnus asks whether anyone believes that Gear case 1 is possible to =
achieve, or are we looking at Gear case 2? Harald says that this is =
theoretically possible by software upgrade or for new phones. Cullen =
says that this may happen, but it is likely to be new/newer devices =
(capable of supporting ICE and the appropriate codecs). Those two things =
are the minimum capabilities, in Cullen's opinion. JDR notes that it is =
possible to use RTCP receive report in a way similar to a stun =
connectivity check, but Matthew notes that there are security issues =
with that substitution. The group agrees that avoiding Voice Hammer and =
cross-protocol attacks is required; if there are ways to do that without =
ICE, fine, but those are key requirements.

Christer presents the slides at:

=
http://trac.tools.ietf.org/wg/rtcweb/trac/attachment/wiki/WikiStart/RTCWEB=
_Presentation_Interim0611.ppt

Use case: Simple Video had no objections
Use case: Multiparty video had no objection on the core requirement, but =
clarification is needed.
	JDR: to be clear, this use case presumes the browser does audio =
mixing
	Christer: yes, but it does not re-emit the mixed stream.
	Cullen: is there a protocol difference from Simple Video?
	Christer: No, not much difference, but important to capture, =
because other use cases have more 	complexity in media streams =
(variable size, etc.) Assumption has full mesh of media 	=
connections.
	Keith Drage: does each browser set up each mesh independently, =
or does the protocol do 	something to set up the mesh?
	Christer: it is still pairwise set up.
	Markus Isomaki: Does the browser have a standard API for this?
	Christer: yes.
	Magnus: we need to clarify different parts of this use case. =
There are no big objections, but the 	difference between this and =
Simple Video is not that clear.
	EKR: There are transport issues and media issues. =46rom the =
transport side, the mesh has 	characteristics similar to the Simple =
Video. The media mixing has more complexity, as it needs
	to describe how to the streams interact.
	Magnus: are there objects to the core requirement? None heard. =
More discussion of the scope 	needed.
	Markus: some kind of audio mixing API is needed. How do we get =
the W3C to agree?
	Christer: there is lots of stuff that has to be communicated.

AI: to the chairs to communicate the need for an audio mixing API to the =
W3C.

Use case: multiparty online game
 - Comments that this is very similar to previous use case where there =
is mixing of other sources of audio (the game in this case)=20
- No objection to this use case=20

Use Case: Video conference system with central server
Keith: How does this use case interact with existing IETF centralized =
conferencing solutions? How much is the centralized sever in this use =
expect to be consistent with existing IETF protocols?
Christer: this leads to a discussion about what protocols the browser =
might need to support, RTCP, BFCP, XCON, etc ...=20
??? : The idea of sending two resolutions of video is not a very broad =
requirement

???: we need to specify things like how to map SSRC to speakers so we =
can see active speakers. We may need things like inter refresh.=20

Conclusion: Need to refine this use case to so that we can get to the =
issues of what is key and what is not=20

Usec ase: Hockey - skipped

Use case: Telephony terminal=20


Moving on to Security Presentation from Eric Rescorla (at 1:48 in =
recording)
Comment from Dan Burnett: Need ability to revoke permission (on slide =
13). No disagreement with this.=20

At 2:16  discussion about wiretap support. JDR said some people want to =
LI some don=92t. Ted pointed out existing Pen and Wire Tap law were not =
retrospective.=20

Mathew - want to clarify slides that we should make it so if you trust =
your calling service, then you know your media is OK. (recording 2:28)=20=


Point raised that allowing a user to see some sort of fingerprint of the =
audio security is fine. If the users has some out of band method to =
communicate that to the far side, they can detect MITM. How they might =
communicate it to the far side not something we would figure out here.=20=


Multiple people  argued for support of RTP and SRTP. Then argued for =
SDES and DTLS-SRTP. There should be a way to get a string that =
fingerprints the media security and a way to display that in the browser =
chrome.=20

Need to deal with tabbed browsing - so someone has a conversation open, =
switches tab, then forgets some tab is involved with a call.=20

RTCWeb Media Privacy presentation: (starting at 3:02 in recording)


Alan, voice is sensitive and important. Also remember that we are =
defining the phone sevice for this century. A bit amazed that we are =
talking about doing non SRTP. Adds negotiation complexities. Going =
through his slides.

Kauffman: Can the signal channel authenticate the ZRTP exchange?=20
Alan: Yes, the hash can be passed on the signal channel and verify the =
whole exchange.

EKR: Why is key-continute work better than a finger print?
Alan: Not in itself, with the SAS one can verify the key continuty chain =
in that instance.
EKR: Still an issue as key continuty will not work if you change =
devices. This enables the attacker to insert a new user identity that is =
similar and which isn=92t is verified. Thus enabling doing a MITM.=20

The SAS is an exceptionall mechanism for a small user group.=20
Ted: Summarizing that there clearly are interest in the media privacy. =
But there are feedback and perceived issues with what can be =
accomplished.=20

NAT Traversal Presentation: (Starting at time ~3:17 into session)
Kauffman presenting.

Magnus: How do you accomplish interoperability if the two end-points are =
using different applications?=20
Kauffman: The services that interop needs to agree on what mechanism to =
use.=20

Cullen: What about pacing?=20
The pacing between multiple tabs doing pace are independent if there is =
ICE or just the proposal.=20

EKR: It is not clearer that a global update is quicker with the =
proposal, rather than in the browser deployement.=20

EKR: Can you actually do this accurately enough for this to work?=20
No one has tested it.  Harald, show me the java script that can do this =
against a regular ICE.

Cullen: Moving on=20

RTP Usage: Starting at time 3:35 of the session
Colin Perkins presenting.

Discussion about multiplexing of audio and video on the same RTP =
session.=20

Arguments between keeping the architecture and have interoperability =
with legacy and saving port state for flows.=20

If QoS is sacrificable then a explicit shimming layer is a solution that =
maintains the session separation.=20

Some arguments for supporting a legacy mode with separate sessions and =
the possibility to put both on the same session.=20

If the general issue of allowing multiple RTP sessions on the same =
transport flow, then this is a general issue and discussion should be in =
AVTCORE and not RTCWEB specific.













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--20cf305642179766f504a6c7d89b
Content-Type: text/plain; charset=ISO-8859-1

On Tue, Jun 28, 2011 at 8:15 AM, Cullen Jennings <fluffy@cisco.com> wrote:

>
>
>          Media processing & codec (20 min)
>          - expect to have at last 1 draft here
>
>
>
To clarify what we're looking for here, we would like to have discussion of
the technical requirements for real time media (e.g. audio, video, real time
text) that arise from the use cases that the group is setting out.
Discussion on the mailing list and/or any drafts folks want to write on this
are highly encouraged, but should focus on the technical requirements.

thanks,

Ted, Cullen, and Magnus

--20cf305642179766f504a6c7d89b
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On Tue, Jun 28, 2011 at 8:15 AM, Cullen Jennings <span dir=3D"ltr">&lt;<a h=
ref=3D"mailto:fluffy@cisco.com">fluffy@cisco.com</a>&gt;</span> wrote:<br><=
div class=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"margin=
: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-lef=
t: 1ex;">
<br>
<br>
 =A0 =A0 =A0 =A0 =A0Media processing &amp; codec (20 min)<br>
 =A0 =A0 =A0 =A0 =A0- expect to have at last 1 draft here<br>
<br><br></blockquote><div><br>To clarify what we&#39;re looking for here, w=
e would like to have discussion of the technical requirements for real time=
 media (e.g. audio, video, real time text) that arise from the use cases th=
at the group is setting out.=A0 Discussion on the mailing list and/or any d=
rafts folks want to write on this are highly encouraged, but should focus o=
n the technical requirements.<br>
<br>thanks,<br><br>Ted, Cullen, and Magnus <br></div></div>

--20cf305642179766f504a6c7d89b--

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Subject: Re: [rtcweb] Non-media data service consensus and requirements
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No, most likely it was I who missed something...  Yes, I remember the 
timer questions,  but I thought Jonathan answered this.

(By the way, I do not have a firm opinion on the subject myself, and I 
don't  understand the issues behind the performance model here.)

Surely, it is too early for the consensus to be formed, but I expect our 
esteemed chairmen to declare it at certain point because the matters 
like that ought to be settled as early as possible.

Igor

On 6/28/2011 9:28 AM, Emil Ivov wrote:
> ÐÐ° 28.06.11 14:52, Igor Faynberg Ð½Ð°Ð¿Ð¸ÑÐ°:
>> At the interim meeting,  there was a strong argument for NOT having ICE
>> as part of the browser, and I remember that no one objected.  My reading
>> is that it has been the decision.
> Well. I did attend the meeting but I remember no such decision or even
> rough consensus. I actually remember people discussing the fact that
> properly implementing ICE in javascript might be tricky due to the timer
> granularity and the pacing requirements in 5245.
>
> Have I missed something?
>
> Emil
>
>> Igor
>>
>>
>> On 6/27/2011 6:43 PM, Emil Ivov wrote:
>>> ...
>>>> For these transactional exchanges the overhead of ICE would be excessive
>>>> and so there will be a very strong temptation to cut corners.
>>> Well, if ICE is part of the browser we could condition sending such data
>>> on the successful termination of ICE processing with the intended
>>> destination. Same as with RTP. Wouldn't this work?
>>>
>>> Emil
>>>
>>>> Assuming that the goal is not to send arbitrary data, then we need to
>>>> dig into the transport requirements more.
>>>>
>>>> For example, is the non-media data to be synchronized with media (e.g.
>>>> real-time text)?
>>>>
>>>> Is there a session associated with the non-media data (e.g. XMPP or MSRP
>>>> exchanges)?
>>>>
>>>> Is there a reliability requirement?
>>>>
>>>> Is it congestion-controlled?
>>>>
>>>> How long-lived are the flows?
>>>>
>>>>
>>>>
>>>>
>>>> ---------------------------------------------------
>>>> From: magnus.westerlund@ericsson.com
>>>> To: rtcweb@ietf.org
>>>> Date: Mon, 27 Jun 2011 09:36:30 +0200
>>>> Subject: [rtcweb] Non-media data service consensus and requirements
>>>>
>>>> WG,
>>>>
>>>> At the interim it was planned to have a bit discussion on the datagram
>>>> service for RTCWEB. The first question to try to resolve if there
>>>> is consensus for including some form of non real-time media (i.e. not
>>>> audio, video) service between peers. This is a bit tangled with the
>>>> actual requirements and use cases. But there was views both for it and
>>>> against it on the mailing list. So lets continue and try to come to a
>>>> conclusion on this discussion.
>>>>
>>>> The use cases mentioned on the mailing list are:
>>>>
>>>> - Dynamic meta data for Conference and other real-time services
>>>>
>>>> - Gaming data with low latency requirements
>>>>
>>>> Does anyone like to add additional use cases?
>>>>
>>>> Based on my personal understanding this points to primarily have the
>>>> RTCWEB provide a unreliable datagram service. This clearly needs
>>>> additional requirements to be secure and safe to deploy, but more about
>>>> this below. I still like to ask the WG here a question.
>>>>
>>>> Are you supporting the inclusion of a unreliable datagram service
>>>> directly between peers? Please provide your view and any additional
>>>> statements of motivation that you desire to provide.
>>>>
>>>> Secondly, there is a question if there needs to have something that
>>>> provides reliable message (of arbitrary size) or byte stream oriented
>>>> data transport between the peers. I personally foresee that people will
>>>> build JS libraries for this on top of a unreliable datagram service. If
>>>> you desire reliable data service as part of the standardized solution
>>>> please provide motivation and use case and requirements.
>>>>
>>>> I also want to take a stab on what I personally see as the requirements
>>>> that exist on unreliable datagram service in the context of RTCWEB.
>>>>
>>>> - Unreliable data transmission
>>>> - Datagram oriented
>>>>      * Size limited by MTU
>>>>        - Path MTU discovery needed
>>>>      * Fragmentation by the application
>>>> - Low latency, i.e. Peer to Peer preferable
>>>> - Congestion Controlled, to be
>>>>      * Network friendly
>>>>      * Not become a Denial of Service tool
>>>> - Security
>>>>     * Confidentiality
>>>>     * Integrity Protected
>>>>     * Source Authenticated (at least bound to the signalling peer)
>>>>     * Ensure consent to receive data
>>>>
>>>> Please debate the above. This is an attempt to ensure that we can
>>>> establish WG consensus on both data service and any requirements.
>>>>
>>>> cheers
>>>>
>>>> Magnus Westerlund
>>>>
>>>> ----------------------------------------------------------------------
>>>> Multimedia Technologies, Ericsson Research EAB/TVM
>>>> ----------------------------------------------------------------------
>>>> Ericsson AB                | Phone  +46 10 7148287
>>>> FÃ¤rÃ¶gatan 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
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb

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Subject: Re: [rtcweb] Non-media data service consensus and requirements
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> No, most likely it was I who missed something...  Yes, I remember the
> timer questions, but I thought Jonathan answered this.

IIRC, Harald suggested that the best way to decide if this was actually 
feasible was with "running code", at least at the proof-of-concept 
level. This seemed like a very good idea to me. The resolution of JS 
timers is not the only issue. One also has to deal with other pages with 
long-running JS scripts that prevent your timers from firing. Various 
advertisers do this on a semi-regular basis, though I don't have good 
numbers on how frequent this is, or how much of a problem it would 
really pose. Running code could be used to actually measure the 
connectivity success rate, for those that want to know.

I don't think the compatibility concerns have been addressed, either. 
With ICE in the browser, you have four or five implementations total 
that have to interoperate, vs. (potentially) one for each website. 
Updates to browsers can be deployed, at scale, to hundreds of millions 
of users in the span of a few weeks (at least with Firefox and Chrome, 
Safari and IE also have well-established short-term update mechanisms, 
though they're usually for security issues rather than new features), 
vs. trying to get each and every website to update their JS library. An 
individual website may be able to deploy updates more quickly than the 
browser vendors, but interoperability isn't about the individual.

And now there are the security concerns with sending datagrams after 
trying to short-cut the ICE process. I don't know if mandating STUN 
requests before sending data to a remote peer is sufficient to address 
these, but clearly some others see a potential problem. Personally, I 
think having p2p datagram support is a much more important feature to 
end users than being able to change the connection handshake protocol in JS.

Everything above is an argument I've heard someone else raise at some 
point. I haven't heard a lot of good responses to these arguments.

From jdrosen@jdrosen.net  Tue Jun 28 12:46:05 2011
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Subject: Re: [rtcweb] Non-media data service consensus and requirements
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Here are some thoughts on additional use cases:

* information on congestion and receiver side environment for purposes 
of improved sender rate adaptation

* active speaker indications

* real-time text (along the lines of 
http://datatracker.ietf.org/doc/rfc5194/)

* video controls - application driven intra-frame requests, for example

* floor control requests/grants


All of these share the common property of being latency sensitive, and 
thus more appropraite for p2p vs. server mediated. The other driver for 
p2p data transport is cost reduction, to avoid the need for server 
mediation for transfer of data. File transfer is the main example of this.

I view the latency-sensitive use cases as a higher priority for us and a 
better match to the work we're doing here.

Thanks,
Jonathan R.

On 6/27/2011 3:36 AM, Magnus Westerlund wrote:
> WG,
>
> At the interim it was planned to have a bit discussion on the datagram
> service for RTCWEB. The first question to try to resolve if there
> is consensus for including some form of non real-time media (i.e. not
> audio, video) service between peers. This is a bit tangled with the
> actual requirements and use cases. But there was views both for it and
> against it on the mailing list. So lets continue and try to come to a
> conclusion on this discussion.
>
> The use cases mentioned on the mailing list are:
>
> - Dynamic meta data for Conference and other real-time services
>
> - Gaming data with low latency requirements
>
> Does anyone like to add additional use cases?
>
> Based on my personal understanding this points to primarily have the
> RTCWEB provide a unreliable datagram service. This clearly needs
> additional requirements to be secure and safe to deploy, but more about
> this below. I still like to ask the WG here a question.
>
> Are you supporting the inclusion of a unreliable datagram service
> directly between peers? Please provide your view and any additional
> statements of motivation that you desire to provide.
>
> Secondly, there is a question if there needs to have something that
> provides reliable message (of arbitrary size) or byte stream oriented
> data transport between the peers. I personally foresee that people will
> build JS libraries for this on top of a unreliable datagram service. If
> you desire reliable data service as part of the standardized solution
> please provide motivation and use case and requirements.
>
> I also want to take a stab on what I personally see as the requirements
> that exist on unreliable datagram service in the context of RTCWEB.
>
> - Unreliable data transmission
> - Datagram oriented
>     * Size limited by MTU
>       - Path MTU discovery needed
>     * Fragmentation by the application
> - Low latency, i.e. Peer to Peer preferable
> - Congestion Controlled, to be
>     * Network friendly
>     * Not become a Denial of Service tool
> - Security
>    * Confidentiality
>    * Integrity Protected
>    * Source Authenticated (at least bound to the signalling peer)
>    * Ensure consent to receive data
>
> Please debate the above. This is an attempt to ensure that we can
> establish WG consensus on both data service and any requirements.
>
> 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

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



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On 06/28/2011 08:34 AM, Cullen Jennings wrote:
> [...] The group agrees that avoiding Voice Hammer and cross-protocol attacks is required; if there are ways to do that without ICE, fine, but those are key requirements.
>    
If ICE relates to X11's ICE (which I doubt on specific screens and 
drivers), then may I hint at the mote method of IPv4 "land" inside IPv6 
(or other protocol) "mote". The concept is ridiculous in single core 
systems with centralized-peripheral, yet may be cents for those that 
keep hardwired bridges. The key point is the is-"land" only knows IPv4, 
so remains unaffected by its float among IPv6.

I wouldn't doubt the mechanics of ICE-walls continue to stretch so far 
into court to quietly determine "this" is the same as "this" in the 
business or not (i.e. stationary battery industry and mobile drive 
systems, or mobile battery industry and stationary drive systems, ... 
endless).

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From dzonatas@gmail.com  Tue Jun 28 13:49:17 2011
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Subject: Re: [rtcweb] Non-media data service consensus and requirements
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Please consider something like "augmented waypoints" for evacuation 
systems (might help resolve stampedes).

On 06/28/2011 12:45 PM, Jonathan Rosenberg wrote:
> Here are some thoughts on additional use cases:
>
> * information on congestion and receiver side environment for purposes 
> of improved sender rate adaptation
>
> * active speaker indications
>
> * real-time text (along the lines of 
> http://datatracker.ietf.org/doc/rfc5194/)
>
> * video controls - application driven intra-frame requests, for example
>
> * floor control requests/grants
>
>
> All of these share the common property of being latency sensitive, and 
> thus more appropraite for p2p vs. server mediated. The other driver 
> for p2p data transport is cost reduction, to avoid the need for server 
> mediation for transfer of data. File transfer is the main example of 
> this.
>
> I view the latency-sensitive use cases as a higher priority for us and 
> a better match to the work we're doing here.
>
> Thanks,
> Jonathan R.
>
> On 6/27/2011 3:36 AM, Magnus Westerlund wrote:
>> WG,
>>
>> At the interim it was planned to have a bit discussion on the datagram
>> service for RTCWEB. The first question to try to resolve if there
>> is consensus for including some form of non real-time media (i.e. not
>> audio, video) service between peers. This is a bit tangled with the
>> actual requirements and use cases. But there was views both for it and
>> against it on the mailing list. So lets continue and try to come to a
>> conclusion on this discussion.
>>
>> The use cases mentioned on the mailing list are:
>>
>> - Dynamic meta data for Conference and other real-time services
>>
>> - Gaming data with low latency requirements
>>
>> Does anyone like to add additional use cases?
>>
>> Based on my personal understanding this points to primarily have the
>> RTCWEB provide a unreliable datagram service. This clearly needs
>> additional requirements to be secure and safe to deploy, but more about
>> this below. I still like to ask the WG here a question.
>>
>> Are you supporting the inclusion of a unreliable datagram service
>> directly between peers? Please provide your view and any additional
>> statements of motivation that you desire to provide.
>>
>> Secondly, there is a question if there needs to have something that
>> provides reliable message (of arbitrary size) or byte stream oriented
>> data transport between the peers. I personally foresee that people will
>> build JS libraries for this on top of a unreliable datagram service. If
>> you desire reliable data service as part of the standardized solution
>> please provide motivation and use case and requirements.
>>
>> I also want to take a stab on what I personally see as the requirements
>> that exist on unreliable datagram service in the context of RTCWEB.
>>
>> - Unreliable data transmission
>> - Datagram oriented
>>     * Size limited by MTU
>>       - Path MTU discovery needed
>>     * Fragmentation by the application
>> - Low latency, i.e. Peer to Peer preferable
>> - Congestion Controlled, to be
>>     * Network friendly
>>     * Not become a Denial of Service tool
>> - Security
>>    * Confidentiality
>>    * Integrity Protected
>>    * Source Authenticated (at least bound to the signalling peer)
>>    * Ensure consent to receive data
>>
>> Please debate the above. This is an attempt to ensure that we can
>> establish WG consensus on both data service and any requirements.
>>
>> cheers
>>
>> Magnus Westerlund
>>
>> ----------------------------------------------------------------------
>> Multimedia Technologies, Ericsson Research EAB/TVM
>> ----------------------------------------------------------------------
>> Ericsson AB                | Phone  +46 10 7148287
>> Fï¿½rï¿½gatan 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 06/28/2011 10:06 AM, Timothy B. Terriberry wrote:
>
> And now there are the security concerns with sending datagrams after 
> trying to short-cut the ICE process.
>

What happens is that the application (of ICE systems) creates the 
delusion the previous path is no longer of need. The ICE system itself 
doesn't do that, as newer adapters are unfamiliar with the previous 
path. The security concern is in the question to end the previous path. 
Some defaulted the previous path "off" after application, which made 
related evolution harder... proven human error when that kind of 
security level (prevent evolution?) is not needed. It works best to 
distinguish versions and with things meant-for standardization (towards 
hardware, or towards plasma-stasis in quantum physics).

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From fluffy@cisco.com  Tue Jun 28 20:28:32 2011
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Subject: [rtcweb] Recordings from last interim meeting
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Here are the recordings from the last interim meeting. The first half of =
the meeting before the break is in the "PartA" files while the second =
half of the meeting after the break is in the "PartB" files. YMMV but =
mac users will probably haver better luck using the .mov files while =
windows users will probably do better with the .mp4 files.

http://dl.dropbox.com/u/17089001/IETF-RTCWeb-80.5-PartA.mov
http://dl.dropbox.com/u/17089001/IETF-RTCWeb-80.5-PartA.mp4

http://dl.dropbox.com/u/17089001/IETF-RTCWeb-80.5-partB.mov
http://dl.dropbox.com/u/17089001/IETF-RTCWeb-80.5-partB.mp4



From randell1@jesup.org  Tue Jun 28 21:15:14 2011
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Subject: Re: [rtcweb] Non-media data service consensus and requirements
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On 6/28/2011 3:45 PM, Jonathan Rosenberg wrote:
> Here are some thoughts on additional use cases:
>
> * information on congestion and receiver side environment for purposes 
> of improved sender rate adaptation

Aka active bandwidth adaptation.  This will generally be better over 
RTCP or other unreliable-but-low-latency channel.

One thing we may want to address is handling congestion across a set of 
streams, instead of only a single stream.  Current mechanisms are 
stream-oriented, which both can cause unexpected conflicts between 
related streams (audio and video for example), and also allows 
"cheating" congestion rules by opening multiple 'connections' (streams).

This was one thing that RTMFP got right, along with allowing reliable 
datagram services.  As mentioned, if it's not provided then N*M people 
will try to implement reliable transport over whatever streams we give 
them (including random RTP streams), and as someone said, they'll make 
mistakes.

>
> * active speaker indications
>
> * real-time text (along the lines of 
> http://datatracker.ietf.org/doc/rfc5194/)
>
> * video controls - application driven intra-frame requests, for example

I suspect these should be dealt with using normal rtcpfb mechanisms, 
there there may be needs for a reliable transport of this request.

>
> * floor control requests/grants
>

And as was mentioned user input events in a VNC/RDP-type remote 
browser-sharing.  (and yes, there are security issues here!)  These are 
both latency-sensitive and must be reliable (or at least a subset of 
events must be reliable, and probably all do).

>
> All of these share the common property of being latency sensitive, and 
> thus more appropraite for p2p vs. server mediated. The other driver 
> for p2p data transport is cost reduction, to avoid the need for server 
> mediation for transfer of data. File transfer is the main example of 
> this.
>
> I view the latency-sensitive use cases as a higher priority for us and 
> a better match to the work we're doing here.

To be clear: I'm in favor of considering non-media unreliable and 
reliable transports.

-- 
Randell Jesup
randell-ietf@jesup.org


From gonzalo.camarillo@ericsson.com  Tue Jun 28 23:47:50 2011
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Subject: Re: [rtcweb] W3C Web Real-Time Communications face-to-face on 23 July 2011 in Quebec City, Canada
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Hi,

we have scheduled this meeting on the Saturday before the IETF (July
23rd) from 14:00 to 18:00. The meeting will take place at the Quebec
City Convention Center (i.e., the IETF venue).

Please, register early so that we can accurately estimate how many
attendees we will have in order to book an appropriate meeting room.
Tentatively, we have reserved room 2103, which seems to be able to seat
around 60 people.

Cheers,

Gonzalo


On 21/06/2011 4:57 PM, Francois Daoust wrote:
> Dear participants in the IETF RTCWEB group,
> 
> The W3C Web Real-Time Communications working group will meet on Saturday 23 July 2011 afternoon, starting at or shortly after 1PM, in Quebec City, Canada, co-located with the IETF Meeting. Exact time range, agenda and location will be announced once known.
> 
> The choice of the date and place is meant to favor synergies between the W3C WebRTC and the IETF RTCWEB groups. Participants in the IETF RTCWEB group are welcome to attend this W3C face-to-face as meeting guests.
> 
> To facilitate logistics purpose, I have set up a registration form, available at:
>   http://www.w3.org/2002/09/wbs/47318/webrtc-f2f-quebec/
> 
> Anyone can fill out the form.
> People willing to attend the F2F need to register before 15 July 2011 (the sooner, the better).
> 
> Please let me know if you have questions.
> 
> Cheers,
> Francois Daoust,
> W3C Team Contact for the W3C WebRTC WG.
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> 


From magnus.westerlund@ericsson.com  Wed Jun 29 00:39:28 2011
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Hi,

As an individual.

On 2011-06-28 21:45, Jonathan Rosenberg wrote:
> Here are some thoughts on additional use cases:
> 
> * information on congestion and receiver side environment for purposes 
> of improved sender rate adaptation

I believe it is important that we have congestion control for the point
to point flows for both RTP and non-media data that are implemented and
enforced in the browser. This is a pure security and personal hygien
question. A rouge application should not be able to consume completely
unproportional bit-rates between two peers and be able to starve out
applications sharing the bottleneck.

Having said that there can clearly be need for application level signals
for changing behavior and adopting to what bit-rate or delay
characteristics that are available. Especially if you have star or other
multi-hop topologies in the application interconnections. For RTP we
already propose some such signals like TMMBR [RFC5104] in our RTP for
RTCWEB document.

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 magnus.westerlund@ericsson.com  Wed Jun 29 00:39:54 2011
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Subject: Re: [rtcweb] WG adoption of Architecture and Overview of RTCWEB document
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WG,

we have received no objections against our proposal and hereby adopt

https://datatracker.ietf.org/doc/draft-alvestrand-rtcweb-overview/

as a WG item.

Cheers

Magnus Westerlund
WG chair

On 2011-06-21 17:22, Magnus Westerlund wrote:
> WG,
> 
> We would like to adopt a document that describes the RTCWEB architecture
> and provides an overview to the functionality that will be specified in
> various documents targeted to specific pieces of the functionalities we
> need for a complete solution.
> 
> There has been several documents that describes architecture, frame
> works or overviews. None of them are complete when it comes to covering
> the different aspects. The documents are:
> 
> https://datatracker.ietf.org/doc/draft-alvestrand-rtcweb-overview/
> https://datatracker.ietf.org/doc/draft-rosenberg-rtcweb-framework/
> https://datatracker.ietf.org/doc/draft-jennings-rtcweb-api/
> 
> The WG chairs' proposal is that we adopt one document and then add in
> aspects that are needed from the other documents. In the process putting
> together an good editor team for the WG document from the persons that
> has contributed so far and want to continue.
> 
> The WG chairs propose that that the WG adopts:
> https://datatracker.ietf.org/doc/draft-alvestrand-rtcweb-overview/
> as a WG item to form the core of an Architecture and Overview document.
> 
> WG participants please provide either support for adoption or provide
> feedback on any issues you see with this approach.
> 
> Please provide your view no later than Tuesday the 28th at 17.00 CEST.
> 
> Magnus Westerlund
> WG Chair
> 
> ----------------------------------------------------------------------
> 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
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From stefan.lk.hakansson@ericsson.com  Wed Jun 29 03:50:30 2011
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Subject: [rtcweb] Comments to use-cases/reqs document
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All,

some comments to
<http://datatracker.ietf.org/doc/draft-ietf-rtcweb-use-cases-and-requirements/?include_text=1>

(yes, I am one of the authors, but the document now reflects what had
been discussed on mail list up to last week)

1. The "Simple video communication service with inter-operator calling"
use case derives the requirement "F25: The browser MUST support a
baseline audio and video codec". While I think this requirement makes
sense it seems a bit odd in this context. To me it rather belongs to a
case where the users use different browsers (and perhaps different
communication devices using different OSes).

2. Some clarifications should be added to "Multiparty video 
communication" and "Video conferencing system with central server" (as 
pointed out in the Minutes from the interim).

3. The "Video Size Change" use case derives F22; but that is already
derived in "Simple Video Communication Service". I propose to remove the 
"Video Size Change" use case.

4. The "Fedex Call" use case derives the DTMF req F19 "The browser must
be able to insert DTMF signals in a media stream". I think we made a
mistake here, what Cullen said was "there should be a way to navigate
the IVR". I think this is a more relevant requirement at this stage. I
think the API reg A16 should be removed as it is not clear if this would 
be done via an API.

5. To me it would make sense to add the NIC change and QoS use cases as
extensions to the "Simple video communication service" use case (as is
done with "inter-operator calling".

6. I think the structure is a bit strange. E.g. the "Multiparty on-line
game with voice communication" use case is not under browser-to-browser
even though it is clearly browser-to-browser.

So, this is what I propose the structure to look like:



     4.  Use-cases
       4.1.  Introduction
       4.2.  Browser-to-browser use-cases
         4.2.1.  Simple Video Communication Service
         4.2.2.  Simple Video Communication Service, access change
         4.2.3.  Simple Video Communication Service, QoS
         4.2.4.  Simple video communication service with
                 inter-operator calling
         4.2.5.  Hockey Game Viewer
         4.2.6.  Multiparty video communication
         4.2.7.  Multiparty on-line game with voice communication
       4.3.  Browser - GW/Server use cases
         4.3.1.  Telephony terminal
         4.3.2.  Fedex Call
         4.3.3.  Video conferencing system with central server

With the following changes:
4.2.1 Extended to say that different browsers (and perhaps operating
systems) are used, F25 derived here

4.2.6 Clarifications in text
4.3.3 Clarifications in text

F19 Changed to talk about navigating in IVR. A16 (DTMF API) removed.

F23 (change netw I/F) derived in 4.2.2

F21 (QoS) derived in 4.2.3

F24 (SIP able to carry session setup data) derived in 4.2.4

Comments?

Stefan

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On Tue, Jun 28, 2011 at 4:33 PM, Magnus Westerlund
<magnus.westerlund@ericsson.com> wrote:
> When it comes to reliable data I interpret you being a supporter for it.

Yes.

> However, can you elaborate on any usages you see for reliable object in
> web-applications to try to see if there are any requirements we are
> missing.

A use case would be an audio/video chat program, with the ability to
transfer files/documents between peers without needing server-side
storage.

From dzonatas@gmail.com  Wed Jun 29 07:28:17 2011
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From: Dzonatas Sol <dzonatas@gmail.com>
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Subject: Re: [rtcweb] Non-media data service consensus and requirements
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On 06/29/2011 12:39 AM, Magnus Westerlund wrote:
> Hi,
>
> As an individual.
>
> On 2011-06-28 21:45, Jonathan Rosenberg wrote:
>    
>> Here are some thoughts on additional use cases:
>>
>> * information on congestion and receiver side environment for purposes
>> of improved sender rate adaptation
>>      
> I believe it is important that we have congestion control for the point
> to point flows for both RTP and non-media data that are implemented and
> enforced in the browser. This is a pure security and personal hygien
> question. A rouge application should not be able to consume completely
> unproportional bit-rates between two peers and be able to starve out
> applications sharing the bottleneck.
>    

I would suggest instead of "for purposes of improved sender rate 
adaption" to "for purposes of manifold adaption", yet that may confuse 
people that relate usage outside such interface topology.

> Having said that there can clearly be need for application level signals
> for changing behavior and adopting to what bit-rate or delay
> characteristics that are available. Especially if you have star or other
> multi-hop topologies in the application interconnections. For RTP we
> already propose some such signals like TMMBR [RFC5104] in our RTP for
> RTCWEB document.
>    

I couldn't find the link off-hand, yet there exists successful non-stop 
busy intersection simulations. The downfall with those simulations 
happens when they assume every vehicle is signal aware. Such simulations 
do not include trains or multi-trailers. This relates, however, in 
multi-hop when viewed from intersection to intersection. One may want to 
demand multi-hop to relieve traffic congestion, which may involve 
disconnection from star-topologies. What star-topologies want is how to 
keep-alive the physical tether (simulated or not) through such temporal 
disconnection. Each star-topology has one centralized physical tether no 
matter how they divide the coverage (cellular). Besides credit systems 
for that temporary digital-divide (gap), the best use-case I have seen 
for such physical tether is games that require centralized physic 
simulation to avoid cheaters (or train wrecks). The law of conservation 
is the alternative for less-than-best credit conversion (since the size 
of each bit is unknown to the observer) while disconnected. ;)

-- 
--- http://twitter.com/Dzonatas_Sol ---
Web Development, Software Engineering
Ag-Biotech, Virtual Reality, Consultant


From fluffy@cisco.com  Wed Jun 29 08:07:58 2011
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From: Cullen Jennings <fluffy@cisco.com>
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Subject: [rtcweb] Media negeotiation and signalling archatecture
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One of the complicated architectural issues for this WG is what path the =
signaling information gets passed over, and in the parts of that that =
path that need to be standardized, what does the signaling information =
looks like. To help get discussion going on that, I have described 3 =
models of how the information could flow in the some pictures in the =
following PDF.=20

=
http://trac.tools.ietf.org/wg/rtcweb/trac/raw-attachment/wiki/WikiStart/RT=
CWeb-Signaling.pdf

I split these up into 3 models that I call High, Mid, and Low based on =
the path the signaling information goes. There has been a fair amount of =
discussion of High and Mid models in the past in this WG thought pretty =
much no discussion of the Low model. The interesting things is that when =
I look at what is getting implemented and prototyped, a lot of it is the =
low model.=20

It certainly possible to support more than one of these but I'm =
interested in the pro and cons of all these but from my point of view, =
one of the key issues is how hard is for people to use. If you look at =
what things have succeeded in HTML, it is typically the things that in =
there simplest form provide a very simple interface to use them. Yes, =
they might have a more complex form that allows richer control but the =
basics are simple. Take the HTML5 video tag for streaming video media =
for example. There are zillion things that could have been passed and =
controlled to this flag yet the proposals that one out were the ones =
that provided something that was incredibly simple to use. Another =
example is the iPhone interface to make a phone call from the browser =
has been very successful from an adoption point of view.=20





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On 06/29/2011 08:07 AM, Cullen Jennings wrote:
>
> http://trac.tools.ietf.org/wg/rtcweb/trac/raw-attachment/wiki/WikiStart/RTCWeb-Signaling.pdf
>
>    

The computer-mouse with built-in printer provides proof of technology in 
usage. The graph paper delegates units of space, bits. I'm surprised 
people have not used the optic-computer-mouse to align what they draw on 
graph paper (to have the process in reverse from screen to paper).

Scalability, so scientists find the scalars and business patents the 
flow to the engineers. Software patents are not bad for such 
standardization, yet this begs the question if low, medium, and high are 
patentable in real-time based on localization. Should we consider the 
word throttle as one scalar? "Materials" is an easier word as 
macro-economies and then micro- prefix means essentially low in such 
relation, without materials.

-- 
--- http://twitter.com/Dzonatas_Sol ---
Web Development, Software Engineering
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From matthew.kaufman@skype.net  Wed Jun 29 16:23:34 2011
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Subject: Re: [rtcweb] Media negeotiation and signalling archatecture
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On Jun 29, 2011, at 8:07 AM, Cullen Jennings wrote:

>=20
> One of the complicated architectural issues for this WG is what path =
the signaling information gets passed over, and in the parts of that =
that path that need to be standardized, what does the signaling =
information looks like. To help get discussion going on that, I have =
described 3 models of how the information could flow in the some =
pictures in the following PDF.=20
>=20
> =
http://trac.tools.ietf.org/wg/rtcweb/trac/raw-attachment/wiki/WikiStart/RT=
CWeb-Signaling.pdf
>=20
> I split these up into 3 models that I call High, Mid, and Low based on =
the path the signaling information goes. There has been a fair amount of =
discussion of High and Mid models in the past in this WG thought pretty =
much no discussion of the Low model. The interesting things is that when =
I look at what is getting implemented and prototyped, a lot of it is the =
low model.=20

I think this deck is a bit in two ways.

First, in the first slide you say (and I agree) that we agree that if =
there's >1 web server we don't care how they talk to each other (I'm =
suspecting something like SIP in most cases)... but then in the "high =
path" you say it "gets complicated to define what happens on the orange =
line"... but I disagree here. If we agree that it is SIP and SDP on the =
orange line then clearly the things that are needed to do NAT traversal =
(ICE) and codec negotiation are all able to be carried in SDP. As =
another alternative you could simply carry something that the browser =
understood natively on that orange path.=20

In fact in the "low path" slide, you talk about the current Chrome =
implementation and the WhatWG proposal, but both of those are actually =
suggesting transporting a JSON blob via the "high path" to do the ICE =
negotiation... and there's no reason to believe that if you can send the =
blob needed to do ICE that you couldn't also do the media type =
negotiation via the same path.

Second, the "high path" is still needed to do the actual call =
signaling... you can't even get the two browsers connected to the two =
different web services interested in setting up a session using ICE if =
there hasn't been communication over the "high path" to set up a call.

Matthew Kaufman




From matthew.kaufman@skype.net  Wed Jun 29 16:38:20 2011
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From: Matthew Kaufman <matthew.kaufman@skype.net>
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--Apple-Mail-7-371012902
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I wrote up a first pass at browser requirements for a media security =
inspector in browsers.

Matthew Kaufman

Begin forwarded message:

> From: internet-drafts@ietf.org
> Date: June 29, 2011 4:35:42 PM PDT
> To: matthew.kaufman@skype.net
> Cc: matthew.kaufman@skype.net
> Subject: New Version Notification for =
draft-kaufman-rtcweb-security-ui-00.txt
>=20
> A new version of I-D, draft-kaufman-rtcweb-security-ui-00.txt has been =
successfully submitted by Matthew Kaufman and posted to the IETF =
repository.
>=20
> Filename:	 draft-kaufman-rtcweb-security-ui
> Revision:	 00
> Title:		 Client Security User Interface Requirements for =
RTCWEB
> Creation date:	 2011-06-30
> WG ID:		 Individual Submission
> Number of pages: 4
>=20
> Abstract:
>   This document calls for a requirement to be imposed on RTCWEB client
>   user interfaces whereby the user may inspect the current media
>   security status.
>=20
>=20
>=20
>=20
> The IETF Secretariat


--Apple-Mail-7-371012902
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<html><head></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">I =
wrote up a first pass at browser requirements for a media security =
inspector in browsers.<div><br></div><div>Matthew =
Kaufman<br><div><div><br><div>Begin forwarded message:</div><br =
class=3D"Apple-interchange-newline"><blockquote type=3D"cite"><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px;"><span style=3D"font-family:'Helvetica'; =
font-size:medium; color:rgba(0, 0, 0, 1);"><b>From: </b></span><span =
style=3D"font-family:'Helvetica'; font-size:medium;"><a =
href=3D"mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a><br><=
/span></div><div style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px;"><span =
style=3D"font-family:'Helvetica'; font-size:medium; color:rgba(0, 0, 0, =
1);"><b>Date: </b></span><span style=3D"font-family:'Helvetica'; =
font-size:medium;">June 29, 2011 4:35:42 PM PDT<br></span></div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px;"><span style=3D"font-family:'Helvetica'; =
font-size:medium; color:rgba(0, 0, 0, 1);"><b>To: </b></span><span =
style=3D"font-family:'Helvetica'; font-size:medium;"><a =
href=3D"mailto:matthew.kaufman@skype.net">matthew.kaufman@skype.net</a><br=
></span></div><div style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px;"><span =
style=3D"font-family:'Helvetica'; font-size:medium; color:rgba(0, 0, 0, =
1);"><b>Cc: </b></span><span style=3D"font-family:'Helvetica'; =
font-size:medium;"><a =
href=3D"mailto:matthew.kaufman@skype.net">matthew.kaufman@skype.net</a><br=
></span></div><div style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px;"><span =
style=3D"font-family:'Helvetica'; font-size:medium; color:rgba(0, 0, 0, =
1);"><b>Subject: </b></span><span style=3D"font-family:'Helvetica'; =
font-size:medium;"><b>New Version Notification for =
draft-kaufman-rtcweb-security-ui-00.txt</b><br></span></div><br><div>A =
new version of I-D, draft-kaufman-rtcweb-security-ui-00.txt has been =
successfully submitted by Matthew Kaufman and posted to the IETF =
repository.<br><br>Filename:<span class=3D"Apple-tab-span" =
style=3D"white-space:pre">	</span> =
draft-kaufman-rtcweb-security-ui<br>Revision:<span =
class=3D"Apple-tab-span" style=3D"white-space:pre">	</span> =
00<br>Title:<span class=3D"Apple-tab-span" style=3D"white-space:pre">	=
</span><span class=3D"Apple-tab-span" style=3D"white-space:pre">	=
</span> Client Security User Interface Requirements for =
RTCWEB<br>Creation date:<span class=3D"Apple-tab-span" =
style=3D"white-space:pre">	</span> 2011-06-30<br>WG ID:<span =
class=3D"Apple-tab-span" style=3D"white-space:pre">	</span><span =
class=3D"Apple-tab-span" style=3D"white-space:pre">	</span> =
Individual Submission<br>Number of pages: 4<br><br>Abstract:<br> =
&nbsp;&nbsp;This document calls for a requirement to be imposed on =
RTCWEB client<br> &nbsp;&nbsp;user interfaces whereby the user may =
inspect the current media<br> &nbsp;&nbsp;security =
status.<br><br><br><br><br>The IETF =
Secretariat<br></div></blockquote></div><br></div></div></body></html>=

--Apple-Mail-7-371012902--

From matthew.kaufman@skype.net  Wed Jun 29 17:50:14 2011
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From: Matthew Kaufman <matthew.kaufman@skype.net>
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I wrote up a method for sending unreliable datagrams to the same port as =
is being used for RTP, RTCP and STUN that does not conflict with this =
usage. It also addresses the security issues involved in allowing =
arbitrary datagrams to be sent.

Matthew Kaufman

Begin forwarded message:

> From: internet-drafts@ietf.org
> Date: June 29, 2011 5:48:52 PM PDT
> To: matthew.kaufman@skype.net
> Cc: matthew.kaufman@skype.net
> Subject: New Version Notification for =
draft-kaufman-rtp-compatible-data-00.txt
>=20
> A new version of I-D, draft-kaufman-rtp-compatible-data-00.txt has =
been successfully submitted by Matthew Kaufman and posted to the IETF =
repository.
>=20
> Filename:	 draft-kaufman-rtp-compatible-data
> Revision:	 00
> Title:		=20
> Creation date:	 2011-06-30
> WG ID:		 Individual Submission
> Number of pages: 5
>=20
> Abstract:
>   This document describes a method for sending unreliable datagrams
>   that are &quot;compatible&quot; with RTP and RTCP, and STUN usage of =
a shared
>   UDP address and port.
>=20
>=20
>=20
>=20
> The IETF Secretariat


--Apple-Mail-8-375326899
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<html><head></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">I =
wrote up a method for sending unreliable datagrams to the same port as =
is being used for RTP, RTCP and STUN that does not conflict with this =
usage. It also addresses the security issues involved in allowing =
arbitrary datagrams to be sent.<div><br></div><div>Matthew =
Kaufman<br><div><br><div>Begin forwarded message:</div><br =
class=3D"Apple-interchange-newline"><blockquote type=3D"cite"><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px;"><span style=3D"font-family:'Helvetica'; =
font-size:medium; color:rgba(0, 0, 0, 1);"><b>From: </b></span><span =
style=3D"font-family:'Helvetica'; font-size:medium;"><a =
href=3D"mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a><br><=
/span></div><div style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px;"><span =
style=3D"font-family:'Helvetica'; font-size:medium; color:rgba(0, 0, 0, =
1);"><b>Date: </b></span><span style=3D"font-family:'Helvetica'; =
font-size:medium;">June 29, 2011 5:48:52 PM PDT<br></span></div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px;"><span style=3D"font-family:'Helvetica'; =
font-size:medium; color:rgba(0, 0, 0, 1);"><b>To: </b></span><span =
style=3D"font-family:'Helvetica'; font-size:medium;"><a =
href=3D"mailto:matthew.kaufman@skype.net">matthew.kaufman@skype.net</a><br=
></span></div><div style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px;"><span =
style=3D"font-family:'Helvetica'; font-size:medium; color:rgba(0, 0, 0, =
1);"><b>Cc: </b></span><span style=3D"font-family:'Helvetica'; =
font-size:medium;"><a =
href=3D"mailto:matthew.kaufman@skype.net">matthew.kaufman@skype.net</a><br=
></span></div><div style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px;"><span =
style=3D"font-family:'Helvetica'; font-size:medium; color:rgba(0, 0, 0, =
1);"><b>Subject: </b></span><span style=3D"font-family:'Helvetica'; =
font-size:medium;"><b>New Version Notification for =
draft-kaufman-rtp-compatible-data-00.txt</b><br></span></div><br><div>A =
new version of I-D, draft-kaufman-rtp-compatible-data-00.txt has been =
successfully submitted by Matthew Kaufman and posted to the IETF =
repository.<br><br>Filename:<span class=3D"Apple-tab-span" =
style=3D"white-space:pre">	</span> =
draft-kaufman-rtp-compatible-data<br>Revision:<span =
class=3D"Apple-tab-span" style=3D"white-space:pre">	</span> =
00<br>Title:<span class=3D"Apple-tab-span" style=3D"white-space:pre">	=
</span><span class=3D"Apple-tab-span" style=3D"white-space:pre">	=
</span> <br>Creation date:<span class=3D"Apple-tab-span" =
style=3D"white-space:pre">	</span> 2011-06-30<br>WG ID:<span =
class=3D"Apple-tab-span" style=3D"white-space:pre">	</span><span =
class=3D"Apple-tab-span" style=3D"white-space:pre">	</span> =
Individual Submission<br>Number of pages: 5<br><br>Abstract:<br> =
&nbsp;&nbsp;This document describes a method for sending unreliable =
datagrams<br> &nbsp;&nbsp;that are &amp;quot;compatible&amp;quot; with =
RTP and RTCP, and STUN usage of a shared<br> &nbsp;&nbsp;UDP address and =
port.<br><br><br><br><br>The IETF =
Secretariat<br></div></blockquote></div><br></div></body></html>=

--Apple-Mail-8-375326899--

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Subject: Re: [rtcweb] W3C Web Real-Time Communications face-to-face on 23 July 2011 in Quebec City, Canada
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Gonzalo,

I (and  other people in my company) had bought a ticket to Quebec in 
advance, according to company policy, with the IETF meeting dates in 
mind.  Changing this now is prohibitively expensive.

I understand that this is a joint meeting with the W3C (and I do think 
it is a very good idea to have it).  But can this be set up on Sunday or 
any other day?
If not, can we have some conferencing arrangements?

With thanks,

Igor



On 6/29/2011 2:47 AM, Gonzalo Camarillo wrote:
> Hi,
>
> we have scheduled this meeting on the Saturday before the IETF (July
> 23rd) from 14:00 to 18:00. The meeting will take place at the Quebec
> City Convention Center (i.e., the IETF venue).
>
> Please, register early so that we can accurately estimate how many
> attendees we will have in order to book an appropriate meeting room.
> Tentatively, we have reserved room 2103, which seems to be able to seat
> around 60 people.
>
> Cheers,
>
> Gonzalo
>
>
> On 21/06/2011 4:57 PM, Francois Daoust wrote:
>> Dear participants in the IETF RTCWEB group,
>>
>> The W3C Web Real-Time Communications working group will meet on Saturday 23 July 2011 afternoon, starting at or shortly after 1PM, in Quebec City, Canada, co-located with the IETF Meeting. Exact time range, agenda and location will be announced once known.
>>
>> The choice of the date and place is meant to favor synergies between the W3C WebRTC and the IETF RTCWEB groups. Participants in the IETF RTCWEB group are welcome to attend this W3C face-to-face as meeting guests.
>>
>> To facilitate logistics purpose, I have set up a registration form, available at:
>>    http://www.w3.org/2002/09/wbs/47318/webrtc-f2f-quebec/
>>
>> Anyone can fill out the form.
>> People willing to attend the F2F need to register before 15 July 2011 (the sooner, the better).
>>
>> Please let me know if you have questions.
>>
>> Cheers,
>> Francois Daoust,
>> W3C Team Contact for the W3C WebRTC WG.
>> _______________________________________________
>> 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 blizzard@mozilla.com  Wed Jun 29 22:48:19 2011
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Subject: Re: [rtcweb] Non-media data service consensus and requirements
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On 6/27/2011 12:36 AM, Magnus Westerlund wrote:
> WG,
>
> At the interim it was planned to have a bit discussion on the datagram
> service for RTCWEB. The first question to try to resolve if there
> is consensus for including some form of non real-time media (i.e. not
> audio, video) service between peers. This is a bit tangled with the
> actual requirements and use cases. But there was views both for it and
> against it on the mailing list. So lets continue and try to come to a
> conclusion on this discussion.
>
> The use cases mentioned on the mailing list are:
>
> - Dynamic meta data for Conference and other real-time services
>
> - Gaming data with low latency requirements
>
> Does anyone like to add additional use cases?

For web browsers we'd (Mozilla) like to see it as a more generic 
transport service for applications.  As part of our web-as-applications 
work we'd like to break some of the assumptions about how applications 
are built.  In particular, having standalone apps that are downloaded 
and installed, but built on web technologies where clients can connect 
with each other is something that's interesting.  (HTTP and WebSockets 
both require connecting via a server intermediary.)  If we had a peer to 
peer connection service, with or without video services, we would be 
able to connect applications together without servers involved, aside 
from initial set-up.  Bringing back the end to end principle actually 
matters for our apps strategy.

And yes, this is different than what other people are doing.  But it's 
one of the reasons why I've been insistent on data transmission as part 
of the peer to peer APIs and protocols.

So if I put my use case down it would be:

- Baselevel communications protocol for application to application 
communication in browsers.

And yes, that's vague.  I could use better words there.

>
> Based on my personal understanding this points to primarily have the
> RTCWEB provide a unreliable datagram service. This clearly needs
> additional requirements to be secure and safe to deploy, but more about
> this below. I still like to ask the WG here a question.
>
> Are you supporting the inclusion of a unreliable datagram service
> directly between peers? Please provide your view and any additional
> statements of motivation that you desire to provide.
>
> Secondly, there is a question if there needs to have something that
> provides reliable message (of arbitrary size) or byte stream oriented
> data transport between the peers. I personally foresee that people will
> build JS libraries for this on top of a unreliable datagram service. If
> you desire reliable data service as part of the standardized solution
> please provide motivation and use case and requirements.

I think that for the app to app case I have to imagine that the most 
common use case will be reliable transport with both messages and 
entities being sent back and forth.

But the file transfer use case, as well as trying to provide some level 
of congestion control around audio & video sessions would seem to 
indicate that we might want to have both, tbh.

But I haven't thought deeply on this particular aspect of the 
discussion.  So I'm easily swayed.

--Chris

From stefan.lk.hakansson@ericsson.com  Wed Jun 29 23:41:21 2011
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From: =?iso-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>
To: Cullen Jennings <fluffy@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Thu, 30 Jun 2011 08:41:18 +0200
Thread-Topic: [rtcweb] Media negeotiation and signalling archatecture
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A couple of quick comments:

* The last slide now says that WhatWG proposal is low path; this is not tru=
e. The WhatWG proposal as I read it is mute on how the SDPs are exchanged, =
it just hands those SDPs over to the application (though high path is indic=
ated: "Communications are coordinated via a signaling channel provided by s=
cript in the page via the server, e.g. using XMLHttpRequest.")

* Notably the WhatWG PeerConnection API does allow both high path and low p=
ath. You can exchange all the SDPs over the server as the spec indicates, b=
ut if your application initially only sets up a PeerConnection with no medi=
a Streams, there will be a peer-to-peer datagram channel available. It can =
be used by the web app to exchange the SDPs for media negotiation when you =
add Streams (then we have a separate thread on that datagram channel - is i=
t reliable or not - or would the app have to add reliability)

* Even if you use the low path approach you still need a high path availabl=
e all the time. It is needed to exchange candidates so that ICE can be used=
 to open the low path - and this would have to be done at the start of the =
session and every time you do a NIC change (a use case you added BTW!)

Stefan

>-----Original Message-----
>From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org]=20
>On Behalf Of Cullen Jennings
>Sent: den 29 juni 2011 17:08
>To: rtcweb@ietf.org
>Subject: [rtcweb] Media negeotiation and signalling archatecture
>
>
>One of the complicated architectural issues for this WG is=20
>what path the signaling information gets passed over, and in=20
>the parts of that that path that need to be standardized, what=20
>does the signaling information looks like. To help get=20
>discussion going on that, I have described 3 models of how the=20
>information could flow in the some pictures in the following PDF.=20
>
>http://trac.tools.ietf.org/wg/rtcweb/trac/raw-attachment/wiki/W
>ikiStart/RTCWeb-Signaling.pdf
>
>I split these up into 3 models that I call High, Mid, and Low=20
>based on the path the signaling information goes. There has=20
>been a fair amount of discussion of High and Mid models in the=20
>past in this WG thought pretty much no discussion of the Low=20
>model. The interesting things is that when I look at what is=20
>getting implemented and prototyped, a lot of it is the low model.=20
>
>It certainly possible to support more than one of these but=20
>I'm interested in the pro and cons of all these but from my=20
>point of view, one of the key issues is how hard is for people=20
>to use. If you look at what things have succeeded in HTML, it=20
>is typically the things that in there simplest form provide a=20
>very simple interface to use them. Yes, they might have a more=20
>complex form that allows richer control but the basics are=20
>simple. Take the HTML5 video tag for streaming video media for=20
>example. There are zillion things that could have been passed=20
>and controlled to this flag yet the proposals that one out=20
>were the ones that provided something that was incredibly=20
>simple to use. Another example is the iPhone interface to make=20
>a phone call from the browser has been very successful from an=20
>adoption point of view.=20
>
>
>
>
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb
>=

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--0016e64356188277d704a6e9d8e2
Content-Type: text/plain; charset=ISO-8859-1

Cullen, what is meant by "current Chrome implementation" in this doc? At
least what we're implementing is not using the Jingle protocol on the net.
We are using libJingle internally, but the API we're building is essentially
the whatwg proposal with some tweaks.

Niklas


On Wed, Jun 29, 2011 at 5:07 PM, Cullen Jennings <fluffy@cisco.com> wrote:

>
> One of the complicated architectural issues for this WG is what path the
> signaling information gets passed over, and in the parts of that that path
> that need to be standardized, what does the signaling information looks
> like. To help get discussion going on that, I have described 3 models of how
> the information could flow in the some pictures in the following PDF.
>
>
> http://trac.tools.ietf.org/wg/rtcweb/trac/raw-attachment/wiki/WikiStart/RTCWeb-Signaling.pdf
>
> I split these up into 3 models that I call High, Mid, and Low based on the
> path the signaling information goes. There has been a fair amount of
> discussion of High and Mid models in the past in this WG thought pretty much
> no discussion of the Low model. The interesting things is that when I look
> at what is getting implemented and prototyped, a lot of it is the low model.
>
> It certainly possible to support more than one of these but I'm interested
> in the pro and cons of all these but from my point of view, one of the key
> issues is how hard is for people to use. If you look at what things have
> succeeded in HTML, it is typically the things that in there simplest form
> provide a very simple interface to use them. Yes, they might have a more
> complex form that allows richer control but the basics are simple. Take the
> HTML5 video tag for streaming video media for example. There are zillion
> things that could have been passed and controlled to this flag yet the
> proposals that one out were the ones that provided something that was
> incredibly simple to use. Another example is the iPhone interface to make a
> phone call from the browser has been very successful from an adoption point
> of view.
>
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--0016e64356188277d704a6e9d8e2
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Cullen, what is meant by &quot;current Chrome implementation&quot; in this =
doc? At least what we&#39;re implementing is not using the Jingle protocol =
on the net. We are using libJingle internally, but the API we&#39;re buildi=
ng is essentially the whatwg proposal with some tweaks.<div>
<br></div><div>Niklas</div><div><br><br><div class=3D"gmail_quote">On Wed, =
Jun 29, 2011 at 5:07 PM, Cullen Jennings <span dir=3D"ltr">&lt;<a href=3D"m=
ailto:fluffy@cisco.com">fluffy@cisco.com</a>&gt;</span> wrote:<br><blockquo=
te class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc so=
lid;padding-left:1ex;">
<br>
One of the complicated architectural issues for this WG is what path the si=
gnaling information gets passed over, and in the parts of that that path th=
at need to be standardized, what does the signaling information looks like.=
 To help get discussion going on that, I have described 3 models of how the=
 information could flow in the some pictures in the following PDF.<br>

<br>
<a href=3D"http://trac.tools.ietf.org/wg/rtcweb/trac/raw-attachment/wiki/Wi=
kiStart/RTCWeb-Signaling.pdf" target=3D"_blank">http://trac.tools.ietf.org/=
wg/rtcweb/trac/raw-attachment/wiki/WikiStart/RTCWeb-Signaling.pdf</a><br>
<br>
I split these up into 3 models that I call High, Mid, and Low based on the =
path the signaling information goes. There has been a fair amount of discus=
sion of High and Mid models in the past in this WG thought pretty much no d=
iscussion of the Low model. The interesting things is that when I look at w=
hat is getting implemented and prototyped, a lot of it is the low model.<br=
>

<br>
It certainly possible to support more than one of these but I&#39;m interes=
ted in the pro and cons of all these but from my point of view, one of the =
key issues is how hard is for people to use. If you look at what things hav=
e succeeded in HTML, it is typically the things that in there simplest form=
 provide a very simple interface to use them. Yes, they might have a more c=
omplex form that allows richer control but the basics are simple. Take the =
HTML5 video tag for streaming video media for example. There are zillion th=
ings that could have been passed and controlled to this flag yet the propos=
als that one out were the ones that provided something that was incredibly =
simple to use. Another example is the iPhone interface to make a phone call=
 from the browser has been very successful from an adoption point of view.<=
br>

<br>
<br>
<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></div>

--0016e64356188277d704a6e9d8e2--

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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: Matthew Kaufman <matthew.kaufman@skype.net>, Cullen Jennings <fluffy@cisco.com>
Date: Thu, 30 Jun 2011 19:04:50 +0200
Thread-Topic: [rtcweb] Media negeotiation and signalling archatecture
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Subject: Re: [rtcweb] Media negeotiation and signalling archatecture
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=20

> -----Original Message-----
> From: rtcweb-bounces@ietf.org=20
> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Matthew Kaufman
> Sent: 30 June 2011 00:23
> To: Cullen Jennings
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Media negeotiation and signalling archatecture
>=20
>=20
> On Jun 29, 2011, at 8:07 AM, Cullen Jennings wrote:
>=20
> >=20
> > One of the complicated architectural issues for this WG is=20
> what path the signaling information gets passed over, and in=20
> the parts of that that path that need to be standardized,=20
> what does the signaling information looks like. To help get=20
> discussion going on that, I have described 3 models of how=20
> the information could flow in the some pictures in the following PDF.=20
> >=20
> >=20
> http://trac.tools.ietf.org/wg/rtcweb/trac/raw-attachment/wiki/
> WikiStart/RTCWeb-Signaling.pdf
> >=20
> > I split these up into 3 models that I call High, Mid, and=20
> Low based on the path the signaling information goes. There=20
> has been a fair amount of discussion of High and Mid models=20
> in the past in this WG thought pretty much no discussion of=20
> the Low model. The interesting things is that when I look at=20
> what is getting implemented and prototyped, a lot of it is=20
> the low model.=20
>=20
> I think this deck is a bit in two ways.
>=20
> First, in the first slide you say (and I agree) that we agree=20
> that if there's >1 web server we don't care how they talk to=20
> each other (I'm suspecting something like SIP in most=20
> cases)... but then in the "high path" you say it "gets=20
> complicated to define what happens on the orange line"... but=20
> I disagree here. If we agree that it is SIP and SDP on the=20
> orange line then clearly the things that are needed to do NAT=20
> traversal (ICE) and codec negotiation are all able to be=20
> carried in SDP.=20
[JRE] And to add to what Matthew says concerning complexity of SIP, much of=
 the complexity is between a UA and its domain registrar/proxy (e.g., SIP o=
utbound, GRUU assignment, Path) and a lot of features that are implemented =
on SIP-PBXs, say, only operate between a UA and its domain registrar/proxy =
(or B2BUA). In the high path, the web server takes on the role of domain re=
gistrar/proxy, so the SIP interface is an inter-proxy, or inter-domain inte=
rface, where a lot of the complexities of SIP don't appear.

John



>As another alternative you could simply carry=20
> something that the browser understood natively on that orange path.=20
>=20
> In fact in the "low path" slide, you talk about the current=20
> Chrome implementation and the WhatWG proposal, but both of=20
> those are actually suggesting transporting a JSON blob via=20
> the "high path" to do the ICE negotiation... and there's no=20
> reason to believe that if you can send the blob needed to do=20
> ICE that you couldn't also do the media type negotiation via=20
> the same path.
>=20
> Second, the "high path" is still needed to do the actual call=20
> signaling... you can't even get the two browsers connected to=20
> the two different web services interested in setting up a=20
> session using ICE if there hasn't been communication over the=20
> "high path" to set up a call.
>=20
> Matthew Kaufman
>=20
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> =

From stewe@stewe.org  Thu Jun 30 13:17:54 2011
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From: Stephan Wenger <stewe@stewe.org>
To: <rtcweb@ietf.org>
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Thread-Topic: Agenda time request
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> This message is in MIME format. Since your mail reader does not understand
this format, some or all of this message may not be legible.

--B_3392284670_6261533
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Hi all,
We are in the process of finishing an I-D on the subject of scalable codecs
for rtcweb.  The draft will be uploaded by the =AD00 deadline.  If there is a
chance, I would like to have a few minutes session time to present it.
Yes, the focus will be on technical aspects; both requirements and how they
can be met.
Regards,
Stephan
=20



--B_3392284670_6261533
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<html><head></head><body style=3D"word-wrap: break-word; -webkit-nbsp-mode: s=
pace; -webkit-line-break: after-white-space; color: rgb(0, 0, 0); font-size:=
 14px; font-family: Calibri, sans-serif; "><div>Hi all,</div><div>We are in =
the process of finishing an I-D on the subject of scalable codecs for rtcweb=
. &nbsp;The draft will be uploaded by the &#8211;00 deadline. &nbsp;If there=
 is a chance, I would like to have a few minutes session time to present it.=
</div><div>Yes, the focus will be on technical aspects; both requirements an=
d how they can be met.</div><div>Regards,</div><div>Stephan</div><div>&nbsp;=
</div></body></html>

--B_3392284670_6261533--



From internet-drafts@ietf.org  Thu Jun 30 22:52:26 2011
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Subject: [rtcweb] I-D Action: draft-ietf-rtcweb-overview-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           : Overview: Real Time Protocols for Brower-based Applicati=
ons
	Author(s)       : Harald T. Alvestrand
	Filename        : draft-ietf-rtcweb-overview-00.txt
	Pages           : 13
	Date            : 2011-06-30

   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 candidate to become 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-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-overview-00.txt
