
From stefan.lk.hakansson@ericsson.com  Fri Jul  1 05:33:47 2011
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From: =?ISO-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>
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Subject: Re: [rtcweb] Fwd: New Version Notification for	draft-kaufman-rtp-compatible-data-00.txt
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Reading the draft I get the impression that the intention is to 
multiplex (multiple) RTP streams with arbitrary datagrams on the same 
5-tuple. In that light the method lacks IMO:

* There is no multiplexing support for different RTP flows - as 
discussed at the Interim meeting session separation should be maintained

* There is no support for rate/congestion control of the datagrams

* There is no support for multiplexing different datagram flows (the 
need for this can be argued though - could be done by the web app)

Stefan
On 2011-06-30 02:50, Matthew Kaufman wrote:
> 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 <mailto:internet-drafts@ietf.org>
>> *Date: *June 29, 2011 5:48:52 PM PDT
>> *To: *matthew.kaufman@skype.net <mailto:matthew.kaufman@skype.net>
>> *Cc: *matthew.kaufman@skype.net <mailto:matthew.kaufman@skype.net>
>> *Subject: **New Version Notification for
>> draft-kaufman-rtp-compatible-data-00.txt*
>>
>> 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.
>>
>> Filename:draft-kaufman-rtp-compatible-data
>> Revision:00
>> Title:
>> Creation date:2011-06-30
>> WG ID:Individual Submission
>> Number of pages: 5
>>
>> 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.
>>
>>
>>
>>
>> The IETF Secretariat
>


From dzonatas@gmail.com  Fri Jul  1 08:12:52 2011
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From: Dzonatas Sol <dzonatas@gmail.com>
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-overview-00.txt
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Hi, I read the entire overview. Only one concern, in 2.4 Terminology:

 >  Real-time media:  Media where generation of content and display of
 >      content are intended to occur closely together in time (on the
 >     order of no more than hundreds of milliseconds).

Can that just read:

 >  Real-time media:  Media where generation of content and display of
 >      content are intended to occur closely together in a given time 
scale.

I've found early specification of date systems may affect assets in ways 
unintended (or implied). There is no simple cure, so I think this is 
clearer without the noted "hundreds of milliseconds", there.

Some have tried to build "authoritative" dates from such real-time 
mechanics. Others pass that date thinking it is part of the asset. I've 
seen it happen, so I'm just thinking it would be nice to have some way 
to say that was not intended, yet I haven't found any.

On 06/30/2011 10:52 PM, internet-drafts@ietf.org wrote:
> A New Internet-Draft is available from the on-line Internet-Drafts directories. This draft is a work item of the Real-Time Communication in WEB-browsers Working Group of the IETF.
>
> 	Title           : Overview: Real Time Protocols for Brower-based Applications
> 	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
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>    


-- 
--- http://twitter.com/Dzonatas_Sol ---
Web Development, Software Engineering
Ag-Biotech, Virtual Reality, Consultant


From ton.kalker@gmail.com  Fri Jul  1 19:22:56 2011
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One minor remark: the definition of echo cancelation in this document
is non-standard. The current definition addresses only a particular
instance of echo. The general definition would be the suppression of
acoustical feedback loops below perceptually noticeable levels.

Ton

Sent from my tPad

e ton.kalker@{huawei.com,gmail.com,ieee.org}
c +1.831.917.1350

On Jun 30, 2011, at 10:52 PM, "internet-drafts@ietf.org"
<internet-drafts@ietf.org> wrote:

> A New Internet-Draft is available from the on-line Internet-Drafts directories. This draft is a work item of the Real-Time Communication in WEB-browsers Working Group of the IETF.
>
>    Title           : Overview: Real Time Protocols for Brower-based Applications
>    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
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From dzonatas@gmail.com  Sun Jul  3 11:28:09 2011
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Before another glossary action happens, let me get this in (can't wait 
for tomorrow):

 >  Real-time media:  Media where generation of content and display of
 >      content are intended to occur closely together in a given time 
scale.

"""
Real-time media:  Media where generation of content and display of 
content are intended to occur closely together in a given time scale 
between known ahead-of-time (AOT) rate and known just-in-time (JIT) rate.
"""

On 07/01/2011 08:10 AM, Dzonatas Sol wrote:
> Hi, I read the entire overview. Only one concern, in 2.4 Terminology:
>
> >  Real-time media:  Media where generation of content and display of
> >      content are intended to occur closely together in time (on the
> >     order of no more than hundreds of milliseconds).
>
> Can that just read:
>
> >  Real-time media:  Media where generation of content and display of
> >      content are intended to occur closely together in a given time 
> scale.
>
> I've found early specification of date systems may affect assets in 
> ways unintended (or implied). There is no simple cure, so I think this 
> is clearer without the noted "hundreds of milliseconds", there.
>
> Some have tried to build "authoritative" dates from such real-time 
> mechanics. Others pass that date thinking it is part of the asset. 
> I've seen it happen, so I'm just thinking it would be nice to have 
> some way to say that was not intended, yet I haven't found any.
>
> On 06/30/2011 10:52 PM, internet-drafts@ietf.org wrote:
>> A New Internet-Draft is available from the on-line Internet-Drafts 
>> directories. This draft is a work item of the Real-Time Communication 
>> in WEB-browsers Working Group of the IETF.
>>
>>     Title           : Overview: Real Time Protocols for Brower-based 
>> Applications
>>     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
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>
>


-- 
--- http://twitter.com/Dzonatas_Sol ---
Web Development, Software Engineering
Ag-Biotech, Virtual Reality, Consultant


From gonzalo.camarillo@ericsson.com  Mon Jul  4 03:01:07 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 Igor,

note that this is not a joint meeting. This is a W3C meeting (i.e.,
under W3C rules) that will be organized in the same venue as the IETF.

With respect to the scheduling of the meeting, it was unfortunately
impossible to find a day that worked for everyone. Saturday was the best
trade-off that could be found.

Cheers,

Gonzalo


On 30/06/2011 6:22 AM, Igor Faynberg wrote:
> 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
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> 


From gonzalo.camarillo@ericsson.com  Mon Jul  4 03:02:38 2011
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Folks,

FYI. This meeting has now also been announced on the general IETF
announce list (see email below).

Cheers,

Gonzalo

-------- Original Message --------
Subject: W3C WebRTC working group meeting: Saturday, July 23 2011
Date: Thu, 30 Jun 2011 20:32:29 +0200
From: Alexa Morris <amorris@amsl.com>
To: ietf-announce@ietf.org <ietf-announce@ietf.org>

The W3C Web Real-Time Communications working group will meet on Saturday
23 July 2011 afternoon, starting at 2PM, in Quebec City, Canada,
co-located with the IETF Meeting.

The choice of the date and place is meant to favor synergies between the
W3C WebRTC and the IETF RTCWEB groups. IETF attendees, especially those
following the RTCWEB WG, are welcome to attend this W3C face-to-face
meeting as meeting guests. Attendees need to register before July 15th
using the following URL:

  http://www.w3.org/2002/09/wbs/47318/webrtc-f2f-quebec/

Note that this is a W3C meeting and, thus, it will be held under W3C
rules.

Regards,
Alexa

-----------
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From internet-drafts@ietf.org  Mon Jul  4 04:37:19 2011
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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-01.txt
	Pages           : 15
	Date            : 2011-07-04

   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-01.txt

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

This Internet-Draft can be retrieved at:
ftp://ftp.ietf.org/internet-drafts/draft-ietf-rtcweb-use-cases-and-requirem=
ents-01.txt

From stefan.lk.hakansson@ericsson.com  Mon Jul  4 04:41:55 2011
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I've uploaded a new version of the use-case and requirements doc. The 
changes are the ones I proposed in an earlier mail 
(<http://www.ietf.org/mail-archive/web/rtcweb/current/msg00227.html>).

I received no responses to that mail; if people have problems with (some 
or all of) the changes I made I guess we will have to revert them in the 
next version

Stefan

-------- Original Message --------
Subject: New Version Notification for 
draft-ietf-rtcweb-use-cases-and-requirements-01.txt
Date: Mon, 4 Jul 2011 13:37:19 +0200
From: internet-drafts@ietf.org <internet-drafts@ietf.org>
To: Stefan Håkansson LK <stefan.lk.hakansson@ericsson.com>
CC: Stefan Håkansson LK <stefan.lk.hakansson@ericsson.com>, Göran 
Eriksson AP <goran.ap.eriksson@ericsson.com>, Christer Holmberg 
<christer.holmberg@ericsson.com>

A new version of I-D, 
draft-ietf-rtcweb-use-cases-and-requirements-01.txt has been 
successfully submitted by Stefan Hakansson and posted to the IETF 
repository.

Filename:	 draft-ietf-rtcweb-use-cases-and-requirements
Revision:	 01
Title:		 Web Real-Time Communication Use-cases and Requirements
Creation date:	 2011-07-04
WG ID:		 rtcweb
Number of pages: 15

Abstract:
    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.

 



The IETF Secretariat

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This is message one of five.

I have submitted new drafts for RTC-Web: signaling and negotiation methods,
media transports, datagram transport for non-media data, and media
processing and codecs.

Cheers,

-Cary

A new version of I-D, draft-cbran-rtcweb-negotiation-00.txt has been
successfully submitted by Cary Bran and posted to the IETF repository.

Filename:     draft-cbran-rtcweb-negotiation
Revision:     00
Title:         RTC-Web Negotiation and Signaling
Creation date:     2011-07-04
WG ID:         Individual Submission
Number of pages: 9

Abstract:
   This document outlines the negotiation and signaling protocols for
   RTC-Web client application implementation.

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


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This is message two of five.

I have submitted new drafts for RTC-Web: signaling and negotiation methods,
media transports, datagram transport for non-media data, and media
processing and codecs.

Cheers,

-Cary

A new version of I-D, draft-cbran-rtcweb-data-00.txt has been successfully
submitted by Cary Bran and posted to the IETF repository.

Filename:     draft-cbran-rtcweb-data
Revision:     00
Title:         RTC-Web Non-Media Data Transport Requirements
Creation date:     2011-07-04
WG ID:         Individual Submission
Number of pages: 6

Abstract:
   This document outlines a protocol and requirements for RTC-Web client
   application to transmit real-time, non-media data.


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


From cbran@cisco.com  Mon Jul  4 12:57:45 2011
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This is message three of five.

I have submitted new drafts for RTC-Web: signaling and negotiation methods,
media transports, datagram transport for non-media data, and media
processing and codecs.

Cheers,

-Cary

A new version of I-D, draft-cbran-rtcweb-nat-00.txt has been successfully
submitted by Cary Bran and posted to the IETF repository.

Filename:     draft-cbran-rtcweb-nat
Revision:     00
Title:         RTC-Web Network Address Translation
Creation date:     2011-07-01
WG ID:         Individual Submission
Number of pages: 8

Abstract:
   This document outlines the network address translation (NAT)
   mechanisms and requirements for RTC-Web client applications.

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


From cbran@cisco.com  Mon Jul  4 12:57:55 2011
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This is message four of five.

I have submitted new drafts for RTC-Web: signaling and negotiation methods,
media transports, datagram transport for non-media data, and media
processing and codecs.

Cheers,

-Cary

A new version of I-D, draft-cbran-rtcweb-media-00.txt has been successfully
submitted by Cary Bran and posted to the IETF repository.

Filename:     draft-cbran-rtcweb-media
Revision:     00
Title:         RTC-Web Media Transport Requirements
Creation date:     2011-06-29
WG ID:         Individual Submission
Number of pages: 11

Abstract:
   This document outlines the media transport protocols and requirements
   for RTC-Web client applications.

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


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This is message five of five.

I have submitted new drafts for RTC-Web: signaling and negotiation methods,
media transports, datagram transport for non-media data, and media
processing and codecs.

Cheers,

-Cary

A new version of I-D, draft-cbran-rtcweb-codec-00.txt has been successfully
submitted by Cary Bran and posted to the IETF repository.

Filename:     draft-cbran-rtcweb-codec
Revision:     00
Title:         RTC-Web Codec and Media Processing Requirements
Creation date:     2011-06-28
WG ID:         Individual Submission
Number of pages: 7

Abstract:
   This document outlines the codec and media processing requirements
   for RTC-Web client application and endpoint devices.

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


From jdrosen@jdrosen.net  Mon Jul  4 15:19:04 2011
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From: Jonathan Rosenberg <jdrosen@jdrosen.net>
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Subject: [rtcweb] Fwd: I-D Action: draft-rosenberg-rtcweb-rtpmux-00.txt
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We put together the draft below to argue in favor of using a single port 
for multiplexing voice and video traffic over RTP.

Looking forward to a lively discussion on this, I am sure.. ;)

Thanks,
Jonathan R.

-------- Original Message --------
Subject: I-D Action: draft-rosenberg-rtcweb-rtpmux-00.txt
Date: Mon, 04 Jul 2011 15:14:25 -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           : Multiplexing of Real-Time Transport Protocol (RTP) 
Traffic for Browser based Real-Time Communications (RTC)
	Author(s)       : Jonathan Rosenberg
                           Cullen Jennings
                           Jon Peterson
                           Matthew Kaufman
                           Eric Rescorla
                           Tim Terriberry
	Filename        : draft-rosenberg-rtcweb-rtpmux-00.txt
	Pages           : 12
	Date            : 2011-07-04

    This document argues that multiplexing of voice and video traffic
    over a single RTP session should be specified as the baseline mode of
    operation for multimedia traffic in RTC web.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-rosenberg-rtcweb-rtpmux-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-rosenberg-rtcweb-rtpmux-00.txt
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Date: Tue, 05 Jul 2011 07:39:15 -0700
From: Stephan Wenger <stewe@stewe.org>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
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Thread-Topic: draft-wenger-rtcweb-layered-codec-00 posted
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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.

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Hi,
As promised, Alex Eleftheriadis and myself have thrown together a quick
draft outlining the advantages and desirability of layered codecs.  You can
find it 
here:http://datatracker.ietf.org/doc/draft-wenger-rtcweb-layered-codec/
Happy reading.
Stephan



--B_3392696359_312233
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	charset="US-ASCII"
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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><div>Hi,</div><div>As promis=
ed, Alex Eleftheriadis and myself have thrown together a quick draft outlini=
ng the advantages and desirability of layered codecs. &nbsp;You can find it =
here:<a href=3D"http://datatracker.ietf.org/doc/draft-wenger-rtcweb-layered-co=
dec">http://datatracker.ietf.org/doc/draft-wenger-rtcweb-layered-codec</a>/<=
/div><div>Happy reading.</div><div>Stephan</div></div></body></html>

--B_3392696359_312233--



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Subject: [rtcweb] draft rtcweb agenda for IETF 81 - version 2
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Here is the rough agenda that Ted, Magnus, and I have been thinking =
of..... bash away ...



Agenda - version 2

          Agenda Bash
          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 ( 20 min )
          draft-ietf-rtcweb-use-cases-and-requirements
         =20
          Architecture and System Overview ( 20 min)
          draft-ietf-rtcweb-overview
         =20
          Security ( 20 min )
          draft-rescorla-rtcweb-security
          draft-kaufman-rtcweb-security-ui
          draft-johnston-rtcweb-media-privacy

          NAT ( 20 min)
          draft-kaufman-rtcweb-traversal      =20
          draft-cbran-rtcweb-nat                =20
         =20


          Start of Session two (Chairs) - 5 min
         =20
          Negotiation & signaling (25 min)
          draft-cbran-rtcweb-negotiation

         =20
          Media (other than multiplexing) (20 min)
          draft-perkins-rtcweb-rtp-usage
          draft-cbran-rtcweb-media
               =20
   =20
          Datagram transports (20 min)
          draft-cbran-rtcweb-data


          Multiplexing (30 min)
          draft-rosenberg-rtcweb-rtpmux
          draft-perkins-rtcweb-rtp-usage=20
          * focus on how/if we multiplex on one UDP port=20


          Media processing & codec (20 min)
          draft-cbran-rtcweb-codec
          draft-alvestrand-dispatch-rtcweb-protocols
          * focus on requirements for codecs=20

         =20
         =20



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From: Stephan Wenger <stewe@stewe.org>
To: Cullen Jennings <fluffy@cisco.com>, <rtcweb@ietf.org>
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Hi Cullen, all,
We would like to have a few minute sin the last subsection to introduce
layered codecs: draft-wenger-rtcweb-layered-codec-00
<http://datatracker.ietf.org/doc/draft-wenger-rtcweb-layered-codec/>
Thanks,
Stephan

On 7.5.2011 09:00 , "Cullen Jennings" <fluffy@cisco.com> wrote:

>
>Here is the rough agenda that Ted, Magnus, and I have been thinking
>of..... bash away ...
>
>
>
>Agenda - version 2
>
>          Agenda Bash
>          Status report (Chairs) - 10 min
>          
>          Liaison Report from W3C WebRTC -(WebRTC Chairs) - 10 min
>          The W3C will have just had a meeting the Saturday before
>          
>          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
>          
>          Use Cases ( 20 min )
>          draft-ietf-rtcweb-use-cases-and-requirements
>          
>          Architecture and System Overview ( 20 min)
>          draft-ietf-rtcweb-overview
>          
>          Security ( 20 min )
>          draft-rescorla-rtcweb-security
>          draft-kaufman-rtcweb-security-ui
>          draft-johnston-rtcweb-media-privacy
>
>          NAT ( 20 min)
>          draft-kaufman-rtcweb-traversal
>          draft-cbran-rtcweb-nat
>          
>
>
>          Start of Session two (Chairs) - 5 min
>          
>          Negotiation & signaling (25 min)
>          draft-cbran-rtcweb-negotiation
>
>          
>          Media (other than multiplexing) (20 min)
>          draft-perkins-rtcweb-rtp-usage
>          draft-cbran-rtcweb-media
>                
>    
>          Datagram transports (20 min)
>          draft-cbran-rtcweb-data
>
>
>          Multiplexing (30 min)
>          draft-rosenberg-rtcweb-rtpmux
>          draft-perkins-rtcweb-rtp-usage
>          * focus on how/if we multiplex on one UDP port
>
>
>          Media processing & codec (20 min)
>          draft-cbran-rtcweb-codec
>          draft-alvestrand-dispatch-rtcweb-protocols
>          * focus on requirements for codecs
>
>          
>          
>
>
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb



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Agree with folks on this thread about this being a way for the receiver =
to tell the send it's preferences - Perhaps the receiver might indicate =
that it would like 1080P video but the best the sender can do is 720P =
because that is the size of camera they have.  In this case the sender =
does what the sender wants ignoring the receiver. There are other cases =
where say the receiver can not decode a bitstream above 768 kbps but the =
sender would typically send a stream at a much higher rate than this. In =
this case the sender needs to do what the receiver wants. You might have =
an old browser that support vp8 and newer version of the browser that =
does both vp8 and vp9 and you use vp8 when that is all you have but =
prefer vp9 if both sides support it. I'm not sure exactly how to phrase =
all this but roughly speaking we need a way for the sender and receiver =
to agree on a format that meets the constraints of both and somewhat =
optimizes the desiree format within these constraints.=20


On Jun 22, 2011, at 12:06 AM, Stefan H=E5kansson LK wrote:

> All,
> =20
> this RFC could perhaps be useful for this purpose of signaling display =
size: http://www.rfc-editor.org/rfc/rfc6236.txt
> =20
> Stefan
>=20
> 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:)
>=20
> Hi Justin, all,
> =20
> 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.
> =20
> Best regards,
> Bert
> =20
> =20
> From: Justin Uberti [mailto:juberti@google.com]=20
> 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:)
> =20
> 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.
> =20
>=20
> 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.
>=20
> Best regards,
> Bert
>=20
>=20
> -----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:)
>=20
>=20
> On Jun 20, 2011, at 1:30 , Harald Alvestrand wrote:
>=20
> > 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.
>=20
> Consider two different video flows using the same codec ...
>=20
> Stream A is 640 x 480 at 1mbps which is this scaled to 320x240 and =
displayed
>=20
> 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
>=20
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> =20
> _______________________________________________
> 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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Resend from the right address...

On 7/5/2011 6:30 PM, Cullen Jennings wrote:
> Agree with folks on this thread about this being a way for the receiver to tell the send it's preferences - Perhaps the receiver might indicate that it would like 1080P video but the best the sender can do is 720P because that is the size of camera they have.  In this case the sender does what the sender wants ignoring the receiver. There are other cases where say the receiver can not decode a bitstream above 768 kbps but the sender would typically send a stream at a much higher rate than this. In this case the sender needs to do what the receiver wants.

These sorts of issues are at least partly addressed in the O/A section 
of the H.264 spec (RFC 6184, update to RFC 3984).  Obviously that only 
applies to H.264.  There are also issues about limitations due to the 
other direction; in particular look at level-asymmetry-allowed and what 
happens if it's not used.

As mentioned, there are issues both of initial negotiation (as implied 
here) and in-call changes.

I think a) the two sides should indicate what they support receiving in 
negotiation, and b) indicate their initial preferred resolution/etc in 
negotiation.  In-call, there should be a mechanism for changing 
preference without a full call renegotiation (I suggest an RTCP-FB 
message), though one could fall back on a full new SDP exchange.

> You might have an old browser that support vp8 and newer version of the browser that does both vp8 and vp9 and you use vp8 when that is all you have but prefer vp9 if both sides support it.

That's just classic SDP and Offer/Answer:

New:
       m=video 1234 RTP/AVP 98 99 100
       a=rtpmap 98 VP9/90000
       a=rtpmap 99 VP8/90000
       a=rtpmap 100 H264/90000

Old:
       m=video 1234 RTP/AVP 99 100
       a=rtpmap 99 VP8/90000
       a=rtpmap 100 H264/90000

> I'm not sure exactly how to phrase all this but roughly speaking we need a way for the sender and receiver to agree on a format that meets the constraints of both and somewhat optimizes the desiree format within these constraints.

RFC 6236 certainly looks like a reasonable option to consider; in fact 
I'd consider it to be the
defacto choice barring some problematic limitation.  It allows asymmetry 
within the limits of the codec
negotiation; if RFC 6184 is used for H.264 then level asymmetry is allowed.

There *might* be an argument for also describing the portions of the 
decoded window that aren't visible
(per the previous conversation).

My biggest issue is that they assume it can only be changed via UPDATE 
or re-INVITE.  Maybe that's
reasonable, but I have to say I'd lean towards solutions that are direct 
end-to-end in-call to cut overhead
and lower reaction time.

   Randell

>
> On Jun 22, 2011, at 12:06 AM, Stefan Hkansson LK wrote:
>
>> 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 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] 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
>>
>> _______________________________________________
>> 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
>
>


-- 
Randell Jesup
randell-ietf@jesup.org

From randell1@jesup.org  Tue Jul  5 20:20:51 2011
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On 6/29/2011 8:50 PM, Matthew Kaufman wrote:
> 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.
http://www.ietf.org/id/draft-kaufman-rtp-compatible-data-00.txt

|    One alternative considered was to encapsulate the arbitrary datagrams
|    within RTP as an RTP payload type.  However the semantics associated
|    with RTP are not always appropriate, depending on the type of
|    arbitrary data being transmitted.

This needs to be expanded on.  Why are they not appropriate?  (Not 
disagreeing, but state it please.)

If this data traverses various SBC/midboxes/NATs-with-ALGs/etc, will the 
apparent "invalid" nature cause possible breakage?

| 5.  Security Considerations
|
|    The contents of these messages SHOULD be encrypted.  Reuse of the
|    keying used for an associated RTP session that is using the same IP
|    address and port MAY be an acceptable method of key specification and
|    cryptosystem choice.

I'd say "The User Message Data of these messages"... to avoid confusion 
about encryption of the
all-zeros 4-byte header.

Also: should it say anything about authentication?

Congestion control needs to be discussed, even if it's to say that it's 
the job of a higher or application layer to define.  I do want to as 
part of rtcweb design make congestion handling be unified, whether or 
not we use a single port (though we probably should use 1 port).

-- 
Randell Jesup
randell-ietf@jesup.org


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Subject: Re: [rtcweb] Media neg-eo-tiation and signalling archa-tecture
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Stefan said:

A couple of quick comments:

* The last slide now says that WhatWG proposal is low path=3B this is not t=
rue. The WhatWG proposal as I read it is mute on how the SDPs are exchanged=
=2C it just hands those SDPs over to the=20
application (though high path is indicated: "Communications are coordinated=
 via a signaling channel provided by script in the page via the server=2C 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=2C=
 but if your application=20
initially only sets up a PeerConnection with no media Streams=2C there will=
 be a peer-to-peer datagram channel available. It can be used by the web ap=
p to exchange the SDPs=20
for media negotiation when you add Streams (then we have a separate thread =
on that datagram channel - is it 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 -=20
and this would have to be done at the start of the session and every time y=
ou do a NIC change (a use case you added BTW!)

Stefan

[BA] I agree that the WhatWG proposal is not inherently low=2C medium or hi=
gh path.   The signalling mechanism is totally up to the signalling callbac=
k function.

I did have a question about how the PeerConnection API manages the gatherin=
g of the ICE candidates=2C though.  It was my (naive?) assumption that beyo=
nd the host candidates (always included?)=2C
the other gathered candidates would based based on the Type information pas=
sed (e.g. STUN/STUNS=2C TURN/TURNS).   If the SDP Blob is not considered "o=
paque" the signalling=20
callback could manipulate/change it=2C to affect the  subsequent ICE proces=
sing.  The peer could do the same thing on its end.  If this were true (is =
it?) then that would allow quite a bit
of flexibility.=20

I was also a bit uncertain about the precise timing of when ICE connectivit=
y checks are kicked off within the API=2C and how things like a NIC change =
are handled.   For example=2C is it
envisaged that it would be possible to implement RTCWEB mobility (e.g. hand=
off of media between interfaces)?=20
  =20

 		 	   		  =

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<body class=3D'hmmessage'><div dir=3D'ltr'>
Stefan said:<br><br><pre>A couple of quick comments:

* The last slide now says that WhatWG proposal is low path=3B this is not t=
rue. The WhatWG proposal as I read it is mute on how the SDPs are exchanged=
=2C it just hands those SDPs over to the <br>application (though high path =
is indicated: "Communications are coordinated via a signaling channel provi=
ded by script in the page via the server=2C 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=2C=
 but if your application <br>initially only sets up a PeerConnection with n=
o media Streams=2C there will be a peer-to-peer datagram channel available.=
 It can be used by the web app to exchange the SDPs <br>for media negotiati=
on when you add Streams (then we have a separate thread on that datagram ch=
annel - is it 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 - <br>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<br><br>[BA] I agree that the WhatWG proposal is not inherently low=
=2C medium or high path.   The signalling mechanism is totally up to the si=
gnalling callback function.<br><br>I did have a question about how the Peer=
Connection API manages the gathering of the ICE candidates=2C though.  It w=
as my (naive?) assumption that beyond the host candidates (always included?=
)=2C<br>the other gathered candidates would based based on the Type informa=
tion passed (e.g. STUN/STUNS=2C TURN/TURNS).   If the SDP Blob is not consi=
dered "opaque" the signalling <br>callback could manipulate/change it=2C to=
 affect the  subsequent ICE processing.  The peer could do the same thing o=
n its end.  If this were true (is it?) then that would allow quite a bit<br=
>of flexibility. <br><br>I was also a bit uncertain about the precise timin=
g of when ICE connectivity checks are kicked off within the API=2C and how =
things like a NIC change are handled.   For example=2C is it<br>envisaged t=
hat it would be possible to implement RTCWEB mobility (e.g. handoff of medi=
a between interfaces)? <br>   <br></pre><br> 		 	   		  </div></body>
</html>=

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From stefan.lk.hakansson@ericsson.com  Tue Jul  5 23:21:12 2011
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Subject: Re: [rtcweb] Media neg-eo-tiation and signalling archa-tecture
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On 2011-07-06 08:06, Bernard Aboba wrote:
> Stefan said:
>
> A couple of quick comments:
>
> * The last slide now says that WhatWG proposal is low path; this is not true. 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 indicated: "Communications are coordinated via a signaling channel provided by script in the page via the server, e.g. using XMLHttpRequest.")
>
> * Notably the WhatWG PeerConnection API does allow both high path and low path. You can exchange all the SDPs over the server as the spec indicates, but if your application
> initially only sets up a PeerConnection with no media 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 it 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 available 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
>
> [BA] I agree that the WhatWG proposal is not inherently low, medium or high path.   The signalling mechanism is totally up to the signalling callback function.
>
> I did have a question about how the PeerConnection API manages the gathering of the ICE candidates, though.  It was my (naive?) assumption that beyond the host candidates (always included?),
> the other gathered candidates would based based on the Type information passed (e.g. STUN/STUNS, TURN/TURNS).   If the SDP Blob is not considered "opaque" the signalling
> callback could manipulate/change it, to affect the  subsequent ICE processing.  The peer could do the same thing on its end.  If this were true (is it?) then that would allow quite a bit
> of flexibility.
[SH] I have the same understanding.
>
> I was also a bit uncertain about the precise timing of when ICE connectivity checks are kicked off within the API, and how things like a NIC change are handled.   For example, is it
> envisaged that it would be possible to implement RTCWEB mobility (e.g. handoff of media between interfaces)?
[SH] I think that is the assumption. Quote from the API spec "...any 
time the network topology changes and the ICE Agent performs an ICE 
Restart".
>
>
>


From randell-ietf@jesup.org  Tue Jul  5 19:54:42 2011
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Subject: Re: [rtcweb] Changing video size (Re: use case:)
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On 7/5/2011 6:30 PM, Cullen Jennings wrote:
> Agree with folks on this thread about this being a way for the receiver to tell the send it's preferences - Perhaps the receiver might indicate that it would like 1080P video but the best the sender can do is 720P because that is the size of camera they have.  In this case the sender does what the sender wants ignoring the receiver. There are other cases where say the receiver can not decode a bitstream above 768 kbps but the sender would typically send a stream at a much higher rate than this. In this case the sender needs to do what the receiver wants.

These sorts of issues are at least partly addressed in the O/A section 
of the H.264 spec (RFC 6184, update to RFC 3984).  Obviously that only 
applies to H.264.  There are also issues about limitations due to the 
other direction; in particular look at level-asymmetry-allowed and what 
happens if it's not used.

As mentioned, there are issues both of initial negotiation (as implied 
here) and in-call changes.

I think a) the two sides should indicate what they support receiving in 
negotiation, and b) indicate their initial preferred resolution/etc in 
negotiation.  In-call, there should be a mechanism for changing 
preference without a full call renegotiation (I suggest an RTCP-FB 
message), though one could fall back on a full new SDP exchange.

> You might have an old browser that support vp8 and newer version of the browser that does both vp8 and vp9 and you use vp8 when that is all you have but prefer vp9 if both sides support it.

That's just classic SDP and Offer/Answer:

New:
       m=video 1234 RTP/AVP 98 99 100
       a=rtpmap 98 VP9/90000
       a=rtpmap 99 VP8/90000
       a=rtpmap 100 H264/90000

Old:
       m=video 1234 RTP/AVP 99 100
       a=rtpmap 99 VP8/90000
       a=rtpmap 100 H264/90000

> I'm not sure exactly how to phrase all this but roughly speaking we need a way for the sender and receiver to agree on a format that meets the constraints of both and somewhat optimizes the desiree format within these constraints.

RFC 6236 certainly looks like a reasonable option to consider; in fact 
I'd consider it to be the
defacto choice barring some problematic limitation.  It allows asymmetry 
within the limits of the codec
negotiation; if RFC 6184 is used for H.264 then level asymmetry is allowed.

There *might* be an argument for also describing the portions of the 
decoded window that aren't visible
(per the previous conversation).

My biggest issue is that they assume it can only be changed via UPDATE 
or re-INVITE.  Maybe that's
reasonable, but I have to say I'd lean towards solutions that are direct 
end-to-end in-call to cut overhead
and lower reaction time.

   Randell

>
> On Jun 22, 2011, at 12:06 AM, Stefan Hkansson LK wrote:
>
>> 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 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] 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
>>
>> _______________________________________________
>> 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
>
>


-- 
Randell Jesup
randell-ietf@jesup.org


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Date: Wed, 06 Jul 2011 15:21:45 -0500
From: Henry Sinnreich <henry.sinnreich@gmail.com>
To: Jonathan Rosenberg <jdrosen@jdrosen.net>, <rtcweb@ietf.org>, Cullen Jennings <fluffy@cisco.com>
Message-ID: <CA3A2E09.1C230%henry.sinnreich@gmail.com>
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+1
Though usually quite skeptical of I-D's not supported by open, available
code and test results, I find this I-D is quite refreshing, realistic and
based on the huge deployment experience of the products supported by the
authors.

The only nit at first reading is not considering audio/video/IM chat as the
normal minimal deployment scenario, either CS or P2P.

Thanks, Henry


On 7/4/11 5:19 PM, "Jonathan Rosenberg" <jdrosen@jdrosen.net> wrote:

> We put together the draft below to argue in favor of using a single port
> for multiplexing voice and video traffic over RTP.
> 
> Looking forward to a lively discussion on this, I am sure.. ;)
> 
> Thanks,
> Jonathan R.
> 
> -------- Original Message --------
> Subject: I-D Action: draft-rosenberg-rtcweb-rtpmux-00.txt
> Date: Mon, 04 Jul 2011 15:14:25 -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           : Multiplexing of Real-Time Transport Protocol (RTP)
> Traffic for Browser based Real-Time Communications (RTC)
> Author(s)       : Jonathan Rosenberg
>                            Cullen Jennings
>                            Jon Peterson
>                            Matthew Kaufman
>                            Eric Rescorla
>                            Tim Terriberry
> Filename        : draft-rosenberg-rtcweb-rtpmux-00.txt
> Pages           : 12
> Date            : 2011-07-04
> 
>     This document argues that multiplexing of voice and video traffic
>     over a single RTP session should be specified as the baseline mode of
>     operation for multimedia traffic in RTC web.
> 
> 
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-rosenberg-rtcweb-rtpmux-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-rosenberg-rtcweb-rtpmux-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
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb



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On Jul 6, 2011, at 2:21 PM, Henry Sinnreich wrote:

> Though usually quite skeptical of I-D's not supported by open, available
> code and test results

I hope to have open, available code, and results by later this year ... 


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Added - and if we can figure out a way to get more time in that section, =
we will...

On Jul 5, 2011, at 10:45 AM, Stephan Wenger wrote:

> Hi Cullen, all,
> We would like to have a few minute sin the last subsection to =
introduce
> layered codecs: draft-wenger-rtcweb-layered-codec-00
> <http://datatracker.ietf.org/doc/draft-wenger-rtcweb-layered-codec/>
> Thanks,
> Stephan
>=20
> On 7.5.2011 09:00 , "Cullen Jennings" <fluffy@cisco.com> wrote:
>=20
>>=20
>> Here is the rough agenda that Ted, Magnus, and I have been thinking
>> of..... bash away ...
>>=20
>>=20
>>=20
>> Agenda - version 2
>>=20
>>         Agenda Bash
>>         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 ( 20 min )
>>         draft-ietf-rtcweb-use-cases-and-requirements
>>=20
>>         Architecture and System Overview ( 20 min)
>>         draft-ietf-rtcweb-overview
>>=20
>>         Security ( 20 min )
>>         draft-rescorla-rtcweb-security
>>         draft-kaufman-rtcweb-security-ui
>>         draft-johnston-rtcweb-media-privacy
>>=20
>>         NAT ( 20 min)
>>         draft-kaufman-rtcweb-traversal
>>         draft-cbran-rtcweb-nat
>>=20
>>=20
>>=20
>>         Start of Session two (Chairs) - 5 min
>>=20
>>         Negotiation & signaling (25 min)
>>         draft-cbran-rtcweb-negotiation
>>=20
>>=20
>>         Media (other than multiplexing) (20 min)
>>         draft-perkins-rtcweb-rtp-usage
>>         draft-cbran-rtcweb-media
>>=20
>>=20
>>         Datagram transports (20 min)
>>         draft-cbran-rtcweb-data
>>=20
>>=20
>>         Multiplexing (30 min)
>>         draft-rosenberg-rtcweb-rtpmux
>>         draft-perkins-rtcweb-rtp-usage
>>         * focus on how/if we multiplex on one UDP port
>>=20
>>=20
>>         Media processing & codec (20 min)
>>         draft-cbran-rtcweb-codec
>>         draft-alvestrand-dispatch-rtcweb-protocols
>>         * focus on requirements for codecs
>>=20
>>=20
>>=20
>>=20
>>=20
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
>=20


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On Jul 6, 2011, at 12:06 AM, Bernard Aboba wrote:

>=20
>=20
> * The last slide now says that WhatWG proposal is low path; this is =
not true. The WhatWG proposal as I read it is mute on how the SDPs are =
exchanged, it just hands those SDPs over to the=20
>=20
> application (though high path is indicated: "Communications are =
coordinated via a signaling channel provided by script in the page via =
the server, e.g. using XMLHttpRequest.")

Hmm - my understanding was only the ICE related information was =
communicated that way and the SDP related information when across the =
channel set up by ICE but I could be totally wrong. It's pretty vague in =
the spec - I got that idea by talking to some of the people involved. =
Anyways, glad to retract that WhatWG is doing low path and just change =
it to not clear.=20



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On Jun 28, 2011, at 6:52 AM, 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.

People objected. WIth my chair hat on I do not yet see consensus on this =
topic one way or the other.=20=

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On Jun 27, 2011, at 1:36 AM, Magnus Westerlund wrote:
>=20
> Does anyone like to add additional use cases?

BFCP=20

I'm interested in the use cases that have real time requirements and =
latency is an issue.

>=20
>=20
> Are you supporting the inclusion of a unreliable datagram service
> directly between peers?
yes

> 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 don't have a big need for this but don't have a problem with it if =
others need it=20

> 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
+1=20

> - Datagram oriented
>   * Size limited by MTU
>     - Path MTU discovery needed
+1

>   * Fragmentation by the application
+1

> - Low latency, i.e. Peer to Peer preferable
+5

> - Congestion Controlled, to be
>   * Network friendly
+1
>   * Not become a Denial of Service tool
+1

> - Security
>  * Confidentiality
+1
>  * Integrity Protected
+1
>  * Source Authenticated (at least bound to the signalling peer)
+1 - I might phrase this a bit differently but I think we are getting at =
the same thing. If we are going to have encryption, there no point in =
encrypting something if you have no idea you our sending encrypted data =
to and from. We need some sort of identity / autehntication

>  * Ensure consent to receive data
+5

In addition to your, I'd like to add that I am trying to maximize the =
odds of it making though the various NATs and Firewalls involved with =
and enterprise or home user.=20

Cullen in my individual contributor role.



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Subject: Re: [rtcweb] Changing video size (Re: use case:)
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On Jul 5, 2011, at 8:55 PM, Randell Jesup wrote:

> RFC 6236 certainly looks like a reasonable option to consider; in fact =
I'd consider it to be the
> defacto choice barring some problematic limitation. =20

yah, make sense to me

> There *might* be an argument for also describing the portions of the =
decoded window that aren't visible
> (per the previous conversation).

I worry about too much "optimization" :-)

>=20
> My biggest issue is that they assume it can only be changed via UPDATE =
or re-INVITE.  Maybe that's
> reasonable, but I have to say I'd lean towards solutions that are =
direct end-to-end in-call to cut overhead
> and lower reaction time.

It one of the arguments people make in favor of the "low path". Or =
perhaps H.245 :-)


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I'm confused about it. So how does the current implementation negotiate =
the codec information?=20


On Jun 30, 2011, at 2:39 AM, Niklas Enbom wrote:

> 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.
>=20
> Niklas
>=20


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My understanding of the API is that the signalling callback is handed the S=
DP blob which includes ICE candidates gathered as directed=2C to manipulate=
 and send as it pleases.  Once an answer is received it is passed down.  Th=
e spec is indeed vague on some points=2C but hey=2C that's what sample code=
 is for : (

> Subject: Re: [rtcweb] Media neg-eo-tiation and signalling archa-tecture
> From: fluffy@cisco.com
> Date: Wed=2C 6 Jul 2011 21:43:25 -0600
> CC: rtcweb@ietf.org
> To: bernard_aboba@hotmail.com
>=20
>=20
> On Jul 6=2C 2011=2C at 12:06 AM=2C Bernard Aboba wrote:
>=20
> >=20
> >=20
> > * The last slide now says that WhatWG proposal is low path=3B this is n=
ot true. The WhatWG proposal as I read it is mute on how the SDPs are excha=
nged=2C it just hands those SDPs over to the=20
> >=20
> > application (though high path is indicated: "Communications are coordin=
ated via a signaling channel provided by script in the page via the server=
=2C e.g. using XMLHttpRequest.")
>=20
> Hmm - my understanding was only the ICE related information was communica=
ted that way and the SDP related information when across the channel set up=
 by ICE but I could be totally wrong. It's pretty vague in the spec - I got=
 that idea by talking to some of the people involved. Anyways=2C glad to re=
tract that WhatWG is doing low path and just change it to not clear.=20
>=20
>=20
 		 	   		  =

--_8672e6dc-4b4e-478c-8ff3-4ca1f309f3ef_
Content-Type: text/html; charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<html>
<head>
<style><!--
.hmmessage P
{
margin:0px=3B
padding:0px
}
body.hmmessage
{
font-size: 10pt=3B
font-family:Tahoma
}
--></style>
</head>
<body class=3D'hmmessage'><div dir=3D'ltr'>
My understanding of the API is that the signalling callback is handed the S=
DP blob which includes ICE candidates gathered as directed=2C to manipulate=
 and send as it pleases.&nbsp=3B Once an answer is received it is passed do=
wn.&nbsp=3B The spec is indeed vague on some points=2C but hey=2C that's wh=
at sample code is for : (<br><br><div>&gt=3B Subject: Re: [rtcweb] Media ne=
g-eo-tiation and signalling archa-tecture<br>&gt=3B From: fluffy@cisco.com<=
br>&gt=3B Date: Wed=2C 6 Jul 2011 21:43:25 -0600<br>&gt=3B CC: rtcweb@ietf.=
org<br>&gt=3B To: bernard_aboba@hotmail.com<br>&gt=3B <br>&gt=3B <br>&gt=3B=
 On Jul 6=2C 2011=2C at 12:06 AM=2C Bernard Aboba wrote:<br>&gt=3B <br>&gt=
=3B &gt=3B <br>&gt=3B &gt=3B <br>&gt=3B &gt=3B * The last slide now says th=
at WhatWG proposal is low path=3B this is not true. The WhatWG proposal as =
I read it is mute on how the SDPs are exchanged=2C it just hands those SDPs=
 over to the <br>&gt=3B &gt=3B <br>&gt=3B &gt=3B application (though high p=
ath is indicated: "Communications are coordinated via a signaling channel p=
rovided by script in the page via the server=2C e.g. using XMLHttpRequest."=
)<br>&gt=3B <br>&gt=3B Hmm - my understanding was only the ICE related info=
rmation was communicated that way and the SDP related information when acro=
ss the channel set up by ICE but I could be totally wrong. It's pretty vagu=
e in the spec - I got that idea by talking to some of the people involved. =
Anyways=2C glad to retract that WhatWG is doing low path and just change it=
 to not clear. <br>&gt=3B <br>&gt=3B <br></div> 		 	   		  </div></body>
</html>=

--_8672e6dc-4b4e-478c-8ff3-4ca1f309f3ef_--

From randell1@jesup.org  Wed Jul  6 22:19:10 2011
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Subject: Re: [rtcweb] Non-media data service consensus and requirements
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On 7/6/2011 11:57 PM, Cullen Jennings wrote:
>> Are you supporting the inclusion of a unreliable datagram service
>> directly between peers?
> yes

I support it as well.

>> 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 don't have a big need for this but don't have a problem with it if others need it

I think we need to provide at least unreliable datagram service, and 
probably reliable datagram service.

>> 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'd rather have one well-debugged standard reliable datagram service 
than N so-so debugged 3rd-party services, which might not "play well 
together".

>> 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
> +1
>
>> - Datagram oriented
>>    * Size limited by MTU
>>      - Path MTU discovery needed
> +1
>
>>    * Fragmentation by the application
> +1
>
>> - Low latency, i.e. Peer to Peer preferable
> +5
>
>> - Congestion Controlled, to be
>>    * Network friendly
> +1

And I'd say (generally to all the part of rtcweb) Unified Congestion 
Control of all streams that are part of the same session.

>>    * Not become a Denial of Service tool
> +1
>
>> - Security
>>   * Confidentiality
> +1
>>   * Integrity Protected
> +1
>>   * Source Authenticated (at least bound to the signalling peer)
> +1 - I might phrase this a bit differently but I think we are getting at the same thing. If we are going to have encryption, there no point in encrypting something if you have no idea you our sending encrypted data to and from. We need some sort of identity / autehntication

I'm not sure I agree on Integrity Protection and Source Authentication.  
There are use cases where we want to have a private conversation (no 
eavesdropping or video leaked to Youtube!), but don't care if someone 
can block or disrupt the call.  (Someone with control of a link or 
midbox can do that already.)  I think those are important optional services.

>>   * Ensure consent to receive data
> +5
>
> In addition to your, I'd like to add that I am trying to maximize the odds of it making though the various NATs and Firewalls involved with and enterprise or home user.

+5

-- 
Randell Jesup
randell-ietf@jesup.org


From alan.b.johnston@gmail.com  Thu Jul  7 06:44:07 2011
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This is a reasonable start to expressing in requirement form some of
the ideas described in draft-johnston-rtcweb-media-privacy and that I
presented on the interim call.  Some additional comments:

At the start, it should be made clear that the UI described is the
browser UI and not anything rendered by the web page.  This is
described in the last requirement of Section 2, but it needs to be
clearly stated up front.

Most of the UI descriptions seem to be described in terms of an
inspector - with additional clicks a user can determine additional
information about the media session, which is reasonable.  However, if
we do allow non-encrypted media sessions (which I really hope we do
not - this goes against the current direction of web security in
moving from http to https), then the encrypted/non encrypted state
needs to be visible as much as possible, in the same way that a user
does not need to click anything to see if http or https is in use.  Of
course, full screen modes and other considerations can limit this.

"  If the transmission is encrypted, the "security characteristics" MUST
   include an indication as to the source of the keying material,
   particularly whether the keying material was delivered out-of-band
   (from a server) or was generated as a result of a pairwise
   negotiation."

I think I understand and agree with this requirement.  However, I
think we can do better in our description of the keying material.  I
think the distinction we are after is whether the keying material was
derived directly between the browsers, or between the browser and the
server.

"   If possible for the cryptosystem in use, the "security
   characteristics" MUST include information regarding the authenticity
   of the far station identity.  (For example, in the case of a self-
   signed certificate with RSA key the contents of the certificate and
   the key fingerprint.)"

I'm not sure about this one.  Unless a PKI is in use, I don't think
there is any useful identity information that can be derived.  I've
been told in the past that "self signed certificates" are actually
just containers for a public key, and as such none of the information
is useful.  Also, I don't understand the value in providing a key
fingerprint to a user.  This might be useful for debugging, but it
isn't of use to users.  Ekr has provided a reference in the past why
this is so.

" If possible for the cryptosystem in use, the "security
   characteristics" SHOULD include a Short Authentication String which
   may be used by the user to authenticate the far station identity and
   keying integrity (specifically, the presence or lack of a man-in-the-
   middle that may be in collusion with the service provider to attempt
   to bypass authentication tests) by communicating this string out-of-
   band with the far party."

I think providing a SAS is very important, so I believe this
requirement to be a MUST.  Also, this wording about "if possible"
might be interpreted to mean "if ZRTP is used".  I think having an SAS
is valuable for any keying method that is browser to browser - I don't
think it has any value for broswer to server keying.  Since the SAS is
just an algorithm based on the secret key used, it should be be
possible to derive one for other keying approaches.  However, we will
need help from crypto experts to make sure the SAS is of sufficient
length and derived correctly to make it useful.   I recall with ZRTP
that we were able to utilize such a short SAS due to the hash
commitment design - other keying methods will need a similar analysis.

Looking forward to the discussion of these issues in Quebec City.

- Alan -

On Wed, Jun 29, 2011 at 6:38 PM, Matthew Kaufman
<matthew.kaufman@skype.net> wrote:
> 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
>
> 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 reposito=
ry.
>
> 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
>
> Abstract:
> =A0=A0This document calls for a requirement to be imposed on RTCWEB clien=
t
> =A0=A0user interfaces whereby the user may inspect the current media
> =A0=A0security status.
>
>
>
>
> The IETF Secretariat
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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I went and looked at some call flows to try and sort out what is going =
on in the current released chrome code and as far as I can tell, Bernard =
you are right, it is passing the data over the HTTP channel. An example =
of what I see is below=20

As far as I can tell this is the JSON encoding of XEP-180 used in jingle =
and a separate JSON encoding of some of SDP. Not sure this matters much =
one way or the other but wanted to pass on what I had found and I will =
correct the slides at some point - my apologies for getting them wrong.=20=


Cullen



{
   "media" : [
      {
         "attributes" : {
            "candidate" : [
               {
                  "component" : 1,
                  "foundation" : 1,
                  "generation" : 0,
                  "ip" : "144.254.150.241",
                  "name" : "rtp",
                  "network_name" : "Intel(R) WiFi Link 1000 BGN",
                  "password" : "OpawIyXLYcJkwxFa",
                  "port" : "61880",
                  "priority" : 1.0,
                  "proto" : "udp",
                  "type" : "local",
                  "username" : "90B0CPvC8p8XPMFt"
               }
            ]
         },
         "label" : 1,
         "rtpmap" : [
            {
               "103" : {
                  "codec" : "audio/ISAC"
               }
            },
            {
               "104" : {
                  "codec" : "audio/ISAC"
               }
            },
            {
               "9" : {
                  "codec" : "audio/G722"
               }
            },
            {
               "102" : {
                  "codec" : "audio/iLBC"
               }
            },
            {
               "0" : {
                  "codec" : "audio/PCMU"
               }
            },
            {
               "8" : {
                  "codec" : "audio/PCMA"
               }
            },
            {
               "99" : {
                  "codec" : "audio/CN"
               }
            },
            {
               "98" : {
                  "codec" : "audio/CN"
               }
            },
            {
               "13" : {
                  "codec" : "audio/CN"
               }
            },
            {
               "127" : {
                  "codec" : "audio/red"
               }
            },
            {
               "106" : {
                  "codec" : "audio/telephone-event"
               }
            }
         ]
      },
      {
         "attributes" : {
            "candidate" : [
               {
                  "component" : 1,
                  "foundation" : 1,
                  "generation" : 0,
                  "ip" : "144.254.150.241",
                  "name" : "video_rtp",
                  "network_name" : "Intel(R) WiFi Link 1000 BGN",
                  "password" : "G4dq4e3f/G7FDJ5h",
                  "port" : "61879",
                  "priority" : 1.0,
                  "proto" : "udp",
                  "type" : "local",
                  "username" : "CBkZYp6/DTTMEFeM"
               }
            ]
         },
         "label" : 2,
         "rtpmap" : [
            {
               "120" : {
                  "codec" : "video/VP8"
               }
            }
         ]
      }
   ]
}


On Jul 6, 2011, at 9:43 PM, Cullen Jennings wrote:

>=20
> On Jul 6, 2011, at 12:06 AM, Bernard Aboba wrote:
>=20
>>=20
>>=20
>> * The last slide now says that WhatWG proposal is low path; this is =
not true. The WhatWG proposal as I read it is mute on how the SDPs are =
exchanged, it just hands those SDPs over to the=20
>>=20
>> application (though high path is indicated: "Communications are =
coordinated via a signaling channel provided by script in the page via =
the server, e.g. using XMLHttpRequest.")
>=20
> Hmm - my understanding was only the ICE related information was =
communicated that way and the SDP related information when across the =
channel set up by ICE but I could be totally wrong. It's pretty vague in =
the spec - I got that idea by talking to some of the people involved. =
Anyways, glad to retract that WhatWG is doing low path and just change =
it to not clear.=20
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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--0016367658843677bb04a78a34c8
Content-Type: text/plain; charset=ISO-8859-1

Not sure in which thread I should answer... In Chrome we're trying to
implement the whatwg proposal with one intentional discrepancy, and that's
the string format pointed out by Cullen. See details in this thread:
http://www.mail-archive.com/whatwg@lists.whatwg.org/msg26163.html. There's
also non-intentional discrepancies that we know of, which we're trying to
remove.

Just be careful when looking at Chrome code to interpret the whatwg spec.
We're trying to interpret that spec in the same way as anyone else, and the
whatwg author is not involved in the Chrome implementation.

Niklas

On Thu, Jul 7, 2011 at 7:52 PM, Cullen Jennings <fluffy@cisco.com> wrote:

>
> I went and looked at some call flows to try and sort out what is going on
> in the current released chrome code and as far as I can tell, Bernard you
> are right, it is passing the data over the HTTP channel. An example of what
> I see is below
>
> As far as I can tell this is the JSON encoding of XEP-180 used in jingle
> and a separate JSON encoding of some of SDP. Not sure this matters much one
> way or the other but wanted to pass on what I had found and I will correct
> the slides at some point - my apologies for getting them wrong.
>
> Cullen
>
>
>
> {
>   "media" : [
>      {
>         "attributes" : {
>            "candidate" : [
>               {
>                  "component" : 1,
>                  "foundation" : 1,
>                  "generation" : 0,
>                  "ip" : "144.254.150.241",
>                  "name" : "rtp",
>                  "network_name" : "Intel(R) WiFi Link 1000 BGN",
>                  "password" : "OpawIyXLYcJkwxFa",
>                  "port" : "61880",
>                  "priority" : 1.0,
>                  "proto" : "udp",
>                  "type" : "local",
>                  "username" : "90B0CPvC8p8XPMFt"
>               }
>            ]
>         },
>         "label" : 1,
>         "rtpmap" : [
>            {
>               "103" : {
>                  "codec" : "audio/ISAC"
>               }
>            },
>            {
>               "104" : {
>                  "codec" : "audio/ISAC"
>               }
>            },
>            {
>               "9" : {
>                  "codec" : "audio/G722"
>               }
>            },
>            {
>               "102" : {
>                  "codec" : "audio/iLBC"
>               }
>            },
>            {
>               "0" : {
>                  "codec" : "audio/PCMU"
>               }
>            },
>            {
>               "8" : {
>                  "codec" : "audio/PCMA"
>               }
>            },
>            {
>               "99" : {
>                  "codec" : "audio/CN"
>               }
>            },
>            {
>               "98" : {
>                  "codec" : "audio/CN"
>               }
>            },
>            {
>               "13" : {
>                  "codec" : "audio/CN"
>               }
>            },
>            {
>               "127" : {
>                  "codec" : "audio/red"
>               }
>            },
>            {
>               "106" : {
>                  "codec" : "audio/telephone-event"
>               }
>            }
>         ]
>      },
>      {
>         "attributes" : {
>            "candidate" : [
>               {
>                  "component" : 1,
>                  "foundation" : 1,
>                  "generation" : 0,
>                  "ip" : "144.254.150.241",
>                  "name" : "video_rtp",
>                  "network_name" : "Intel(R) WiFi Link 1000 BGN",
>                  "password" : "G4dq4e3f/G7FDJ5h",
>                  "port" : "61879",
>                  "priority" : 1.0,
>                  "proto" : "udp",
>                  "type" : "local",
>                  "username" : "CBkZYp6/DTTMEFeM"
>               }
>            ]
>         },
>         "label" : 2,
>         "rtpmap" : [
>            {
>               "120" : {
>                  "codec" : "video/VP8"
>               }
>            }
>         ]
>      }
>   ]
> }
>
>
> On Jul 6, 2011, at 9:43 PM, Cullen Jennings wrote:
>
> >
> > On Jul 6, 2011, at 12:06 AM, Bernard Aboba wrote:
> >
> >>
> >>
> >> * The last slide now says that WhatWG proposal is low path; this is not
> true. 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 indicated: "Communications are
> coordinated via a signaling channel provided by script in the page via the
> server, e.g. using XMLHttpRequest.")
> >
> > Hmm - my understanding was only the ICE related information was
> communicated that way and the SDP related information when across the
> channel set up by ICE but I could be totally wrong. It's pretty vague in the
> spec - I got that idea by talking to some of the people involved. Anyways,
> glad to retract that WhatWG is doing low path and just change it to not
> clear.
> >
> >
> > _______________________________________________
> > 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
>

--0016367658843677bb04a78a34c8
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Not sure in which thread I should answer... In Chrome we&#39;re trying to i=
mplement the whatwg proposal with one intentional discrepancy, and that&#39=
;s the string format pointed out by Cullen. See details in this thread:=A0<=
span class=3D"Apple-style-span" style=3D"font-family: arial, sans-serif; fo=
nt-size: 13px; background-color: rgb(255, 255, 255); "><a href=3D"http://ww=
w.mail-archive.com/whatwg@lists.whatwg.org/msg26163.html" target=3D"_blank"=
 style=3D"color: rgb(0, 0, 204); ">http://www.mail-archive.com/whatwg@lists=
.whatwg.org/msg26163.html</a>. There&#39;s also non-intentional discrepanci=
es that we know of, which we&#39;re trying to remove.</span><div>
<font class=3D"Apple-style-span" face=3D"arial, sans-serif"><br></font></di=
v><div><font class=3D"Apple-style-span" face=3D"arial, sans-serif">Just be =
careful when looking at Chrome code to interpret the whatwg spec. We&#39;re=
 trying to interpret that spec in the same way as anyone else, and the what=
wg author is not involved in the Chrome implementation.</font></div>
<div><font class=3D"Apple-style-span" face=3D"arial, sans-serif"><br></font=
></div><div><font class=3D"Apple-style-span" face=3D"arial, sans-serif">Nik=
las<br></font><div><br><div class=3D"gmail_quote">On Thu, Jul 7, 2011 at 7:=
52 PM, Cullen Jennings <span dir=3D"ltr">&lt;<a href=3D"mailto:fluffy@cisco=
.com">fluffy@cisco.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;"><br>
I went and looked at some call flows to try and sort out what is going on i=
n the current released chrome code and as far as I can tell, Bernard you ar=
e right, it is passing the data over the HTTP channel. An example of what I=
 see is below<br>

<br>
As far as I can tell this is the JSON encoding of XEP-180 used in jingle an=
d a separate JSON encoding of some of SDP. Not sure this matters much one w=
ay or the other but wanted to pass on what I had found and I will correct t=
he slides at some point - my apologies for getting them wrong.<br>

<br>
Cullen<br>
<br>
<br>
<br>
{<br>
 =A0 &quot;media&quot; : [<br>
 =A0 =A0 =A0{<br>
 =A0 =A0 =A0 =A0 &quot;attributes&quot; : {<br>
 =A0 =A0 =A0 =A0 =A0 =A0&quot;candidate&quot; : [<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 {<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&quot;component&quot; : 1,<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&quot;foundation&quot; : 1,<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&quot;generation&quot; : 0,<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&quot;ip&quot; : &quot;144.254.150.241&=
quot;,<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&quot;name&quot; : &quot;rtp&quot;,<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&quot;network_name&quot; : &quot;Intel(=
R) WiFi Link 1000 BGN&quot;,<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&quot;password&quot; : &quot;OpawIyXLYc=
JkwxFa&quot;,<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&quot;port&quot; : &quot;61880&quot;,<b=
r>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&quot;priority&quot; : 1.0,<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&quot;proto&quot; : &quot;udp&quot;,<br=
>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&quot;type&quot; : &quot;local&quot;,<b=
r>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&quot;username&quot; : &quot;90B0CPvC8p=
8XPMFt&quot;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 }<br>
 =A0 =A0 =A0 =A0 =A0 =A0]<br>
 =A0 =A0 =A0 =A0 },<br>
 =A0 =A0 =A0 =A0 &quot;label&quot; : 1,<br>
 =A0 =A0 =A0 =A0 &quot;rtpmap&quot; : [<br>
 =A0 =A0 =A0 =A0 =A0 =A0{<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 &quot;103&quot; : {<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&quot;codec&quot; : &quot;audio/ISAC&qu=
ot;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 }<br>
 =A0 =A0 =A0 =A0 =A0 =A0},<br>
 =A0 =A0 =A0 =A0 =A0 =A0{<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 &quot;104&quot; : {<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&quot;codec&quot; : &quot;audio/ISAC&qu=
ot;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 }<br>
 =A0 =A0 =A0 =A0 =A0 =A0},<br>
 =A0 =A0 =A0 =A0 =A0 =A0{<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 &quot;9&quot; : {<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&quot;codec&quot; : &quot;audio/G722&qu=
ot;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 }<br>
 =A0 =A0 =A0 =A0 =A0 =A0},<br>
 =A0 =A0 =A0 =A0 =A0 =A0{<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 &quot;102&quot; : {<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&quot;codec&quot; : &quot;audio/iLBC&qu=
ot;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 }<br>
 =A0 =A0 =A0 =A0 =A0 =A0},<br>
 =A0 =A0 =A0 =A0 =A0 =A0{<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 &quot;0&quot; : {<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&quot;codec&quot; : &quot;audio/PCMU&qu=
ot;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 }<br>
 =A0 =A0 =A0 =A0 =A0 =A0},<br>
 =A0 =A0 =A0 =A0 =A0 =A0{<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 &quot;8&quot; : {<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&quot;codec&quot; : &quot;audio/PCMA&qu=
ot;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 }<br>
 =A0 =A0 =A0 =A0 =A0 =A0},<br>
 =A0 =A0 =A0 =A0 =A0 =A0{<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 &quot;99&quot; : {<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&quot;codec&quot; : &quot;audio/CN&quot=
;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 }<br>
 =A0 =A0 =A0 =A0 =A0 =A0},<br>
 =A0 =A0 =A0 =A0 =A0 =A0{<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 &quot;98&quot; : {<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&quot;codec&quot; : &quot;audio/CN&quot=
;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 }<br>
 =A0 =A0 =A0 =A0 =A0 =A0},<br>
 =A0 =A0 =A0 =A0 =A0 =A0{<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 &quot;13&quot; : {<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&quot;codec&quot; : &quot;audio/CN&quot=
;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 }<br>
 =A0 =A0 =A0 =A0 =A0 =A0},<br>
 =A0 =A0 =A0 =A0 =A0 =A0{<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 &quot;127&quot; : {<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&quot;codec&quot; : &quot;audio/red&quo=
t;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 }<br>
 =A0 =A0 =A0 =A0 =A0 =A0},<br>
 =A0 =A0 =A0 =A0 =A0 =A0{<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 &quot;106&quot; : {<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&quot;codec&quot; : &quot;audio/telepho=
ne-event&quot;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 }<br>
 =A0 =A0 =A0 =A0 =A0 =A0}<br>
 =A0 =A0 =A0 =A0 ]<br>
 =A0 =A0 =A0},<br>
 =A0 =A0 =A0{<br>
 =A0 =A0 =A0 =A0 &quot;attributes&quot; : {<br>
 =A0 =A0 =A0 =A0 =A0 =A0&quot;candidate&quot; : [<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 {<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&quot;component&quot; : 1,<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&quot;foundation&quot; : 1,<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&quot;generation&quot; : 0,<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&quot;ip&quot; : &quot;144.254.150.241&=
quot;,<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&quot;name&quot; : &quot;video_rtp&quot=
;,<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&quot;network_name&quot; : &quot;Intel(=
R) WiFi Link 1000 BGN&quot;,<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&quot;password&quot; : &quot;G4dq4e3f/G=
7FDJ5h&quot;,<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&quot;port&quot; : &quot;61879&quot;,<b=
r>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&quot;priority&quot; : 1.0,<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&quot;proto&quot; : &quot;udp&quot;,<br=
>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&quot;type&quot; : &quot;local&quot;,<b=
r>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&quot;username&quot; : &quot;CBkZYp6/DT=
TMEFeM&quot;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 }<br>
 =A0 =A0 =A0 =A0 =A0 =A0]<br>
 =A0 =A0 =A0 =A0 },<br>
 =A0 =A0 =A0 =A0 &quot;label&quot; : 2,<br>
 =A0 =A0 =A0 =A0 &quot;rtpmap&quot; : [<br>
 =A0 =A0 =A0 =A0 =A0 =A0{<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 &quot;120&quot; : {<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0&quot;codec&quot; : &quot;video/VP8&quo=
t;<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 }<br>
 =A0 =A0 =A0 =A0 =A0 =A0}<br>
 =A0 =A0 =A0 =A0 ]<br>
 =A0 =A0 =A0}<br>
 =A0 ]<br>
<div><div></div><div class=3D"h5">}<br>
<br>
<br>
On Jul 6, 2011, at 9:43 PM, Cullen Jennings wrote:<br>
<br>
&gt;<br>
&gt; On Jul 6, 2011, at 12:06 AM, Bernard Aboba wrote:<br>
&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; * The last slide now says that WhatWG proposal is low path; this i=
s not true. The WhatWG proposal as I read it is mute on how the SDPs are ex=
changed, it just hands those SDPs over to the<br>
&gt;&gt;<br>
&gt;&gt; application (though high path is indicated: &quot;Communications a=
re coordinated via a signaling channel provided by script in the page via t=
he server, e.g. using XMLHttpRequest.&quot;)<br>
&gt;<br>
&gt; Hmm - my understanding was only the ICE related information was commun=
icated that way and the SDP related information when across the channel set=
 up by ICE but I could be totally wrong. It&#39;s pretty vague in the spec =
- I got that idea by talking to some of the people involved. Anyways, glad =
to retract that WhatWG is doing low path and just change it to not clear.<b=
r>

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

--0016367658843677bb04a78a34c8--

From fluffy@cisco.com  Fri Jul  8 07:11:35 2011
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From: Cullen Jennings <fluffy@cisco.com>
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To: Niklas Enbom <niklase@google.com>
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Media neg-eo-tiation and signalling archa-tecture
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Makes sense - thanks. I'm trying to interpret the spec by looking at the =
code. I'll stop doing that :-)

On Jul 8, 2011, at 1:58 AM, Niklas Enbom wrote:

> Not sure in which thread I should answer... In Chrome we're trying to =
implement the whatwg proposal with one intentional discrepancy, and =
that's the string format pointed out by Cullen. See details in this =
thread: =
http://www.mail-archive.com/whatwg@lists.whatwg.org/msg26163.html. =
There's also non-intentional discrepancies that we know of, which we're =
trying to remove.
>=20
> Just be careful when looking at Chrome code to interpret the whatwg =
spec. We're trying to interpret that spec in the same way as anyone =
else, and the whatwg author is not involved in the Chrome =
implementation.
>=20
> Niklas
>=20
> On Thu, Jul 7, 2011 at 7:52 PM, Cullen Jennings <fluffy@cisco.com> =
wrote:
>=20
> I went and looked at some call flows to try and sort out what is going =
on in the current released chrome code and as far as I can tell, Bernard =
you are right, it is passing the data over the HTTP channel. An example =
of what I see is below
>=20
> As far as I can tell this is the JSON encoding of XEP-180 used in =
jingle and a separate JSON encoding of some of SDP. Not sure this =
matters much one way or the other but wanted to pass on what I had found =
and I will correct the slides at some point - my apologies for getting =
them wrong.
>=20
> Cullen
>=20
>=20
>=20
> {
>   "media" : [
>      {
>         "attributes" : {
>            "candidate" : [
>               {
>                  "component" : 1,
>                  "foundation" : 1,
>                  "generation" : 0,
>                  "ip" : "144.254.150.241",
>                  "name" : "rtp",
>                  "network_name" : "Intel(R) WiFi Link 1000 BGN",
>                  "password" : "OpawIyXLYcJkwxFa",
>                  "port" : "61880",
>                  "priority" : 1.0,
>                  "proto" : "udp",
>                  "type" : "local",
>                  "username" : "90B0CPvC8p8XPMFt"
>               }
>            ]
>         },
>         "label" : 1,
>         "rtpmap" : [
>            {
>               "103" : {
>                  "codec" : "audio/ISAC"
>               }
>            },
>            {
>               "104" : {
>                  "codec" : "audio/ISAC"
>               }
>            },
>            {
>               "9" : {
>                  "codec" : "audio/G722"
>               }
>            },
>            {
>               "102" : {
>                  "codec" : "audio/iLBC"
>               }
>            },
>            {
>               "0" : {
>                  "codec" : "audio/PCMU"
>               }
>            },
>            {
>               "8" : {
>                  "codec" : "audio/PCMA"
>               }
>            },
>            {
>               "99" : {
>                  "codec" : "audio/CN"
>               }
>            },
>            {
>               "98" : {
>                  "codec" : "audio/CN"
>               }
>            },
>            {
>               "13" : {
>                  "codec" : "audio/CN"
>               }
>            },
>            {
>               "127" : {
>                  "codec" : "audio/red"
>               }
>            },
>            {
>               "106" : {
>                  "codec" : "audio/telephone-event"
>               }
>            }
>         ]
>      },
>      {
>         "attributes" : {
>            "candidate" : [
>               {
>                  "component" : 1,
>                  "foundation" : 1,
>                  "generation" : 0,
>                  "ip" : "144.254.150.241",
>                  "name" : "video_rtp",
>                  "network_name" : "Intel(R) WiFi Link 1000 BGN",
>                  "password" : "G4dq4e3f/G7FDJ5h",
>                  "port" : "61879",
>                  "priority" : 1.0,
>                  "proto" : "udp",
>                  "type" : "local",
>                  "username" : "CBkZYp6/DTTMEFeM"
>               }
>            ]
>         },
>         "label" : 2,
>         "rtpmap" : [
>            {
>               "120" : {
>                  "codec" : "video/VP8"
>               }
>            }
>         ]
>      }
>   ]
> }
>=20
>=20
> On Jul 6, 2011, at 9:43 PM, Cullen Jennings wrote:
>=20
> >
> > On Jul 6, 2011, at 12:06 AM, Bernard Aboba wrote:
> >
> >>
> >>
> >> * The last slide now says that WhatWG proposal is low path; this is =
not true. 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 indicated: "Communications are =
coordinated via a signaling channel provided by script in the page via =
the server, e.g. using XMLHttpRequest.")
> >
> > Hmm - my understanding was only the ICE related information was =
communicated that way and the SDP related information when across the =
channel set up by ICE but I could be totally wrong. It's pretty vague in =
the spec - I got that idea by talking to some of the people involved. =
Anyways, glad to retract that WhatWG is doing low path and just change =
it to not clear.
> >
> >
> > _______________________________________________
> > 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
>=20


From stefan.lk.hakansson@ericsson.com  Sun Jul 10 05:06:58 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>, Niklas Enbom <niklase@google.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Media neg-eo-tiation and signalling archa-tecture
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We also followed the WhatWG proposal as well as possible in our implementat=
ion (that you can download and test here: https://labs.ericsson.com/apis/we=
b-real-time-communication/downloads). We're using SDP offer/answers as prop=
osed in the whatwg spec; however there is a separate SDP dialogue per strea=
m (and not per session) to avoid glaring conditions.

The work also generated quite some feedback to whatwg: http://lists.whatwg.=
org/htdig.cgi/whatwg-whatwg.org/2011-June/031950.html

Stefan

________________________________________
From: rtcweb-bounces@ietf.org [rtcweb-bounces@ietf.org] On Behalf Of Cullen=
 Jennings [fluffy@cisco.com]
Sent: Friday, July 08, 2011 4:11 PM
To: Niklas Enbom
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Media neg-eo-tiation and signalling archa-tecture

Makes sense - thanks. I'm trying to interpret the spec by looking at the co=
de. I'll stop doing that :-)

On Jul 8, 2011, at 1:58 AM, Niklas Enbom wrote:

> Not sure in which thread I should answer... In Chrome we're trying to imp=
lement the whatwg proposal with one intentional discrepancy, and that's the=
 string format pointed out by Cullen. See details in this thread: http://ww=
w.mail-archive.com/whatwg@lists.whatwg.org/msg26163.html. There's also non-=
intentional discrepancies that we know of, which we're trying to remove.
>
> Just be careful when looking at Chrome code to interpret the whatwg spec.=
 We're trying to interpret that spec in the same way as anyone else, and th=
e whatwg author is not involved in the Chrome implementation.
>
> Niklas
>
> On Thu, Jul 7, 2011 at 7:52 PM, Cullen Jennings <fluffy@cisco.com> wrote:
>
> I went and looked at some call flows to try and sort out what is going on=
 in the current released chrome code and as far as I can tell, Bernard you =
are right, it is passing the data over the HTTP channel. An example of what=
 I see is below
>
> As far as I can tell this is the JSON encoding of XEP-180 used in jingle =
and a separate JSON encoding of some of SDP. Not sure this matters much one=
 way or the other but wanted to pass on what I had found and I will correct=
 the slides at some point - my apologies for getting them wrong.
>
> Cullen
>
>
>
> {
>   "media" : [
>      {
>         "attributes" : {
>            "candidate" : [
>               {
>                  "component" : 1,
>                  "foundation" : 1,
>                  "generation" : 0,
>                  "ip" : "144.254.150.241",
>                  "name" : "rtp",
>                  "network_name" : "Intel(R) WiFi Link 1000 BGN",
>                  "password" : "OpawIyXLYcJkwxFa",
>                  "port" : "61880",
>                  "priority" : 1.0,
>                  "proto" : "udp",
>                  "type" : "local",
>                  "username" : "90B0CPvC8p8XPMFt"
>               }
>            ]
>         },
>         "label" : 1,
>         "rtpmap" : [
>            {
>               "103" : {
>                  "codec" : "audio/ISAC"
>               }
>            },
>            {
>               "104" : {
>                  "codec" : "audio/ISAC"
>               }
>            },
>            {
>               "9" : {
>                  "codec" : "audio/G722"
>               }
>            },
>            {
>               "102" : {
>                  "codec" : "audio/iLBC"
>               }
>            },
>            {
>               "0" : {
>                  "codec" : "audio/PCMU"
>               }
>            },
>            {
>               "8" : {
>                  "codec" : "audio/PCMA"
>               }
>            },
>            {
>               "99" : {
>                  "codec" : "audio/CN"
>               }
>            },
>            {
>               "98" : {
>                  "codec" : "audio/CN"
>               }
>            },
>            {
>               "13" : {
>                  "codec" : "audio/CN"
>               }
>            },
>            {
>               "127" : {
>                  "codec" : "audio/red"
>               }
>            },
>            {
>               "106" : {
>                  "codec" : "audio/telephone-event"
>               }
>            }
>         ]
>      },
>      {
>         "attributes" : {
>            "candidate" : [
>               {
>                  "component" : 1,
>                  "foundation" : 1,
>                  "generation" : 0,
>                  "ip" : "144.254.150.241",
>                  "name" : "video_rtp",
>                  "network_name" : "Intel(R) WiFi Link 1000 BGN",
>                  "password" : "G4dq4e3f/G7FDJ5h",
>                  "port" : "61879",
>                  "priority" : 1.0,
>                  "proto" : "udp",
>                  "type" : "local",
>                  "username" : "CBkZYp6/DTTMEFeM"
>               }
>            ]
>         },
>         "label" : 2,
>         "rtpmap" : [
>            {
>               "120" : {
>                  "codec" : "video/VP8"
>               }
>            }
>         ]
>      }
>   ]
> }
>
>
> On Jul 6, 2011, at 9:43 PM, Cullen Jennings wrote:
>
> >
> > On Jul 6, 2011, at 12:06 AM, Bernard Aboba wrote:
> >
> >>
> >>
> >> * The last slide now says that WhatWG proposal is low path; this is no=
t true. The WhatWG proposal as I read it is mute on how the SDPs are exchan=
ged, it just hands those SDPs over to the
> >>
> >> application (though high path is indicated: "Communications are coordi=
nated via a signaling channel provided by script in the page via the server=
, e.g. using XMLHttpRequest.")
> >
> > Hmm - my understanding was only the ICE related information was communi=
cated that way and the SDP related information when across the channel set =
up by ICE but I could be totally wrong. It's pretty vague in the spec - I g=
ot that idea by talking to some of the people involved. Anyways, glad to re=
tract that WhatWG is doing low path and just change it to not clear.
> >
> >
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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

From harald@alvestrand.no  Mon Jul 11 06:44:02 2011
Return-Path: <harald@alvestrand.no>
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Subject: Re: [rtcweb] Media neg-eo-tiation and signalling archa-tecture
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On 07/07/11 05:43, Cullen Jennings wrote:
> On Jul 6, 2011, at 12:06 AM, Bernard Aboba wrote:
>
>>
>> * The last slide now says that WhatWG proposal is low path; this is not true. 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 indicated: "Communications are coordinated via a signaling channel provided by script in the page via the server, e.g. using XMLHttpRequest.")
> Hmm - my understanding was only the ICE related information was communicated that way and the SDP related information when across the channel set up by ICE but I could be totally wrong. It's pretty vague in the spec - I got that idea by talking to some of the people involved. Anyways, glad to retract that WhatWG is doing low path and just change it to not clear.
Google's current implementation of the WhatWG spec is "high path". If 
you dump the requests that the example client programs pass to the 
mediation server (which is talking fairly normal HTTP), you will see the 
JSON objects that contain the codec negotiation information.

>
> _______________________________________________
> 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 06/22/11 05:59, Justin Uberti wrote:
> 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.
There might be some things it can do, even if it can neither recode nor 
do SVC-style tricks.
For instance, if it's a distribution point (one incoming video stream, 
many outgoing copies), it can keep track of the maximal video size 
requested by any of its downstreams, and signal the upstream to send a 
resolution that is no higher than the largest resolution requested.

I agree that we can't demand that the sender send exactly the resolution 
requested; in *many* cases, it's reasonable to send both higher and 
lower resolutions than the requested ones.

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


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  <body bgcolor="#ffffff" text="#000000">
    On 06/22/11 05:59, Justin Uberti wrote:
    <blockquote
      cite="mid:BANLkTik=N6qtPRUesk3mi62rV1Lvixu54w@mail.gmail.com"
      type="cite">
      <div class="gmail_quote">
        <div>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.</div>
      </div>
    </blockquote>
    There might be some things it can do, even if it can neither recode
    nor do SVC-style tricks.<br>
    For instance, if it's a distribution point (one incoming video
    stream, many outgoing copies), it can keep track of the maximal
    video size requested by any of its downstreams, and signal the
    upstream to send a resolution that is no higher than the largest
    resolution requested.<br>
    <br>
    I agree that we can't demand that the sender send exactly the
    resolution requested; in *many* cases, it's reasonable to send both
    higher and lower resolutions than the requested ones.<br>
    <br>
    <blockquote
      cite="mid:BANLkTik=N6qtPRUesk3mi62rV1Lvixu54w@mail.gmail.com"
      type="cite">
      <div class="gmail_quote">
        <div><br>
        </div>
        <blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt
          0.8ex; border-left: 1px solid rgb(204, 204, 204);
          padding-left: 1ex;">
          <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 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.<br>
          <br>
          Best regards,<br>
          <font color="#888888">Bert<br>
          </font>
          <div class="im"><br>
            <br>
            -----Original Message-----<br>
            From: <a moz-do-not-send="true"
              href="mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a>
            [mailto:<a moz-do-not-send="true"
              href="mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a>]
            On Behalf Of Cullen Jennings<br>
            Sent: 22 June 2011 00:11<br>
            To: Aron Rosenberg<br>
            Cc: <a moz-do-not-send="true" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
          </div>
          <div>
            <div class="h5">Subject: Re: [rtcweb] Changing video 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 moz-do-not-send="true"
                href="mailto: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 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.<br>
              &gt;&gt;<br>
              &gt;&gt;<br>
              &gt;&gt; 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.<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 displayed<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 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.<br>
              <br>
              <br>
              <br>
              <br>
              _______________________________________________<br>
              rtcweb mailing list<br>
              <a moz-do-not-send="true" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
              <a moz-do-not-send="true"
                href="https://www.ietf.org/mailman/listinfo/rtcweb"
                target="_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
              _______________________________________________<br>
              rtcweb mailing list<br>
              <a moz-do-not-send="true" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
              <a moz-do-not-send="true"
                href="https://www.ietf.org/mailman/listinfo/rtcweb"
                target="_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
            </div>
          </div>
        </blockquote>
      </div>
      <br>
      <pre wrap="">
<fieldset class="mimeAttachmentHeader"></fieldset>
_______________________________________________
rtcweb mailing list
<a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
    </blockquote>
    <br>
  </body>
</html>

--------------070604010309080705030302--

From harald@alvestrand.no  Mon Jul 11 07:17:38 2011
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Date: Mon, 11 Jul 2011 16:17:43 +0200
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Subject: Re: [rtcweb] Comments to use-cases/reqs document
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Just one comment-on-comment:

On 06/29/11 12:50, Stefan Hkansson LK wrote:
> All,
>
> some comments to
> <http://datatracker.ietf.org/doc/draft-ietf-rtcweb-use-cases-and-requirements/?include_text=1> 
>
>
>
> 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.

I disagree with this proposal.
While the "resizing" requirement is in "simple video communication 
service" as
"The users can change the display sizes during the session", I believe 
the basic use case does not derive the requirement that the change in 
video size be communicated to the other party.

I would propose the following changes:
- Reinstate the text below:

4.2.4.  Video Size Change

4.2.4.1.  Description

    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 regenerates the video codec paramters with Alice's
    browser to change the resolution of the video Alice sends to match
    the smaller size.

4.2.4.2.  Derived Requirements

    F22 ( It SHOULD be possible to modify video codec parameters during a
    session.)

- Change the text of F22 to be:

   It SHOULD be possible for one party to request a change in video 
codec parameters, such as size,
   during a session, and for the other party to react to this request.



From harald@alvestrand.no  Mon Jul 11 07:21:30 2011
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From: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-overview-00.txt
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Speaking with editor hat on:

Unless someone else chimes in, I am not going to act on these requests.

I regard the tying of "real-time" to "hundreds of milliseconds" as 
important - I have been in discussions where people have started arguing 
about single-millisecond delays and in discussions where people argue 
that "lag is not important as long as delivery occurs at a constant 
rate" - both are, to my mind, inappropriate and/or irrelevant for our 
use cases.

With regard to "in a given time scale between known ahead-of-time (AOT) 
rate and known just-in-time (JIT) rate", I don't understand its intention.

On 07/03/11 20:27, Dzonatas Sol wrote:
> Before another glossary action happens, let me get this in (can't wait 
> for tomorrow):
>
> >  Real-time media:  Media where generation of content and display of
> >      content are intended to occur closely together in a given time 
> scale.
>
> """
> Real-time media:  Media where generation of content and display of 
> content are intended to occur closely together in a given time scale 
> between known ahead-of-time (AOT) rate and known just-in-time (JIT) rate.
> """
>
> On 07/01/2011 08:10 AM, Dzonatas Sol wrote:
>> Hi, I read the entire overview. Only one concern, in 2.4 Terminology:
>>
>> >  Real-time media:  Media where generation of content and display of
>> >      content are intended to occur closely together in time (on the
>> >     order of no more than hundreds of milliseconds).
>>
>> Can that just read:
>>
>> >  Real-time media:  Media where generation of content and display of
>> >      content are intended to occur closely together in a given time 
>> scale.
>>
>> I've found early specification of date systems may affect assets in 
>> ways unintended (or implied). There is no simple cure, so I think 
>> this is clearer without the noted "hundreds of milliseconds", there.
>>
>> Some have tried to build "authoritative" dates from such real-time 
>> mechanics. Others pass that date thinking it is part of the asset. 
>> I've seen it happen, so I'm just thinking it would be nice to have 
>> some way to say that was not intended, yet I haven't found any.
>>
>> On 06/30/2011 10:52 PM, internet-drafts@ietf.org wrote:
>>> A New Internet-Draft is available from the on-line Internet-Drafts 
>>> directories. This draft is a work item of the Real-Time 
>>> Communication in WEB-browsers Working Group of the IETF.
>>>
>>>     Title           : Overview: Real Time Protocols for Brower-based 
>>> Applications
>>>     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
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>
>>
>>
>
>


From matthew.kaufman@skype.net  Mon Jul 11 09:59:41 2011
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On Jul 1, 2011, at 5:10 AM, Stefan H=E5kansson LK wrote:

> Reading the draft I get the impression that the intention is to =
multiplex (multiple) RTP streams with arbitrary datagrams on the same =
5-tuple. In that light the method lacks IMO:
>=20
> * There is no multiplexing support for different RTP flows - as =
discussed at the Interim meeting session separation should be maintained

This draft doesn't address that.

>=20
> * There is no support for rate/congestion control of the datagrams
>=20

Correct. This is a tool for getting datagrams sent and received. =
Congestion control, reliability, etc. are all a high-layer problem than =
this.

> * There is no support for multiplexing different datagram flows (the =
need for this can be argued though - could be done by the web app)
>=20

I would argue that this is an application problem, or possibly the =
responsibility of a layer that goes above this.

Matthew Kaufman


From dzonatas@gmail.com  Mon Jul 11 21:51:05 2011
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-overview-00.txt
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Chair,

The scale factor is known here:

http://mathworld.wolfram.com/AffineTransformation.html

Thanks.

P.S. Next year is ok for review on this, thanks again. Engineers may 
want to test few "same" types of values, or "same" rates, like 
milliseconds, and revise.

On 07/11/2011 07:21 AM, Harald Alvestrand wrote:
> Speaking with editor hat on:
>
> Unless someone else chimes in, I am not going to act on these requests.
>
> I regard the tying of "real-time" to "hundreds of milliseconds" as 
> important - I have been in discussions where people have started 
> arguing about single-millisecond delays and in discussions where 
> people argue that "lag is not important as long as delivery occurs at 
> a constant rate" - both are, to my mind, inappropriate and/or 
> irrelevant for our use cases.
>
> With regard to "in a given time scale between known ahead-of-time 
> (AOT) rate and known just-in-time (JIT) rate", I don't understand its 
> intention.
>
> On 07/03/11 20:27, Dzonatas Sol wrote:
>> Before another glossary action happens, let me get this in (can't 
>> wait for tomorrow):
>>
>> >  Real-time media:  Media where generation of content and display of
>> >      content are intended to occur closely together in a given time 
>> scale.
>>
>> """
>> Real-time media:  Media where generation of content and display of 
>> content are intended to occur closely together in a given time scale 
>> between known ahead-of-time (AOT) rate and known just-in-time (JIT) 
>> rate.
>> """
>>
>> On 07/01/2011 08:10 AM, Dzonatas Sol wrote:
>>> Hi, I read the entire overview. Only one concern, in 2.4 Terminology:
>>>
>>> >  Real-time media:  Media where generation of content and display of
>>> >      content are intended to occur closely together in time (on the
>>> >     order of no more than hundreds of milliseconds).
>>>
>>> Can that just read:
>>>
>>> >  Real-time media:  Media where generation of content and display of
>>> >      content are intended to occur closely together in a given 
>>> time scale.
>>>
>>> I've found early specification of date systems may affect assets in 
>>> ways unintended (or implied). There is no simple cure, so I think 
>>> this is clearer without the noted "hundreds of milliseconds", there.
>>>
>>> Some have tried to build "authoritative" dates from such real-time 
>>> mechanics. Others pass that date thinking it is part of the asset. 
>>> I've seen it happen, so I'm just thinking it would be nice to have 
>>> some way to say that was not intended, yet I haven't found any.
>>>
>>> On 06/30/2011 10:52 PM, internet-drafts@ietf.org wrote:
>>>> A New Internet-Draft is available from the on-line Internet-Drafts 
>>>> directories. This draft is a work item of the Real-Time 
>>>> Communication in WEB-browsers Working Group of the IETF.
>>>>
>>>>     Title           : Overview: Real Time Protocols for 
>>>> Brower-based Applications
>>>>     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
>>>> _______________________________________________
>>>> 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
>


-- 
--- http://twitter.com/Dzonatas_Sol ---
Web Development, Software Engineering
Ag-Biotech, Virtual Reality, Consultant


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Subject: [rtcweb] Media processing and codecs (Re: draft rtcweb agenda for IETF 81 - version 2)
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On 07/05/11 18:00, Cullen Jennings wrote:
>            Media processing&  codec (20 min)
>            draft-cbran-rtcweb-codec
>            draft-alvestrand-dispatch-rtcweb-protocols
>            * focus on requirements for codecs
I think the technical content of 
draft-alvestrand-dispatch-rtcweb-protocols in the area of codecs is 
adequately represented by draft-cbran-rtcweb-codec, so that (older) 
draft does not need to be listed.

There are sections of -overview that are relevant to the discussion, in 
particular the text about the reasons why having mandatory-to-implement 
codecs are good.

                     Harald


From stefan.lk.hakansson@ericsson.com  Tue Jul 12 04:58:58 2011
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From: =?iso-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>
To: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Tue, 12 Jul 2011 13:58:56 +0200
Thread-Topic: [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-require=
ments/?include_text=3D1>
>>
>>
>>
>> 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.
>
>I disagree with this proposal.
>While the "resizing" requirement is in "simple video communication
>service" as
>"The users can change the display sizes during the session", I believe
>the basic use case does not derive the requirement that the change in
>video size be communicated to the other party.
>
>I would propose the following changes:
>- Reinstate the text below:
>
>4.2.4.  Video Size Change
>
>4.2.4.1.  Description
>
>    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 regenerates the video codec paramters with Alice's
>    browser to change the resolution of the video Alice sends to match
>    the smaller size.
My preference is always to keep the number of use cases down. Could we not =
just add a sentence to the current "Simple Video Communication Service" use=
 case so it reads something like "The users can change the sizes of the vid=
eo displays during the session. Changes of the display size are communicate=
d from the displaying browser to the one sending the stream so that the vid=
eo codec parameters can be changed accordingly."

But my preference is not strong - I'm happy to reinsert the use case if peo=
ple think so.
>4.2.4.2.  Derived Requirements
>
>    F22 ( It SHOULD be possible to modify video codec parameters during a
>    session.)
F22 now reads "The browser SHOULD use encoding of streams suitable for the =
current rendering (e.g. video display size) and SHOULD change parameters if=
 the rendering changes during the session"
>
>- Change the text of F22 to be:
>
>   It SHOULD be possible for one party to request a change in video
>codec parameters, such as size,
>   during a session, and for the other party to react to this request.
I think "party" is very vague. What does it mean - the browser, the user, t=
he web app?

Stefan


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

From harald@alvestrand.no  Tue Jul 12 07:17:19 2011
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On 07/12/11 13:58, Stefan Hkansson LK wrote:
>>> All,
>>>
>>> some comments to
>>> <http://datatracker.ietf.org/doc/draft-ietf-rtcweb-use-cases-and-requirements/?include_text=1>
>>>
>>>
>>>
>>> 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.
>> I disagree with this proposal.
>> While the "resizing" requirement is in "simple video communication
>> service" as
>> "The users can change the display sizes during the session", I believe
>> the basic use case does not derive the requirement that the change in
>> video size be communicated to the other party.
>>
>> I would propose the following changes:
>> - Reinstate the text below:
>>
>> 4.2.4.  Video Size Change
>>
>> 4.2.4.1.  Description
>>
>>     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 regenerates the video codec paramters with Alice's
>>     browser to change the resolution of the video Alice sends to match
>>     the smaller size.
> My preference is always to keep the number of use cases down.
We may have a difference of style here - I prefer to have explicit use 
cases that show only a single action or attribute that needs to be 
supported, which means that the number of use cases tends to be high.

I think this is especially important for the situation where a WG has to 
decide to drop an use case because no agreed-upon implementation can be 
found.
>   Could we not just add a sentence to the current "Simple Video Communication Service" use case so it reads something like "The users can change the sizes of the video displays during the session. Changes of the display size are communicated from the displaying browser to the one sending the stream so that the video codec parameters can be changed accordingly."
As per above - I would prefer the "Simple Video Communication Service" 
to be as simple as possible. Even resizing the window locally is more 
than "as simple as possible" - the Gtalk video client for my phone 
doesn't support that.
> But my preference is not strong - I'm happy to reinsert the use case if people think so.
>> 4.2.4.2.  Derived Requirements
>>
>>     F22 ( It SHOULD be possible to modify video codec parameters during a
>>     session.)
> F22 now reads "The browser SHOULD use encoding of streams suitable for the current rendering (e.g. video display size) and SHOULD change parameters if the rendering changes during the session"
>> - Change the text of F22 to be:
>>
>>    It SHOULD be possible for one party to request a change in video
>> codec parameters, such as size,
>>    during a session, and for the other party to react to this request.
> I think "party" is very vague. What does it mean - the browser, the user, the web app?
Since this is the IETF WG - the "party" is the browser, user and web app 
together; somehow the desire for a change has to be communicated across 
the network, and that's the requirement I think needs to be in the IETF 
requirements list.

The corresponding API requirement (which I'm not sure how to formulate - 
all this may go on in the browser and need no explicit API) needs to go 
into the W3C requirements list.


From stewe@stewe.org  Tue Jul 12 07:36:05 2011
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From: Stephan Wenger <stewe@stewe.org>
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Hi Harald,
If you define "simple" as "minimum functionality", then there should also
be a use case "not quite so simple; offer at least the user experience you
get from a Skype client today".  That use case may well turn out to be
more popular than the "simple" use case according to your definition.
However, my preference would be not to assume minimalistic use cases, but
rather tune the level of mandation of the requirements.  (So I'm in
Stefan's camp -- keeping the number of use cases reasonable)
Stephan


On 7.12.2011 07:17 , "Harald Alvestrand" <harald@alvestrand.no> wrote:

>On 07/12/11 13:58, Stefan H=E5kansson LK wrote:
>>>> All,
>>>>
>>>> some comments to
>>>>=20
>>>><http://datatracker.ietf.org/doc/draft-ietf-rtcweb-use-cases-and-requir
>>>>ements/?include_text=3D1>
>>>>
>>>>
>>>>
>>>> 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.
>>> I disagree with this proposal.
>>> While the "resizing" requirement is in "simple video communication
>>> service" as
>>> "The users can change the display sizes during the session", I believe
>>> the basic use case does not derive the requirement that the change in
>>> video size be communicated to the other party.
>>>
>>> I would propose the following changes:
>>> - Reinstate the text below:
>>>
>>> 4.2.4.  Video Size Change
>>>
>>> 4.2.4.1.  Description
>>>
>>>     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 regenerates the video codec paramters with Alice's
>>>     browser to change the resolution of the video Alice sends to match
>>>     the smaller size.
>> My preference is always to keep the number of use cases down.
>We may have a difference of style here - I prefer to have explicit use
>cases that show only a single action or attribute that needs to be
>supported, which means that the number of use cases tends to be high.
>
>I think this is especially important for the situation where a WG has to
>decide to drop an use case because no agreed-upon implementation can be
>found.
>>   Could we not just add a sentence to the current "Simple Video
>>Communication Service" use case so it reads something like "The users
>>can change the sizes of the video displays during the session. Changes
>>of the display size are communicated from the displaying browser to the
>>one sending the stream so that the video codec parameters can be changed
>>accordingly."
>As per above - I would prefer the "Simple Video Communication Service"
>to be as simple as possible. Even resizing the window locally is more
>than "as simple as possible" - the Gtalk video client for my phone
>doesn't support that.
>> But my preference is not strong - I'm happy to reinsert the use case if
>>people think so.
>>> 4.2.4.2.  Derived Requirements
>>>
>>>     F22 ( It SHOULD be possible to modify video codec parameters
>>>during a
>>>     session.)
>> F22 now reads "The browser SHOULD use encoding of streams suitable for
>>the current rendering (e.g. video display size) and SHOULD change
>>parameters if the rendering changes during the session"
>>> - Change the text of F22 to be:
>>>
>>>    It SHOULD be possible for one party to request a change in video
>>> codec parameters, such as size,
>>>    during a session, and for the other party to react to this request.
>> I think "party" is very vague. What does it mean - the browser, the
>>user, the web app?
>Since this is the IETF WG - the "party" is the browser, user and web app
>together; somehow the desire for a change has to be communicated across
>the network, and that's the requirement I think needs to be in the IETF
>requirements list.
>
>The corresponding API requirement (which I'm not sure how to formulate -
>all this may go on in the browser and need no explicit API) needs to go
>into the W3C requirements list.
>
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb



From harald@alvestrand.no  Tue Jul 12 08:02:52 2011
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On 07/12/11 16:35, Stephan Wenger wrote:
> Hi Harald,
> If you define "simple" as "minimum functionality", then there should also
> be a use case "not quite so simple; offer at least the user experience you
> get from a Skype client today".  That use case may well turn out to be
> more popular than the "simple" use case according to your definition.
> However, my preference would be not to assume minimalistic use cases, but
> rather tune the level of mandation of the requirements.  (So I'm in
> Stefan's camp -- keeping the number of use cases reasonable)
I'm all for "reasonable", but my "reasonable" may be different from 
yours.... the sum of all requirements should indeed allow us to build 
equivalents to Skype or Google Talk on top of this platform (except for 
the parts that don't make sense, of course :-) ), but I find it easier 
to talk about auto-zoom to keep head size constant (for instance - at 
least one Skype beta supported that) as a separate use case, not merged 
with the "basic use case".

> Stephan
>
>
> On 7.12.2011 07:17 , "Harald Alvestrand"<harald@alvestrand.no>  wrote:
>
>> On 07/12/11 13:58, Stefan Hkansson LK wrote:
>>>>> All,
>>>>>
>>>>> some comments to
>>>>>
>>>>> <http://datatracker.ietf.org/doc/draft-ietf-rtcweb-use-cases-and-requir
>>>>> ements/?include_text=1>
>>>>>
>>>>>
>>>>>
>>>>> 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.
>>>> I disagree with this proposal.
>>>> While the "resizing" requirement is in "simple video communication
>>>> service" as
>>>> "The users can change the display sizes during the session", I believe
>>>> the basic use case does not derive the requirement that the change in
>>>> video size be communicated to the other party.
>>>>
>>>> I would propose the following changes:
>>>> - Reinstate the text below:
>>>>
>>>> 4.2.4.  Video Size Change
>>>>
>>>> 4.2.4.1.  Description
>>>>
>>>>      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 regenerates the video codec paramters with Alice's
>>>>      browser to change the resolution of the video Alice sends to match
>>>>      the smaller size.
>>> My preference is always to keep the number of use cases down.
>> We may have a difference of style here - I prefer to have explicit use
>> cases that show only a single action or attribute that needs to be
>> supported, which means that the number of use cases tends to be high.
>>
>> I think this is especially important for the situation where a WG has to
>> decide to drop an use case because no agreed-upon implementation can be
>> found.
>>>    Could we not just add a sentence to the current "Simple Video
>>> Communication Service" use case so it reads something like "The users
>>> can change the sizes of the video displays during the session. Changes
>>> of the display size are communicated from the displaying browser to the
>>> one sending the stream so that the video codec parameters can be changed
>>> accordingly."
>> As per above - I would prefer the "Simple Video Communication Service"
>> to be as simple as possible. Even resizing the window locally is more
>> than "as simple as possible" - the Gtalk video client for my phone
>> doesn't support that.
>>> But my preference is not strong - I'm happy to reinsert the use case if
>>> people think so.
>>>> 4.2.4.2.  Derived Requirements
>>>>
>>>>      F22 ( It SHOULD be possible to modify video codec parameters
>>>> during a
>>>>      session.)
>>> F22 now reads "The browser SHOULD use encoding of streams suitable for
>>> the current rendering (e.g. video display size) and SHOULD change
>>> parameters if the rendering changes during the session"
>>>> - Change the text of F22 to be:
>>>>
>>>>     It SHOULD be possible for one party to request a change in video
>>>> codec parameters, such as size,
>>>>     during a session, and for the other party to react to this request.
>>> I think "party" is very vague. What does it mean - the browser, the
>>> user, the web app?
>> Since this is the IETF WG - the "party" is the browser, user and web app
>> together; somehow the desire for a change has to be communicated across
>> the network, and that's the requirement I think needs to be in the IETF
>> requirements list.
>>
>> The corresponding API requirement (which I'm not sure how to formulate -
>> all this may go on in the browser and need no explicit API) needs to go
>> into the W3C requirements list.
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>
>


From stewe@stewe.org  Tue Jul 12 08:09:40 2011
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That's fine, but then you need to add at least one more use case:
point-to-point, and not quite simple...
Stephan


On 7.12.2011 08:02 , "Harald Alvestrand" <harald@alvestrand.no> wrote:

>On 07/12/11 16:35, Stephan Wenger wrote:
>> Hi Harald,
>> If you define "simple" as "minimum functionality", then there should
>>also
>> be a use case "not quite so simple; offer at least the user experience
>>you
>> get from a Skype client today".  That use case may well turn out to be
>> more popular than the "simple" use case according to your definition.
>> However, my preference would be not to assume minimalistic use cases,
>>but
>> rather tune the level of mandation of the requirements.  (So I'm in
>> Stefan's camp -- keeping the number of use cases reasonable)
>I'm all for "reasonable", but my "reasonable" may be different from
>yours.... the sum of all requirements should indeed allow us to build
>equivalents to Skype or Google Talk on top of this platform (except for
>the parts that don't make sense, of course :-) ), but I find it easier
>to talk about auto-zoom to keep head size constant (for instance - at
>least one Skype beta supported that) as a separate use case, not merged
>with the "basic use case".
>
>> Stephan
>>
>>
>> On 7.12.2011 07:17 , "Harald Alvestrand"<harald@alvestrand.no>  wrote:
>>
>>> On 07/12/11 13:58, Stefan H=E5kansson LK wrote:
>>>>>> All,
>>>>>>
>>>>>> some comments to
>>>>>>
>>>>>>=20
>>>>>><http://datatracker.ietf.org/doc/draft-ietf-rtcweb-use-cases-and-requ
>>>>>>ir
>>>>>> ements/?include_text=3D1>
>>>>>>
>>>>>>
>>>>>>
>>>>>> 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.
>>>>> I disagree with this proposal.
>>>>> While the "resizing" requirement is in "simple video communication
>>>>> service" as
>>>>> "The users can change the display sizes during the session", I
>>>>>believe
>>>>> the basic use case does not derive the requirement that the change in
>>>>> video size be communicated to the other party.
>>>>>
>>>>> I would propose the following changes:
>>>>> - Reinstate the text below:
>>>>>
>>>>> 4.2.4.  Video Size Change
>>>>>
>>>>> 4.2.4.1.  Description
>>>>>
>>>>>      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 regenerates the video codec paramters with Alice's
>>>>>      browser to change the resolution of the video Alice sends to
>>>>>match
>>>>>      the smaller size.
>>>> My preference is always to keep the number of use cases down.
>>> We may have a difference of style here - I prefer to have explicit use
>>> cases that show only a single action or attribute that needs to be
>>> supported, which means that the number of use cases tends to be high.
>>>
>>> I think this is especially important for the situation where a WG has
>>>to
>>> decide to drop an use case because no agreed-upon implementation can be
>>> found.
>>>>    Could we not just add a sentence to the current "Simple Video
>>>> Communication Service" use case so it reads something like "The users
>>>> can change the sizes of the video displays during the session. Changes
>>>> of the display size are communicated from the displaying browser to
>>>>the
>>>> one sending the stream so that the video codec parameters can be
>>>>changed
>>>> accordingly."
>>> As per above - I would prefer the "Simple Video Communication Service"
>>> to be as simple as possible. Even resizing the window locally is more
>>> than "as simple as possible" - the Gtalk video client for my phone
>>> doesn't support that.
>>>> But my preference is not strong - I'm happy to reinsert the use case
>>>>if
>>>> people think so.
>>>>> 4.2.4.2.  Derived Requirements
>>>>>
>>>>>      F22 ( It SHOULD be possible to modify video codec parameters
>>>>> during a
>>>>>      session.)
>>>> F22 now reads "The browser SHOULD use encoding of streams suitable for
>>>> the current rendering (e.g. video display size) and SHOULD change
>>>> parameters if the rendering changes during the session"
>>>>> - Change the text of F22 to be:
>>>>>
>>>>>     It SHOULD be possible for one party to request a change in video
>>>>> codec parameters, such as size,
>>>>>     during a session, and for the other party to react to this
>>>>>request.
>>>> I think "party" is very vague. What does it mean - the browser, the
>>>> user, the web app?
>>> Since this is the IETF WG - the "party" is the browser, user and web
>>>app
>>> together; somehow the desire for a change has to be communicated across
>>> the network, and that's the requirement I think needs to be in the IETF
>>> requirements list.
>>>
>>> The corresponding API requirement (which I'm not sure how to formulate
>>>-
>>> all this may go on in the browser and need no explicit API) needs to go
>>> into the W3C requirements list.
>>>
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>>
>



From henry.sinnreich@gmail.com  Tue Jul 12 08:56:27 2011
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+1

The use cases and especially requirements should be kept as simple/minimal
as possible but no more. Actually the most interesting use cases may not be
known at present; a good test if rtcweb is really innovative.
 =20
Quite happy that Harald and Cullen seem to keep firm control on this.

As for the video/screens parameters, one of my gripes is the use of
antiquated SDP instead of standard metadata formats for all screens, I/O
devices, etc.=20
SDP is a big can of worms to throw out.

Legacy RTP as well, but this seems to be covered for now.

Finally, just to put all my gripes in one place: Why not use HIP with
ICE/HIP instead if SIP-ICE? This is so self evident.

Thanks, Henry



On 7/12/11 10:02 AM, "Harald Alvestrand" <harald@alvestrand.no> wrote:

> On 07/12/11 16:35, Stephan Wenger wrote:
>> Hi Harald,
>> If you define "simple" as "minimum functionality", then there should als=
o
>> be a use case "not quite so simple; offer at least the user experience y=
ou
>> get from a Skype client today".  That use case may well turn out to be
>> more popular than the "simple" use case according to your definition.
>> However, my preference would be not to assume minimalistic use cases, bu=
t
>> rather tune the level of mandation of the requirements.  (So I'm in
>> Stefan's camp -- keeping the number of use cases reasonable)
> I'm all for "reasonable", but my "reasonable" may be different from
> yours.... the sum of all requirements should indeed allow us to build
> equivalents to Skype or Google Talk on top of this platform (except for
> the parts that don't make sense, of course :-) ), but I find it easier
> to talk about auto-zoom to keep head size constant (for instance - at
> least one Skype beta supported that) as a separate use case, not merged
> with the "basic use case".
>=20
>> Stephan
>>=20
>>=20
>> On 7.12.2011 07:17 , "Harald Alvestrand"<harald@alvestrand.no>  wrote:
>>=20
>>> On 07/12/11 13:58, Stefan H=E5kansson LK wrote:
>>>>>> All,
>>>>>>=20
>>>>>> some comments to
>>>>>>=20
>>>>>> <http://datatracker.ietf.org/doc/draft-ietf-rtcweb-use-cases-and-req=
uir
>>>>>> ements/?include_text=3D1>
>>>>>>=20
>>>>>>=20
>>>>>>=20
>>>>>> 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.
>>>>> I disagree with this proposal.
>>>>> While the "resizing" requirement is in "simple video communication
>>>>> service" as
>>>>> "The users can change the display sizes during the session", I believ=
e
>>>>> the basic use case does not derive the requirement that the change in
>>>>> video size be communicated to the other party.
>>>>>=20
>>>>> I would propose the following changes:
>>>>> - Reinstate the text below:
>>>>>=20
>>>>> 4.2.4.  Video Size Change
>>>>>=20
>>>>> 4.2.4.1.  Description
>>>>>=20
>>>>>      Alice and Bob are in a video call in their browsers and have
>>>>>      negotiate a high resolution video.  Bob decides to change the si=
ze
>>>>> of
>>>>>      the windows his browser is displaying video to a small size.
>>>>>=20
>>>>>      Bob's browser regenerates the video codec paramters with Alice's
>>>>>      browser to change the resolution of the video Alice sends to mat=
ch
>>>>>      the smaller size.
>>>> My preference is always to keep the number of use cases down.
>>> We may have a difference of style here - I prefer to have explicit use
>>> cases that show only a single action or attribute that needs to be
>>> supported, which means that the number of use cases tends to be high.
>>>=20
>>> I think this is especially important for the situation where a WG has t=
o
>>> decide to drop an use case because no agreed-upon implementation can be
>>> found.
>>>>    Could we not just add a sentence to the current "Simple Video
>>>> Communication Service" use case so it reads something like "The users
>>>> can change the sizes of the video displays during the session. Changes
>>>> of the display size are communicated from the displaying browser to th=
e
>>>> one sending the stream so that the video codec parameters can be chang=
ed
>>>> accordingly."
>>> As per above - I would prefer the "Simple Video Communication Service"
>>> to be as simple as possible. Even resizing the window locally is more
>>> than "as simple as possible" - the Gtalk video client for my phone
>>> doesn't support that.
>>>> But my preference is not strong - I'm happy to reinsert the use case i=
f
>>>> people think so.
>>>>> 4.2.4.2.  Derived Requirements
>>>>>=20
>>>>>      F22 ( It SHOULD be possible to modify video codec parameters
>>>>> during a
>>>>>      session.)
>>>> F22 now reads "The browser SHOULD use encoding of streams suitable for
>>>> the current rendering (e.g. video display size) and SHOULD change
>>>> parameters if the rendering changes during the session"
>>>>> - Change the text of F22 to be:
>>>>>=20
>>>>>     It SHOULD be possible for one party to request a change in video
>>>>> codec parameters, such as size,
>>>>>     during a session, and for the other party to react to this reques=
t.
>>>> I think "party" is very vague. What does it mean - the browser, the
>>>> user, the web app?
>>> Since this is the IETF WG - the "party" is the browser, user and web ap=
p
>>> together; somehow the desire for a change has to be communicated across
>>> the network, and that's the requirement I think needs to be in the IETF
>>> requirements list.
>>>=20
>>> The corresponding API requirement (which I'm not sure how to formulate =
-
>>> all this may go on in the browser and need no explicit API) needs to go
>>> into the W3C requirements list.
>>>=20
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>=20
>>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb



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Date: Tue, 12 Jul 2011 10:23:15 -0700
From: Dzonatas Sol <dzonatas@gmail.com>
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The crystal radio is "not quite simple" if we use standard memory 
technology (or hard-ware version).

Given signal-to-noise, the maximal fractal surface may seem to be the 
union of point-to-point. That is falsifiable, the millisecond is not 
unless one proves one. ASCII presents "the choice" to support that. 
Standard? "Not my business."

The event horizon for real-time appears only simulated (by us), and I 
like the religions topics on this, for science by science.

For example,

We could express: millisecond := <square>alpha+beta</square>

We could write that with MathML, yet the published MathML is not in 
plain English, enough. Reason? Maybe. How do you know if that means 
"square root of the value of alpha plus beta" or "alpha squared plus 
beta squared, and that the square root of that value". Which quote is in 
present tense? The obvious.

The constance is the power of two, a magic number in computer science, 
yet for mathematics, a real number? 1 + 1.

Elementary support for that number? Some say not their business.

Let's revisit the scale factor of affine transformations: 
http://mathworld.wolfram.com/AffineTransformation.html

How would re-write that equation to remove the magic numbers in order to 
have the pure scalar value? Replace the first two and last two with 
"alpha" and "omega", and format with XML to MathML.

Is the truth, the pure scalar value, of two relative to the affine 
transformation by Wolfram? Ballard's method is E=mcc, so not always. 
Mathematicians think relative to "Energy equals mass times speed of 
light to the power of two."

Use-case for "real-time": law of two.

On 07/12/2011 08:09 AM, Stephan Wenger wrote:
> That's fine, but then you need to add at least one more use case:
> point-to-point, and not quite simple...
> Stephan
>
>
> On 7.12.2011 08:02 , "Harald Alvestrand"<harald@alvestrand.no>  wrote:
>
>    
>> On 07/12/11 16:35, Stephan Wenger wrote:
>>      
>>> Hi Harald,
>>> If you define "simple" as "minimum functionality", then there should
>>> also
>>> be a use case "not quite so simple; offer at least the user experience
>>> you
>>> get from a Skype client today".  That use case may well turn out to be
>>> more popular than the "simple" use case according to your definition.
>>> However, my preference would be not to assume minimalistic use cases,
>>> but
>>> rather tune the level of mandation of the requirements.  (So I'm in
>>> Stefan's camp -- keeping the number of use cases reasonable)
>>>        
>> I'm all for "reasonable", but my "reasonable" may be different from
>> yours.... the sum of all requirements should indeed allow us to build
>> equivalents to Skype or Google Talk on top of this platform (except for
>> the parts that don't make sense, of course :-) ), but I find it easier
>> to talk about auto-zoom to keep head size constant (for instance - at
>> least one Skype beta supported that) as a separate use case, not merged
>> with the "basic use case".
>>
>>      
>>> Stephan
>>>
>>>
>>> On 7.12.2011 07:17 , "Harald Alvestrand"<harald@alvestrand.no>   wrote:
>>>
>>>        
>>>> On 07/12/11 13:58, Stefan H�kansson LK wrote:
>>>>          
>>>>>>> All,
>>>>>>>
>>>>>>> some comments to
>>>>>>>
>>>>>>>
>>>>>>> <http://datatracker.ietf.org/doc/draft-ietf-rtcweb-use-cases-and-requ
>>>>>>> ir
>>>>>>> ements/?include_text=1>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> 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.
>>>>>>>                
>>>>>> I disagree with this proposal.
>>>>>> While the "resizing" requirement is in "simple video communication
>>>>>> service" as
>>>>>> "The users can change the display sizes during the session", I
>>>>>> believe
>>>>>> the basic use case does not derive the requirement that the change in
>>>>>> video size be communicated to the other party.
>>>>>>
>>>>>> I would propose the following changes:
>>>>>> - Reinstate the text below:
>>>>>>
>>>>>> 4.2.4.  Video Size Change
>>>>>>
>>>>>> 4.2.4.1.  Description
>>>>>>
>>>>>>       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 regenerates the video codec paramters with Alice's
>>>>>>       browser to change the resolution of the video Alice sends to
>>>>>> match
>>>>>>       the smaller size.
>>>>>>              
>>>>> My preference is always to keep the number of use cases down.
>>>>>            
>>>> We may have a difference of style here - I prefer to have explicit use
>>>> cases that show only a single action or attribute that needs to be
>>>> supported, which means that the number of use cases tends to be high.
>>>>
>>>> I think this is especially important for the situation where a WG has
>>>> to
>>>> decide to drop an use case because no agreed-upon implementation can be
>>>> found.
>>>>          
>>>>>     Could we not just add a sentence to the current "Simple Video
>>>>> Communication Service" use case so it reads something like "The users
>>>>> can change the sizes of the video displays during the session. Changes
>>>>> of the display size are communicated from the displaying browser to
>>>>> the
>>>>> one sending the stream so that the video codec parameters can be
>>>>> changed
>>>>> accordingly."
>>>>>            
>>>> As per above - I would prefer the "Simple Video Communication Service"
>>>> to be as simple as possible. Even resizing the window locally is more
>>>> than "as simple as possible" - the Gtalk video client for my phone
>>>> doesn't support that.
>>>>          
>>>>> But my preference is not strong - I'm happy to reinsert the use case
>>>>> if
>>>>> people think so.
>>>>>            
>>>>>> 4.2.4.2.  Derived Requirements
>>>>>>
>>>>>>       F22 ( It SHOULD be possible to modify video codec parameters
>>>>>> during a
>>>>>>       session.)
>>>>>>              
>>>>> F22 now reads "The browser SHOULD use encoding of streams suitable for
>>>>> the current rendering (e.g. video display size) and SHOULD change
>>>>> parameters if the rendering changes during the session"
>>>>>            
>>>>>> - Change the text of F22 to be:
>>>>>>
>>>>>>      It SHOULD be possible for one party to request a change in video
>>>>>> codec parameters, such as size,
>>>>>>      during a session, and for the other party to react to this
>>>>>> request.
>>>>>>              
>>>>> I think "party" is very vague. What does it mean - the browser, the
>>>>> user, the web app?
>>>>>            
>>>> Since this is the IETF WG - the "party" is the browser, user and web
>>>> app
>>>> together; somehow the desire for a change has to be communicated across
>>>> the network, and that's the requirement I think needs to be in the IETF
>>>> requirements list.
>>>>
>>>> The corresponding API requirement (which I'm not sure how to formulate
>>>> -
>>>> all this may go on in the browser and need no explicit API) needs to go
>>>> into the W3C requirements list.
>>>>
>>>> _______________________________________________
>>>> 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
>
>    


-- 
--- http://twitter.com/Dzonatas_Sol ---
Web Development, Software Engineering
Ag-Biotech, Virtual Reality, Consultant


From stefan.lk.hakansson@ericsson.com  Wed Jul 13 06:14:08 2011
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>>
>> * There is no support for rate/congestion control of the datagrams
>>
>
>Correct. This is a tool for getting datagrams sent and received. Congestio=
n control, reliability, etc. are all a high-layer problem than this.
Could you elaborate a bit on how that is supposed to work? Would there be a=
nother protocol layer on top?
>> * There is no support for multiplexing different datagram flows (the nee=
d for this can be argued though - could be done by the web app)
>>
>
>I would argue that this is an application problem, or possibly the respons=
ibility of a layer that goes above this.
As indicated, I tend to agree to that this should be handled by the applica=
tion.

Stefan=

From magnus.westerlund@ericsson.com  Wed Jul 13 08:57:51 2011
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Subject: [rtcweb] Consensus Call on Non-media data service consensus and requirements
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Hi,

I have reviewed the input both the last 2 weeks and the discussion back
in April.

I see a strong support but with at least 2 people disagreeing to a basic
p2p datagram functionality. The use cases are various and some just
state that they see it as important functionality to provide to empower
the web application.

Based on this I declare a rough consensus on that we should provide a
non-media data service that is unreliable datagram oriented directly
between the peers.

One of objections against this was lack of clear requirements for what
the service. The straw men requirements I included has gotten some
discussion. Mostly support for them, but it is clear to me that we need
to further develop them. I would recommend the proponents for driving
proposals towards meeting this functionality to further discuss the
requirements taking the input so far into consideration.

When it comes to reliable data transfer between peers there has been 4-5
that wanted the functionality, 2 additional that explicitly stated they
where okay with it. No additional that has stated against it.

My interpretation is that we are close to having a rough consensus for
reliable data service, but we have so far seen no proponent willing to
suggest a solution for this. I would also note that a solution is likely
a functionality block that isn't dependent on more than the
signaling/negotiation and the NAT traversal and thus can be added a
later stage or be worked on with a completion date further into the
future than other pieces already.

So for reliable data I would recommend that someone takes on the role of
proponent and provides a draft with their perceived requirements and a
straw man proposal for how to solve these requirements so we have
something more tangible to discuss.

Cheers

Magnus

On 2011-06-27 09:36, 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
> Frgatan 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
Frgatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


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Instead of "NAT traversal", can we reduce-reduce that term to 
"synopsis". I've deleted my justification for that several times.

On 07/13/2011 08:57 AM, Magnus Westerlund wrote:
> Hi,
>
> I have reviewed the input both the last 2 weeks and the discussion back
> in April.
>
> I see a strong support but with at least 2 people disagreeing to a basic
> p2p datagram functionality. The use cases are various and some just
> state that they see it as important functionality to provide to empower
> the web application.
>
> Based on this I declare a rough consensus on that we should provide a
> non-media data service that is unreliable datagram oriented directly
> between the peers.
>
> One of objections against this was lack of clear requirements for what
> the service. The straw men requirements I included has gotten some
> discussion. Mostly support for them, but it is clear to me that we need
> to further develop them. I would recommend the proponents for driving
> proposals towards meeting this functionality to further discuss the
> requirements taking the input so far into consideration.
>
> When it comes to reliable data transfer between peers there has been 4-5
> that wanted the functionality, 2 additional that explicitly stated they
> where okay with it. No additional that has stated against it.
>
> My interpretation is that we are close to having a rough consensus for
> reliable data service, but we have so far seen no proponent willing to
> suggest a solution for this. I would also note that a solution is likely
> a functionality block that isn't dependent on more than the
> signaling/negotiation and the NAT traversal and thus can be added a
> later stage or be worked on with a completion date further into the
> future than other pieces already.
>
> So for reliable data I would recommend that someone takes on the role of
> proponent and provides a draft with their perceived requirements and a
> straw man proposal for how to solve these requirements so we have
> something more tangible to discuss.
>
> Cheers
>
> Magnus
>
> On 2011-06-27 09:36, 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
>> F�r�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
>>
>>      
>
>    


-- 
--- http://twitter.com/Dzonatas_Sol ---
Web Development, Software Engineering
Ag-Biotech, Virtual Reality, Consultant


From dzonatas@gmail.com  Wed Jul 13 11:28:55 2011
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One more thing, the constraint on that reduce-reduce is the high path.

P.S. "noise-reduction", we do not get upset over "soft" patents for 
noise reduction, please do...

On 07/13/2011 10:08 AM, Dzonatas Sol wrote:
> Instead of "NAT traversal", can we reduce-reduce that term to 
> "synopsis". I've deleted my justification for that several times.
>
> On 07/13/2011 08:57 AM, Magnus Westerlund wrote:
>> Hi,
>>
>> I have reviewed the input both the last 2 weeks and the discussion back
>> in April.
>>
>> I see a strong support but with at least 2 people disagreeing to a basic
>> p2p datagram functionality. The use cases are various and some just
>> state that they see it as important functionality to provide to empower
>> the web application.
>>
>> Based on this I declare a rough consensus on that we should provide a
>> non-media data service that is unreliable datagram oriented directly
>> between the peers.
>>
>> One of objections against this was lack of clear requirements for what
>> the service. The straw men requirements I included has gotten some
>> discussion. Mostly support for them, but it is clear to me that we need
>> to further develop them. I would recommend the proponents for driving
>> proposals towards meeting this functionality to further discuss the
>> requirements taking the input so far into consideration.
>>
>> When it comes to reliable data transfer between peers there has been 4-5
>> that wanted the functionality, 2 additional that explicitly stated they
>> where okay with it. No additional that has stated against it.
>>
>> My interpretation is that we are close to having a rough consensus for
>> reliable data service, but we have so far seen no proponent willing to
>> suggest a solution for this. I would also note that a solution is likely
>> a functionality block that isn't dependent on more than the
>> signaling/negotiation and the NAT traversal and thus can be added a
>> later stage or be worked on with a completion date further into the
>> future than other pieces already.
>>
>> So for reliable data I would recommend that someone takes on the role of
>> proponent and provides a draft with their perceived requirements and a
>> straw man proposal for how to solve these requirements so we have
>> something more tangible to discuss.
>>
>> Cheers
>>
>> Magnus
>>
>> On 2011-06-27 09:36, 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
>>> F�r�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
>>>
>>
>
>


-- 
--- http://twitter.com/Dzonatas_Sol ---
Web Development, Software Engineering
Ag-Biotech, Virtual Reality, Consultant


From dzonatas@gmail.com  Wed Jul 13 15:12:18 2011
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Updated: medium path flow called either "petrifaction" or "cloud". I 
know there are better words, oh ya, "traits".

Trinary machines... enjoy.

On 07/13/2011 11:28 AM, Dzonatas Sol wrote:
> One more thing, the constraint on that reduce-reduce is the high path.
>
> P.S. "noise-reduction", we do not get upset over "soft" patents for 
> noise reduction, please do...
>
> On 07/13/2011 10:08 AM, Dzonatas Sol wrote:
>> Instead of "NAT traversal", can we reduce-reduce that term to 
>> "synopsis". I've deleted my justification for that several times.
>>
>> On 07/13/2011 08:57 AM, Magnus Westerlund wrote:
>>> Hi,
>>>
>>> I have reviewed the input both the last 2 weeks and the discussion back
>>> in April.
>>>
>>> I see a strong support but with at least 2 people disagreeing to a 
>>> basic
>>> p2p datagram functionality. The use cases are various and some just
>>> state that they see it as important functionality to provide to empower
>>> the web application.
>>>
>>> Based on this I declare a rough consensus on that we should provide a
>>> non-media data service that is unreliable datagram oriented directly
>>> between the peers.
>>>
>>> One of objections against this was lack of clear requirements for what
>>> the service. The straw men requirements I included has gotten some
>>> discussion. Mostly support for them, but it is clear to me that we need
>>> to further develop them. I would recommend the proponents for driving
>>> proposals towards meeting this functionality to further discuss the
>>> requirements taking the input so far into consideration.
>>>
>>> When it comes to reliable data transfer between peers there has been 
>>> 4-5
>>> that wanted the functionality, 2 additional that explicitly stated they
>>> where okay with it. No additional that has stated against it.
>>>
>>> My interpretation is that we are close to having a rough consensus for
>>> reliable data service, but we have so far seen no proponent willing to
>>> suggest a solution for this. I would also note that a solution is 
>>> likely
>>> a functionality block that isn't dependent on more than the
>>> signaling/negotiation and the NAT traversal and thus can be added a
>>> later stage or be worked on with a completion date further into the
>>> future than other pieces already.
>>>
>>> So for reliable data I would recommend that someone takes on the 
>>> role of
>>> proponent and provides a draft with their perceived requirements and a
>>> straw man proposal for how to solve these requirements so we have
>>> something more tangible to discuss.
>>>
>>> Cheers
>>>
>>> Magnus
>>>
>>> On 2011-06-27 09:36, 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
>>>> F�r�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
>>>>
>>>
>>
>>
>
>


-- 
--- http://twitter.com/Dzonatas_Sol ---
Web Development, Software Engineering
Ag-Biotech, Virtual Reality, Consultant


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On 2011-07-13 19:08, Dzonatas Sol wrote:
> Instead of "NAT traversal", can we reduce-reduce that term to 
> "synopsis". I've deleted my justification for that several times.
> 

Dzonatas, your comments doesn't make sense to me. We need to use terms
that are as commonly well understood as possible.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
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From gonzalo.camarillo@ericsson.com  Thu Jul 14 01:16:42 2011
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Hi,

note that the deadline for registering to this meeting is tomorrow (see
email below).

Cheers,

Gonzalo

On 04/07/2011 1:02 PM, Gonzalo Camarillo wrote:
> Folks,
> 
> FYI. This meeting has now also been announced on the general IETF
> announce list (see email below).
> 
> Cheers,
> 
> Gonzalo
> 
> -------- Original Message --------
> Subject: W3C WebRTC working group meeting: Saturday, July 23 2011
> Date: Thu, 30 Jun 2011 20:32:29 +0200
> From: Alexa Morris <amorris@amsl.com>
> To: ietf-announce@ietf.org <ietf-announce@ietf.org>
> 
> The W3C Web Real-Time Communications working group will meet on Saturday
> 23 July 2011 afternoon, starting at 2PM, in Quebec City, Canada,
> co-located with the IETF Meeting.
> 
> The choice of the date and place is meant to favor synergies between the
> W3C WebRTC and the IETF RTCWEB groups. IETF attendees, especially those
> following the RTCWEB WG, are welcome to attend this W3C face-to-face
> meeting as meeting guests. Attendees need to register before July 15th
> using the following URL:
> 
>   http://www.w3.org/2002/09/wbs/47318/webrtc-f2f-quebec/
> 
> Note that this is a W3C meeting and, thus, it will be held under W3C
> rules.
> 
> Regards,
> Alexa
> 
> -----------
> Alexa Morris / Executive Director / IETF
> 48377 Fremont Blvd., Suite 117, Fremont, CA  94538
> Phone: +1.510.492.4089 / Fax: +1.510.492.4001
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> 


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My bad for thinking out-loud about star-topology.

On 07/14/2011 12:15 AM, Magnus Westerlund wrote:
> On 2011-07-13 19:08, Dzonatas Sol wrote:
>    
>> Instead of "NAT traversal", can we reduce-reduce that term to
>> "synopsis". I've deleted my justification for that several times.
>>
>>      
> Dzonatas, your comments doesn't make sense to me. We need to use terms
> that are as commonly well understood as possible.
>
> 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
> ----------------------------------------------------------------------
>
>
>    


-- 
--- http://twitter.com/Dzonatas_Sol ---
Web Development, Software Engineering
Ag-Biotech, Virtual Reality, Consultant


From matthew.kaufman@skype.net  Thu Jul 14 13:03:55 2011
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Does anyone on this list have a contact for the admin of the w3 list? I 
have a posting that's more relevant for that list that appears to have 
gone into a black hole.

Matthew Kaufman

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I'd like to push back on the reliable service. I've yet to hear a use =
case for it that was real time. It's very hard to do a reliable real =
time protocol and we have seen zero proposal for this. For non real time =
data, just dump it in dropbox of whatever your equivalent is and don't =
do it peer to peer. Unless someone has a real need, and wants to put =
forward a proposal, I don't see a need to focus energy on this right =
now. I'd rather work on the thing everyone agrees they do need which is =
the unreliable transfer.=20

Cullen <in my individaul contribut role>=20

On Jul 13, 2011, at 8:57 , Magnus Westerlund wrote:

> Hi,
>=20
> I have reviewed the input both the last 2 weeks and the discussion =
back
> in April.
>=20
> I see a strong support but with at least 2 people disagreeing to a =
basic
> p2p datagram functionality. The use cases are various and some just
> state that they see it as important functionality to provide to =
empower
> the web application.
>=20
> Based on this I declare a rough consensus on that we should provide a
> non-media data service that is unreliable datagram oriented directly
> between the peers.
>=20
> One of objections against this was lack of clear requirements for what
> the service. The straw men requirements I included has gotten some
> discussion. Mostly support for them, but it is clear to me that we =
need
> to further develop them. I would recommend the proponents for driving
> proposals towards meeting this functionality to further discuss the
> requirements taking the input so far into consideration.
>=20
> When it comes to reliable data transfer between peers there has been =
4-5
> that wanted the functionality, 2 additional that explicitly stated =
they
> where okay with it. No additional that has stated against it.
>=20
> My interpretation is that we are close to having a rough consensus for
> reliable data service, but we have so far seen no proponent willing to
> suggest a solution for this. I would also note that a solution is =
likely
> a functionality block that isn't dependent on more than the
> signaling/negotiation and the NAT traversal and thus can be added a
> later stage or be worked on with a completion date further into the
> future than other pieces already.
>=20
> So for reliable data I would recommend that someone takes on the role =
of
> proponent and provides a draft with their perceived requirements and a
> straw man proposal for how to solve these requirements so we have
> something more tangible to discuss.
>=20
> Cheers
>=20
> Magnus
>=20
> On 2011-06-27 09:36, Magnus Westerlund wrote:
>> 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=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
>=20
>=20
> --=20
>=20
> Magnus Westerlund
>=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


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



From tim@phonefromhere.com  Fri Jul 15 10:30:06 2011
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From: Tim Panton <tim@phonefromhere.com>
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To: Cullen Jennings <fluffy@cisco.com>
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] realiable data service
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If we don't specify a  reliable transport then all the page authors will =
end adding a reliable layer in javascript, it will be as ugly as DTMF in =
RTP !
Or worse it will be spray and pray.

Most of the use-cases I can think of need  sequenced and reliable =
datagrams. If I'm playing a game I don't want it to
drop or mis-order any one of the actions in the unholster, point, shoot  =
sequence :-)

I don't think it is very hard to implement, we could take RFC 5456's =
sequencing and retry mechanism for example.
or indeed Plan9's IL .

Tim.
=20

On 15 Jul 2011, at 16:51, Cullen Jennings wrote:

>=20
> I'd like to push back on the reliable service. I've yet to hear a use =
case for it that was real time. It's very hard to do a reliable real =
time protocol and we have seen zero proposal for this. For non real time =
data, just dump it in dropbox of whatever your equivalent is and don't =
do it peer to peer. Unless someone has a real need, and wants to put =
forward a proposal, I don't see a need to focus energy on this right =
now. I'd rather work on the thing everyone agrees they do need which is =
the unreliable transfer.=20
>=20
> Cullen <in my individaul contribut role>=20
>=20
> On Jul 13, 2011, at 8:57 , Magnus Westerlund wrote:
>=20
>> Hi,
>>=20
>> I have reviewed the input both the last 2 weeks and the discussion =
back
>> in April.
>>=20
>> I see a strong support but with at least 2 people disagreeing to a =
basic
>> p2p datagram functionality. The use cases are various and some just
>> state that they see it as important functionality to provide to =
empower
>> the web application.
>>=20
>> Based on this I declare a rough consensus on that we should provide a
>> non-media data service that is unreliable datagram oriented directly
>> between the peers.
>>=20
>> One of objections against this was lack of clear requirements for =
what
>> the service. The straw men requirements I included has gotten some
>> discussion. Mostly support for them, but it is clear to me that we =
need
>> to further develop them. I would recommend the proponents for driving
>> proposals towards meeting this functionality to further discuss the
>> requirements taking the input so far into consideration.
>>=20
>> When it comes to reliable data transfer between peers there has been =
4-5
>> that wanted the functionality, 2 additional that explicitly stated =
they
>> where okay with it. No additional that has stated against it.
>>=20
>> My interpretation is that we are close to having a rough consensus =
for
>> reliable data service, but we have so far seen no proponent willing =
to
>> suggest a solution for this. I would also note that a solution is =
likely
>> a functionality block that isn't dependent on more than the
>> signaling/negotiation and the NAT traversal and thus can be added a
>> later stage or be worked on with a completion date further into the
>> future than other pieces already.
>>=20
>> So for reliable data I would recommend that someone takes on the role =
of
>> proponent and provides a draft with their perceived requirements and =
a
>> straw man proposal for how to solve these requirements so we have
>> something more tangible to discuss.
>>=20
>> Cheers
>>=20
>> Magnus
>>=20
>> On 2011-06-27 09:36, Magnus Westerlund wrote:
>>> 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=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
>>=20
>>=20
>> --=20
>>=20
>> Magnus Westerlund
>>=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
>=20
> Cullen Jennings
> For corporate legal information go to:
> http://www.cisco.com/web/about/doing_business/legal/cri/index.html
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From emcho@sip-communicator.org  Fri Jul 15 11:16:00 2011
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From: Emil Ivov <emcho@jitsi.org>
Date: Fri, 15 Jul 2011 20:15:44 +0200
To: Cullen Jennings <fluffy@cisco.com>
Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] realiable data service
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On 15 juil. 2011, at 17:51, Cullen Jennings <fluffy@cisco.com> wrote:

>=20
> I'd like to push back on the reliable service. I've yet to hear a use case=
 for it that was real time.=20

Adding to the DTMF and gaming mentioned by Tim, there's also:

- remote desktop control
- shared whiteboards

I'd also like to mention again the file transfer case. Although it is not ex=
actly real-time, it does require a reliable datagram transport for NAT trave=
rsal, and it is part of most real-time applications.

Cheers,
Emil

--
http://jitsi.org

> It's very hard to do a reliable real time protocol and we have seen zero p=
roposal for this. For non real time data, just dump it in dropbox of whateve=
r your equivalent is and don't do it peer to peer. Unless someone has a real=
 need, and wants to put forward a proposal, I don't see a need to focus ener=
gy on this right now. I'd rather work on the thing everyone agrees they do n=
eed which is the unreliable transfer.=20
>=20
> Cullen <in my individaul contribut role>=20
>=20
> On Jul 13, 2011, at 8:57 , Magnus Westerlund wrote:
>=20
>> Hi,
>>=20
>> I have reviewed the input both the last 2 weeks and the discussion back
>> in April.
>>=20
>> I see a strong support but with at least 2 people disagreeing to a basic
>> p2p datagram functionality. The use cases are various and some just
>> state that they see it as important functionality to provide to empower
>> the web application.
>>=20
>> Based on this I declare a rough consensus on that we should provide a
>> non-media data service that is unreliable datagram oriented directly
>> between the peers.
>>=20
>> One of objections against this was lack of clear requirements for what
>> the service. The straw men requirements I included has gotten some
>> discussion. Mostly support for them, but it is clear to me that we need
>> to further develop them. I would recommend the proponents for driving
>> proposals towards meeting this functionality to further discuss the
>> requirements taking the input so far into consideration.
>>=20
>> When it comes to reliable data transfer between peers there has been 4-5
>> that wanted the functionality, 2 additional that explicitly stated they
>> where okay with it. No additional that has stated against it.
>>=20
>> My interpretation is that we are close to having a rough consensus for
>> reliable data service, but we have so far seen no proponent willing to
>> suggest a solution for this. I would also note that a solution is likely
>> a functionality block that isn't dependent on more than the
>> signaling/negotiation and the NAT traversal and thus can be added a
>> later stage or be worked on with a completion date further into the
>> future than other pieces already.
>>=20
>> So for reliable data I would recommend that someone takes on the role of
>> proponent and provides a draft with their perceived requirements and a
>> straw man proposal for how to solve these requirements so we have
>> something more tangible to discuss.
>>=20
>> Cheers
>>=20
>> Magnus
>>=20
>> On 2011-06-27 09:36, Magnus Westerlund wrote:
>>> 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
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>=20
>>=20
>>=20
>> --=20
>>=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
> Cullen Jennings
> For corporate legal information go to:
> http://www.cisco.com/web/about/doing_business/legal/cri/index.html
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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I'm sure I mentioned one case on another WG that oftens gets dropped. 
The simple idea is to deliver content ahead of time by subscription to 
an event. The content is then deemed reliably deployed at by the 
schedule event. Last minute subscriptions are turned over to unreliable. 
The subscription can be as simple as SMTP, so that simplicity alone is 
why people drop and not look at it seriously enough.

If you have over 10,000 people for an event with massive amounts of 
pre-compiled content, then it makes sense to look at it more seriously. 
The complexity is only what people perceive as static content.

I hate to be the nag about such, yet protein topologies and 
crystallization techniques already have vocab that describe such 
ahead-of-time content delivery. The vocab is not accessible to 
everybody, so this is often overlooked as something else.

I don't see why we need to force pre-compiled content over real-time 
services, especially unreliable, and expect something reliable. People 
still do that, and it doesn't seem to be going away.

...as close as we can get to such use-case: start with the solution. =)

On 07/15/2011 08:51 AM, Cullen Jennings wrote:
> I'd like to push back on the reliable service. I've yet to hear a use case for it that was real time. It's very hard to do a reliable real time protocol and we have seen zero proposal for this. For non real time data, just dump it in dropbox of whatever your equivalent is and don't do it peer to peer. Unless someone has a real need, and wants to put forward a proposal, I don't see a need to focus energy on this right now. I'd rather work on the thing everyone agrees they do need which is the unreliable transfer.
>
> Cullen<in my individaul contribut role>
>
> On Jul 13, 2011, at 8:57 , Magnus Westerlund wrote:
>
>    
>> Hi,
>>
>> I have reviewed the input both the last 2 weeks and the discussion back
>> in April.
>>
>> I see a strong support but with at least 2 people disagreeing to a basic
>> p2p datagram functionality. The use cases are various and some just
>> state that they see it as important functionality to provide to empower
>> the web application.
>>
>> Based on this I declare a rough consensus on that we should provide a
>> non-media data service that is unreliable datagram oriented directly
>> between the peers.
>>
>> One of objections against this was lack of clear requirements for what
>> the service. The straw men requirements I included has gotten some
>> discussion. Mostly support for them, but it is clear to me that we need
>> to further develop them. I would recommend the proponents for driving
>> proposals towards meeting this functionality to further discuss the
>> requirements taking the input so far into consideration.
>>
>> When it comes to reliable data transfer between peers there has been 4-5
>> that wanted the functionality, 2 additional that explicitly stated they
>> where okay with it. No additional that has stated against it.
>>
>> My interpretation is that we are close to having a rough consensus for
>> reliable data service, but we have so far seen no proponent willing to
>> suggest a solution for this. I would also note that a solution is likely
>> a functionality block that isn't dependent on more than the
>> signaling/negotiation and the NAT traversal and thus can be added a
>> later stage or be worked on with a completion date further into the
>> future than other pieces already.
>>
>> So for reliable data I would recommend that someone takes on the role of
>> proponent and provides a draft with their perceived requirements and a
>> straw man proposal for how to solve these requirements so we have
>> something more tangible to discuss.
>>
>> Cheers
>>
>> Magnus
>>
>> On 2011-06-27 09:36, 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
>>> F�r�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
>>>
>>>        
>>
>> -- 
>>
>> Magnus Westerlund
>>
>> ----------------------------------------------------------------------
>> Multimedia Technologies, Ericsson Research EAB/TVM
>> ----------------------------------------------------------------------
>> Ericsson AB                | Phone  +46 10 7148287
>> F�r�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
>>      
>
> Cullen Jennings
> For corporate legal information go to:
> http://www.cisco.com/web/about/doing_business/legal/cri/index.html
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>    


-- 
--- http://twitter.com/Dzonatas_Sol ---
Web Development, Software Engineering
Ag-Biotech, Virtual Reality, Consultant


From dzonatas@gmail.com  Sat Jul 16 10:41:03 2011
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Date: Sat, 16 Jul 2011 10:41:00 -0700
From: Dzonatas Sol <dzonatas@gmail.com>
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] realiable data service
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Another (independent) perspective is http://www.relayforlife.org 
<http://www.relayforlife.org/> on Second Life for current capability 
(and... accessibility). The common tether is the (already noted) Olympics.

On 07/16/2011 10:26 AM, Dzonatas Sol wrote:
> I'm sure I mentioned one case on another WG that oftens gets dropped. 
> The simple idea is to deliver content ahead of time by subscription to 
> an event. The content is then deemed reliably deployed at by the 
> schedule event. Last minute subscriptions are turned over to 
> unreliable. The subscription can be as simple as SMTP, so that 
> simplicity alone is why people drop and not look at it seriously enough.
>
> If you have over 10,000 people for an event with massive amounts of 
> pre-compiled content, then it makes sense to look at it more 
> seriously. The complexity is only what people perceive as static content.
>
> I hate to be the nag about such, yet protein topologies and 
> crystallization techniques already have vocab that describe such 
> ahead-of-time content delivery. The vocab is not accessible to 
> everybody, so this is often overlooked as something else.
>
> I don't see why we need to force pre-compiled content over real-time 
> services, especially unreliable, and expect something reliable. People 
> still do that, and it doesn't seem to be going away.
>
> ...as close as we can get to such use-case: start with the solution. =)
>
> On 07/15/2011 08:51 AM, Cullen Jennings wrote:
>> I'd like to push back on the reliable service. I've yet to hear a use 
>> case for it that was real time. It's very hard to do a reliable real 
>> time protocol and we have seen zero proposal for this. For non real 
>> time data, just dump it in dropbox of whatever your equivalent is and 
>> don't do it peer to peer. Unless someone has a real need, and wants 
>> to put forward a proposal, I don't see a need to focus energy on this 
>> right now. I'd rather work on the thing everyone agrees they do need 
>> which is the unreliable transfer.
>>
>> Cullen<in my individaul contribut role>
>>
>> On Jul 13, 2011, at 8:57 , Magnus Westerlund wrote:
>>
>>> Hi,
>>>
>>> I have reviewed the input both the last 2 weeks and the discussion back
>>> in April.
>>>
>>> I see a strong support but with at least 2 people disagreeing to a 
>>> basic
>>> p2p datagram functionality. The use cases are various and some just
>>> state that they see it as important functionality to provide to empower
>>> the web application.
>>>
>>> Based on this I declare a rough consensus on that we should provide a
>>> non-media data service that is unreliable datagram oriented directly
>>> between the peers.
>>>
>>> One of objections against this was lack of clear requirements for what
>>> the service. The straw men requirements I included has gotten some
>>> discussion. Mostly support for them, but it is clear to me that we need
>>> to further develop them. I would recommend the proponents for driving
>>> proposals towards meeting this functionality to further discuss the
>>> requirements taking the input so far into consideration.
>>>
>>> When it comes to reliable data transfer between peers there has been 
>>> 4-5
>>> that wanted the functionality, 2 additional that explicitly stated they
>>> where okay with it. No additional that has stated against it.
>>>
>>> My interpretation is that we are close to having a rough consensus for
>>> reliable data service, but we have so far seen no proponent willing to
>>> suggest a solution for this. I would also note that a solution is 
>>> likely
>>> a functionality block that isn't dependent on more than the
>>> signaling/negotiation and the NAT traversal and thus can be added a
>>> later stage or be worked on with a completion date further into the
>>> future than other pieces already.
>>>
>>> So for reliable data I would recommend that someone takes on the 
>>> role of
>>> proponent and provides a draft with their perceived requirements and a
>>> straw man proposal for how to solve these requirements so we have
>>> something more tangible to discuss.
>>>
>>> Cheers
>>>
>>> Magnus
>>>
>>> On 2011-06-27 09:36, 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
>>>> F�r�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
>>>>
>>>
>>> -- 
>>>
>>> Magnus Westerlund
>>>
>>> ----------------------------------------------------------------------
>>> Multimedia Technologies, Ericsson Research EAB/TVM
>>> ----------------------------------------------------------------------
>>> Ericsson AB                | Phone  +46 10 7148287
>>> F�r�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
>>
>> Cullen Jennings
>> For corporate legal information go to:
>> http://www.cisco.com/web/about/doing_business/legal/cri/index.html
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>
>


-- 
--- http://twitter.com/Dzonatas_Sol ---
Web Development, Software Engineering
Ag-Biotech, Virtual Reality, Consultant


From randell1@jesup.org  Sat Jul 16 11:38:17 2011
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On 7/15/2011 1:29 PM, Tim Panton wrote:
> If we don't specify a  reliable transport then all the page authors will end adding a reliable layer in javascript, it will be as ugly as DTMF in RTP !
> Or worse it will be spray and pray.
>
> Most of the use-cases I can think of need  sequenced and reliable datagrams. If I'm playing a game I don't want it to
> drop or mis-order any one of the actions in the unholster, point, shoot  sequence :-)

I must agree that if we don't provide a reliable transport in 
conjunction with the
other streams, you'll get a hodgepodge of incompatible, semi-functional
reliable transports layered on top of unreliable streams.  Not that one 
can't do a good
reliable transport on top of an unreliable layer, but people have 
varying abilities
to get it right.    Also, an integrated solution might have options for 
better
performance than a user-layered solution would (for example, prioritizing
certain packets from a reliable stream like restransmits or acks).

As for the need - one of the first real-time apps on the internet was 
netrek.  This
predates the  web by years (I was in a netrek league in 1991), in the 
days when
a big international company would have a 64Kbps connection.  The 
original TCP-based
protocol broke down badly when exposed to congestion and the real net.  
The solution
was to break the data streams into two: one unreliable one for most 
positional updates,
and a reliable stream for game-state-critical data (like "you're dead").

Saying "if you want reliable data, use a proxy" is also silly IMHO.

Perhaps some here from or formerly from Adobe can speak to the utility 
of reliable streams
in RTFMP.

In VoIP there's a consistent issue with the need to do some things (like 
DTMF) reliably when
your primary streams are unreliable.  Witness DTMF and the mess over RFC 
2833 vs SIP INFO.

> I don't think it is very hard to implement, we could take RFC 5456's sequencing and retry mechanism for example.
> or indeed Plan9's IL .

-- 
Randell Jesup
randell-ietf@jesup.org


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On 07/16/2011 11:37 AM, Randell Jesup wrote:
> Not that one can't do a good
> reliable transport on top of an unreliable layer, but people have 
> varying abilities
> to get it right.

If we assume nobody has the ability to "get it right" then; unreliable 
is the only solution.


> In VoIP there's a consistent issue with the need to do some things 
> (like DTMF) reliably when
> your primary streams are unreliable.  Witness DTMF and the mess over 
> RFC 2833 vs SIP INFO.

DTMF over JIT is one reason why some rather not allow others to get that 
right. Those who expect perfect flow rather just drop all other reliable 
use-cases. There is a famous case for this I avoid detail here.

For some reason we can't just say order reliable before unreliable 
because it is a waste over the given sequence (where that is already 
known). Here, the difference engine is easier to talk about.

If people really want anything like DTMF over JIT, then they may want to 
get into more formal methods of traffic-shapers, especially the schema 
to describe such for vocab.

Shape is reliable, also known as envelopes.

-- 
--- http://twitter.com/Dzonatas_Sol ---
Web Development, Software Engineering
Ag-Biotech, Virtual Reality, Consultant


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On 07/16/2011 11:37 AM, Randell Jesup wrote:
>
> As for the need - one of the first real-time apps on the internet was 
> netrek.  This
> predates the  web by years (I was in a netrek league in 1991), in the 
> days when
> a big international company would have a 64Kbps connection. 

This one is hard to beat in regards to "firsts": http://bfi.org/

I'm not gonna argue on this list when ether- existed in history; 
however, we can grow this now, and we moved on to polymers. If someone 
has a string of lazy monomers that suddenly produce one real-time event 
together, is that "on-the-wire"? Better question is how we can get this 
terminology kept in school where others want to exploit it as a chain 
reaction (it's not). Quantum mechanics is still seems anti-social to 
some. We have patience, and setup simulations.

Some are afraid to approach this technology because they think they'll 
infringe on our business. Our business is more about the continual flow; 
rest assured, so some of these patents do not expire, like "instant" 
clean water.


-- 
--- http://twitter.com/Dzonatas_Sol ---
Web Development, Software Engineering
Ag-Biotech, Virtual Reality, Consultant


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On Fri, Jul 15, 2011 at 8:51 AM, Cullen Jennings <fluffy@cisco.com> wrote:

>
> I'd like to push back on the reliable service. I've yet to hear a use case
> for it that was real time. It's very hard to do a reliable real time
> protocol and we have seen zero proposal for this. For non real time data,
> just dump it in dropbox of whatever your equivalent is and don't do it peer
> to peer.


Can I see your dropbox incantation for remote camera movement?  I am
particularly interested in how you make it useful without a reliable, near
real-time message that says "go check dropbox for $foo"....

To put this another way, a reliable protocol associated with a real time
stream has some pretty obvious uses (gaming has also already been
mentioned).  Whether it is truly "real time" or not is potentially subject
to debate, but the results of the semantic nit-picking won't tell us much.
The reliable protocol is a useful part of the real time system, even if its
retransmission and congestion control methods make it less "real time" than
rtp.

Just my two cents, also as an individual contributor,

Ted

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On Fri, Jul 15, 2011 at 8:51 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>
I&#39;d like to push back on the reliable service. I&#39;ve yet to hear a u=
se case for it that was real time. It&#39;s very hard to do a reliable real=
 time protocol and we have seen zero proposal for this. For non real time d=
ata, just dump it in dropbox of whatever your equivalent is and don&#39;t d=
o it peer to peer. </blockquote>
<div><br>Can I see your dropbox incantation for remote camera movement?=A0 =
I am particularly interested in how you make it useful without a reliable, =
near real-time message that says &quot;go check dropbox for $foo&quot;....<=
br>
<br>To put this another way, a reliable protocol associated with a real tim=
e stream has some pretty obvious uses (gaming has also already been mention=
ed).=A0 Whether it is truly &quot;real time&quot; or not is potentially sub=
ject to debate, but the results of the semantic nit-picking won&#39;t tell =
us much.=A0 The reliable protocol is a useful part of the real time system,=
 even if its retransmission and congestion control methods make it less &qu=
ot;real time&quot; than rtp.<br>
<br>Just my two cents, also as an individual contributor,<br><br>Ted<br><br=
>=A0<br></div></div>

--20cf303f6a2851b42804a84bc5d4--

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Hi Cullen,

I agree with your push back.

Focus is very important. Audio and video already present a huge challenge.
(signalling, network, web developer friendly API, security, browser
integration...)

Also, I think that there is a real risk in introducing "duplicate"
functionality in the browser as it may confuse web developers. There is a
lot going on in HTML5...

In my mind, this is "version 2.0" stuff.

/Serge


On Fri, Jul 15, 2011 at 17:51, Cullen Jennings <fluffy@cisco.com> wrote:

>
> I'd like to push back on the reliable service. I've yet to hear a use cas=
e
> for it that was real time. It's very hard to do a reliable real time
> protocol and we have seen zero proposal for this. For non real time data,
> just dump it in dropbox of whatever your equivalent is and don't do it pe=
er
> to peer. Unless someone has a real need, and wants to put forward a
> proposal, I don't see a need to focus energy on this right now. I'd rathe=
r
> work on the thing everyone agrees they do need which is the unreliable
> transfer.
>
> Cullen <in my individaul contribut role>
>
> On Jul 13, 2011, at 8:57 , Magnus Westerlund wrote:
>
> > Hi,
> >
> > I have reviewed the input both the last 2 weeks and the discussion back
> > in April.
> >
> > I see a strong support but with at least 2 people disagreeing to a basi=
c
> > p2p datagram functionality. The use cases are various and some just
> > state that they see it as important functionality to provide to empower
> > the web application.
> >
> > Based on this I declare a rough consensus on that we should provide a
> > non-media data service that is unreliable datagram oriented directly
> > between the peers.
> >
> > One of objections against this was lack of clear requirements for what
> > the service. The straw men requirements I included has gotten some
> > discussion. Mostly support for them, but it is clear to me that we need
> > to further develop them. I would recommend the proponents for driving
> > proposals towards meeting this functionality to further discuss the
> > requirements taking the input so far into consideration.
> >
> > When it comes to reliable data transfer between peers there has been 4-=
5
> > that wanted the functionality, 2 additional that explicitly stated they
> > where okay with it. No additional that has stated against it.
> >
> > My interpretation is that we are close to having a rough consensus for
> > reliable data service, but we have so far seen no proponent willing to
> > suggest a solution for this. I would also note that a solution is likel=
y
> > a functionality block that isn't dependent on more than the
> > signaling/negotiation and the NAT traversal and thus can be added a
> > later stage or be worked on with a completion date further into the
> > future than other pieces already.
> >
> > So for reliable data I would recommend that someone takes on the role o=
f
> > proponent and provides a draft with their perceived requirements and a
> > straw man proposal for how to solve these requirements so we have
> > something more tangible to discuss.
> >
> > Cheers
> >
> > Magnus
> >
> > On 2011-06-27 09:36, 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 abou=
t
> >> 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 wil=
l
> >> build JS libraries for this on top of a unreliable datagram service. I=
f
> >> 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 requirement=
s
> >> 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=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
> >>
> >
> >
> > --
> >
> > Magnus Westerlund
> >
> > ----------------------------------------------------------------------
> > 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
>
>
> Cullen Jennings
> For corporate legal information go to:
> http://www.cisco.com/web/about/doing_business/legal/cri/index.html
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


Google Sweden AB | Kungsbron 2, SE-111 22 Stockholm | Org. nr. 556656-6880
Apparently, this footer is required in Europe. Apologies. This email may be
confidential or privileged.  If you received this communication by mistake,
please don't forward it to anyone else,please erase all copies and
attachments, and please let me know that it went to the wrong person.
 Thanks.

--000e0cd484bcd3775d04a852c796
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Hi Cullen,<div><br></div><div>I agree with your push back.</div><div><br></=
div><div>Focus is very important. Audio and video already present a huge ch=
allenge. (signalling, network, web developer friendly API, security, browse=
r integration...)</div>

<div><br></div><div>Also, I think that there is a real risk in introducing =
&quot;duplicate&quot; functionality in the browser as it may confuse web de=
velopers. There is a lot going on in HTML5...</div><div><br></div><div>

In my mind, this is &quot;version 2.0&quot; stuff.</div><div><br></div><div=
>/Serge</div><div><br></div><div><br><div class=3D"gmail_quote">On Fri, Jul=
 15, 2011 at 17:51, Cullen Jennings <span dir=3D"ltr">&lt;<a href=3D"mailto=
:fluffy@cisco.com">fluffy@cisco.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;"><br>
I&#39;d like to push back on the reliable service. I&#39;ve yet to hear a u=
se case for it that was real time. It&#39;s very hard to do a reliable real=
 time protocol and we have seen zero proposal for this. For non real time d=
ata, just dump it in dropbox of whatever your equivalent is and don&#39;t d=
o it peer to peer. Unless someone has a real need, and wants to put forward=
 a proposal, I don&#39;t see a need to focus energy on this right now. I&#3=
9;d rather work on the thing everyone agrees they do need which is the unre=
liable transfer.<br>


<br>
Cullen &lt;in my individaul contribut role&gt;<br>
<br>
On Jul 13, 2011, at 8:57 , Magnus Westerlund wrote:<br>
<br>
&gt; Hi,<br>
&gt;<br>
&gt; I have reviewed the input both the last 2 weeks and the discussion bac=
k<br>
&gt; in April.<br>
&gt;<br>
&gt; I see a strong support but with at least 2 people disagreeing to a bas=
ic<br>
&gt; p2p datagram functionality. The use cases are various and some just<br=
>
&gt; state that they see it as important functionality to provide to empowe=
r<br>
&gt; the web application.<br>
&gt;<br>
&gt; Based on this I declare a rough consensus on that we should provide a<=
br>
&gt; non-media data service that is unreliable datagram oriented directly<b=
r>
&gt; between the peers.<br>
&gt;<br>
&gt; One of objections against this was lack of clear requirements for what=
<br>
&gt; the service. The straw men requirements I included has gotten some<br>
&gt; discussion. Mostly support for them, but it is clear to me that we nee=
d<br>
&gt; to further develop them. I would recommend the proponents for driving<=
br>
&gt; proposals towards meeting this functionality to further discuss the<br=
>
&gt; requirements taking the input so far into consideration.<br>
&gt;<br>
&gt; When it comes to reliable data transfer between peers there has been 4=
-5<br>
&gt; that wanted the functionality, 2 additional that explicitly stated the=
y<br>
&gt; where okay with it. No additional that has stated against it.<br>
&gt;<br>
&gt; My interpretation is that we are close to having a rough consensus for=
<br>
&gt; reliable data service, but we have so far seen no proponent willing to=
<br>
&gt; suggest a solution for this. I would also note that a solution is like=
ly<br>
&gt; a functionality block that isn&#39;t dependent on more than the<br>
&gt; signaling/negotiation and the NAT traversal and thus can be added a<br=
>
&gt; later stage or be worked on with a completion date further into the<br=
>
&gt; future than other pieces already.<br>
&gt;<br>
&gt; So for reliable data I would recommend that someone takes on the role =
of<br>
&gt; proponent and provides a draft with their perceived requirements and a=
<br>
&gt; straw man proposal for how to solve these requirements so we have<br>
&gt; something more tangible to discuss.<br>
&gt;<br>
&gt; Cheers<br>
&gt;<br>
&gt; Magnus<br>
&gt;<br>
&gt; On 2011-06-27 09:36, Magnus Westerlund wrote:<br>
&gt;&gt; WG,<br>
&gt;&gt;<br>
&gt;&gt; At the interim it was planned to have a bit discussion on the data=
gram<br>
&gt;&gt; service for RTCWEB. The first question to try to resolve if there<=
br>
&gt;&gt; is consensus for including some form of non real-time media (i.e. =
not<br>
&gt;&gt; audio, video) service between peers. This is a bit tangled with th=
e<br>
&gt;&gt; actual requirements and use cases. But there was views both for it=
 and<br>
&gt;&gt; against it on the mailing list. So lets continue and try to come t=
o a<br>
&gt;&gt; conclusion on this discussion.<br>
&gt;&gt;<br>
&gt;&gt; The use cases mentioned on the mailing list are:<br>
&gt;&gt;<br>
&gt;&gt; - Dynamic meta data for Conference and other real-time services<br=
>
&gt;&gt;<br>
&gt;&gt; - Gaming data with low latency requirements<br>
&gt;&gt;<br>
&gt;&gt; Does anyone like to add additional use cases?<br>
&gt;&gt;<br>
&gt;&gt; Based on my personal understanding this points to primarily have t=
he<br>
&gt;&gt; RTCWEB provide a unreliable datagram service. This clearly needs<b=
r>
&gt;&gt; additional requirements to be secure and safe to deploy, but more =
about<br>
&gt;&gt; this below. I still like to ask the WG here a question.<br>
&gt;&gt;<br>
&gt;&gt; Are you supporting the inclusion of a unreliable datagram service<=
br>
&gt;&gt; directly between peers? Please provide your view and any additiona=
l<br>
&gt;&gt; statements of motivation that you desire to provide.<br>
&gt;&gt;<br>
&gt;&gt; Secondly, there is a question if there needs to have something tha=
t<br>
&gt;&gt; provides reliable message (of arbitrary size) or byte stream orien=
ted<br>
&gt;&gt; data transport between the peers. I personally foresee that people=
 will<br>
&gt;&gt; build JS libraries for this on top of a unreliable datagram servic=
e. If<br>
&gt;&gt; you desire reliable data service as part of the standardized solut=
ion<br>
&gt;&gt; please provide motivation and use case and requirements.<br>
&gt;&gt;<br>
&gt;&gt; I also want to take a stab on what I personally see as the require=
ments<br>
&gt;&gt; that exist on unreliable datagram service in the context of RTCWEB=
.<br>
&gt;&gt;<br>
&gt;&gt; - Unreliable data transmission<br>
&gt;&gt; - Datagram oriented<br>
&gt;&gt; =A0 * Size limited by MTU<br>
&gt;&gt; =A0 =A0 - Path MTU discovery needed<br>
&gt;&gt; =A0 * Fragmentation by the application<br>
&gt;&gt; - Low latency, i.e. Peer to Peer preferable<br>
&gt;&gt; - Congestion Controlled, to be<br>
&gt;&gt; =A0 * Network friendly<br>
&gt;&gt; =A0 * Not become a Denial of Service tool<br>
&gt;&gt; - Security<br>
&gt;&gt; =A0* Confidentiality<br>
&gt;&gt; =A0* Integrity Protected<br>
&gt;&gt; =A0* Source Authenticated (at least bound to the signalling peer)<=
br>
&gt;&gt; =A0* Ensure consent to receive data<br>
&gt;&gt;<br>
&gt;&gt; Please debate the above. This is an attempt to ensure that we can<=
br>
&gt;&gt; establish WG consensus on both data service and any requirements.<=
br>
&gt;&gt;<br>
&gt;&gt; cheers<br>
&gt;&gt;<br>
&gt;&gt; Magnus Westerlund<br>
&gt;&gt;<br>
&gt;&gt; ------------------------------------------------------------------=
----<br>
&gt;&gt; Multimedia Technologies, Ericsson Research EAB/TVM<br>
&gt;&gt; ------------------------------------------------------------------=
----<br>
&gt;&gt; Ericsson AB =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Phone =A0<a href=3D"t=
el:%2B46%2010%207148287" value=3D"+46107148287">+46 10 7148287</a><br>
&gt;&gt; F=E4r=F6gatan 6 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Mobile <a href=3D=
"tel:%2B46%2073%200949079" value=3D"+46730949079">+46 73 0949079</a><br>
&gt;&gt; SE-164 80 Stockholm, Sweden| mailto: <a href=3D"mailto:magnus.west=
erlund@ericsson.com">magnus.westerlund@ericsson.com</a><br>
&gt;&gt; ------------------------------------------------------------------=
----<br>
&gt;&gt;<br>
&gt;&gt; _______________________________________________<br>
&gt;&gt; rtcweb mailing list<br>
&gt;&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt;&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D=
"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
&gt;&gt;<br>
&gt;<br>
&gt;<br>
&gt; --<br>
&gt;<br>
&gt; Magnus Westerlund<br>
&gt;<br>
&gt; ----------------------------------------------------------------------=
<br>
&gt; Multimedia Technologies, Ericsson Research EAB/TVM<br>
&gt; ----------------------------------------------------------------------=
<br>
&gt; Ericsson AB =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Phone =A0<a href=3D"tel:%=
2B46%2010%207148287" value=3D"+46107148287">+46 10 7148287</a><br>
&gt; F=E4r=F6gatan 6 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Mobile <a href=3D"tel=
:%2B46%2073%200949079" value=3D"+46730949079">+46 73 0949079</a><br>
&gt; SE-164 80 Stockholm, Sweden| mailto: <a href=3D"mailto:magnus.westerlu=
nd@ericsson.com">magnus.westerlund@ericsson.com</a><br>
&gt; ----------------------------------------------------------------------=
<br>
&gt;<br>
&gt; _______________________________________________<br>
&gt; rtcweb mailing list<br>
&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br>
<br>
Cullen Jennings<br>
For corporate legal information go to:<br>
<a href=3D"http://www.cisco.com/web/about/doing_business/legal/cri/index.ht=
ml" target=3D"_blank">http://www.cisco.com/web/about/doing_business/legal/c=
ri/index.html</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></blockquote><div><br></div=
><font class=3D"Apple-style-span" color=3D"#999999" size=3D"1">Google Swede=
n AB | Kungsbron 2, SE-111 22 Stockholm | Org. nr. 556656-6880=A0</font><br=
>

<div><span class=3D"Apple-style-span" style=3D"color: rgb(153, 153, 153); f=
ont-size: x-small; ">Apparently, this footer is required in Europe. Apologi=
es. This email may be confidential or privileged. =A0If you received this c=
ommunication by mistake, please don&#39;t forward it to anyone else,please =
erase all copies and attachments, and please let me know that it went to th=
e wrong person. =A0Thanks.</span>=A0</div>

</div>
</div>

--000e0cd484bcd3775d04a852c796--

From silviapfeiffer1@gmail.com  Mon Jul 18 01:17:14 2011
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From: Silvia Pfeiffer <silviapfeiffer1@gmail.com>
Date: Mon, 18 Jul 2011 18:16:52 +1000
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To: Serge Lachapelle <sergel@google.com>
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Subject: Re: [rtcweb] realiable data service
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FWIW I agree that we should focus on audio and video communication and
not introduce extra complexity by trying to solve real-time data
communication at the same time.

If it falls out as a side effect that what we are developing satisfies
other use cases beyond audio and video, too, then that's fine. But we
should not develop for it. In my experience, real-time data typically
has different requirements than audio and video, such as a much
smaller data size, such as a much higher requirement on reliability of
data delivery (at least video delivery can be slightly lossy without
issues), such as a much smaller requirement on isosynchronous data
delivery, or a lesser requirement on low jitter.

There are already other means of delivering data between browsers such
as Web sockets and XMPP, so let's not confuse the landscape more by
trying to be all things to all people.

Regards,
Silvia.


On Mon, Jul 18, 2011 at 5:16 PM, Serge Lachapelle <sergel@google.com> wrote=
:
> Hi Cullen,
> I agree with your push back.
> Focus is very important. Audio and video already present a huge challenge=
.
> (signalling, network, web developer friendly API, security, browser
> integration...)
> Also, I think that there is a real risk in introducing "duplicate"
> functionality in the browser as it may confuse web developers. There is a
> lot going on in HTML5...
> In my mind, this is "version 2.0" stuff.
> /Serge
>
> On Fri, Jul 15, 2011 at 17:51, Cullen Jennings <fluffy@cisco.com> wrote:
>>
>> I'd like to push back on the reliable service. I've yet to hear a use ca=
se
>> for it that was real time. It's very hard to do a reliable real time
>> protocol and we have seen zero proposal for this. For non real time data=
,
>> just dump it in dropbox of whatever your equivalent is and don't do it p=
eer
>> to peer. Unless someone has a real need, and wants to put forward a
>> proposal, I don't see a need to focus energy on this right now. I'd rath=
er
>> work on the thing everyone agrees they do need which is the unreliable
>> transfer.
>>
>> Cullen <in my individaul contribut role>
>>
>> On Jul 13, 2011, at 8:57 , Magnus Westerlund wrote:
>>
>> > Hi,
>> >
>> > I have reviewed the input both the last 2 weeks and the discussion bac=
k
>> > in April.
>> >
>> > I see a strong support but with at least 2 people disagreeing to a bas=
ic
>> > p2p datagram functionality. The use cases are various and some just
>> > state that they see it as important functionality to provide to empowe=
r
>> > the web application.
>> >
>> > Based on this I declare a rough consensus on that we should provide a
>> > non-media data service that is unreliable datagram oriented directly
>> > between the peers.
>> >
>> > One of objections against this was lack of clear requirements for what
>> > the service. The straw men requirements I included has gotten some
>> > discussion. Mostly support for them, but it is clear to me that we nee=
d
>> > to further develop them. I would recommend the proponents for driving
>> > proposals towards meeting this functionality to further discuss the
>> > requirements taking the input so far into consideration.
>> >
>> > When it comes to reliable data transfer between peers there has been 4=
-5
>> > that wanted the functionality, 2 additional that explicitly stated the=
y
>> > where okay with it. No additional that has stated against it.
>> >
>> > My interpretation is that we are close to having a rough consensus for
>> > reliable data service, but we have so far seen no proponent willing to
>> > suggest a solution for this. I would also note that a solution is like=
ly
>> > a functionality block that isn't dependent on more than the
>> > signaling/negotiation and the NAT traversal and thus can be added a
>> > later stage or be worked on with a completion date further into the
>> > future than other pieces already.
>> >
>> > So for reliable data I would recommend that someone takes on the role =
of
>> > proponent and provides a draft with their perceived requirements and a
>> > straw man proposal for how to solve these requirements so we have
>> > something more tangible to discuss.
>> >
>> > Cheers
>> >
>> > Magnus
>> >
>> > On 2011-06-27 09:36, Magnus Westerlund wrote:
>> >> 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
>> >> =A0 * Size limited by MTU
>> >> =A0 =A0 - Path MTU discovery needed
>> >> =A0 * Fragmentation by the application
>> >> - Low latency, i.e. Peer to Peer preferable
>> >> - Congestion Controlled, to be
>> >> =A0 * Network friendly
>> >> =A0 * Not become a Denial of Service tool
>> >> - Security
>> >> =A0* Confidentiality
>> >> =A0* Integrity Protected
>> >> =A0* Source Authenticated (at least bound to the signalling peer)
>> >> =A0* 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 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Phone =A0+46 10 7148287
>> >> F=E4r=F6gatan 6 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Mobile +46 73 094907=
9
>> >> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
>> >> ---------------------------------------------------------------------=
-
>> >>
>> >> _______________________________________________
>> >> rtcweb mailing list
>> >> rtcweb@ietf.org
>> >> https://www.ietf.org/mailman/listinfo/rtcweb
>> >>
>> >
>> >
>> > --
>> >
>> > Magnus Westerlund
>> >
>> > ----------------------------------------------------------------------
>> > Multimedia Technologies, Ericsson Research EAB/TVM
>> > ----------------------------------------------------------------------
>> > Ericsson AB =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Phone =A0+46 10 7148287
>> > F=E4r=F6gatan 6 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Mobile +46 73 0949079
>> > SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
>> > ----------------------------------------------------------------------
>> >
>> > _______________________________________________
>> > rtcweb mailing list
>> > rtcweb@ietf.org
>> > https://www.ietf.org/mailman/listinfo/rtcweb
>>
>>
>> Cullen Jennings
>> For corporate legal information go to:
>> http://www.cisco.com/web/about/doing_business/legal/cri/index.html
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>
> Google Sweden AB | Kungsbron 2, SE-111 22 Stockholm | Org. nr. 556656-688=
0
> Apparently, this footer is required in Europe. Apologies. This email may =
be
> confidential or privileged. =A0If you received this communication by mist=
ake,
> please don't forward it to anyone else,please erase all copies and
> attachments, and please let me know that it went to the wrong person.
> =A0Thanks.
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

From tim@phonefromhere.com  Mon Jul 18 02:28:51 2011
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From: Tim Panton <tim@phonefromhere.com>
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I understand the pushback on supporting  reliable datagrams from Cullen, =
Silvia, Serge and others.

I'm pushing for it simply because I've had it in our browser rt-audio =
component, and used it extensively.
Without the ability to transport reliable datagrams in 'near realtime' =
the projects=20
we have done would have been far less compelling for the user. The =
Datagrams are often the=20
key to providing a special user experience, beyond conventional =
telephony. If we don't provide this,
we risk producing the tools for a round of  browser based deskphone =
emulators. We can do better.

As an example, the demo I show in this blog would have been much harder =
and less responsive without
realtime data coming back up the RFC 5456 channel :

=
https://babyis60.wordpress.com/2009/10/24/my-astricon-googlewave-ibook-sky=
pe-demo/

So I guess I am asking: Are we aiming for  browser based deskphone =
emulators or a new sort of
rich media experience for browser users?=20

I know what I want.

Tim.




From magnus.westerlund@ericsson.com  Mon Jul 18 06:56:11 2011
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On 2011-07-15 19:29, Tim Panton wrote:
> I don't think it is very hard to implement, we could take RFC 5456's
> sequencing and retry mechanism for example. or indeed Plan9's IL .

I would warn about thinking this is easy. It is easy to get something
that works on sunny days, it is hard to build something that is secure,
efficient, robust and fair.

I would like to point out that a reliable protocol do need a number of
functions. If those are achieve from a lower layer or need to be
implemented in this protocol is a choice. However, functionalities such as:

- Congestion Control
- Flow Control
- Retransmission
- RTT estimation
- Buffer handling
- Parameter and functionality negotiation

Are likely all need for a safe and stable and future safe protocol.

There are good reasons for choosing something there is experience in
that it works.

For example I took a quick look at RFC5456 which appears to not have
congestion control. I think it is partly relying on that the transported
flow will adopt them self if trunking, or if not, that they become
useless and the call terminates because of that.

cheers

Magnus Westerlund

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


From tim@phonefromhere.com  Mon Jul 18 07:20:05 2011
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On 18 Jul 2011, at 14:56, Magnus Westerlund wrote:

> On 2011-07-15 19:29, Tim Panton wrote:
>> I don't think it is very hard to implement, we could take RFC 5456's
>> sequencing and retry mechanism for example. or indeed Plan9's IL .
>=20
> I would warn about thinking this is easy. It is easy to get something
> that works on sunny days, it is hard to build something that is =
secure,
> efficient, robust and fair.
>=20

Agreed, getting all 4 in any protocol is hard, indeed I don't think I've =
seen it=20
done yet.

> I would like to point out that a reliable protocol do need a number of
> functions. If those are achieve from a lower layer or need to be
> implemented in this protocol is a choice. However, functionalities =
such as:
>=20
> - Congestion Control
> - Flow Control
> - Retransmission
> - RTT estimation
> - Buffer handling
> - Parameter and functionality negotiation

Agreed, but only Retransmission is specific to reliable datagrams (as =
opposed to
unreliable ones).=20

>=20
> Are likely all need for a safe and stable and future safe protocol.
>=20
> There are good reasons for choosing something there is experience in
> that it works.
>=20
> For example I took a quick look at RFC5456 which appears to not have
> congestion control. I think it is partly relying on that the =
transported
> flow will adopt them self if trunking, or if not, that they become
> useless and the call terminates because of that.

RFC5456 has 2 sorts of packet
 - miniframes, which typically form the bulk of the traffic are handled =
unreliably.=20
 - fullframes which are mostly used for signalling  but some of which =
are reliable datagrams (e.g. the TEXT frame type).
I was saying we could (in theory) ignore the miniframes and adopt the =
fullframe sequence,ack,retry mechanisms for
the reliable datagram transport.
The core protocol has been 'repurposed' in this way before : =
http://www.dundi.com/dundi.txt so it is definitely possible.

I take your point about the lack of congestion control, but the ACK =
mechanism could be used to detect incipient congestion.


>=20
> cheers
>=20
> Magnus Westerlund
>=20
>=20


Tim.=

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On 7/18/2011 9:56 AM, Magnus Westerlund wrote:
> On 2011-07-15 19:29, Tim Panton wrote:
>> I don't think it is very hard to implement, we could take RFC 5456's
>> sequencing and retry mechanism for example. or indeed Plan9's IL .
> I would warn about thinking this is easy. It is easy to get something
> that works on sunny days, it is hard to build something that is secure,
> efficient, robust and fair.

Agreed, absolutely - which is why I don't want 100 bad designs (let 
alone implementations)
in Javascript.


> I would like to point out that a reliable protocol do need a number of
> functions. If those are achieve from a lower layer or need to be
> implemented in this protocol is a choice. However, functionalities such as:
>
> - Congestion Control
> - Flow Control
> - Retransmission
> - RTT estimation
> - Buffer handling
> - Parameter and functionality negotiation
>
> Are likely all need for a safe and stable and future safe protocol.
>
> There are good reasons for choosing something there is experience in
> that it works.

Agreed also.  There are a few options, one of this is:
1) Encapsulated TCP-over-UDP, where the UDP channel is set up by the same
     mechanisms we use to set up the RTP channels, such as 
http://code.google.com/p/udptunnel/
     This is a big advantage in that normally devices behind NATs can't 
easily set up a direct TCP
     connection, and we can leverage the ability to set up p2p UDP 
channels to get it.

     The UDP channel could participate in the proposed shared bandwidth 
handling/
     congestion control, which is an advantage over a separate TCP 
channel.  The downside is that
     TCP's AIMD may cause interactions with the underlying bandwidth 
handling, so it may need
     to be excluded - some analysis is needed, once we know what we're 
doing on congestion
     control for the UDP channels.

     TCP's time characteristics are very well understood, though not 
optimal.

     The MTU would be slightly reduced - shouldn't be a problem.

It's not a datagram service, which is a minor downside, which offers the 
second option:
2) SCTP-over-UDP.  This is datagram-oriented and provides reliable delivery.
      It supports ECN and congestion control.

      One nice thing is that SCTP allows for unreliable delivery but 
with sequencing, or reliable
      delivery without in-sequence delivery, dependent on the 
application needs.  This can help
      the application optimize the realtime aspects of the reliable channel.

-- 
Randell Jesup
randell-ietf@jesup.org


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Forgive the top-posting, but I want to make a meta-comment.  If we were
planning on building something from scratch, I would strongly agree that it
would be quite tough going, especially given our timelines.  But this
working group is not allowed to re-use existing work, it is strongly
encouraged to do so.  Picking a protocol to implement which has the
characteristics below is a lot less difficult than re-starting the exercise
from scratch.

Providing XMPP, DTLS, or SCTP between the endpoints certainly has
challenges, but it would *not* involve starting from scratch.

regards,

Ted


On Mon, Jul 18, 2011 at 6:56 AM, Magnus Westerlund <
magnus.westerlund@ericsson.com> wrote:

> On 2011-07-15 19:29, Tim Panton wrote:
> > I don't think it is very hard to implement, we could take RFC 5456's
> > sequencing and retry mechanism for example. or indeed Plan9's IL .
>
> I would warn about thinking this is easy. It is easy to get something
> that works on sunny days, it is hard to build something that is secure,
> efficient, robust and fair.
>
> I would like to point out that a reliable protocol do need a number of
> functions. If those are achieve from a lower layer or need to be
> implemented in this protocol is a choice. However, functionalities such a=
s:
>
> - Congestion Control
> - Flow Control
> - Retransmission
> - RTT estimation
> - Buffer handling
> - Parameter and functionality negotiation
>
> Are likely all need for a safe and stable and future safe protocol.
>
> There are good reasons for choosing something there is experience in
> that it works.
>
> For example I took a quick look at RFC5456 which appears to not have
> congestion control. I think it is partly relying on that the transported
> flow will adopt them self if trunking, or if not, that they become
> useless and the call terminates because of that.
>
> cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> 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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<div>Forgive the top-posting, but I want to make a meta-comment. =A0If we w=
ere planning on building something from scratch, I would strongly agree tha=
t it would be quite tough going, especially given our timelines. =A0But thi=
s working group is not allowed to re-use existing work, it is strongly enco=
uraged to do so. =A0Picking a protocol to implement which has the character=
istics below is a lot less difficult than re-starting the exercise from scr=
atch.</div>
<div><br></div><div>Providing XMPP, DTLS, or SCTP between the endpoints cer=
tainly has challenges, but it would *not* involve starting from scratch.</d=
iv><div><br></div><div>regards,</div><div><br></div><div>Ted</div><div>
<br></div><div><br></div>On Mon, Jul 18, 2011 at 6:56 AM, Magnus Westerlund=
 <span dir=3D"ltr">&lt;<a href=3D"mailto:magnus.westerlund@ericsson.com">ma=
gnus.westerlund@ericsson.com</a>&gt;</span> wrote:<br><div class=3D"gmail_q=
uote">
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex;"><div class=3D"im">On 2011-07-15 19:29, Tim =
Panton wrote:<br>
&gt; I don&#39;t think it is very hard to implement, we could take RFC 5456=
&#39;s<br>
&gt; sequencing and retry mechanism for example. or indeed Plan9&#39;s IL .=
<br>
<br>
</div>I would warn about thinking this is easy. It is easy to get something=
<br>
that works on sunny days, it is hard to build something that is secure,<br>
efficient, robust and fair.<br>
<br>
I would like to point out that a reliable protocol do need a number of<br>
functions. If those are achieve from a lower layer or need to be<br>
implemented in this protocol is a choice. However, functionalities such as:=
<br>
<br>
- Congestion Control<br>
- Flow Control<br>
- Retransmission<br>
- RTT estimation<br>
- Buffer handling<br>
- Parameter and functionality negotiation<br>
<br>
Are likely all need for a safe and stable and future safe protocol.<br>
<br>
There are good reasons for choosing something there is experience in<br>
that it works.<br>
<br>
For example I took a quick look at RFC5456 which appears to not have<br>
congestion control. I think it is partly relying on that the transported<br=
>
flow will adopt them self if trunking, or if not, that they become<br>
useless and the call terminates because of that.<br>
<div><div></div><div class=3D"h5"><br>
cheers<br>
<br>
Magnus Westerlund<br>
<br>
----------------------------------------------------------------------<br>
Multimedia Technologies, Ericsson Research EAB/TVM<br>
----------------------------------------------------------------------<br>
Ericsson AB =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Phone =A0<a href=3D"tel:%2B46%=
2010%207148287" value=3D"+46107148287">+46 10 7148287</a><br>
F=E4r=F6gatan 6 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Mobile <a href=3D"tel:%2B4=
6%2073%200949079" value=3D"+46730949079">+46 73 0949079</a><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>
</div></div></blockquote></div><br>

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On Jul 18, 2011, at 7:19 , Tim Panton wrote:

>=20
> On 18 Jul 2011, at 14:56, Magnus Westerlund wrote:
>=20
>> On 2011-07-15 19:29, Tim Panton wrote:
>>> I don't think it is very hard to implement, we could take RFC 5456's
>>> sequencing and retry mechanism for example. or indeed Plan9's IL .
>>=20
>> I would warn about thinking this is easy. It is easy to get something
>> that works on sunny days, it is hard to build something that is =
secure,
>> efficient, robust and fair.
>>=20
>=20
> Agreed, getting all 4 in any protocol is hard, indeed I don't think =
I've seen it=20
> done yet.

FWIW - TLS over TCP with application level reconnect on failure seem to =
meet all 4. Where it fails if you can't easily do it in user space. So =
I'd add two more. Needs to work with high degree of nat/fw traversal =
success and be easy to deploy.=20



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On Jul 17, 2011, at 15:54 , Ted Hardie wrote:

> On Fri, Jul 15, 2011 at 8:51 AM, Cullen Jennings <fluffy@cisco.com> =
wrote:
>=20
> I'd like to push back on the reliable service. I've yet to hear a use =
case for it that was real time. It's very hard to do a reliable real =
time protocol and we have seen zero proposal for this. For non real time =
data, just dump it in dropbox of whatever your equivalent is and don't =
do it peer to peer.
>=20
> Can I see your dropbox incantation for remote camera movement?  I am =
particularly interested in how you make it useful without a reliable, =
near real-time message that says "go check dropbox for $foo"....

Ok, Ok, - I deserver to made fun of.  There are something you can't do =
with drop box or XMPP but lets take the PTU control problem. I have seen =
protocols that have semantics like "start panning right" or "pan right =
10 degrees relative to current position". These protocol all have an =
lousy user experience compared to the ones that have semantics of "go to =
absolutely position Y at time X". IMHO, the best way to do a real time =
PTU control protocol is just periodical send the absolute position you =
want the camera to be at and send it more than once, or each time the =
users presses the button, if you don't want to do an ACK. It's hard to =
been something like that if latency is important. If latency is not =
important, XMPP is easier. (thought I would still send absolute not =
relative). I'll note more than one system is doing PTU control over RTP =
using an approach much like sending DTMF via RTP.

>=20
> To put this another way, a reliable protocol associated with a real =
time stream has some pretty obvious uses (gaming has also already been =
mentioned).

I'm pushing back on it is obvious that there are use cases for this. =
That there is simultaneous need for the semantics of "you must deliver =
this data no matter how long it takes" and "this data need to get there =
in less than X time or it is useless" is not obvious to me.=20

That said, if someone has a proposal to go do this, as I said before, I =
have no object to doing this. It be fine to have reliability but I'm not =
seeing it as critical for use cases like gaming and PTU control.=20


>  Whether it is truly "real time" or not is potentially subject to =
debate, but the results of the semantic nit-picking won't tell us much.  =
The reliable protocol is a useful part of the real time system, even if =
its retransmission and congestion control methods make it less "real =
time" than rtp.

agree

>=20
> Just my two cents, also as an individual contributor,
>=20
> Ted
>=20
> =20


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On 18 Jul 2011, at 18:12, Cullen Jennings wrote:

>>=20
>>=20
>> To put this another way, a reliable protocol associated with a real =
time stream has some pretty obvious uses (gaming has also already been =
mentioned).
>=20
> I'm pushing back on it is obvious that there are use cases for this. =
That there is simultaneous need for the semantics of "you must deliver =
this data no matter how long it takes" and "this data need to get there =
in less than X time or it is useless" is not obvious to me.=20

To my mind,  _sequence_ is vital.=20

The protocol should promise to deliver these datagrams to the =
application layer in the order that they were
sent. If that requires an occasional round-trip delay to cope with =
packet loss (perhaps with a flag to say it was late),
that's acceptable (but sub-optimal), out-of-order datagrams aren't. =20

In the vast majority of cases we can expect it to deliver the datagram =
in close sync with the associated media.

Tim.=

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From: Cullen Jennings <fluffy@cisco.com>
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To: Tim Panton <tim@phonefromhere.com>
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Subject: Re: [rtcweb] realiable data service
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On Jul 15, 2011, at 10:29 , Tim Panton wrote:

> If we don't specify a  reliable transport then all the page authors =
will end adding a reliable layer in javascript, it will be as ugly as =
DTMF in RTP !

Tim, I agree with your concern here. My largest concern of not =
specifying something is that it just means we will see a million =
different things done in JavaScript. That's pretty much what we have in =
the IETF today. Lots of protocols implement a different reliability =
layer on top of UDP. However, it has not been all that bad and it has =
been very hard to get the transport guys to go and define one that =
everyone can use (I've presented in transport asking for this in the =
past). The end result has not really been all that bad. The protocols =
work even thought each one did there own reliability systems so I have =
not been all that worried about it. I'm not sure DTMF is a fair example =
to compare to...

> Or worse it will be spray and pray.

sort of like we do with audio and video in RTP :-)=20


> Most of the use-cases I can think of need  sequenced and reliable =
datagrams. If I'm playing a game I don't want it to
> drop or mis-order any one of the actions in the unholster, point, =
shoot  sequence :-)

A game might have a UI that requires you to do but due to round trip =
latency times, most games  are going to send over the write something =
more like an object to type bullet exists at this location in Phase/Time =
space and let both sides simulate from that and decide if there was a =
hit or not. The phase/time space is just the location of the bullet, =
information about it such as it's type, and it's velocity vector as well =
as the point in time. That might get acked or retransmitted or just =
ignored on loss depending on the reliability of the game but a FPS is =
going to have crappy UE if it both side to multiple lock step =
transaction to keep state like unholster, point, shoot in sync between =
the sides.=20

>=20
> I don't think it is very hard to implement, we could take RFC 5456's =
sequencing and retry mechanism for example.
> or indeed Plan9's IL .

Again, no object to someone coming up with a good proposal (I suspect =
that 5456 would get more than a bit of push back) but it's not on my =
high priority list because the games I can think of could probably work =
fine with a proposal more like the above than needing a reliable =
protocol. That said, I do share your concern about we will just see lots =
of crappy implementations of this a reliable protocol for file transfer =
on top of whatever we do for this RTCWeb work.=20

>=20
> Tim.
>=20
>=20
> On 15 Jul 2011, at 16:51, Cullen Jennings wrote:
>=20
>>=20
>> I'd like to push back on the reliable service. I've yet to hear a use =
case for it that was real time. It's very hard to do a reliable real =
time protocol and we have seen zero proposal for this. For non real time =
data, just dump it in dropbox of whatever your equivalent is and don't =
do it peer to peer. Unless someone has a real need, and wants to put =
forward a proposal, I don't see a need to focus energy on this right =
now. I'd rather work on the thing everyone agrees they do need which is =
the unreliable transfer.=20
>>=20
>> Cullen <in my individaul contribut role>=20
>>=20
>> On Jul 13, 2011, at 8:57 , Magnus Westerlund wrote:
>>=20
>>> Hi,
>>>=20
>>> I have reviewed the input both the last 2 weeks and the discussion =
back
>>> in April.
>>>=20
>>> I see a strong support but with at least 2 people disagreeing to a =
basic
>>> p2p datagram functionality. The use cases are various and some just
>>> state that they see it as important functionality to provide to =
empower
>>> the web application.
>>>=20
>>> Based on this I declare a rough consensus on that we should provide =
a
>>> non-media data service that is unreliable datagram oriented directly
>>> between the peers.
>>>=20
>>> One of objections against this was lack of clear requirements for =
what
>>> the service. The straw men requirements I included has gotten some
>>> discussion. Mostly support for them, but it is clear to me that we =
need
>>> to further develop them. I would recommend the proponents for =
driving
>>> proposals towards meeting this functionality to further discuss the
>>> requirements taking the input so far into consideration.
>>>=20
>>> When it comes to reliable data transfer between peers there has been =
4-5
>>> that wanted the functionality, 2 additional that explicitly stated =
they
>>> where okay with it. No additional that has stated against it.
>>>=20
>>> My interpretation is that we are close to having a rough consensus =
for
>>> reliable data service, but we have so far seen no proponent willing =
to
>>> suggest a solution for this. I would also note that a solution is =
likely
>>> a functionality block that isn't dependent on more than the
>>> signaling/negotiation and the NAT traversal and thus can be added a
>>> later stage or be worked on with a completion date further into the
>>> future than other pieces already.
>>>=20
>>> So for reliable data I would recommend that someone takes on the =
role of
>>> proponent and provides a draft with their perceived requirements and =
a
>>> straw man proposal for how to solve these requirements so we have
>>> something more tangible to discuss.
>>>=20
>>> Cheers
>>>=20
>>> Magnus
>>>=20
>>> On 2011-06-27 09:36, Magnus Westerlund wrote:
>>>> 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=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
>>>=20
>>>=20
>>> --=20
>>>=20
>>> Magnus Westerlund
>>>=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
>>=20
>> Cullen Jennings
>> For corporate legal information go to:
>> http://www.cisco.com/web/about/doing_business/legal/cri/index.html
>>=20
>>=20
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


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



From fluffy@cisco.com  Mon Jul 18 11:30:37 2011
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From: Cullen Jennings <fluffy@cisco.com>
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Subject: [rtcweb] Rohan Red Cross Use Case
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I'd like to add an use case along the following lines...

Rohan works for international aid agency and deploys wireless networks =
in places where disasters have wiped out all the existing =
infrastructure. There's just been an earthquake in Haiti. Betty works =
for Canada Red Cross and need to get the Red Cross location in Haiti =
connected to other aid groups in Haiti to help arrange complex logistics =
of things like what order various aid agencies are even going to use the =
limited resources of the runway at the airport. Rohan and Betty both use =
the AidNet web application for communicating and have cached copies =
using the HTML5 offline support on their computers or, if they can't get =
their computers into the country, can at least get copy in via USB key =
and find a local computer with a browser. Rohan sets up a wireless =
access point on the tallest thing he can find and managed to provide =
wireless coverage for many square miles. Betty connects to that and =
manages to call Rohan to coordinate and let Rohan know the location of =
RedCross so that better wireless can be arranged there.=20

This one more or less one of the use cases for P2PSIP work to be able to =
set up an ad hoc voice communications system before one even had access =
to DNS server or any global internet infrastructure. I think that the =
whole P2PSIP system could be done inside browser (or a system similar to =
it). The underling transport of P2PSIP is very close to the proposal =
here including the use of ICE. One would probably need to define a new =
transport that an Reload overlay could use but other than that, I think =
one could meet this use case using something along the lines of Reload =
running in browser.=20

It gets more interesting as you add work such as what is a happening in =
the VIPR WG. The browser on the mobile could have access the the call =
logs for calls made over the cellular voice part of the phone. That =
would allow the browser to run the VIPR related protocols and be able to =
receive VoIP calls for that number.





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On 07/18/2011 10:12 AM, Cullen Jennings wrote:
> There are something you can't do with drop box or XMPP but lets take 
> the PTU control problem. I have seen protocols that have semantics 
> like "start panning right" or "pan right 10 degrees relative to 
> current position". These protocol all have an lousy user experience 
> compared to the ones that have semantics of "go to absolutely position 
> Y at time X".

Gets kind of hard in consideration of gesture recognition as semantics, 
so I think the example above shows why semantic terminology leads to 
bloat without given sequences. The difference is useful where error 
correction is more obvious; for example, avoidance of RTT vs 
retransmission (over wireless). Lossless compression is valuable here, 
yet the semantics of compression let people think lossy is ideal, which 
lossy leads later to extra links (the bloat). Tweaked the example above: 
"track X for duration Y until Z"


-- 
--- http://twitter.com/Dzonatas_Sol ---
Web Development, Software Engineering
Ag-Biotech, Virtual Reality, Consultant


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On Jul 15, 2011, at 8:51 AM, Cullen Jennings wrote:

>=20
> I'd like to push back on the reliable service. I've yet to hear a use =
case for it that was real time. It's very hard to do a reliable real =
time protocol and we have seen zero proposal for this. For non real time =
data, just dump it in dropbox of whatever your equivalent is and don't =
do it peer to peer. Unless someone has a real need, and wants to put =
forward a proposal, I don't see a need to focus energy on this right =
now. I'd rather work on the thing everyone agrees they do need which is =
the unreliable transfer.=20

I'm personally torn on this one. I think that if we want to get =
something actually specified and deal with the majority of the use =
cases, we really need to focus on A/V and make data as simple as =
possible... that means unreliable datagrams with nothing but browser =
enforcement of rate via a simple congestion control algorithm.

On the other hand, if we go that route, people building things with the =
API will be forced to do it in Javascript... and all the arguments for =
putting ICE in the browser code instead of Javascript apply here, =
specifically that getting it right for N=3D# of browser implementations =
case is much more likely than getting it right for N=3D# of Javascript =
coders case.

But to get there probably requires solving many of the same set of =
problems that MFP and RTMFP solved, and that's a whole lot of =
specification and code to implement it, and we haven't even started on =
the agreement of which building blocks to start with (DCCP? SCTP? TCP =
over UDP?) much less gotten to a complete specification.

Unfortunately the right answer is probably to, once again, go to TSVAREA =
and state that a protocol that fits set of requirements would be really =
useful, and then not put it in browsers until it exists. There might be =
some value to making that ask in the form of a requirements document, =
but it needs to not be a "requirements of browsers for RTCWEB", instead =
it needs to be "requirements for a data transport protocol if a =
sophisticated one were to be added to RTCWEB in the future".

As I do remember most of the reasons we came up with MFP and RTMFP, I =
would be glad to contribute to such a requirements document.

But until then, I think we need *something* that lets data get from A to =
B over the direct unreliable channel.

Matthew Kaufman=

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On Jul 18, 2011, at 11:52 , Matthew Kaufman wrote:

>=20
> On Jul 15, 2011, at 8:51 AM, Cullen Jennings wrote:
>=20
>>=20
>> I'd like to push back on the reliable service. I've yet to hear a use =
case for it that was real time. It's very hard to do a reliable real =
time protocol and we have seen zero proposal for this. For non real time =
data, just dump it in dropbox of whatever your equivalent is and don't =
do it peer to peer. Unless someone has a real need, and wants to put =
forward a proposal, I don't see a need to focus energy on this right =
now. I'd rather work on the thing everyone agrees they do need which is =
the unreliable transfer.=20
>=20
> I'm personally torn on this one. I think that if we want to get =
something actually specified and deal with the majority of the use =
cases, we really need to focus on A/V and make data as simple as =
possible... that means unreliable datagrams with nothing but browser =
enforcement of rate via a simple congestion control algorithm.
>=20
> On the other hand, if we go that route, people building things with =
the API will be forced to do it in Javascript... and all the arguments =
for putting ICE in the browser code instead of Javascript apply here, =
specifically that getting it right for N=3D# of browser implementations =
case is much more likely than getting it right for N=3D# of Javascript =
coders case.
>=20
> But to get there probably requires solving many of the same set of =
problems that MFP and RTMFP solved, and that's a whole lot of =
specification and code to implement it, and we haven't even started on =
the agreement of which building blocks to start with (DCCP? SCTP? TCP =
over UDP?) much less gotten to a complete specification.
>=20
> Unfortunately the right answer is probably to, once again, go to =
TSVAREA and state that a protocol that fits set of requirements would be =
really useful, and then not put it in browsers until it exists. There =
might be some value to making that ask in the form of a requirements =
document, but it needs to not be a "requirements of browsers for =
RTCWEB", instead it needs to be "requirements for a data transport =
protocol if a sophisticated one were to be added to RTCWEB in the =
future".
>=20
> As I do remember most of the reasons we came up with MFP and RTMFP, I =
would be glad to contribute to such a requirements document.
>=20
> But until then, I think we need *something* that lets data get from A =
to B over the direct unreliable channel.
>=20
> Matthew Kaufman


I find myself on about the same page. I'd be happy to help go write up =
some requirement slides for the transport area.=20



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On 7/18/2011 1:12 PM, Cullen Jennings wrote:
> On Jul 17, 2011, at 15:54 , Ted Hardie wrote:
>
>> On Fri, Jul 15, 2011 at 8:51 AM, Cullen Jennings<fluffy@cisco.com>  wrote:
>>
>> I'd like to push back on the reliable service. I've yet to hear a use case for it that was real time. It's very hard to do a reliable real time protocol and we have seen zero proposal for this. For non real time data, just dump it in dropbox of whatever your equivalent is and don't do it peer to peer.
>>
>> Can I see your dropbox incantation for remote camera movement?  I am particularly interested in how you make it useful without a reliable, near real-time message that says "go check dropbox for $foo"....
> Ok, Ok, - I deserver to made fun of.  There are something you can't do with drop box or XMPP but lets take the PTU control problem. I have seen protocols that have semantics like "start panning right" or "pan right 10 degrees relative to current position". These protocol all have an lousy user experience compared to the ones that have semantics of "go to absolutely position Y at time X". IMHO, the best way to do a real time PTU control protocol is just periodical send the absolute position you want the camera to be at and send it more than once, or each time the users presses the button, if you don't want to do an ACK. It's hard to been something like that if latency is important. If latency is not important, XMPP is easier. (thought I would still send absolute not relative). I'll note more than one system is doing PTU control over RTP using an approach much like sending DTMF via RTP.

I agree it's easier/more accurate to say "move to X".  However, people 
are used to various
"continuous" motion controls, like zoom and trying to frame a subject, 
or follow motion.  From
a UI point of view, a good low-delay continuous reactive system is often 
more functional, depending
the problem space.  People will find a way to fit their preferred user 
experience onto the protocol,
even if it's not a good fit for the protocol, because they don't care 
about the protocol - they
care about their UI.

>> To put this another way, a reliable protocol associated with a real time stream has some pretty obvious uses (gaming has also already been mentioned).
> I'm pushing back on it is obvious that there are use cases for this. That there is simultaneous need for the semantics of "you must deliver this data no matter how long it takes" and "this data need to get there in less than X time or it is useless" is not obvious to me.

It's rare that "no matter how long it takes" is hugely delayed - if it 
is, you have other, bigger
problems.  Delayed - could well be; usually only a little, with some 
jitter.  And SCTP would
be even better as far as that's concerned.

Talk to some RT networked game designers.  I'll bet they all use a mix 
of reliable and unreliable
protocols (perhaps not like TCP, more like SCTP).  But even though 
reliable == not guaranteed,
I'll bet they all want it to get there as fast as possible in the normal 
case.

People are ignoring the huge positive of having this data connection 
bundled with the other
streams and connected with them.  Having to set up a parallel reliable 
stream adds all sorts
of complexity to the application when firewall traversal/IP address 
changes/etc are included, and
then you have the issue of it being a second distinct stream from a 
congestion point of view - it
makes it very hard to control the relative bandwidth of the reliable 
stream versus all the unreliable
streams - you're likely to get ~ 50% for the reliable stream if you're 
overcommitted, with the % varying
quite a bit.  And if the stream has to go through a relay or server 
while media goes P2P
you have an issue for people designing services to use this protocol - 
increased cost and bandwidth
for one.  Even a chat-with-file-transfer - if the file xfer has to go 
through the server, you have to maintain
bandwidth for that case, and transfer speed may be limited by RTT of 
going through a remote server
(think Taiwan user to Taiwan user with a relay in the US).

All that said - yes, there are complexity issues to consider, which was 
why I was suggesting leveraging
an existing tunneled protocol like SCTP or even TCP-over-UDP.

-- 
Randell Jesup
randell-ietf@jesup.org


From magnus.westerlund@ericsson.com  Tue Jul 19 07:01:59 2011
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Subject: Re: [rtcweb] realiable data service
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On 2011-07-18 20:21, Cullen Jennings wrote:
> 
> Tim, I agree with your concern here. My largest concern of not
> specifying something is that it just means we will see a million
> different things done in JavaScript. That's pretty much what we have
> in the IETF today. Lots of protocols implement a different
> reliability layer on top of UDP. 

I would note that the the JS library at least is going to run on top of
a congestion controlled and secured datagram service. Thus, any
implementor can primarily hurt themselves rather than the network.

> However, it has not been all that
> bad and it has been very hard to get the transport guys to go and
> define one that everyone can use (I've presented in transport asking
> for this in the past). The end result has not really been all that
> bad. The protocols work even thought each one did there own
> reliability systems so I have not been all that worried about it. I'm
> not sure DTMF is a fair example to compare to...

I can't resist this bate as a transport guy. Well, things don't happen
for free, and no one was willing to stick in there for the long haul to
actually define it.

There also has been a lot of political resistance, and the fact that
transport development in OS etc happens at glacier speed, unless someone
well connected enough truly wants something.

I think a reasonable, re-use as much as possible, sane proposal has some
chance. And we WG chairs should try to get early feedback from TSV-Dir
etc so we avoid late surprises.

Please note that this is not an a comment in any direction around the
support for reliable data service. Just trying to provide some
background and feel for the land.

cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Frgatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
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From henry.sinnreich@gmail.com  Tue Jul 19 07:16:56 2011
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Date: Tue, 19 Jul 2011 09:16:42 -0500
From: Henry Sinnreich <henry.sinnreich@gmail.com>
To: Serge Lachapelle <sergel@google.com>, Cullen Jennings <fluffy@cisco.com>
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+1
Maybe it would be useful for the proponents of this and many other
=B3desirable=B2 features to think from the perspective if they would have to pa=
y
themselves for such developments and also make the code freely available.
Cullen at least will make his code available.

Thanks, Henry


On 7/18/11 2:16 AM, "Serge Lachapelle" <sergel@google.com> wrote:

> Hi Cullen,
>=20
> I agree with your push back.
>=20
> Focus is very important. Audio and video already present a huge challenge=
.
> (signalling, network, web developer friendly API, security, browser
> integration...)
>=20
> Also, I think that there is a real risk in introducing "duplicate"
> functionality in the browser as it may confuse web developers. There is a=
 lot
> going on in HTML5...
>=20
> In my mind, this is "version 2.0" stuff.
>=20
> /Serge
>=20
>=20
> On Fri, Jul 15, 2011 at 17:51, Cullen Jennings <fluffy@cisco.com> wrote:
>>=20
>> I'd like to push back on the reliable service. I've yet to hear a use ca=
se
>> for it that was real time. It's very hard to do a reliable real time pro=
tocol
>> and we have seen zero proposal for this. For non real time data, just du=
mp it
>> in dropbox of whatever your equivalent is and don't do it peer to peer.
>> Unless someone has a real need, and wants to put forward a proposal, I d=
on't
>> see a need to focus energy on this right now. I'd rather work on the thi=
ng
>> everyone agrees they do need which is the unreliable transfer.
>>=20
>> Cullen <in my individaul contribut role>
>>=20
>> On Jul 13, 2011, at 8:57 , Magnus Westerlund wrote:
>>=20
>>> > Hi,
>>> >
>>> > I have reviewed the input both the last 2 weeks and the discussion ba=
ck
>>> > in April.
>>> >
>>> > I see a strong support but with at least 2 people disagreeing to a ba=
sic
>>> > p2p datagram functionality. The use cases are various and some just
>>> > state that they see it as important functionality to provide to empow=
er
>>> > the web application.
>>> >
>>> > Based on this I declare a rough consensus on that we should provide a
>>> > non-media data service that is unreliable datagram oriented directly
>>> > between the peers.
>>> >
>>> > One of objections against this was lack of clear requirements for wha=
t
>>> > the service. The straw men requirements I included has gotten some
>>> > discussion. Mostly support for them, but it is clear to me that we ne=
ed
>>> > to further develop them. I would recommend the proponents for driving
>>> > proposals towards meeting this functionality to further discuss the
>>> > requirements taking the input so far into consideration.
>>> >
>>> > When it comes to reliable data transfer between peers there has been =
4-5
>>> > that wanted the functionality, 2 additional that explicitly stated th=
ey
>>> > where okay with it. No additional that has stated against it.
>>> >
>>> > My interpretation is that we are close to having a rough consensus fo=
r
>>> > reliable data service, but we have so far seen no proponent willing t=
o
>>> > suggest a solution for this. I would also note that a solution is lik=
ely
>>> > a functionality block that isn't dependent on more than the
>>> > signaling/negotiation and the NAT traversal and thus can be added a
>>> > later stage or be worked on with a completion date further into the
>>> > future than other pieces already.
>>> >
>>> > So for reliable data I would recommend that someone takes on the role=
 of
>>> > proponent and provides a draft with their perceived requirements and =
a
>>> > straw man proposal for how to solve these requirements so we have
>>> > something more tangible to discuss.
>>> >
>>> > Cheers
>>> >
>>> > Magnus
>>> >
>>> > On 2011-06-27 09:36, Magnus Westerlund wrote:
>>>> >> WG,
>>>> >>
>>>> >> At the interim it was planned to have a bit discussion on the datag=
ram
>>>> >> 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. n=
ot
>>>> >> 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 th=
e
>>>> >> RTCWEB provide a unreliable datagram service. This clearly needs
>>>> >> additional requirements to be secure and safe to deploy, but more a=
bout
>>>> >> 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 orient=
ed
>>>> >> 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 soluti=
on
>>>> >> please provide motivation and use case and requirements.
>>>> >>
>>>> >> I also want to take a stab on what I personally see as the requirem=
ents
>>>> >> that exist on unreliable datagram service in the context of RTCWEB.
>>>> >>
>>>> >> - Unreliable data transmission
>>>> >> - Datagram oriented
>>>> >> =A0 * Size limited by MTU
>>>> >> =A0 =A0 - Path MTU discovery needed
>>>> >> =A0 * Fragmentation by the application
>>>> >> - Low latency, i.e. Peer to Peer preferable
>>>> >> - Congestion Controlled, to be
>>>> >> =A0 * Network friendly
>>>> >> =A0 * Not become a Denial of Service tool
>>>> >> - Security
>>>> >> =A0* Confidentiality
>>>> >> =A0* Integrity Protected
>>>> >> =A0* Source Authenticated (at least bound to the signalling peer)
>>>> >> =A0* 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 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Phone =A0


--B_3393911806_989140
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<HTML>
<HEAD>
<TITLE>Re: [rtcweb] realiable data service</TITLE>
</HEAD>
<BODY>
<FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STYLE=3D'font-size:13pt=
'>+1<BR>
Maybe it would be useful for the proponents of this and many other &#8220;d=
esirable&#8221; features to think from the perspective if they would have to=
 pay themselves for such developments and also make the code freely availabl=
e.<BR>
Cullen at least will make his code available.<BR>
<BR>
Thanks, Henry<BR>
<BR>
<BR>
On 7/18/11 2:16 AM, &quot;Serge Lachapelle&quot; &lt;<a href=3D"sergel@google=
.com">sergel@google.com</a>&gt; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><=
SPAN STYLE=3D'font-size:13pt'>Hi Cullen,<BR>
<BR>
I agree with your push back.<BR>
<BR>
Focus is very important. Audio and video already present a huge challenge. =
(signalling, network, web developer friendly API, security, browser integrat=
ion...)<BR>
<BR>
Also, I think that there is a real risk in introducing &quot;duplicate&quot=
; functionality in the browser as it may confuse web developers. There is a =
lot going on in HTML5...<BR>
<BR>
In my mind, this is &quot;version 2.0&quot; stuff.<BR>
<BR>
/Serge<BR>
<BR>
<BR>
On Fri, Jul 15, 2011 at 17:51, Cullen Jennings &lt;<a href=3D"fluffy@cisco.co=
m">fluffy@cisco.com</a>&gt; wrote:<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><=
SPAN STYLE=3D'font-size:13pt'><BR>
I'd like to push back on the reliable service. I've yet to hear a use case =
for it that was real time. It's very hard to do a reliable real time protoco=
l and we have seen zero proposal for this. For non real time data, just dump=
 it in dropbox of whatever your equivalent is and don't do it peer to peer. =
Unless someone has a real need, and wants to put forward a proposal, I don't=
 see a need to focus energy on this right now. I'd rather work on the thing =
everyone agrees they do need which is the unreliable transfer.<BR>
<BR>
Cullen &lt;in my individaul contribut role&gt;<BR>
<BR>
On Jul 13, 2011, at 8:57 , Magnus Westerlund wrote:<BR>
<BR>
&gt; Hi,<BR>
&gt;<BR>
&gt; I have reviewed the input both the last 2 weeks and the discussion bac=
k<BR>
&gt; in April.<BR>
&gt;<BR>
&gt; I see a strong support but with at least 2 people disagreeing to a bas=
ic<BR>
&gt; p2p datagram functionality. The use cases are various and some just<BR=
>
&gt; state that they see it as important functionality to provide to empowe=
r<BR>
&gt; the web application.<BR>
&gt;<BR>
&gt; Based on this I declare a rough consensus on that we should provide a<=
BR>
&gt; non-media data service that is unreliable datagram oriented directly<B=
R>
&gt; between the peers.<BR>
&gt;<BR>
&gt; One of objections against this was lack of clear requirements for what=
<BR>
&gt; the service. The straw men requirements I included has gotten some<BR>
&gt; discussion. Mostly support for them, but it is clear to me that we nee=
d<BR>
&gt; to further develop them. I would recommend the proponents for driving<=
BR>
&gt; proposals towards meeting this functionality to further discuss the<BR=
>
&gt; requirements taking the input so far into consideration.<BR>
&gt;<BR>
&gt; When it comes to reliable data transfer between peers there has been 4=
-5<BR>
&gt; that wanted the functionality, 2 additional that explicitly stated the=
y<BR>
&gt; where okay with it. No additional that has stated against it.<BR>
&gt;<BR>
&gt; My interpretation is that we are close to having a rough consensus for=
<BR>
&gt; reliable data service, but we have so far seen no proponent willing to=
<BR>
&gt; suggest a solution for this. I would also note that a solution is like=
ly<BR>
&gt; a functionality block that isn't dependent on more than the<BR>
&gt; signaling/negotiation and the NAT traversal and thus can be added a<BR=
>
&gt; later stage or be worked on with a completion date further into the<BR=
>
&gt; future than other pieces already.<BR>
&gt;<BR>
&gt; So for reliable data I would recommend that someone takes on the role =
of<BR>
&gt; proponent and provides a draft with their perceived requirements and a=
<BR>
&gt; straw man proposal for how to solve these requirements so we have<BR>
&gt; something more tangible to discuss.<BR>
&gt;<BR>
&gt; Cheers<BR>
&gt;<BR>
&gt; Magnus<BR>
&gt;<BR>
&gt; On 2011-06-27 09:36, Magnus Westerlund wrote:<BR>
&gt;&gt; WG,<BR>
&gt;&gt;<BR>
&gt;&gt; At the interim it was planned to have a bit discussion on the data=
gram<BR>
&gt;&gt; service for RTCWEB. The first question to try to resolve if there<=
BR>
&gt;&gt; is consensus for including some form of non real-time media (i.e. =
not<BR>
&gt;&gt; audio, video) service between peers. This is a bit tangled with th=
e<BR>
&gt;&gt; actual requirements and use cases. But there was views both for it=
 and<BR>
&gt;&gt; against it on the mailing list. So lets continue and try to come t=
o a<BR>
&gt;&gt; conclusion on this discussion.<BR>
&gt;&gt;<BR>
&gt;&gt; The use cases mentioned on the mailing list are:<BR>
&gt;&gt;<BR>
&gt;&gt; - Dynamic meta data for Conference and other real-time services<BR=
>
&gt;&gt;<BR>
&gt;&gt; - Gaming data with low latency requirements<BR>
&gt;&gt;<BR>
&gt;&gt; Does anyone like to add additional use cases?<BR>
&gt;&gt;<BR>
&gt;&gt; Based on my personal understanding this points to primarily have t=
he<BR>
&gt;&gt; RTCWEB provide a unreliable datagram service. This clearly needs<B=
R>
&gt;&gt; additional requirements to be secure and safe to deploy, but more =
about<BR>
&gt;&gt; this below. I still like to ask the WG here a question.<BR>
&gt;&gt;<BR>
&gt;&gt; Are you supporting the inclusion of a unreliable datagram service<=
BR>
&gt;&gt; directly between peers? Please provide your view and any additiona=
l<BR>
&gt;&gt; statements of motivation that you desire to provide.<BR>
&gt;&gt;<BR>
&gt;&gt; Secondly, there is a question if there needs to have something tha=
t<BR>
&gt;&gt; provides reliable message (of arbitrary size) or byte stream orien=
ted<BR>
&gt;&gt; data transport between the peers. I personally foresee that people=
 will<BR>
&gt;&gt; build JS libraries for this on top of a unreliable datagram servic=
e. If<BR>
&gt;&gt; you desire reliable data service as part of the standardized solut=
ion<BR>
&gt;&gt; please provide motivation and use case and requirements.<BR>
&gt;&gt;<BR>
&gt;&gt; I also want to take a stab on what I personally see as the require=
ments<BR>
&gt;&gt; that exist on unreliable datagram service in the context of RTCWEB=
.<BR>
&gt;&gt;<BR>
&gt;&gt; - Unreliable data transmission<BR>
&gt;&gt; - Datagram oriented<BR>
&gt;&gt; =A0 * Size limited by MTU<BR>
&gt;&gt; =A0 =A0 - Path MTU discovery needed<BR>
&gt;&gt; =A0 * Fragmentation by the application<BR>
&gt;&gt; - Low latency, i.e. Peer to Peer preferable<BR>
&gt;&gt; - Congestion Controlled, to be<BR>
&gt;&gt; =A0 * Network friendly<BR>
&gt;&gt; =A0 * Not become a Denial of Service tool<BR>
&gt;&gt; - Security<BR>
&gt;&gt; =A0* Confidentiality<BR>
&gt;&gt; =A0* Integrity Protected<BR>
&gt;&gt; =A0* Source Authenticated (at least bound to the signalling peer)<BR=
>
&gt;&gt; =A0* Ensure consent to receive data<BR>
&gt;&gt;<BR>
&gt;&gt; Please debate the above. This is an attempt to ensure that we can<=
BR>
&gt;&gt; establish WG consensus on both data service and any requirements.<=
BR>
&gt;&gt;<BR>
&gt;&gt; cheers<BR>
&gt;&gt;<BR>
&gt;&gt; Magnus Westerlund<BR>
&gt;&gt;<BR>
&gt;&gt; ------------------------------------------------------------------=
----<BR>
&gt;&gt; Multimedia Technologies, Ericsson Research EAB/TVM<BR>
&gt;&gt; ------------------------------------------------------------------=
----<BR>
&gt;&gt; Ericsson AB =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Phone =A0<BR>
</SPAN></FONT></BLOCKQUOTE></BLOCKQUOTE>
</BODY>
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Subject: [rtcweb] To multiplex or not!
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Hi,

This email is as an individual contributor.

I want to get started on the discussion of the Multiplexing of the
various protocols over single lower layer transport flow, such as a UDP
flow. I will attempt to split up the questions into different emails.

The first question I think is reasonably easy to get answered, but I
think it is time we determine if my belief in the answer is correct or not.

The traffic between two RTCWEB peers from the various components, such
as RTP sessions, datagram service:

a) MUST be sent as Individual flows for each component.

b) MUST be multiplexed into a single transport flow.

c) SHOULD be multiplexed into a single transport flow, but the RTCWEB
peer MUST be able to send them as individual flows.

I would love if people can indicate their choice or preferences.

I personally prefer A as it it is simplest in all aspect except the NAT
traversal.
- It allows for flow based QoS.
- It is the what the implementation that exist mostly do
- Signaling protocols that exist support it, no extra functionality
- People are used to the concept
- It minimizes the difference to legacy.

Thus it is the quickest road to define something with the least formal
push back and concern over maturity of any solution.

The downside with B and C is that we do have to solve the multiplexing
and get an agreement that gets through all the hurdles.

Of these two opens I do prefer C.  Although it results in the extra
complexities of having both alternatives, it will give us both a
fallback, flow based QoS and better legacy support.

Now it is your time to make your opinion heard!

Cheers

Magnus Westerlund

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


From magnus.westerlund@ericsson.com  Tue Jul 19 08:11:44 2011
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Subject: [rtcweb] How to multiplex between peers
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WG,

As individual contributor.

I would recommend that people read both
https://datatracker.ietf.org/doc/draft-rosenberg-rtcweb-rtpmux/?include_text=1
and
https://datatracker.ietf.org/doc/draft-perkins-rtcweb-rtp-usage/?include_text=1
before getting into this debate.

Assuming that the consensus of "To multiplex or not!" thread is either
of the options requiring multiplexing we get to the question of how to
do the multiplexing.

Lets start with a very fundamental question:

Explicit vs Implicit multiplexing

So the explicit is when something in the received packets has the
explicit purpose to identifies a context that resolves the protocol, the
session context etc.

Implicit is when we have no such explicit field, instead we have to rely
on that investigating values of particular bits in the payload of the
packet. For example trying to identify RTP based on the first byte of
the RTP header where the version two bits shall be 10b.

I am strongly arguing against implicit identifying the protocol for the
following reasons.

1. We have 4 or more protocols (The possible set of protocols include:
STUN, RTP/RTCP, Datagram, and DTLS-SRTP. There is likely future
extensions, like reliable protocol)  that needs to be identified and
demultiplexed. A single extension in one of these protocols can cause
the whole house of cards to come tumbling down.

2. It will require one to find an RTP internal session multiplexing
mechanism. This is likely not easy or causes a fork of RTP into two
non-compatible versions. In addition such a mechanism needs defining
which takes time, likely substantial time as there is quite a lot of RTP
mechanisms to consider.

3. We likely need a multiplexing layer anyway if we want a fall back
over WebSockets or HTTP.

Thus we see this a very fragile mechanism that creates additional work
overhead and large uncertainties in when it would be available.

We have in our draft:
https://datatracker.ietf.org/doc/draft-perkins-rtcweb-rtp-usage/ looked
at two possible suggestions for explicit multiplexing. Either a thin
shim layer primarily consisting of a identifier that through negotiation
is bound to a context, like an particular RTP session or a datagram
channel.

The other alternative we considered was using DCCP over UDP for RTP and
datagram. That way we got a complete port space for multiplexing
different purposes for flows. We also got a transport framework with its
own negotiation, congestion control, a several other features.

We invite people to consider these proposals or suggest better
alternatives of explicit multiplexing. But I hope we can avoid a serious
mistake and select implicit multiplexing.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Frgatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
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From juberti@google.com  Tue Jul 19 08:21:37 2011
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From: Justin Uberti <juberti@google.com>
Date: Tue, 19 Jul 2011 08:20:56 -0700
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On Tue, Jul 19, 2011 at 7:28 AM, Magnus Westerlund <
magnus.westerlund@ericsson.com> wrote:

> Hi,
>
> This email is as an individual contributor.
>
> I want to get started on the discussion of the Multiplexing of the
> various protocols over single lower layer transport flow, such as a UDP
> flow. I will attempt to split up the questions into different emails.
>
> The first question I think is reasonably easy to get answered, but I
> think it is time we determine if my belief in the answer is correct or no=
t.
>
> The traffic between two RTCWEB peers from the various components, such
> as RTP sessions, datagram service:
>

To be clear, we are talking about different RTP sessions, as opposed to
different RTP sources within a single RTP session, correct?


> a) MUST be sent as Individual flows for each component.
>
> b) MUST be multiplexed into a single transport flow.
>
> c) SHOULD be multiplexed into a single transport flow, but the RTCWEB
> peer MUST be able to send them as individual flows.
>
> I would love if people can indicate their choice or preferences.
>
> I personally prefer A as it it is simplest in all aspect except the NAT
> traversal.
> - It allows for flow based QoS.
>

For video applications, it's not clear that flow-based QoS is sufficient, a=
s
some packets are of higher importance than others. Diffserv markings allow
per-packet QoS to be applied, but this could be done even in a multiplexing
scenario.


> - It is the what the implementation that exist mostly do
> - Signaling protocols that exist support it, no extra functionality
> - People are used to the concept
> - It minimizes the difference to legacy.
>

I agree that existing implementations will need to change to support this.
However, is there a large enough base of deployed endpoints that support a)
multiple RTP sessions and b) the other RTCWEB constrains (ICE, etc) to make
this a significant concern?


> Thus it is the quickest road to define something with the least formal
> push back and concern over maturity of any solution.
>
> The downside with B and C is that we do have to solve the multiplexing
> and get an agreement that gets through all the hurdles.
>
> Of these two opens I do prefer C.  Although it results in the extra
> complexities of having both alternatives, it will give us both a
> fallback, flow based QoS and better legacy support.
>



> Now it is your time to make your opinion heard!
>
> 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
>

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<br><br><div class=3D"gmail_quote">On Tue, Jul 19, 2011 at 7:28 AM, Magnus =
Westerlund <span dir=3D"ltr">&lt;<a href=3D"mailto:magnus.westerlund@ericss=
on.com">magnus.westerlund@ericsson.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;">

Hi,<br>
<br>
This email is as an individual contributor.<br>
<br>
I want to get started on the discussion of the Multiplexing of the<br>
various protocols over single lower layer transport flow, such as a UDP<br>
flow. I will attempt to split up the questions into different emails.<br>
<br>
The first question I think is reasonably easy to get answered, but I<br>
think it is time we determine if my belief in the answer is correct or not.=
<br>
<br>
The traffic between two RTCWEB peers from the various components, such<br>
as RTP sessions, datagram service:<br></blockquote><div><br></div><div>To b=
e clear, we are talking about different RTP sessions, as opposed to differe=
nt RTP sources within a single RTP session, correct?</div><div><br></div>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex;">
<br>
a) MUST be sent as Individual flows for each component.<br>
<br>
b) MUST be multiplexed into a single transport flow.<br>
<br>
c) SHOULD be multiplexed into a single transport flow, but the RTCWEB<br>
peer MUST be able to send them as individual flows.<br>
<br>
I would love if people can indicate their choice or preferences.<br>
<br>
I personally prefer A as it it is simplest in all aspect except the NAT<br>
traversal.<br>
- It allows for flow based QoS.<br></blockquote><div><br></div><div>For vid=
eo applications, it&#39;s not clear that flow-based QoS is sufficient, as s=
ome packets are of higher importance than others. Diffserv markings allow p=
er-packet QoS to be applied, but this could be done even in a multiplexing =
scenario.</div>

<div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8=
ex;border-left:1px #ccc solid;padding-left:1ex;">
- It is the what the implementation that exist mostly do<br>
- Signaling protocols that exist support it, no extra functionality<br>
- People are used to the concept<br>
- It minimizes the difference to legacy.<br></blockquote><div><br></div><di=
v>I agree that existing implementations will need to change to support this=
. However, is there a large enough base of deployed endpoints that support =
a) multiple RTP sessions and b) the other RTCWEB constrains (ICE, etc) to m=
ake this a significant concern?=C2=A0</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>
Thus it is the quickest road to define something with the least formal<br>
push back and concern over maturity of any solution.<br>
<br>
The downside with B and C is that we do have to solve the multiplexing<br>
and get an agreement that gets through all the hurdles.<br>
<br>
Of these two opens I do prefer C. =C2=A0Although it results in the extra<br=
>
complexities of having both alternatives, it will give us both a<br>
fallback, flow based QoS and better legacy support.<br></blockquote><div><b=
r></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>
Now it is your time to make your opinion heard!<br>
<br>
Cheers<br>
<br>
Magnus Westerlund<br>
<br>
----------------------------------------------------------------------<br>
Multimedia Technologies, Ericsson Research EAB/TVM<br>
----------------------------------------------------------------------<br>
Ericsson AB =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0| Phone =
=C2=A0<a href=3D"tel:%2B46%2010%207148287" value=3D"+46107148287">+46 10 71=
48287</a><br>
F=C3=A4r=C3=B6gatan 6 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0| Mobile <a href=3D"tel:%2B46%2073%200949079" value=3D"+46730949079">+46=
 73 0949079</a><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>

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On 2011-07-19 17:20, Justin Uberti wrote:
> 
> On Tue, Jul 19, 2011 at 7:28 AM, Magnus Westerlund
> <magnus.westerlund@ericsson.com <mailto:magnus.westerlund@ericsson.com>>
> wrote:
> 
>     The traffic between two RTCWEB peers from the various components, such
>     as RTP sessions, datagram service:
> 
> 
> To be clear, we are talking about different RTP sessions, as opposed to
> different RTP sources within a single RTP session, correct?

Exactly!


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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On 07/19/2011 07:28 AM, Magnus Westerlund wrote:
> a) MUST be sent as Individual flows for each component.
>
> b) MUST be multiplexed into a single transport flow.
>
> c) SHOULD be multiplexed into a single transport flow, but the RTCWEB
> peer MUST be able to send them as individual flows.
>
>    

One clarification please, if there are known ranges of A as one 
spectrum, does this WG still consider that multiplexed? IPv6, for 
example, allows us to multiplex in a TCP-less way, simply by fulfillment 
of the flows to more than one individual address. Those addresses could 
constitute one spectrum in a stateful manner, or not.

-- 
--- http://twitter.com/Dzonatas_Sol ---
Web Development, Software Engineering
Ag-Biotech, Virtual Reality, Consultant


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On 7/19/2011 8:11 AM, Magnus Westerlund wrote:
> 1. We have 4 or more protocols (The possible set of protocols include:
> STUN, RTP/RTCP, Datagram, and DTLS-SRTP. There is likely future
> extensions, like reliable protocol)  that needs to be identified and
> demultiplexed. A single extension in one of these protocols can cause
> the whole house of cards to come tumbling down.

Multiplexing RTP, RTCP, and STUN on the same port already needs to keep 
working for lots of other reasons, we can leverage those reasons.

draft-kaufman-rtp-compatible-data suggests that the data service can be 
multiplexed with these as well, in a way that is protected by the same 
things that protect the above.

> 2. It will require one to find an RTP internal session multiplexing
> mechanism. This is likely not easy or causes a fork of RTP into two
> non-compatible versions. In addition such a mechanism needs defining
> which takes time, likely substantial time as there is quite a lot of RTP
> mechanisms to consider.

There's one, SSRC. Since we already allow two or more audio streams to 
be multiplexed using SSRC, and two or more video streams to be 
multiplexed using SSRC, it really isn't a stretch to allow one audio and 
one video to be multiplexed using SSRC when the two ends agree that this 
works for them.

> 3. We likely need a multiplexing layer anyway if we want a fall back
> over WebSockets or HTTP.

I agree that we currently don't have good agreement of how a TCP 
fallback works for RTCWEB, but it is clearly necessary.

> Thus we see this a very fragile mechanism that creates additional work
> overhead and large uncertainties in when it would be available.

Not any more fragile than RTP+RTCP+STUN on the same port already is.

> We have in our draft:
> https://datatracker.ietf.org/doc/draft-perkins-rtcweb-rtp-usage/ looked
> at two possible suggestions for explicit multiplexing. Either a thin
> shim layer primarily consisting of a identifier that through negotiation
> is bound to a context, like an particular RTP session or a datagram
> channel.

Adding a shim layer causes two problems:
1. You need to be able to run the system in a way that has the shim 
layer removed when you want to do legacy interoperability.
2. Existing protocol stacks and middleboxes cannot be reused as easily.

> The other alternative we considered was using DCCP over UDP for RTP and
> datagram. That way we got a complete port space for multiplexing
> different purposes for flows. We also got a transport framework with its
> own negotiation, congestion control, a several other features.

Same problem, no matter what the shim layer is.

I've already considered explicit multiplexing. It has lots of 
advantages. But it also fails several of the tests laid out in the 
requirements, and delays implementation and adoption.

So I think we're down to implicit multiplexing, or no multiplexing at 
all. And when we run out of IPv4 NAT mappings, you'll understand why 
multiplexing should be in there.


Matthew Kaufman


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At Google, we created a TCP/ICE-UDP layering to solve this exact problem,
using a user-mode implementation of TCP. It has some rough edges, but has
worked well in practice, and the code is freely available.

We know people will want/need a reliable messaging mechanism for p2p data.
While we can debate the details of this mechanism at length, I suspect
anything we choose will result in a better end-state than if we require
application developers to implement their own.

On Tue, Jul 19, 2011 at 7:16 AM, Henry Sinnreich
<henry.sinnreich@gmail.com>wrote:

>  +1
> Maybe it would be useful for the proponents of this and many other
> =E2=80=9Cdesirable=E2=80=9D features to think from the perspective if the=
y would have to pay
> themselves for such developments and also make the code freely available.
> Cullen at least will make his code available.
>
> Thanks, Henry
>
>
>
> On 7/18/11 2:16 AM, "Serge Lachapelle" <sergel@google.com> wrote:
>
> Hi Cullen,
>
> I agree with your push back.
>
> Focus is very important. Audio and video already present a huge challenge=
.
> (signalling, network, web developer friendly API, security, browser
> integration...)
>
> Also, I think that there is a real risk in introducing "duplicate"
> functionality in the browser as it may confuse web developers. There is a
> lot going on in HTML5...
>
> In my mind, this is "version 2.0" stuff.
>
> /Serge
>
>
> On Fri, Jul 15, 2011 at 17:51, Cullen Jennings <fluffy@cisco.com> wrote:
>
>
> I'd like to push back on the reliable service. I've yet to hear a use cas=
e
> for it that was real time. It's very hard to do a reliable real time
> protocol and we have seen zero proposal for this. For non real time data,
> just dump it in dropbox of whatever your equivalent is and don't do it pe=
er
> to peer. Unless someone has a real need, and wants to put forward a
> proposal, I don't see a need to focus energy on this right now. I'd rathe=
r
> work on the thing everyone agrees they do need which is the unreliable
> transfer.
>
> Cullen <in my individaul contribut role>
>
> On Jul 13, 2011, at 8:57 , Magnus Westerlund wrote:
>
> > Hi,
> >
> > I have reviewed the input both the last 2 weeks and the discussion back
> > in April.
> >
> > I see a strong support but with at least 2 people disagreeing to a basi=
c
> > p2p datagram functionality. The use cases are various and some just
> > state that they see it as important functionality to provide to empower
> > the web application.
> >
> > Based on this I declare a rough consensus on that we should provide a
> > non-media data service that is unreliable datagram oriented directly
> > between the peers.
> >
> > One of objections against this was lack of clear requirements for what
> > the service. The straw men requirements I included has gotten some
> > discussion. Mostly support for them, but it is clear to me that we need
> > to further develop them. I would recommend the proponents for driving
> > proposals towards meeting this functionality to further discuss the
> > requirements taking the input so far into consideration.
> >
> > When it comes to reliable data transfer between peers there has been 4-=
5
> > that wanted the functionality, 2 additional that explicitly stated they
> > where okay with it. No additional that has stated against it.
> >
> > My interpretation is that we are close to having a rough consensus for
> > reliable data service, but we have so far seen no proponent willing to
> > suggest a solution for this. I would also note that a solution is likel=
y
> > a functionality block that isn't dependent on more than the
> > signaling/negotiation and the NAT traversal and thus can be added a
> > later stage or be worked on with a completion date further into the
> > future than other pieces already.
> >
> > So for reliable data I would recommend that someone takes on the role o=
f
> > proponent and provides a draft with their perceived requirements and a
> > straw man proposal for how to solve these requirements so we have
> > something more tangible to discuss.
> >
> > Cheers
> >
> > Magnus
> >
> > On 2011-06-27 09:36, 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 abou=
t
> >> 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 wil=
l
> >> build JS libraries for this on top of a unreliable datagram service. I=
f
> >> 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 requirement=
s
> >> 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
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--000e0cd56f2692b0a904a86e3270
Content-Type: text/html; charset=UTF-8
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At Google, we created a TCP/ICE-UDP layering to solve this exact problem, u=
sing a user-mode implementation of TCP. It has some rough edges, but has wo=
rked well in practice, and the code is freely available.<div><br></div><div=
>

We know people will want/need a reliable messaging mechanism for p2p data. =
While we can debate the details of this mechanism at length, I suspect anyt=
hing we choose will result in a better end-state than if we require applica=
tion developers to implement their own.<br>

<br><div class=3D"gmail_quote">On Tue, Jul 19, 2011 at 7:16 AM, Henry Sinnr=
eich <span dir=3D"ltr">&lt;<a href=3D"mailto:henry.sinnreich@gmail.com">hen=
ry.sinnreich@gmail.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_=
quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1=
ex;">





<div>
<font face=3D"Calibri, Verdana, Helvetica, Arial"><span style=3D"font-size:=
13pt">+1<br>
Maybe it would be useful for the proponents of this and many other =E2=80=
=9Cdesirable=E2=80=9D features to think from the perspective if they would =
have to pay themselves for such developments and also make the code freely =
available.<br>


Cullen at least will make his code available.<br>
<br>
Thanks, Henry<div><div></div><div class=3D"h5"><br>
<br>
<br>
On 7/18/11 2:16 AM, &quot;Serge Lachapelle&quot; &lt;<a href=3D"http://serg=
el@google.com" target=3D"_blank">sergel@google.com</a>&gt; wrote:<br>
<br>
</div></div></span></font><div><div></div><div class=3D"h5"><blockquote><fo=
nt face=3D"Calibri, Verdana, Helvetica, Arial"><span style=3D"font-size:13p=
t">Hi Cullen,<br>
<br>
I agree with your push back.<br>
<br>
Focus is very important. Audio and video already present a huge challenge. =
(signalling, network, web developer friendly API, security, browser integra=
tion...)<br>
<br>
Also, I think that there is a real risk in introducing &quot;duplicate&quot=
; functionality in the browser as it may confuse web developers. There is a=
 lot going on in HTML5...<br>
<br>
In my mind, this is &quot;version 2.0&quot; stuff.<br>
<br>
/Serge<br>
<br>
<br>
On Fri, Jul 15, 2011 at 17:51, Cullen Jennings &lt;<a href=3D"http://fluffy=
@cisco.com" target=3D"_blank">fluffy@cisco.com</a>&gt; wrote:<br>
</span></font><blockquote><font face=3D"Calibri, Verdana, Helvetica, Arial"=
><span style=3D"font-size:13pt"><br>
I&#39;d like to push back on the reliable service. I&#39;ve yet to hear a u=
se case for it that was real time. It&#39;s very hard to do a reliable real=
 time protocol and we have seen zero proposal for this. For non real time d=
ata, just dump it in dropbox of whatever your equivalent is and don&#39;t d=
o it peer to peer. Unless someone has a real need, and wants to put forward=
 a proposal, I don&#39;t see a need to focus energy on this right now. I&#3=
9;d rather work on the thing everyone agrees they do need which is the unre=
liable transfer.<br>


<br>
Cullen &lt;in my individaul contribut role&gt;<br>
<br>
On Jul 13, 2011, at 8:57 , Magnus Westerlund wrote:<br>
<br>
&gt; Hi,<br>
&gt;<br>
&gt; I have reviewed the input both the last 2 weeks and the discussion bac=
k<br>
&gt; in April.<br>
&gt;<br>
&gt; I see a strong support but with at least 2 people disagreeing to a bas=
ic<br>
&gt; p2p datagram functionality. The use cases are various and some just<br=
>
&gt; state that they see it as important functionality to provide to empowe=
r<br>
&gt; the web application.<br>
&gt;<br>
&gt; Based on this I declare a rough consensus on that we should provide a<=
br>
&gt; non-media data service that is unreliable datagram oriented directly<b=
r>
&gt; between the peers.<br>
&gt;<br>
&gt; One of objections against this was lack of clear requirements for what=
<br>
&gt; the service. The straw men requirements I included has gotten some<br>
&gt; discussion. Mostly support for them, but it is clear to me that we nee=
d<br>
&gt; to further develop them. I would recommend the proponents for driving<=
br>
&gt; proposals towards meeting this functionality to further discuss the<br=
>
&gt; requirements taking the input so far into consideration.<br>
&gt;<br>
&gt; When it comes to reliable data transfer between peers there has been 4=
-5<br>
&gt; that wanted the functionality, 2 additional that explicitly stated the=
y<br>
&gt; where okay with it. No additional that has stated against it.<br>
&gt;<br>
&gt; My interpretation is that we are close to having a rough consensus for=
<br>
&gt; reliable data service, but we have so far seen no proponent willing to=
<br>
&gt; suggest a solution for this. I would also note that a solution is like=
ly<br>
&gt; a functionality block that isn&#39;t dependent on more than the<br>
&gt; signaling/negotiation and the NAT traversal and thus can be added a<br=
>
&gt; later stage or be worked on with a completion date further into the<br=
>
&gt; future than other pieces already.<br>
&gt;<br>
&gt; So for reliable data I would recommend that someone takes on the role =
of<br>
&gt; proponent and provides a draft with their perceived requirements and a=
<br>
&gt; straw man proposal for how to solve these requirements so we have<br>
&gt; something more tangible to discuss.<br>
&gt;<br>
&gt; Cheers<br>
&gt;<br>
&gt; Magnus<br>
&gt;<br>
&gt; On 2011-06-27 09:36, Magnus Westerlund wrote:<br>
&gt;&gt; WG,<br>
&gt;&gt;<br>
&gt;&gt; At the interim it was planned to have a bit discussion on the data=
gram<br>
&gt;&gt; service for RTCWEB. The first question to try to resolve if there<=
br>
&gt;&gt; is consensus for including some form of non real-time media (i.e. =
not<br>
&gt;&gt; audio, video) service between peers. This is a bit tangled with th=
e<br>
&gt;&gt; actual requirements and use cases. But there was views both for it=
 and<br>
&gt;&gt; against it on the mailing list. So lets continue and try to come t=
o a<br>
&gt;&gt; conclusion on this discussion.<br>
&gt;&gt;<br>
&gt;&gt; The use cases mentioned on the mailing list are:<br>
&gt;&gt;<br>
&gt;&gt; - Dynamic meta data for Conference and other real-time services<br=
>
&gt;&gt;<br>
&gt;&gt; - Gaming data with low latency requirements<br>
&gt;&gt;<br>
&gt;&gt; Does anyone like to add additional use cases?<br>
&gt;&gt;<br>
&gt;&gt; Based on my personal understanding this points to primarily have t=
he<br>
&gt;&gt; RTCWEB provide a unreliable datagram service. This clearly needs<b=
r>
&gt;&gt; additional requirements to be secure and safe to deploy, but more =
about<br>
&gt;&gt; this below. I still like to ask the WG here a question.<br>
&gt;&gt;<br>
&gt;&gt; Are you supporting the inclusion of a unreliable datagram service<=
br>
&gt;&gt; directly between peers? Please provide your view and any additiona=
l<br>
&gt;&gt; statements of motivation that you desire to provide.<br>
&gt;&gt;<br>
&gt;&gt; Secondly, there is a question if there needs to have something tha=
t<br>
&gt;&gt; provides reliable message (of arbitrary size) or byte stream orien=
ted<br>
&gt;&gt; data transport between the peers. I personally foresee that people=
 will<br>
&gt;&gt; build JS libraries for this on top of a unreliable datagram servic=
e. If<br>
&gt;&gt; you desire reliable data service as part of the standardized solut=
ion<br>
&gt;&gt; please provide motivation and use case and requirements.<br>
&gt;&gt;<br>
&gt;&gt; I also want to take a stab on what I personally see as the require=
ments<br>
&gt;&gt; that exist on unreliable datagram service in the context of RTCWEB=
.<br>
&gt;&gt;<br>
&gt;&gt; - Unreliable data transmission<br>
&gt;&gt; - Datagram oriented<br>
&gt;&gt; =C2=A0 * Size limited by MTU<br>
&gt;&gt; =C2=A0 =C2=A0 - Path MTU discovery needed<br>
&gt;&gt; =C2=A0 * Fragmentation by the application<br>
&gt;&gt; - Low latency, i.e. Peer to Peer preferable<br>
&gt;&gt; - Congestion Controlled, to be<br>
&gt;&gt; =C2=A0 * Network friendly<br>
&gt;&gt; =C2=A0 * Not become a Denial of Service tool<br>
&gt;&gt; - Security<br>
&gt;&gt; =C2=A0* Confidentiality<br>
&gt;&gt; =C2=A0* Integrity Protected<br>
&gt;&gt; =C2=A0* Source Authenticated (at least bound to the signalling pee=
r)<br>
&gt;&gt; =C2=A0* Ensure consent to receive data<br>
&gt;&gt;<br>
&gt;&gt; Please debate the above. This is an attempt to ensure that we can<=
br>
&gt;&gt; establish WG consensus on both data service and any requirements.<=
br>
&gt;&gt;<br>
&gt;&gt; cheers<br>
&gt;&gt;<br>
&gt;&gt; Magnus Westerlund<br>
&gt;&gt;<br>
&gt;&gt; ------------------------------------------------------------------=
----<br>
&gt;&gt; Multimedia Technologies, Ericsson Research EAB/TVM<br>
&gt;&gt; ------------------------------------------------------------------=
----<br>
&gt;&gt; Ericsson AB =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0| Phone =C2=A0<br>
</span></font></blockquote></blockquote>
</div></div></div>


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

--000e0cd56f2692b0a904a86e3270--

From randell1@jesup.org  Tue Jul 19 09:38:23 2011
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On 7/19/2011 11:58 AM, Justin Uberti wrote:
> At Google, we created a TCP/ICE-UDP layering to solve this exact 
> problem, using a user-mode implementation of TCP. It has some rough 
> edges, but has worked well in practice, and the code is freely available.

Right; pretty much what I'd anticipate, and working open code is good.  
SCTP as mentioned is a good choice too, and there exist (it appears) 
SCTP-over-UDP implementations, which may ease things.  Also the 
TCP-over-UDP library I pointed to.

> We know people will want/need a reliable messaging mechanism for p2p 
> data. While we can debate the details of this mechanism at length, I 
> suspect anything we choose will result in a better end-state than if 
> we require application developers to implement their own.

As is obvious from my other messages, I agree wholeheartedly.

> On Tue, Jul 19, 2011 at 7:16 AM, Henry Sinnreich 
> <henry.sinnreich@gmail.com <mailto:henry.sinnreich@gmail.com>> wrote:
>
>     +1
>     Maybe it would be useful for the proponents of this and many other
>     “desirable” features to think from the perspective if they would
>     have to pay themselves for such developments and also make the
>     code freely available.
>     Cullen at least will make his code available.
>

See above.  No one here wants to develop *another* reliable protocol if 
we can avoid it.  I understand your reluctance.

The goal is to produce something that will meet the needs and get used.  
If we don't meet the
needs of the prospective users it's either an total waste of time (if 
they ignore this effort and stick
with Flash/plugins/etc) or we get all sorts of 
hack/bad/problematic/incompatible implementations
by individual developers using the UDP (or RTP!) streams we open as a 
base transport.

To your point, another way this effort could fail is by biting off too 
much or taking too long to
converge, which is a real danger.  That's one reason to leverage 
existing protocols.

-- 
Randell Jesup
randell-ietf@jesup.org


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On 7/19/2011 7:28 AM, Magnus Westerlund wrote:
> a) MUST be sent as Individual flows for each component.
>
> b) MUST be multiplexed into a single transport flow.
>
> c) SHOULD be multiplexed into a single transport flow, but the RTCWEB
> peer MUST be able to send them as individual flows.
>
> I would love if people can indicate their choice or preferences.

C. But I'm not sure that the "must" needs to be anything more than "you 
just make two peer connection objects, one for each". (In other words, 
any API that allows for multiplexing almost certainly allows for 
non-multiplexing)... this assumes that the multiplexing is implicit, of 
course. With explicit multiplexing it is even harder... the API needs to 
support turning on and off the multiplexing shim as well, something I'm 
not in favor of.

Matthew Kaufman

From emil@sip-communicator.org  Tue Jul 19 09:47:34 2011
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=D0=9D=D0=B0 19.07.11 16:28, Magnus Westerlund =D0=BD=D0=B0=D0=BF=D0=B8=D1=
=81=D0=B0:
<snip>

> a) MUST be sent as Individual flows for each component.

+1

> b) MUST be multiplexed into a single transport flow.
>=20
> c) SHOULD be multiplexed into a single transport flow, but the RTCWEB
> peer MUST be able to send them as individual flows.
>=20
> I would love if people can indicate their choice or preferences.
>=20
> I personally prefer A as it it is simplest in all aspect except the NAT=

> traversal.

It definitely simplifies it on paper - it would be far easier to draw an
ICE negotiation diagram that only deals with a single stream component.

I am not sure however to what extent this would actually simplify
implementations. They would still need to handle multiple candidates and
pacing and such ... and of course they'd also need to implement  a
de-multiplexing layer.

<snip>
> - People are used to the concept
> - It minimizes the difference to legacy.

Personally I consider this as the most important part.

Cheers,
Emil

>=20
> Thus it is the quickest road to define something with the least formal
> push back and concern over maturity of any solution.
>=20
> The downside with B and C is that we do have to solve the multiplexing
> and get an agreement that gets through all the hurdles.
>=20
> Of these two opens I do prefer C.  Although it results in the extra
> complexities of having both alternatives, it will give us both a
> fallback, flow based QoS and better legacy support.
>=20
> Now it is your time to make your opinion heard!
>=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 emil@sip-communicator.org  Tue Jul 19 09:55:23 2011
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=D0=9D=D0=B0 19.07.11 17:20, Justin Uberti =D0=BD=D0=B0=D0=BF=D0=B8=D1=81=
=D0=B0:
> =20
>=20
>     - It is the what the implementation that exist mostly do
>     - Signaling protocols that exist support it, no extra functionality=

>     - People are used to the concept
>     - It minimizes the difference to legacy.
>=20
>=20
> I agree that existing implementations will need to change to support
> this. However, is there a large enough base of deployed endpoints that
> support a) multiple RTP sessions and b) the other RTCWEB constrains
> (ICE, etc) to make this a significant concern?=20

You are referring to SIP/XMPP clients using RTP and ICE, right? Well,
given that this has been IETF's default way of handling RTC for the past
few years I would assume that there are a bunch yes.

Besides, if we ever make the ICE part optional (like XEP-0177 does for
XMPP) then we'd have media-layer interoperability with any VoIP client
out there and this would definitely be a great win.

Emil

--=20
http://jitsi.org


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=D0=9D=D0=B0 19.07.11 18:37, Randell Jesup =D0=BD=D0=B0=D0=BF=D0=B8=D1=81=
=D0=B0:
> On 7/19/2011 11:58 AM, Justin Uberti wrote:
>> At Google, we created a TCP/ICE-UDP layering to solve this exact=20
>> problem, using a user-mode implementation of TCP. It has some rough=20
>> edges, but has worked well in practice, and the code is freely availab=
le.
>=20
> Right; pretty much what I'd anticipate

+1 most definitely. Again, I have yet to see a massively used RTC
application that doesn't have file transfer. So if this is not part of
RTCWEB we will basically be asking developers to reimplement Pseudo TCP
in JavaScript.

Emil

>, and working open code is good. =20
> SCTP as mentioned is a good choice too, and there exist (it appears)=20
> SCTP-over-UDP implementations, which may ease things.  Also the=20
> TCP-over-UDP library I pointed to.
>=20
>> We know people will want/need a reliable messaging mechanism for p2p=20
>> data. While we can debate the details of this mechanism at length, I=20
>> suspect anything we choose will result in a better end-state than if=20
>> we require application developers to implement their own.
>=20
> As is obvious from my other messages, I agree wholeheartedly.
>=20
>> On Tue, Jul 19, 2011 at 7:16 AM, Henry Sinnreich=20
>> <henry.sinnreich@gmail.com <mailto:henry.sinnreich@gmail.com>> wrote:
>>
>>     +1
>>     Maybe it would be useful for the proponents of this and many other=

>>     =E2=80=9Cdesirable=E2=80=9D features to think from the perspective=
 if they would
>>     have to pay themselves for such developments and also make the
>>     code freely available.
>>     Cullen at least will make his code available.
>>
>=20
> See above.  No one here wants to develop *another* reliable protocol if=
=20
> we can avoid it.  I understand your reluctance.
>=20
> The goal is to produce something that will meet the needs and get used.=
 =20
> If we don't meet the
> needs of the prospective users it's either an total waste of time (if=20
> they ignore this effort and stick
> with Flash/plugins/etc) or we get all sorts of=20
> hack/bad/problematic/incompatible implementations
> by individual developers using the UDP (or RTP!) streams we open as a=20
> base transport.
>=20
> To your point, another way this effort could fail is by biting off too =

> much or taking too long to
> converge, which is a real danger.  That's one reason to leverage=20
> existing protocols.
>=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


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Subject: Re: [rtcweb] To multiplex or not!
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On 7/19/2011 10:28 AM, Magnus Westerlund wrote:
> The traffic between two RTCWEB peers from the various components, such
> as RTP sessions, datagram service:
>
> a) MUST be sent as Individual flows for each component.
>
> b) MUST be multiplexed into a single transport flow.
>
> c) SHOULD be multiplexed into a single transport flow, but the RTCWEB
> peer MUST be able to send them as individual flows.

C).   I understand the appeal of a).  However, we already have minor 
issues with handling
congestion across multiple streams - i.e. congestion notification goes 
to one flow, but not
generally to the other flows that make up the total traffic between 
these two points.  This
leads to sub-optimal response to congestion, and as the number of flows 
increases this
would get worse.  Now, this really isn't mandated by the 
multiplex/non-multiplex nature; it's
just easier and more obvious when they're multiplexed.

The other big issue is as you add more than a single or two streams, the 
overhead and
side-issues around ICE/etc get more ... interesting.  In particular you 
can get (I assume) cases
where different streams that are meant to be synchronized and follow the 
same route
go over different paths/interfaces.

If we only had to deal with 1 audio stream, or 1 audio and 1 video 
stream, the balance would be
different.  But in the world this spec will work in I strongly believe 
more than 2 streams
will often exist.  (Witness Google's "Huddle").

-- 
Randell Jesup
randell-ietf@jesup.org


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--000e0cd39bee2cbe0904a86f2f2b
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There are many very established ways to transfer files from a web browser.
Dropbox, Google Docs, yousendit, mobileme, not counting the media specific
ways, such as Flickr, Spotify, Youtube, etc... All offering great APIs for
uploads, ACLs, etc...

Seems like re-inventing the wheel.

/Serge

On Tue, Jul 19, 2011 at 19:02, Emil Ivov <emcho@jitsi.org> wrote:

>
>
> =D0=9D=D0=B0 19.07.11 18:37, Randell Jesup =D0=BD=D0=B0=D0=BF=D0=B8=D1=81=
=D0=B0:
> > On 7/19/2011 11:58 AM, Justin Uberti wrote:
> >> At Google, we created a TCP/ICE-UDP layering to solve this exact
> >> problem, using a user-mode implementation of TCP. It has some rough
> >> edges, but has worked well in practice, and the code is freely
> available.
> >
> > Right; pretty much what I'd anticipate
>
> +1 most definitely. Again, I have yet to see a massively used RTC
> application that doesn't have file transfer. So if this is not part of
> RTCWEB we will basically be asking developers to reimplement Pseudo TCP
> in JavaScript.
>
> Emil
>
> >, and working open code is good.
> > SCTP as mentioned is a good choice too, and there exist (it appears)
> > SCTP-over-UDP implementations, which may ease things.  Also the
> > TCP-over-UDP library I pointed to.
> >
> >> We know people will want/need a reliable messaging mechanism for p2p
> >> data. While we can debate the details of this mechanism at length, I
> >> suspect anything we choose will result in a better end-state than if
> >> we require application developers to implement their own.
> >
> > As is obvious from my other messages, I agree wholeheartedly.
> >
> >> On Tue, Jul 19, 2011 at 7:16 AM, Henry Sinnreich
> >> <henry.sinnreich@gmail.com <mailto:henry.sinnreich@gmail.com>> wrote:
> >>
> >>     +1
> >>     Maybe it would be useful for the proponents of this and many other
> >>     =E2=80=9Cdesirable=E2=80=9D features to think from the perspective=
 if they would
> >>     have to pay themselves for such developments and also make the
> >>     code freely available.
> >>     Cullen at least will make his code available.
> >>
> >
> > See above.  No one here wants to develop *another* reliable protocol if
> > we can avoid it.  I understand your reluctance.
> >
> > The goal is to produce something that will meet the needs and get used.
> > If we don't meet the
> > needs of the prospective users it's either an total waste of time (if
> > they ignore this effort and stick
> > with Flash/plugins/etc) or we get all sorts of
> > hack/bad/problematic/incompatible implementations
> > by individual developers using the UDP (or RTP!) streams we open as a
> > base transport.
> >
> > To your point, another way this effort could fail is by biting off too
> > much or taking too long to
> > converge, which is a real danger.  That's one reason to leverage
> > existing protocols.
> >
>
> --
> 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
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>



--=20
Serge Lachapelle | Product Manager | sergel@google.com | +46 732 01 22 32
Google Sweden AB | Kungsbron 2, SE-111 22 Stockholm | Org. nr. 556656-6880

Apparently, this footer is required in Europe. Apologies. This email may be
confidential or privileged.  If you received this communication by mistake,
please don't forward it to anyone else,please erase all copies and
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 Thanks.

--000e0cd39bee2cbe0904a86f2f2b
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

There are many very established ways to transfer files from a web browser. =
Dropbox, Google Docs, yousendit, mobileme, not counting the media specific =
ways, such as Flickr, Spotify, Youtube, etc... All offering great APIs for =
uploads, ACLs, etc...<div>

<br></div><div>Seems like re-inventing the wheel.</div>
<div><br><div>/Serge<br><div><br><div class=3D"gmail_quote">On Tue, Jul 19,=
 2011 at 19:02, Emil Ivov <span dir=3D"ltr">&lt;<a href=3D"mailto:emcho@jit=
si.org" target=3D"_blank">emcho@jitsi.org</a>&gt;</span> wrote:<br><blockqu=
ote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc s=
olid;padding-left:1ex">



<br>
<br>
=D0=9D=D0=B0 19.07.11 18:37, Randell Jesup =D0=BD=D0=B0=D0=BF=D0=B8=D1=81=
=D0=B0:<br>
<div>&gt; On 7/19/2011 11:58 AM, Justin Uberti wrote:<br>
&gt;&gt; At Google, we created a TCP/ICE-UDP layering to solve this exact<b=
r>
&gt;&gt; problem, using a user-mode implementation of TCP. It has some roug=
h<br>
&gt;&gt; edges, but has worked well in practice, and the code is freely ava=
ilable.<br>
&gt;<br>
&gt; Right; pretty much what I&#39;d anticipate<br>
<br>
</div>+1 most definitely. Again, I have yet to see a massively used RTC<br>
application that doesn&#39;t have file transfer. So if this is not part of<=
br>
RTCWEB we will basically be asking developers to reimplement Pseudo TCP<br>
in JavaScript.<br>
<br>
Emil<br>
<div><div></div><div><br>
&gt;, and working open code is good.<br>
&gt; SCTP as mentioned is a good choice too, and there exist (it appears)<b=
r>
&gt; SCTP-over-UDP implementations, which may ease things. =C2=A0Also the<b=
r>
&gt; TCP-over-UDP library I pointed to.<br>
&gt;<br>
&gt;&gt; We know people will want/need a reliable messaging mechanism for p=
2p<br>
&gt;&gt; data. While we can debate the details of this mechanism at length,=
 I<br>
&gt;&gt; suspect anything we choose will result in a better end-state than =
if<br>
&gt;&gt; we require application developers to implement their own.<br>
&gt;<br>
&gt; As is obvious from my other messages, I agree wholeheartedly.<br>
&gt;<br>
&gt;&gt; On Tue, Jul 19, 2011 at 7:16 AM, Henry Sinnreich<br>
&gt;&gt; &lt;<a href=3D"mailto:henry.sinnreich@gmail.com" target=3D"_blank"=
>henry.sinnreich@gmail.com</a> &lt;mailto:<a href=3D"mailto:henry.sinnreich=
@gmail.com" target=3D"_blank">henry.sinnreich@gmail.com</a>&gt;&gt; wrote:<=
br>



&gt;&gt;<br>
&gt;&gt; =C2=A0 =C2=A0 +1<br>
&gt;&gt; =C2=A0 =C2=A0 Maybe it would be useful for the proponents of this =
and many other<br>
&gt;&gt; =C2=A0 =C2=A0 =E2=80=9Cdesirable=E2=80=9D features to think from t=
he perspective if they would<br>
&gt;&gt; =C2=A0 =C2=A0 have to pay themselves for such developments and als=
o make the<br>
&gt;&gt; =C2=A0 =C2=A0 code freely available.<br>
&gt;&gt; =C2=A0 =C2=A0 Cullen at least will make his code available.<br>
&gt;&gt;<br>
&gt;<br>
&gt; See above. =C2=A0No one here wants to develop *another* reliable proto=
col if<br>
&gt; we can avoid it. =C2=A0I understand your reluctance.<br>
&gt;<br>
&gt; The goal is to produce something that will meet the needs and get used=
.<br>
&gt; If we don&#39;t meet the<br>
&gt; needs of the prospective users it&#39;s either an total waste of time =
(if<br>
&gt; they ignore this effort and stick<br>
&gt; with Flash/plugins/etc) or we get all sorts of<br>
&gt; hack/bad/problematic/incompatible implementations<br>
&gt; by individual developers using the UDP (or RTP!) streams we open as a<=
br>
&gt; base transport.<br>
&gt;<br>
&gt; To your point, another way this effort could fail is by biting off too=
<br>
&gt; much or taking too long to<br>
&gt; converge, which is a real danger. =C2=A0That&#39;s one reason to lever=
age<br>
&gt; existing protocols.<br>
&gt;<br>
<br>
--<br>
</div></div><font color=3D"#888888">Emil Ivov, Ph.D. =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 67000 Strasbourg,<b=
r>
Project Lead =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0=
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 France<br>
Jitsi<br>
<a href=3D"mailto:emcho@jitsi.org" target=3D"_blank">emcho@jitsi.org</a> =
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0PHONE: <a href=3D"tel:%2B33.1.77.62.43.30" value=3D"+33177624330"=
 target=3D"_blank">+33.1.77.62.43.30</a><br>
<a href=3D"http://jitsi.org" target=3D"_blank">http://jitsi.org</a> =C2=A0 =
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 FAX: =
=C2=A0 <a href=3D"tel:%2B33.1.77.62.47.31" value=3D"+33177624731" target=3D=
"_blank">+33.1.77.62.47.31</a><br>
</font><div><div></div><div><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br><br clear=3D"all"><br>-- <br><div><div s=
tyle=3D"line-height:1.5em;padding-top:10px;margin-top:10px;color:rgb(85, 85=
, 85);font-family:sans-serif;font-size:small">
<span style=3D"border-top-width:2px;border-right-width:0px;border-bottom-wi=
dth:0px;border-left-width:0px;border-top-style:solid;border-right-style:sol=
id;border-bottom-style:solid;border-left-style:solid;border-top-color:rgb(2=
13, 15, 37);border-right-color:rgb(213, 15, 37);border-bottom-color:rgb(213=
, 15, 37);border-left-color:rgb(213, 15, 37);padding-top:2px;margin-top:2px=
">Serge Lachapelle=C2=A0|</span><span style=3D"border-top-width:2px;border-=
right-width:0px;border-bottom-width:0px;border-left-width:0px;border-top-st=
yle:solid;border-right-style:solid;border-bottom-style:solid;border-left-st=
yle:solid;border-top-color:rgb(51, 105, 232);border-right-color:rgb(51, 105=
, 232);border-bottom-color:rgb(51, 105, 232);border-left-color:rgb(51, 105,=
 232);padding-top:2px;margin-top:2px">=C2=A0Product Manager=C2=A0|</span><s=
pan style=3D"border-top-width:2px;border-right-width:0px;border-bottom-widt=
h:0px;border-left-width:0px;border-top-style:solid;border-right-style:solid=
;border-bottom-style:solid;border-left-style:solid;border-top-color:rgb(0, =
153, 57);border-right-color:rgb(0, 153, 57);border-bottom-color:rgb(0, 153,=
 57);border-left-color:rgb(0, 153, 57);padding-top:2px;margin-top:2px">=C2=
=A0<a href=3D"mailto:sergel@google.com" target=3D"_blank">sergel@google.com=
</a>=C2=A0|</span><span style=3D"border-top-width:2px;border-right-width:0p=
x;border-bottom-width:0px;border-left-width:0px;border-top-style:solid;bord=
er-right-style:solid;border-bottom-style:solid;border-left-style:solid;bord=
er-top-color:rgb(238, 178, 17);border-right-color:rgb(238, 178, 17);border-=
bottom-color:rgb(238, 178, 17);border-left-color:rgb(238, 178, 17);padding-=
top:2px;margin-top:2px">=C2=A0<a href=3D"tel:%2B46%20732%2001%2022%2032" va=
lue=3D"+46732012232" target=3D"_blank">+46 732 01 22 32</a></span></div>



</div><font color=3D"#999999" size=3D"1">Google Sweden AB | Kungsbron 2, SE=
-111 22 Stockholm | Org. nr. 556656-6880=C2=A0</font><br><br><font color=3D=
"#999999" size=3D"1">Apparently, this footer is required in Europe. Apologi=
es. This email may be confidential or privileged. =C2=A0If you received thi=
s communication by mistake, please don&#39;t forward it to anyone else,plea=
se erase all copies and attachments, and please let me know that it went to=
 the wrong person. =C2=A0Thanks.</font><br>




</div></div></div>

--000e0cd39bee2cbe0904a86f2f2b--

From emil@sip-communicator.org  Tue Jul 19 10:21:38 2011
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Date: Tue, 19 Jul 2011 19:21:29 +0200
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] realiable data service
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=D0=9D=D0=B0 19.07.11 19:09, Serge Lachapelle =D0=BD=D0=B0=D0=BF=D0=B8=D1=
=81=D0=B0:
> There are many very established ways to transfer files from a web
> browser. Dropbox, Google Docs, yousendit, mobileme, not counting the
> media specific ways, such as Flickr, Spotify, Youtube, etc... All
> offering great APIs for uploads, ACLs, etc...

I am sorry, I should have been more specific when referring to the file
transfer use case. What I meant was the possibility to let users send
files directly to each other.

All of the services you quote above have a public sharing or a backup
aspect that requires files to be stored on a server. Sending files
however, does not have this constraint so forcing the flow to go through
a public relay is only unnecessary overhead.

> Seems like re-inventing the wheel.

Yeah ... but people do that every now and then ;)

http://code.google.com/p/libjingle/source/browse/trunk/talk/p2p/base/pseu=
dotcp.h

Emil


>=20
> /Serge
>=20
> On Tue, Jul 19, 2011 at 19:02, Emil Ivov <emcho@jitsi.org
> <mailto:emcho@jitsi.org>> wrote:
>=20
>=20
>=20
>     =D0=9D=D0=B0 19.07.11 18:37, Randell Jesup =D0=BD=D0=B0=D0=BF=D0=B8=
=D1=81=D0=B0:
>     > On 7/19/2011 11:58 AM, Justin Uberti wrote:
>     >> At Google, we created a TCP/ICE-UDP layering to solve this exact=

>     >> problem, using a user-mode implementation of TCP. It has some ro=
ugh
>     >> edges, but has worked well in practice, and the code is freely
>     available.
>     >
>     > Right; pretty much what I'd anticipate
>=20
>     +1 most definitely. Again, I have yet to see a massively used RTC
>     application that doesn't have file transfer. So if this is not part=
 of
>     RTCWEB we will basically be asking developers to reimplement Pseudo=
 TCP
>     in JavaScript.
>=20
>     Emil
>=20
>     >, and working open code is good.
>     > SCTP as mentioned is a good choice too, and there exist (it appea=
rs)
>     > SCTP-over-UDP implementations, which may ease things.  Also the
>     > TCP-over-UDP library I pointed to.
>     >
>     >> We know people will want/need a reliable messaging mechanism for=
 p2p
>     >> data. While we can debate the details of this mechanism at lengt=
h, I
>     >> suspect anything we choose will result in a better end-state tha=
n if
>     >> we require application developers to implement their own.
>     >
>     > As is obvious from my other messages, I agree wholeheartedly.
>     >
>     >> On Tue, Jul 19, 2011 at 7:16 AM, Henry Sinnreich
>     >> <henry.sinnreich@gmail.com <mailto:henry.sinnreich@gmail.com>
>     <mailto:henry.sinnreich@gmail.com
>     <mailto:henry.sinnreich@gmail.com>>> wrote:
>     >>
>     >>     +1
>     >>     Maybe it would be useful for the proponents of this and many=

>     other
>     >>     =E2=80=9Cdesirable=E2=80=9D features to think from the persp=
ective if they would
>     >>     have to pay themselves for such developments and also make t=
he
>     >>     code freely available.
>     >>     Cullen at least will make his code available.
>     >>
>     >
>     > See above.  No one here wants to develop *another* reliable
>     protocol if
>     > we can avoid it.  I understand your reluctance.
>     >
>     > The goal is to produce something that will meet the needs and get=

>     used.
>     > If we don't meet the
>     > needs of the prospective users it's either an total waste of time=
 (if
>     > they ignore this effort and stick
>     > with Flash/plugins/etc) or we get all sorts of
>     > hack/bad/problematic/incompatible implementations
>     > by individual developers using the UDP (or RTP!) streams we open =
as a
>     > base transport.
>     >
>     > To your point, another way this effort could fail is by biting of=
f too
>     > much or taking too long to
>     > converge, which is a real danger.  That's one reason to leverage
>     > existing protocols.
>     >
>=20
>     --
>     Emil Ivov, Ph.D.                       67000 Strasbourg,
>     Project Lead                           France
>     Jitsi
>     emcho@jitsi.org <mailto:emcho@jitsi.org>                     =20
>      PHONE: +33.1.77.62.43.30 <tel:%2B33.1.77.62.43.30>
>     http://jitsi.org                       FAX:   +33.1.77.62.47.31
>     <tel:%2B33.1.77.62.47.31>
>=20
>     _______________________________________________
>     rtcweb mailing list
>     rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>     https://www.ietf.org/mailman/listinfo/rtcweb
>=20
>=20
>=20
>=20
> --=20
> Serge Lachapelle | Product Manager | sergel@google.com
> <mailto:sergel@google.com> | +46 732 01 22 32
> <tel:%2B46%20732%2001%2022%2032>
> Google Sweden AB | Kungsbron 2, SE-111 22 Stockholm | Org. nr. 556656-6=
880=20
>=20
> Apparently, this footer is required in Europe. Apologies. This email ma=
y
> be confidential or privileged.  If you received this communication by
> mistake, please don't forward it to anyone else,please erase all copies=

> and attachments, and please let me know that it went to the wrong
> person.  Thanks.

--=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 dzonatas@gmail.com  Tue Jul 19 10:37:07 2011
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Subject: Re: [rtcweb] realiable data service
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I suspect that TCP becomes the new semantic word for all browsers, as 
otherwise the movement of TCP to user-mode is already evident, and 
redundantly done (as suggested). TCP was the default way for developers 
to find stream-modes. Portability, like this, misrepresents needs beyond 
beginner layers.

Despite change, UDP over HTTP over S/MIME appears proper and quick 
enough (i.e. "can harden"). Path of least resistance... SSRC already 
offers that bit of available change and overlap. The question for 
standard besides mux is only expect RTP first or use OPTIONS to 
downgrade HTTP. We've used 1xx to bypass HTTP/TCP layers before, so in 
fairness I don't think this is correct for rtcweb in regards to states.

Implementation at software level seems futile if that concept is really 
that hard. We need non-encrypted unshrunken headers. Either that or 402 
out of no-where simply bytes for energy alone... a security disaster. I 
usually stay quiet about that, yet please note that harvesters are this 
artificially intelligent now. =)

On 07/19/2011 08:58 AM, Justin Uberti wrote:
> At Google, we created a TCP/ICE-UDP layering to solve this exact 
> problem, using a user-mode implementation of TCP. It has some rough 
> edges, but has worked well in practice, and the code is freely available.
>
> We know people will want/need a reliable messaging mechanism for p2p 
> data. While we can debate the details of this mechanism at length, I 
> suspect anything we choose will result in a better end-state than if 
> we require application developers to implement their own.
>
> On Tue, Jul 19, 2011 at 7:16 AM, Henry Sinnreich 
> <henry.sinnreich@gmail.com <mailto:henry.sinnreich@gmail.com>> wrote:
>
>     +1
>     Maybe it would be useful for the proponents of this and many other
>     “desirable” features to think from the perspective if they would
>     have to pay themselves for such developments and also make the
>     code freely available.
>     Cullen at least will make his code available.
>
>     Thanks, Henry
>
>
>
>     On 7/18/11 2:16 AM, "Serge Lachapelle" <sergel@google.com
>     <http://sergel@google.com>> wrote:
>
>         Hi Cullen,
>
>         I agree with your push back.
>
>         Focus is very important. Audio and video already present a
>         huge challenge. (signalling, network, web developer friendly
>         API, security, browser integration...)
>
>         Also, I think that there is a real risk in introducing
>         "duplicate" functionality in the browser as it may confuse web
>         developers. There is a lot going on in HTML5...
>
>         In my mind, this is "version 2.0" stuff.
>
>         /Serge
>
>
>         On Fri, Jul 15, 2011 at 17:51, Cullen Jennings
>         <fluffy@cisco.com <http://fluffy@cisco.com>> wrote:
>
>
>             I'd like to push back on the reliable service. I've yet to
>             hear a use case for it that was real time. It's very hard
>             to do a reliable real time protocol and we have seen zero
>             proposal for this. For non real time data, just dump it in
>             dropbox of whatever your equivalent is and don't do it
>             peer to peer. Unless someone has a real need, and wants to
>             put forward a proposal, I don't see a need to focus energy
>             on this right now. I'd rather work on the thing everyone
>             agrees they do need which is the unreliable transfer.
>
>             Cullen <in my individaul contribut role>
>
>             On Jul 13, 2011, at 8:57 , Magnus Westerlund wrote:
>
>             > Hi,
>             >
>             > I have reviewed the input both the last 2 weeks and the
>             discussion back
>             > in April.
>             >
>             > I see a strong support but with at least 2 people
>             disagreeing to a basic
>             > p2p datagram functionality. The use cases are various and
>             some just
>             > state that they see it as important functionality to
>             provide to empower
>             > the web application.
>             >
>             > Based on this I declare a rough consensus on that we
>             should provide a
>             > non-media data service that is unreliable datagram
>             oriented directly
>             > between the peers.
>             >
>             > One of objections against this was lack of clear
>             requirements for what
>             > the service. The straw men requirements I included has
>             gotten some
>             > discussion. Mostly support for them, but it is clear to
>             me that we need
>             > to further develop them. I would recommend the proponents
>             for driving
>             > proposals towards meeting this functionality to further
>             discuss the
>             > requirements taking the input so far into consideration.
>             >
>             > When it comes to reliable data transfer between peers
>             there has been 4-5
>             > that wanted the functionality, 2 additional that
>             explicitly stated they
>             > where okay with it. No additional that has stated against it.
>             >
>             > My interpretation is that we are close to having a rough
>             consensus for
>             > reliable data service, but we have so far seen no
>             proponent willing to
>             > suggest a solution for this. I would also note that a
>             solution is likely
>             > a functionality block that isn't dependent on more than the
>             > signaling/negotiation and the NAT traversal and thus can
>             be added a
>             > later stage or be worked on with a completion date
>             further into the
>             > future than other pieces already.
>             >
>             > So for reliable data I would recommend that someone takes
>             on the role of
>             > proponent and provides a draft with their perceived
>             requirements and a
>             > straw man proposal for how to solve these requirements so
>             we have
>             > something more tangible to discuss.
>             >
>             > Cheers
>             >
>             > Magnus
>             >
>             > On 2011-06-27 09:36, 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
>
>
>     _______________________________________________
>     rtcweb mailing list
>     rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>     https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>    


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On 07/19/2011 10:21 AM, Emil Ivov wrote:
>
>
>> Seems like re-inventing the wheel.
>>      
> Yeah ... but people do that every now and then ;)
>
> http://code.google.com/p/libjingle/source/browse/trunk/talk/p2p/base/pseudotcp.h
>
> Emil
>    

Exactly why mainframes and clouds go hand and hand now: TCP is not 
sustainable.

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On 07/19/2011 09:37 AM, Randell Jesup wrote:
>
> To your point, another way this effort could fail is by biting off too 
> much or taking too long to
> converge, which is a real danger.  That's one reason to leverage 
> existing protocols.
>

Baby-safe is what I keep in mind every day. I'm not perfectr, neither is 
danger.

Maybe some systems want optimizational values out of their protocols 
because they see no need for the consumption of computational energy. 
All the riches, what the world want at even given moment. Practical reality.


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Subject: Re: [rtcweb] To multiplex or not!
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I'd like to see it that devices MUST support multiplexing and I'd =
probably be in favor of also MUST support non multiplexed flows. The =
multiplexing is a big advantage for any device that needs to deal with =
NATs so I suspect many "legacy" devices would be moving to this even if =
there was no need to deal with the browser like devices. I also think =
that allowing this type of multiplexing will make firewall traversal =
easier where a device can configure a firewall to open just one =
specified port and use it for RTP.=20

The number places where RSVP is usable does not really overlap with =
places one uses ICE so it does not bother me in the slightest that RSVP =
won't work with this. It's pretty easy to imagine an extension to RSVP =
to support reservations based on SSRC inside the flow. Diff Serv based =
qos will work which is what more commonly used in situations where we =
are using NATs.=20

The primary reason I want multiplexing is the time to set up all the ICE =
connections. The fact of the matter is that NATs limit how quickly you =
can create new flows. If all you need is something simple with just 2 =
flows, one for audio and one for video, this is workable. But as soon as =
you move to trying to have a transition strategy from V4 to V6 and =
multiple person video or even just multi screen video, the user =
experience goes downhill rapidly.=20

Much of the IETF thinking around ICE was done on the assumption that ICE =
could be done before ringing started or during ringing but the =
deployments I am seeing are moving more towards not doing ICE until =
after a call is answered. There are many reasons for this but one of =
them is that if Alice calls Bob, and Bob is on a mobile internet device, =
you don't want to reveal Bob's location to Alice before Bob even decides =
to answer the call. Given an IP address more or less reveals location =
these days that a bit of issues to do ICE before answering. This puts =
more pressure on being able to do ICE quickly for a good user =
experience.=20

Cullen <with my individual contributor hat on>

On Jul 19, 2011, at 7:28 , Magnus Westerlund wrote:

> Hi,
>=20
> This email is as an individual contributor.
>=20
> I want to get started on the discussion of the Multiplexing of the
> various protocols over single lower layer transport flow, such as a =
UDP
> flow. I will attempt to split up the questions into different emails.
>=20
> The first question I think is reasonably easy to get answered, but I
> think it is time we determine if my belief in the answer is correct or =
not.
>=20
> The traffic between two RTCWEB peers from the various components, such
> as RTP sessions, datagram service:
>=20
> a) MUST be sent as Individual flows for each component.
>=20
> b) MUST be multiplexed into a single transport flow.
>=20
> c) SHOULD be multiplexed into a single transport flow, but the RTCWEB
> peer MUST be able to send them as individual flows.
>=20
> I would love if people can indicate their choice or preferences.
>=20
> I personally prefer A as it it is simplest in all aspect except the =
NAT
> traversal.
> - It allows for flow based QoS.
> - It is the what the implementation that exist mostly do
> - Signaling protocols that exist support it, no extra functionality
> - People are used to the concept
> - It minimizes the difference to legacy.
>=20
> Thus it is the quickest road to define something with the least formal
> push back and concern over maturity of any solution.
>=20
> The downside with B and C is that we do have to solve the multiplexing
> and get an agreement that gets through all the hurdles.
>=20
> Of these two opens I do prefer C.  Although it results in the extra
> complexities of having both alternatives, it will give us both a
> fallback, flow based QoS and better legacy support.
>=20
> Now it is your time to make your opinion heard!
>=20
> Cheers
>=20
> Magnus Westerlund
>=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


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



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On 07/19/2011 05:05 PM, Cullen Jennings wrote:
> I'd like to see it that devices MUST support multiplexing and I'd probably be in favor of also MUST support non multiplexed flows. The multiplexing is a big advantage for any device that needs to deal with NATs so I suspect many "legacy" devices would be moving to this even if there was no need to deal with the browser like devices. I also think that allowing this type of multiplexing will make firewall traversal easier where a device can configure a firewall to open just one specified port and use it for RTP.
>
> The number places where RSVP is usable does not really overlap with places one uses ICE so it does not bother me in the slightest that RSVP won't work with this. It's pretty easy to imagine an extension to RSVP to support reservations based on SSRC inside the flow. Diff Serv based qos will work which is what more commonly used in situations where we are using NATs.
>
> The primary reason I want multiplexing is the time to set up all the ICE connections. The fact of the matter is that NATs limit how quickly you can create new flows. If all you need is something simple with just 2 flows, one for audio and one for video, this is workable. But as soon as you move to trying to have a transition strategy from V4 to V6 and multiple person video or even just multi screen video, the user experience goes downhill rapidly.
>    

V9 is quick, yet the impression gives the vaporware experience because 
it is solid state less-than-best solution. For real-time, probably best 
to just expose if V4 is fast or slow, and we can assume the opposite 
about V6.


> Much of the IETF thinking around ICE was done on the assumption that ICE could be done before ringing started or during ringing but the deployments I am seeing are moving more towards not doing ICE until after a call is answered. There are many reasons for this but one of them is that if Alice calls Bob, and Bob is on a mobile internet device, you don't want to reveal Bob's location to Alice before Bob even decides to answer the call. Given an IP address more or less reveals location these days that a bit of issues to do ICE before answering. This puts more pressure on being able to do ICE quickly for a good user experience.
>    

V9 crystallization technique simply wants to know if anything changed 
while 'do ICE quickly' is done. Consider that quality of service for 
dynamic-compilers, especially those that need what weight of energy 
before any real-time process is done.

If we know the process and data does not contain any location, sex, or 
age, then pressure relief exists with that process and data. I see three 
flows for ICE, not 2, and one separate individual flow.

Safe to multiplex those flows? Not by these standards in this WG.


> Cullen<with my individual contributor hat on>
>
> On Jul 19, 2011, at 7:28 , Magnus Westerlund wrote:
>
>    
>> Hi,
>>
>> This email is as an individual contributor.
>>
>> I want to get started on the discussion of the Multiplexing of the
>> various protocols over single lower layer transport flow, such as a UDP
>> flow. I will attempt to split up the questions into different emails.
>>
>> The first question I think is reasonably easy to get answered, but I
>> think it is time we determine if my belief in the answer is correct or not.
>>
>> The traffic between two RTCWEB peers from the various components, such
>> as RTP sessions, datagram service:
>>
>> a) MUST be sent as Individual flows for each component.
>>
>> b) MUST be multiplexed into a single transport flow.
>>
>> c) SHOULD be multiplexed into a single transport flow, but the RTCWEB
>> peer MUST be able to send them as individual flows.
>>
>> I would love if people can indicate their choice or preferences.
>>
>> I personally prefer A as it it is simplest in all aspect except the NAT
>> traversal.
>> - It allows for flow based QoS.
>> - It is the what the implementation that exist mostly do
>> - Signaling protocols that exist support it, no extra functionality
>> - People are used to the concept
>> - It minimizes the difference to legacy.
>>
>> Thus it is the quickest road to define something with the least formal
>> push back and concern over maturity of any solution.
>>
>> The downside with B and C is that we do have to solve the multiplexing
>> and get an agreement that gets through all the hurdles.
>>
>> Of these two opens I do prefer C.  Although it results in the extra
>> complexities of having both alternatives, it will give us both a
>> fallback, flow based QoS and better legacy support.
>>
>> Now it is your time to make your opinion heard!
>>
>> Cheers
>>
>> Magnus Westerlund
>>
>> ----------------------------------------------------------------------
>> Multimedia Technologies, Ericsson Research EAB/TVM
>> ----------------------------------------------------------------------
>> Ericsson AB                | Phone  +46 10 7148287
>> F�r�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
>>      
>
> Cullen Jennings
> For corporate legal information go to:
> http://www.cisco.com/web/about/doing_business/legal/cri/index.html
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>    


-- 
--- http://twitter.com/Dzonatas_Sol ---
Web Development, Software Engineering
Ag-Biotech, Virtual Reality, Consultant


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inline ...

On Jul 19, 2011, at 8:11 , Magnus Westerlund wrote:

> WG,
>=20
> As individual contributor.
>=20
> I would recommend that people read both
> =
https://datatracker.ietf.org/doc/draft-rosenberg-rtcweb-rtpmux/?include_te=
xt=3D1
> and
> =
https://datatracker.ietf.org/doc/draft-perkins-rtcweb-rtp-usage/?include_t=
ext=3D1
> before getting into this debate.
>=20
> Assuming that the consensus of "To multiplex or not!" thread is either
> of the options requiring multiplexing we get to the question of how to
> do the multiplexing.
>=20
> Lets start with a very fundamental question:
>=20
> Explicit vs Implicit multiplexing
>=20
> So the explicit is when something in the received packets has the
> explicit purpose to identifies a context that resolves the protocol, =
the
> session context etc.
>=20
> Implicit is when we have no such explicit field, instead we have to =
rely
> on that investigating values of particular bits in the payload of the
> packet. For example trying to identify RTP based on the first byte of
> the RTP header where the version two bits shall be 10b.
>=20
> I am strongly arguing against implicit identifying the protocol for =
the
> following reasons.
>=20
> 1. We have 4 or more protocols (The possible set of protocols include:
> STUN, RTP/RTCP, Datagram, and DTLS-SRTP. There is likely future
> extensions, like reliable protocol)  that needs to be identified and
> demultiplexed. A single extension in one of these protocols can cause
> the whole house of cards to come tumbling down.

Uh, we already demux STUN, RTP/RTC, DTLS-SRTP based on first byte and =
this is no big deal and is sort of water under the bridge at this point. =
But this first byte multiplex issues is a red herring to this topic, =
what happens with the demux of these protocols does not really have much =
to do with how to demux the different RTP sessions.=20

>=20
> 2. It will require one to find an RTP internal session multiplexing
> mechanism. This is likely not easy or causes a fork of RTP into two
> non-compatible versions. In addition such a mechanism needs defining
> which takes time, likely substantial time as there is quite a lot of =
RTP
> mechanisms to consider.

A bunch of us our proposing SSRC. We not seeing any problems with it.  I =
rather having this discussion be concrete around the specific proposal =
on the table and not just some abstract it might be hard to find =
something.=20

>=20
> 3. We likely need a multiplexing layer anyway if we want a fall back
> over WebSockets or HTTP.

Not really - I think this is best done by making another TURN like =
protocol that is compatible uses WebSockets or HTTP as the underling =
transport. An architecture like this means that  that protocol is =
selected unilateral by the endpoint and used to get through that =
endpoints  firewalls. Both clients don't have to agree to do the same =
thing - just like If I am using a TURN and you are not, we can still  =
have a call. Avoid a bilateral solution where possible make it much =
easier to roll out.=20

>=20
> Thus we see this a very fragile mechanism that creates additional work
> overhead and large uncertainties in when it would be available.

This all sounds sort of doom and gloom but lets talk about real =
problems.=20

>=20
> We have in our draft:
> https://datatracker.ietf.org/doc/draft-perkins-rtcweb-rtp-usage/ =
looked
> at two possible suggestions for explicit multiplexing. Either a thin
> shim layer primarily consisting of a identifier that through =
negotiation
> is bound to a context, like an particular RTP session or a datagram
> channel.

The problem with this is that if you are talking to a device that does =
not support this you forever have to go through some intermediary =
gateway that strips the byte. Just like turns servers, this reduces the =
user experience by adding latency and jitter and is expensive to run due =
to the bandwidth required.=20

>=20
> The other alternative we considered was using DCCP over UDP for RTP =
and
> datagram. That way we got a complete port space for multiplexing
> different purposes for flows. We also got a transport framework with =
its
> own negotiation, congestion control, a several other features.

This has the same problem as above with the added problem of I am not =
aware of a good (meaning widely tested) DCCP implementation in user land =
and to be deployable this pretty much needs to be all user land.=20

>=20
> We invite people to consider these proposals or suggest better
> alternatives of explicit multiplexing. But I hope we can avoid a =
serious
> mistake and select implicit multiplexing.

I'd like people to start talking about what the actual serious mistakes =
of implicit multiplexing as specified in the SSRC proposal are. Ok, Have =
I thrown the gauntlet down enough :-) Really, I'm just not getting what =
the problems are - if I understood the problems, I might have a =
different view point on this.=20

Cullen <with my individual contributor hat on>=

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On Jul 18, 2011, at 15:11 , Randell Jesup wrote:

> All that said - yes, there are complexity issues to consider, which =
was why I was suggesting leveraging
> an existing tunneled protocol like SCTP or even TCP-over-UDP.

I agree with bunch of what you are saying but the previous several times =
I've seen the use SCTP over UDP conversation, it usually ends about the =
point the we ask for a user land implementation. Whatever we do more or =
less has to be user land or already exist in the OS that people run =
browsers on.=20

If someone has a user land implementation of SCTP or TCP, I imagine that =
might change the outcome a bit from previous times. I have not looked at =
the TCP over UPD library Justin mentioned but, at casual glance, I =
noticed is around 400 lines of code which is very small, which is cool. =
But given the lines of code in say the BSD TCP stack, it does make me =
wonder what's missing. That said, I don't think we need something with =
the perforce of the BSD TCP stack as long as what is there is TCP =
friendly and robust.=20







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On Jul 18, 2011, at 2:28 , Tim Panton wrote:

> I understand the pushback on supporting  reliable datagrams from =
Cullen, Silvia, Serge and others.
>=20
> I'm pushing for it simply because I've had it in our browser rt-audio =
component, and used it extensively.
> Without the ability to transport reliable datagrams in 'near realtime' =
the projects=20
> we have done would have been far less compelling for the user. The =
Datagrams are often the=20
> key to providing a special user experience, beyond conventional =
telephony. If we don't provide this,
> we risk producing the tools for a round of  browser based deskphone =
emulators. We can do better.
>=20
> As an example, the demo I show in this blog would have been much =
harder and less responsive without
> realtime data coming back up the RFC 5456 channel :
>=20
> =
https://babyis60.wordpress.com/2009/10/24/my-astricon-googlewave-ibook-sky=
pe-demo/
>=20

cool - I had not seen that before=20

> So I guess I am asking: Are we aiming for  browser based deskphone =
emulators or a new sort of
> rich media experience for browser users?=20

Hey, I just want my black phone :-) Ok, I'm sold that a reliable data =
mechanisms be used and if people build their own, many of them will be =
bad.=20

>=20
> I know what I want.
>=20
> Tim.
>=20
>=20
>=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



From stefan.lk.hakansson@ericsson.com  Tue Jul 19 23:36:07 2011
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On 2011-07-19 18:40, Matthew Kaufman wrote:
> On 7/19/2011 7:28 AM, Magnus Westerlund wrote:
>> a) MUST be sent as Individual flows for each component.
>>
>> b) MUST be multiplexed into a single transport flow.
>>
>> c) SHOULD be multiplexed into a single transport flow, but the RTCWEB
>> peer MUST be able to send them as individual flows.
>>
>> I would love if people can indicate their choice or preferences.
>
> C. But I'm not sure that the "must" needs to be anything more than "you
> just make two peer connection objects, one for each". (In other words,
> any API that allows for multiplexing almost certainly allows for
> non-multiplexing)... this assumes that the multiplexing is implicit, of
> course. With explicit multiplexing it is even harder... the API needs to
> support turning on and off the multiplexing shim as well, something I'm
> not in favor of.
The APIs being discussed now deal with "MediaStreams" that contain both 
audio (zero to many) and video (zero to many). This has been found to be 
a reasonable level of abstraction for the web app developer. The web app 
developer would have difficulties to understand how many RTP streams a 
MediaStream would generate. So it is not so simple to deal with this by 
creating several peer conn objects.

Besides, it has been discussed that A/V synch should be maintained 
within a MediaStream. This is something you would lose if you split it 
up and transport it using separate peer conn objects.

Stefan
>
> Matthew Kaufman
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From magnus.westerlund@ericsson.com  Wed Jul 20 00:44:04 2011
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On 2011-07-19 18:40, Matthew Kaufman wrote:
> On 7/19/2011 7:28 AM, Magnus Westerlund wrote:
>> a) MUST be sent as Individual flows for each component.
>>
>> b) MUST be multiplexed into a single transport flow.
>>
>> c) SHOULD be multiplexed into a single transport flow, but the RTCWEB
>> peer MUST be able to send them as individual flows.
>>
>> I would love if people can indicate their choice or preferences.
> 
> C. But I'm not sure that the "must" needs to be anything more than "you 
> just make two peer connection objects, one for each". (In other words, 
> any API that allows for multiplexing almost certainly allows for 
> non-multiplexing)... this assumes that the multiplexing is implicit, of 
> course. With explicit multiplexing it is even harder... the API needs to 
> support turning on and off the multiplexing shim as well, something I'm 
> not in favor of.

I am far from certain that it is that simple. First of all, two
different RTP sessions, one for audio and one for video that goes over
different UDP flows, still has an association, for example the used
CNAME to indicate that these flows can be synchronized and are from the
same end-point. If the API definition is such that a stream object
implies a synchronization context when sending, then moving the streams
to two different objects fails to have that separation in semantics.

I think the MUST is important in the aspect that it is a clear agreement
to get this to work. How, that is for us to work out together with W3C.

Thus, I wouldn't rule out that actual need for some option that you can
set to get non-multiplexed. It might be conceptually much simpler for a
developer than to actually understand how you need to spit streams and
then add additional binding properties to these media streams which
could be an alternative design.

Cheers

Magnus Westerlund

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


From gonzalo.camarillo@ericsson.com  Wed Jul 20 00:52:27 2011
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Folks,

as you know if you are following this WG, there have been discussions on
multiplexing RTP sessions and how that would affect legacy RTP systems.

Those discussions have not concluded yet. Nevertheless, note that
changes to the RTP architecture and other fundamental changes to RTP
(should you require any) would need to be made in AVTCORE, which is
where the RTP expertise is. The RTCWeb charter says the following:

> If there is identification of missing protocols or
> functionalities, such work can be requested to be done in another
> working group with a suitable charter or by requests for chartering it
> in this WG or another WG

Cheers,

Gonzalo


From stefan.lk.hakansson@ericsson.com  Wed Jul 20 02:37:07 2011
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Subject: Re: [rtcweb] To multiplex or not!
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FWIW I'm in camp A (but of course C is fine as well). Flow based QoS is 
part of modern cellular networks, and I would like rtcweb to work well 
on them (our experience is also that when the throughput is reduced you 
prefer the video to be degraded rather than having the audio being 
chopped - so you would need them to be individual flows).

Stefan
On 2011-07-19 16:28, Magnus Westerlund wrote:
> Hi,
>
> This email is as an individual contributor.
>
> I want to get started on the discussion of the Multiplexing of the
> various protocols over single lower layer transport flow, such as a UDP
> flow. I will attempt to split up the questions into different emails.
>
> The first question I think is reasonably easy to get answered, but I
> think it is time we determine if my belief in the answer is correct or not.
>
> The traffic between two RTCWEB peers from the various components, such
> as RTP sessions, datagram service:
>
> a) MUST be sent as Individual flows for each component.
>
> b) MUST be multiplexed into a single transport flow.
>
> c) SHOULD be multiplexed into a single transport flow, but the RTCWEB
> peer MUST be able to send them as individual flows.
>
> I would love if people can indicate their choice or preferences.
>
> I personally prefer A as it it is simplest in all aspect except the NAT
> traversal.
> - It allows for flow based QoS.
> - It is the what the implementation that exist mostly do
> - Signaling protocols that exist support it, no extra functionality
> - People are used to the concept
> - It minimizes the difference to legacy.
>
> Thus it is the quickest road to define something with the least formal
> push back and concern over maturity of any solution.
>
> The downside with B and C is that we do have to solve the multiplexing
> and get an agreement that gets through all the hurdles.
>
> Of these two opens I do prefer C.  Although it results in the extra
> complexities of having both alternatives, it will give us both a
> fallback, flow based QoS and better legacy support.
>
> Now it is your time to make your opinion heard!
>
> Cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> Frgatan 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 stefan.lk.hakansson@ericsson.com  Wed Jul 20 02:43:54 2011
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Subject: Re: [rtcweb] realiable data service
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On 2011-07-19 19:09, Serge Lachapelle wrote:
> There are many very established ways to transfer files from a web
> browser. Dropbox, Google Docs, yousendit, mobileme, not counting the
> media specific ways, such as Flickr, Spotify, Youtube, etc... All
> offering great APIs for uploads, ACLs, etc...
>
> Seems like re-inventing the wheel.
I tend to agree. What the data channel of rtcweb could provide is 
shorter latency - but to me that does not seem relevant for a file transfer.
>
> /Serge
>
> On Tue, Jul 19, 2011 at 19:02, Emil Ivov <emcho@jitsi.org
> <mailto:emcho@jitsi.org>> wrote:
>
>
>
>     На 19.07.11 18:37, Randell Jesup написа:
>      > On 7/19/2011 11:58 AM, Justin Uberti wrote:
>      >> At Google, we created a TCP/ICE-UDP layering to solve this exact
>      >> problem, using a user-mode implementation of TCP. It has some rough
>      >> edges, but has worked well in practice, and the code is freely
>     available.
>      >
>      > Right; pretty much what I'd anticipate
>
>     +1 most definitely. Again, I have yet to see a massively used RTC
>     application that doesn't have file transfer. So if this is not part of
>     RTCWEB we will basically be asking developers to reimplement Pseudo TCP
>     in JavaScript.
>
>     Emil
>
>      >, and working open code is good.
>      > SCTP as mentioned is a good choice too, and there exist (it appears)
>      > SCTP-over-UDP implementations, which may ease things.  Also the
>      > TCP-over-UDP library I pointed to.
>      >
>      >> We know people will want/need a reliable messaging mechanism for p2p
>      >> data. While we can debate the details of this mechanism at length, I
>      >> suspect anything we choose will result in a better end-state than if
>      >> we require application developers to implement their own.
>      >
>      > As is obvious from my other messages, I agree wholeheartedly.
>      >
>      >> On Tue, Jul 19, 2011 at 7:16 AM, Henry Sinnreich
>      >> <henry.sinnreich@gmail.com <mailto:henry.sinnreich@gmail.com>
>     <mailto:henry.sinnreich@gmail.com
>     <mailto:henry.sinnreich@gmail.com>>> wrote:
>      >>
>      >>     +1
>      >>     Maybe it would be useful for the proponents of this and many
>     other
>      >>     “desirable” features to think from the perspective if they would
>      >>     have to pay themselves for such developments and also make the
>      >>     code freely available.
>      >>     Cullen at least will make his code available.
>      >>
>      >
>      > See above.  No one here wants to develop *another* reliable
>     protocol if
>      > we can avoid it.  I understand your reluctance.
>      >
>      > The goal is to produce something that will meet the needs and get
>     used.
>      > If we don't meet the
>      > needs of the prospective users it's either an total waste of time (if
>      > they ignore this effort and stick
>      > with Flash/plugins/etc) or we get all sorts of
>      > hack/bad/problematic/incompatible implementations
>      > by individual developers using the UDP (or RTP!) streams we open as a
>      > base transport.
>      >
>      > To your point, another way this effort could fail is by biting
>     off too
>      > much or taking too long to
>      > converge, which is a real danger.  That's one reason to leverage
>      > existing protocols.
>      >
>
>     --
>     Emil Ivov, Ph.D.                       67000 Strasbourg,
>     Project Lead                           France
>     Jitsi
>     emcho@jitsi.org <mailto:emcho@jitsi.org>
>       PHONE: +33.1.77.62.43.30 <tel:%2B33.1.77.62.43.30>
>     http://jitsi.org                       FAX: +33.1.77.62.47.31
>     <tel:%2B33.1.77.62.47.31>
>
>     _______________________________________________
>     rtcweb mailing list
>     rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>     https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
>
> --
> Serge Lachapelle | Product Manager |sergel@google.com
> <mailto:sergel@google.com> |+46 732 01 22 32
> <tel:%2B46%20732%2001%2022%2032>
> Google Sweden AB | Kungsbron 2, SE-111 22 Stockholm | Org. nr. 556656-6880
>
> Apparently, this footer is required in Europe. Apologies. This email may
> be confidential or privileged.  If you received this communication by
> mistake, please don't forward it to anyone else,please erase all copies
> and attachments, and please let me know that it went to the wrong
> person.  Thanks.


From harald@alvestrand.no  Wed Jul 20 02:49:25 2011
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Subject: Re: [rtcweb] To multiplex or not!
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Magnus, I cannot parse your list of choices.

In the case where you have

Sender A
Recipient B
Three video streams going from A to B
Three audio streams going from A to B
One video stream going from B to A
One audio stream going from B to A

with the application not requiring the network to treat the packets any=20
differently, and with the application using the CNAME mechanism to match =

audio streams with corresponding video streams

it is completely unclear to me whether your choice A) requires 8=20
transport flows, 4 transport flows or 2 transport flows, and it is=20
completely unclear to me what C) actually means. (I assume "transport=20
flow" =3D "RTP sessions" here)

Please - can you clarify what your question is intended to mean?

                  Harald


On 07/19/11 16:28, Magnus Westerlund wrote:
> Hi,
>
> This email is as an individual contributor.
>
> I want to get started on the discussion of the Multiplexing of the
> various protocols over single lower layer transport flow, such as a UDP=

> flow. I will attempt to split up the questions into different emails.
>
> The first question I think is reasonably easy to get answered, but I
> think it is time we determine if my belief in the answer is correct or =
not.
>
> The traffic between two RTCWEB peers from the various components, such
> as RTP sessions, datagram service:
>
> a) MUST be sent as Individual flows for each component.
>
> b) MUST be multiplexed into a single transport flow.
>
> c) SHOULD be multiplexed into a single transport flow, but the RTCWEB
> peer MUST be able to send them as individual flows.
>
> I would love if people can indicate their choice or preferences.
>
> I personally prefer A as it it is simplest in all aspect except the NAT=

> traversal.
> - It allows for flow based QoS.
> - It is the what the implementation that exist mostly do
> - Signaling protocols that exist support it, no extra functionality
> - People are used to the concept
> - It minimizes the difference to legacy.
>
> Thus it is the quickest road to define something with the least formal
> push back and concern over maturity of any solution.
>
> The downside with B and C is that we do have to solve the multiplexing
> and get an agreement that gets through all the hurdles.
>
> Of these two opens I do prefer C.  Although it results in the extra
> complexities of having both alternatives, it will give us both a
> fallback, flow based QoS and better legacy support.
>
> Now it is your time to make your opinion heard!
>
> Cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> 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
>



From emil@sip-communicator.org  Wed Jul 20 04:08:59 2011
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=D0=9D=D0=B0 20.07.11 02:05, Cullen Jennings =D0=BD=D0=B0=D0=BF=D0=B8=D1=81=
=D0=B0:
> I'd like to see it that devices MUST support multiplexing and I'd
> probably be in favor of also MUST support non multiplexed flows. The
> multiplexing is a big advantage for any device that needs to deal
> with NATs so I suspect many "legacy" devices would be moving to this
> even if there was no need to deal with the browser like devices. I
> also think that allowing this type of multiplexing will make firewall
> traversal easier where a device can configure a firewall to open just
> one specified port and use it for RTP.

I don't really agree with this part. Devices would still need to handle
candidates that come from  STUN, TURN, UPnP, host IPv4, host IPv6, and
potentially VPN, MIPv6 and other tunnelling mechanisms. Whether or not
this only happens for a single component/stream doesn't really simplify
code does it?

According to my personal experience with our work on ice4j.org, managing
multiple checklists was far less resource consuming than implementing
TURN tunnels and persmissions, STUN multiplexing, synchronisation and
even getting pacing right.

<snip>

> The primary reason I want multiplexing is the time to set up all the
> ICE connections. The fact of the matter is that NATs limit how
> quickly you can create new flows. If all you need is something simple
> with just 2 flows, one for audio and one for video, this is workable.
> But as soon as you move to trying to have a transition strategy from
> V4 to V6 and multiple person video or even just multi screen video,
> the user experience goes downhill rapidly.

I agree more with this argument, although freezing candidates
significantly brings down the overhead incurred by having multiple
streams and components and I am not sure that the actual win here would
be good enough to pay for implementing multiplexing and for breaking
interoperability with those that don't have it.

> Much of the IETF thinking around ICE was done on the assumption that
> ICE could be done before ringing started or during ringing but the
> deployments I am seeing are moving more towards not doing ICE until
> after a call is answered. There are many reasons for this but one of
> them is that if Alice calls Bob, and Bob is on a mobile internet
> device, you don't want to reveal Bob's location to Alice before Bob
> even decides to answer the call. Given an IP address more or less
> reveals location these days that a bit of issues to do ICE before
> answering. This puts more pressure on being able to do ICE quickly
> for a good user experience.

Agreed, but is multiplexing the only way of bringing the time down? The
XMPP crowd for example has been using a mechanism that allows candidates
to be sent as they become available rather than wait for all harvesting
to complete. Maybe this is worth exploring and integrating in the
official ICE (although I don't see how it would work with SIP).

Cheers,
Emil

>=20
> Cullen <with my individual contributor hat on>
>=20
> On Jul 19, 2011, at 7:28 , Magnus Westerlund wrote:
>=20
>> Hi,
>>=20
>> This email is as an individual contributor.
>>=20
>> I want to get started on the discussion of the Multiplexing of the=20
>> various protocols over single lower layer transport flow, such as a
>> UDP flow. I will attempt to split up the questions into different
>> emails.
>>=20
>> The first question I think is reasonably easy to get answered, but
>> I think it is time we determine if my belief in the answer is
>> correct or not.
>>=20
>> The traffic between two RTCWEB peers from the various components,
>> such as RTP sessions, datagram service:
>>=20
>> a) MUST be sent as Individual flows for each component.
>>=20
>> b) MUST be multiplexed into a single transport flow.
>>=20
>> c) SHOULD be multiplexed into a single transport flow, but the
>> RTCWEB peer MUST be able to send them as individual flows.
>>=20
>> I would love if people can indicate their choice or preferences.
>>=20
>> I personally prefer A as it it is simplest in all aspect except the
>> NAT traversal. - It allows for flow based QoS. - It is the what the
>> implementation that exist mostly do - Signaling protocols that
>> exist support it, no extra functionality - People are used to the
>> concept - It minimizes the difference to legacy.
>>=20
>> Thus it is the quickest road to define something with the least
>> formal push back and concern over maturity of any solution.
>>=20
>> The downside with B and C is that we do have to solve the
>> multiplexing and get an agreement that gets through all the
>> hurdles.
>>=20
>> Of these two opens I do prefer C.  Although it results in the
>> extra complexities of having both alternatives, it will give us
>> both a fallback, flow based QoS and better legacy support.
>>=20
>> Now it is your time to make your opinion heard!
>>=20
>> Cheers
>>=20
>> Magnus Westerlund
>>=20
>> ----------------------------------------------------------------------=

>>
>>=20
Multimedia Technologies, Ericsson Research EAB/TVM
>> ----------------------------------------------------------------------=

>>
>>=20
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
_______________________________________________
>> rtcweb mailing list rtcweb@ietf.org=20
>> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
>=20
> Cullen Jennings For corporate legal information go to:=20
> http://www.cisco.com/web/about/doing_business/legal/cri/index.html
>=20
>=20
> _______________________________________________ rtcweb mailing list=20
> 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


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=D0=9D=D0=B0 20.07.11 02:43, Cullen Jennings =D0=BD=D0=B0=D0=BF=D0=B8=D1=81=
=D0=B0:
>>> We have in our draft:=20
>>> https://datatracker.ietf.org/doc/draft-perkins-rtcweb-rtp-usage/
>>> looked at two possible suggestions for explicit multiplexing.
>>> Either a thin shim layer primarily consisting of a identifier
>>> that through negotiation is bound to a context, like an
>>> particular RTP session or a datagram channel.
>
> The problem with this is that if you are talking to a device that
> does not support this you forever have to go through some
> intermediary gateway that strips the byte. Just like turns servers,
> this reduces the user experience by adding latency and jitter and is
> expensive to run due to the bandwidth required.

How is this different from implicit multiplexing and talking to devices
that don't support it?

Emil


--=20
http://jitsi.org


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Subject: Re: [rtcweb] realiable data service
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=D0=9D=D0=B0 20.07.11 11:43, Stefan H=C3=A5kansson LK =D0=BD=D0=B0=D0=BF=D0=
=B8=D1=81=D0=B0:
> On 2011-07-19 19:09, Serge Lachapelle wrote:
>> There are many very established ways to transfer files from a web
>> browser. Dropbox, Google Docs, yousendit, mobileme, not counting the
>> media specific ways, such as Flickr, Spotify, Youtube, etc... All
>> offering great APIs for uploads, ACLs, etc...
>>
>> Seems like re-inventing the wheel.
> I tend to agree. What the data channel of rtcweb could provide is=20
> shorter latency - but to me that does not seem relevant for a file tran=
sfer.

I am not so sure about that. Transferring a file directly between you
and your peer is often way faster than relaying through the servers of
well known services. The time necessary to transfer several tens of megs
can very greatly ... from a couple of minutes to an hour or more. Users
are definitely very sensitive to this.

Besides, this is not only about the latency. It's mostly about the
reources that such relayed transfers require on the server side. Most
providers want to bring those to a minimum. So much so, that some even
put relatively low caps on the traffic you can send through the
signalling channel.

For example, the desktop sharing implementation we have in Jitsi uses
signalling (SIP or XMPP) to convey key and mouse events. When using
Google's XMPP service users would regularly experience blockages caused
by the Google service dropping XMPP IQs when they go above a certain
frequency. And this is in no way a criticism for the service itself ...
it is however definitely a reason to have a reliable means of
communicating directly between peers (or at least as directly as possible=
).

Emil


>>
>> /Serge
>>
>> On Tue, Jul 19, 2011 at 19:02, Emil Ivov <emcho@jitsi.org
>> <mailto:emcho@jitsi.org>> wrote:
>>
>>
>>
>>     =D0=9D=D0=B0 19.07.11 18:37, Randell Jesup =D0=BD=D0=B0=D0=BF=D0=B8=
=D1=81=D0=B0:
>>      > On 7/19/2011 11:58 AM, Justin Uberti wrote:
>>      >> At Google, we created a TCP/ICE-UDP layering to solve this exa=
ct
>>      >> problem, using a user-mode implementation of TCP. It has some =
rough
>>      >> edges, but has worked well in practice, and the code is freely=

>>     available.
>>      >
>>      > Right; pretty much what I'd anticipate
>>
>>     +1 most definitely. Again, I have yet to see a massively used RTC
>>     application that doesn't have file transfer. So if this is not par=
t of
>>     RTCWEB we will basically be asking developers to reimplement Pseud=
o TCP
>>     in JavaScript.
>>
>>     Emil
>>
>>      >, and working open code is good.
>>      > SCTP as mentioned is a good choice too, and there exist (it app=
ears)
>>      > SCTP-over-UDP implementations, which may ease things.  Also the=

>>      > TCP-over-UDP library I pointed to.
>>      >
>>      >> We know people will want/need a reliable messaging mechanism f=
or p2p
>>      >> data. While we can debate the details of this mechanism at len=
gth, I
>>      >> suspect anything we choose will result in a better end-state t=
han if
>>      >> we require application developers to implement their own.
>>      >
>>      > As is obvious from my other messages, I agree wholeheartedly.
>>      >
>>      >> On Tue, Jul 19, 2011 at 7:16 AM, Henry Sinnreich
>>      >> <henry.sinnreich@gmail.com <mailto:henry.sinnreich@gmail.com>
>>     <mailto:henry.sinnreich@gmail.com
>>     <mailto:henry.sinnreich@gmail.com>>> wrote:
>>      >>
>>      >>     +1
>>      >>     Maybe it would be useful for the proponents of this and ma=
ny
>>     other
>>      >>     =E2=80=9Cdesirable=E2=80=9D features to think from the per=
spective if they would
>>      >>     have to pay themselves for such developments and also make=
 the
>>      >>     code freely available.
>>      >>     Cullen at least will make his code available.
>>      >>
>>      >
>>      > See above.  No one here wants to develop *another* reliable
>>     protocol if
>>      > we can avoid it.  I understand your reluctance.
>>      >
>>      > The goal is to produce something that will meet the needs and g=
et
>>     used.
>>      > If we don't meet the
>>      > needs of the prospective users it's either an total waste of ti=
me (if
>>      > they ignore this effort and stick
>>      > with Flash/plugins/etc) or we get all sorts of
>>      > hack/bad/problematic/incompatible implementations
>>      > by individual developers using the UDP (or RTP!) streams we ope=
n as a
>>      > base transport.
>>      >
>>      > To your point, another way this effort could fail is by biting
>>     off too
>>      > much or taking too long to
>>      > converge, which is a real danger.  That's one reason to leverag=
e
>>      > existing protocols.
>>      >
>>
>>     --
>>     Emil Ivov, Ph.D.                       67000 Strasbourg,
>>     Project Lead                           France
>>     Jitsi
>>     emcho@jitsi.org <mailto:emcho@jitsi.org>
>>       PHONE: +33.1.77.62.43.30 <tel:%2B33.1.77.62.43.30>
>>     http://jitsi.org                       FAX: +33.1.77.62.47.31
>>     <tel:%2B33.1.77.62.47.31>
>>
>>     _______________________________________________
>>     rtcweb mailing list
>>     rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>>     https://www.ietf.org/mailman/listinfo/rtcweb
>>
>>
>>
>>
>> --
>> Serge Lachapelle | Product Manager |sergel@google.com
>> <mailto:sergel@google.com> |+46 732 01 22 32
>> <tel:%2B46%20732%2001%2022%2032>
>> Google Sweden AB | Kungsbron 2, SE-111 22 Stockholm | Org. nr. 556656-=
6880
>>
>> Apparently, this footer is required in Europe. Apologies. This email m=
ay
>> be confidential or privileged.  If you received this communication by
>> mistake, please don't forward it to anyone else,please erase all copie=
s
>> and attachments, and please let me know that it went to the wrong
>> person.  Thanks.
>=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


From magnus.westerlund@ericsson.com  Wed Jul 20 04:10:13 2011
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On 2011-07-20 11:49, Harald Alvestrand wrote:
> Magnus, I cannot parse your list of choices.
> 
> In the case where you have
> 
> Sender A
> Recipient B
> Three video streams going from A to B
> Three audio streams going from A to B
> One video stream going from B to A
> One audio stream going from B to A
> 
> with the application not requiring the network to treat the packets any 
> differently, and with the application using the CNAME mechanism to match 
> audio streams with corresponding video streams
> 
> it is completely unclear to me whether your choice A) requires 8 
> transport flows, 4 transport flows or 2 transport flows, and it is 
> completely unclear to me what C) actually means. (I assume "transport 
> flow" = "RTP sessions" here)
> 
> Please - can you clarify what your question is intended to mean?

With choice A I do mean that one for each RTP session use one UDP flow
in each direction. And that these two UDP flows will be symmetric, i.e.
a bi-directional UDP flow.

This in contrast to B and C when you use some multiplexing to map all
RTP sessions, independently of how many you have onto one bi-directional
UDP flow.

The above example you give is at least two RTP sessions, one for audio
and one for video. It might be more than these two if the reason you are
sending multiple video streams is because the streams should be
differently treated, for example in a RTP mixer. Where you perform
composition of video streams of the participants and simply switch a
demonstration camera. Or it may be that entity A above is transport
relay and it is forwarding the camera from three different conference
participants and all video streams are the same RTP session.

It is the application that has do make these choices about if it needs
more than one RTP session or not.

Cheers

Magnus Westerlund

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Frgatan 6                | Mobile +46 73 0949079
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From magnus.westerlund@ericsson.com  Wed Jul 20 05:11:02 2011
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As an Individual

Follow up inline

On 2011-07-20 02:43, Cullen Jennings wrote:
> On Jul 19, 2011, at 8:11 , Magnus Westerlund wrote:
>> 
>> 1. We have 4 or more protocols (The possible set of protocols
>> include: STUN, RTP/RTCP, Datagram, and DTLS-SRTP. There is likely
>> future extensions, like reliable protocol)  that needs to be
>> identified and demultiplexed. A single extension in one of these
>> protocols can cause the whole house of cards to come tumbling
>> down.
> 
> Uh, we already demux STUN, RTP/RTC, DTLS-SRTP based on first byte and
> this is no big deal and is sort of water under the bridge at this
> point. But this first byte multiplex issues is a red herring to this
> topic, what happens with the demux of these protocols does not really
> have much to do with how to demux the different RTP sessions.

I agree that we currently have a design that multiplex the three above
protocols onto the same lower layer when one uses ICE and DTSL-SRTP.
However, the proposal in the WG is pilling on to this, but suggesting to
add at least one protocol to the mix, maybe more if we do reliable in
the future. The more independent protocols there is in the mix the
bigger the risk is that we get a failure in the demultiplexing.

>From my perspective this is not a red-herring at all. It is about good
and future proof design.

Both this issue and the RTP session multiplexing needs to be considered.

> 
>> 
>> 2. It will require one to find an RTP internal session
>> multiplexing mechanism. This is likely not easy or causes a fork of
>> RTP into two non-compatible versions. In addition such a mechanism
>> needs defining which takes time, likely substantial time as there
>> is quite a lot of RTP mechanisms to consider.
> 
> A bunch of us our proposing SSRC. We not seeing any problems with it.
> I rather having this discussion be concrete around the specific
> proposal on the table and not just some abstract it might be hard to
> find something.

The proposal on the table is to change the semantics of the SSRC field.
So that it contains one part that is session identifier and another that
is stream identifier. This have substantial impact on the RTP/RTCP
specification and the implementation. To the degree where what you have
is not any longer compatible with RFC 3550. It looks like RTP/RTCP but
it isn't any more.

We have documented the impacts of your proposal in Sections 2.2.1 and
3.3.1 of
https://datatracker.ietf.org/doc/draft-perkins-rtcweb-rtp-usage/?include_text=1

So from my perspective the proposal on the table is a complete no-go.
And I have been thinking trying to find a proposal that wouldn't result
in all these issues and still be internal to RTP. And I haven't found
one yet. That is why I flag this.

If there is a solution that doesn't have as serious impact I do want
AVTCORE to standardize it.

> 
>> 
>> 3. We likely need a multiplexing layer anyway if we want a fall
>> back over WebSockets or HTTP.
> 
> Not really - I think this is best done by making another TURN like
> protocol that is compatible uses WebSockets or HTTP as the underling
> transport. An architecture like this means that  that protocol is
> selected unilateral by the endpoint and used to get through that
> endpoints  firewalls. Both clients don't have to agree to do the same
> thing - just like If I am using a TURN and you are not, we can still
> have a call. Avoid a bilateral solution where possible make it much
> easier to roll out.

Well, it still needs to put out compatible flows. And if we have a
multiplexing layer that becomes equally easy.

> 
>> 
>> Thus we see this a very fragile mechanism that creates additional
>> work overhead and large uncertainties in when it would be
>> available.
> 
> This all sounds sort of doom and gloom but lets talk about real
> problems.
> 

Timing is a real problem. If we want to do multiplexing in RTCWEB 1.0 we
need to pick what is ready and available. We can't go around proposing
new things that are complicated. Changing the SSRC field semantics in
RTP/RTCP is a complicated thing.

We will have tons of discussion of what the requirements really are for
a more generally applicable solution. Not one that barely meets RTCWEBs
needs, when we know the minute this goes out the door a lot of other RTP
using services and applications would like to use it. The teleprecense
work in CLUE is just one example. There will be a bunch. Thus AVTCORE
needs to have a serious discussion of what the requirements are. Then we
will discuss what the architectural implications are of any proposal.
Review what existing functionality we are ready to break and likely need
to replace. What the issues are with gatewaying between old and new
semantics.

Cullen, you are an experienced IETF, you have been an AD for RAI. What
do think the probability is that this discussion has concluded and
resulted in an IESG approved specification before December 2012?

>> 
>> We have in our draft: 
>> https://datatracker.ietf.org/doc/draft-perkins-rtcweb-rtp-usage/
>> looked at two possible suggestions for explicit multiplexing.
>> Either a thin shim layer primarily consisting of a identifier that
>> through negotiation is bound to a context, like an particular RTP
>> session or a datagram channel.
> 
> The problem with this is that if you are talking to a device that
> does not support this you forever have to go through some
> intermediary gateway that strips the byte. Just like turns servers,
> this reduces the user experience by adding latency and jitter and is
> expensive to run due to the bandwidth required.

Isn't that gateway going to be present anyway in most case. I think the
ICE connectivity checks are one thing that will force a gateway in many
cases. In the cases it isn't needed, then you can skip the multiplexing
in that case to reach the legacy.

Would it not be better to have a clean explicit design for how to
multiplex protocols between peers that can be used in RTCWEB and other
future applications that doesn't drag our past mistakes into the design
and forever prevent simple expansion with new functionality.

The cost, a low one from my perspective is to say that legacy don't get
the benefits of multiplexing. A feature they anyway are not going to get
unless they use a gateway or are updated to be RTCWEB compliant.



> 
>> 
>> The other alternative we considered was using DCCP over UDP for RTP
>> and datagram. That way we got a complete port space for
>> multiplexing different purposes for flows. We also got a transport
>> framework with its own negotiation, congestion control, a several
>> other features.
> 
> This has the same problem as above with the added problem of I am not
> aware of a good (meaning widely tested) DCCP implementation in user
> land and to be deployable this pretty much needs to be all user land.


Yes, I do agree that it needs to be a user-land implementation in the
browser until DCCP becomes available in the OSes. But that is true also
for all the alternatives that so far been discussed. TFRC in RTP will be
as much user-land as DCCP over UDP with TFRC.

The big difference is that we either have a well reviewed and by the
community and IESG approved specification in DCCP. Or we create
something our selves that needs to go through all that review and will
be even less tested than DCCP is.


> 
> 
>> 
>> We invite people to consider these proposals or suggest better 
>> alternatives of explicit multiplexing. But I hope we can avoid a
>> serious mistake and select implicit multiplexing.
> 
> I'd like people to start talking about what the actual serious
> mistakes of implicit multiplexing as specified in the SSRC proposal
> are. Ok, Have I thrown the gauntlet down enough :-) Really, I'm just
> not getting what the problems are - if I understood the problems, I
> might have a different view point on this.
> 

Please do read the latest version of
https://datatracker.ietf.org/doc/draft-perkins-rtcweb-rtp-usage/?include_text=1

We have expanded on the discussion, both about the particular proposal
and the general issues with multiplexing is RTP.

If you can give some hint what in that argumentation you don't
understand we can try to explain in some other way.

Cheers

Magnus Westerlund

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----------------------------------------------------------------------
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From harald@alvestrand.no  Wed Jul 20 05:58:22 2011
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On 07/20/11 13:10, Magnus Westerlund wrote:
> On 2011-07-20 11:49, Harald Alvestrand wrote:
>> Magnus, I cannot parse your list of choices.
>>
>> In the case where you have
>>
>> Sender A
>> Recipient B
>> Three video streams going from A to B
>> Three audio streams going from A to B
>> One video stream going from B to A
>> One audio stream going from B to A
>>
>> with the application not requiring the network to treat the packets an=
y
>> differently, and with the application using the CNAME mechanism to mat=
ch
>> audio streams with corresponding video streams
>>
>> it is completely unclear to me whether your choice A) requires 8
>> transport flows, 4 transport flows or 2 transport flows, and it is
>> completely unclear to me what C) actually means. (I assume "transport
>> flow" =3D "RTP sessions" here)
>>
>> Please - can you clarify what your question is intended to mean?
> With choice A I do mean that one for each RTP session use one UDP flow
> in each direction. And that these two UDP flows will be symmetric, i.e.=

> a bi-directional UDP flow.
>
> This in contrast to B and C when you use some multiplexing to map all
> RTP sessions, independently of how many you have onto one bi-directiona=
l
> UDP flow.
>
> The above example you give is at least two RTP sessions, one for audio
> and one for video. It might be more than these two if the reason you ar=
e
> sending multiple video streams is because the streams should be
> differently treated, for example in a RTP mixer. Where you perform
> composition of video streams of the participants and simply switch a
> demonstration camera. Or it may be that entity A above is transport
> relay and it is forwarding the camera from three different conference
> participants and all video streams are the same RTP session.
>
> It is the application that has do make these choices about if it needs
> more than one RTP session or not.
Magnus, if the number of UDP flows is constant (even if it's slightly=20
larger than 1) when the number of video streams changes under your=20
solution A, I think this is a viable option.

If the number of UDP flows was linear with the number of video streams,=20
I believe that a number of scenarios we want RTCWEB to support,=20
including stuff like the Google+ Hangouts, would be basically unsustainab=
le.

So - with the understanding that an application is going to put as many=20
video streams it wants down one RTP session if that's appropriate for=20
the purposes of the application - I am in favour of the "no=20
multiplexing" solution.

                      Harald
> Cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> 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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On 07/20/11 03:13, Cullen Jennings wrote:
> On Jul 18, 2011, at 15:11 , Randell Jesup wrote:
>
>> All that said - yes, there are complexity issues to consider, which was why I was suggesting leveraging
>> an existing tunneled protocol like SCTP or even TCP-over-UDP.
> I agree with bunch of what you are saying but the previous several times I've seen the use SCTP over UDP conversation, it usually ends about the point the we ask for a user land implementation. Whatever we do more or less has to be user land or already exist in the OS that people run browsers on.
>
> If someone has a user land implementation of SCTP or TCP, I imagine that might change the outcome a bit from previous times. I have not looked at the TCP over UPD library Justin mentioned but, at casual glance, I noticed is around 400 lines of code which is very small, which is cool. But given the lines of code in say the BSD TCP stack, it does make me wonder what's missing. That said, I don't think we need something with the perforce of the BSD TCP stack as long as what is there is TCP friendly and robust.
I believe this is the current URL:

http://code.google.com/p/libjingle/source/browse/trunk/talk/session/tunnel/pseudotcpchannel.cc 
(500 or so lines).

The real protocol implementation is here:
http://code.google.com/p/libjingle/source/browse/trunk/talk/p2p/base/pseudotcp.cc
which is another 1000 lines (not counting the various .h files).

We're not THAT good at writing compact code :-)
>
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


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> Besides, it has been discussed that A/V synch should be maintained
> within a MediaStream. This is something you would lose if you split it
> up and transport it using separate peer conn objects.

Actually, one of the major design goals of the MediaStream API is to 
allow the synchronization of multiple, separate MediaStreams, and I've 
been working with roc to ensure that this is, in fact, relatively 
straightforward to do.

Which is not an argument against a more explicit API for dealing with 
multiplexing (that argument belongs on a different list anyway).

From fluffy@cisco.com  Wed Jul 20 07:11:10 2011
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On Jul 20, 2011, at 4:08 , Emil Ivov wrote:

> =D0=9D=D0=B0 20.07.11 02:05, Cullen Jennings =D0=BD=D0=B0=D0=BF=D0=B8=D1=
=81=D0=B0:
>> I'd like to see it that devices MUST support multiplexing and I'd
>> probably be in favor of also MUST support non multiplexed flows. The
>> multiplexing is a big advantage for any device that needs to deal
>> with NATs so I suspect many "legacy" devices would be moving to this
>> even if there was no need to deal with the browser like devices. I
>> also think that allowing this type of multiplexing will make firewall
>> traversal easier where a device can configure a firewall to open just
>> one specified port and use it for RTP.
>=20
> I don't really agree with this part. Devices would still need to =
handle
> candidates that come from  STUN, TURN, UPnP, host IPv4, host IPv6, and
> potentially VPN, MIPv6 and other tunnelling mechanisms. Whether or not
> this only happens for a single component/stream doesn't really =
simplify
> code does it?
>=20
> According to my personal experience with our work on ice4j.org, =
managing
> multiple checklists was far less resource consuming than implementing
> TURN tunnels and persmissions, STUN multiplexing, synchronisation and
> even getting pacing right.


Ahh, I think we have come to the key issues here - perhaps I have not =
been explaining myself very well. So in your experience roughly how long =
does it take to set up a for two scenarios using ICE as is today.=20

First scenario: I have two devices that want to set up a single audio =
stream and they each have the candidates from: local v4 IP address, v4 =
stun, v4 turn, local v6, turn v6. Obviously they could have a lot more =
but that seems like a reasonable starting point.=20

Second scenario: same as first address wise but instead of a single =
audio stream they want to set up a single audio stream  to a conference =
bridge plus 7 video streams for the video from the 5 people on the =
bridge plus a presentation stream and stream of video from active =
speaker. I don't really care much about the scenario other than there =
are 8 streams being set up. But this type of scenario is becoming very =
common for multi party video chat as it allows you to see perhaps small =
versions of everyone plus a large version of active speaker.=20

My experience is the answer to the first scenario is not as quick as you =
would like and the answer to how long the second takes is about 8 times =
longer than the first one. You might do a bit better than that depending =
on how clever your implementation is but it still a lot longer.=20

I totally agree with you that multiplexing may not hugely simplify the =
code and you still have to deal with the hard parts to implement. WHat =
is helps with is it makes the second scenario take exactly the same time =
as the first scenario because instead of taking 8 times a long. It's =
purely a user experience thing. Does that make sense?=20


>=20
> <snip>
>=20
>> The primary reason I want multiplexing is the time to set up all the
>> ICE connections. The fact of the matter is that NATs limit how
>> quickly you can create new flows. If all you need is something simple
>> with just 2 flows, one for audio and one for video, this is workable.
>> But as soon as you move to trying to have a transition strategy from
>> V4 to V6 and multiple person video or even just multi screen video,
>> the user experience goes downhill rapidly.
>=20
> I agree more with this argument, although freezing candidates
> significantly brings down the overhead incurred by having multiple
> streams and components and I am not sure that the actual win here =
would
> be good enough to pay for implementing multiplexing and for breaking
> interoperability with those that don't have it.

Use multiplexing would be negotiated over the SDP, just like rtcp/rtp =
mux is negotiated today, so it would not break interoperability with =
stuff that did not support it.=20

>=20
>> Much of the IETF thinking around ICE was done on the assumption that
>> ICE could be done before ringing started or during ringing but the
>> deployments I am seeing are moving more towards not doing ICE until
>> after a call is answered. There are many reasons for this but one of
>> them is that if Alice calls Bob, and Bob is on a mobile internet
>> device, you don't want to reveal Bob's location to Alice before Bob
>> even decides to answer the call. Given an IP address more or less
>> reveals location these days that a bit of issues to do ICE before
>> answering. This puts more pressure on being able to do ICE quickly
>> for a good user experience.
>=20
> Agreed, but is multiplexing the only way of bringing the time down? =
The
> XMPP crowd for example has been using a mechanism that allows =
candidates
> to be sent as they become available rather than wait for all =
harvesting
> to complete. Maybe this is worth exploring and integrating in the
> official ICE (although I don't see how it would work with SIP).

I am a fan of sending candidates as they become available but I don't =
think this helps much as most devices can gather the addresses before =
the user even initiates a call and the gather it typically very quick =
relative to the rest of the ICE process.=20

>=20
> Cheers,
> Emil
>=20
>>=20
>> Cullen <with my individual contributor hat on>
>>=20
>> On Jul 19, 2011, at 7:28 , Magnus Westerlund wrote:
>>=20
>>> Hi,
>>>=20
>>> This email is as an individual contributor.
>>>=20
>>> I want to get started on the discussion of the Multiplexing of the=20=

>>> various protocols over single lower layer transport flow, such as a
>>> UDP flow. I will attempt to split up the questions into different
>>> emails.
>>>=20
>>> The first question I think is reasonably easy to get answered, but
>>> I think it is time we determine if my belief in the answer is
>>> correct or not.
>>>=20
>>> The traffic between two RTCWEB peers from the various components,
>>> such as RTP sessions, datagram service:
>>>=20
>>> a) MUST be sent as Individual flows for each component.
>>>=20
>>> b) MUST be multiplexed into a single transport flow.
>>>=20
>>> c) SHOULD be multiplexed into a single transport flow, but the
>>> RTCWEB peer MUST be able to send them as individual flows.
>>>=20
>>> I would love if people can indicate their choice or preferences.
>>>=20
>>> I personally prefer A as it it is simplest in all aspect except the
>>> NAT traversal. - It allows for flow based QoS. - It is the what the
>>> implementation that exist mostly do - Signaling protocols that
>>> exist support it, no extra functionality - People are used to the
>>> concept - It minimizes the difference to legacy.
>>>=20
>>> Thus it is the quickest road to define something with the least
>>> formal push back and concern over maturity of any solution.
>>>=20
>>> The downside with B and C is that we do have to solve the
>>> multiplexing and get an agreement that gets through all the
>>> hurdles.
>>>=20
>>> Of these two opens I do prefer C.  Although it results in the
>>> extra complexities of having both alternatives, it will give us
>>> both a fallback, flow based QoS and better legacy support.
>>>=20
>>> Now it is your time to make your opinion heard!
>>>=20
>>> Cheers
>>>=20
>>> Magnus Westerlund
>>>=20
>>> =
----------------------------------------------------------------------
>>>=20
>>>=20
> Multimedia Technologies, Ericsson Research EAB/TVM
>>> =
----------------------------------------------------------------------
>>>=20
>>>=20
> 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=20
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>=20
>>=20
>> Cullen Jennings For corporate legal information go to:=20
>> http://www.cisco.com/web/about/doing_business/legal/cri/index.html
>>=20
>>=20
>> _______________________________________________ rtcweb mailing list=20=

>> rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb
>>=20
>=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
>=20


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



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On Jul 20, 2011, at 4:08 , Emil Ivov wrote:

> =D0=9D=D0=B0 20.07.11 02:43, Cullen Jennings =D0=BD=D0=B0=D0=BF=D0=B8=D1=
=81=D0=B0:
>>>> We have in our draft:=20
>>>> https://datatracker.ietf.org/doc/draft-perkins-rtcweb-rtp-usage/
>>>> looked at two possible suggestions for explicit multiplexing.
>>>> Either a thin shim layer primarily consisting of a identifier
>>>> that through negotiation is bound to a context, like an
>>>> particular RTP session or a datagram channel.
>>=20
>> The problem with this is that if you are talking to a device that
>> does not support this you forever have to go through some
>> intermediary gateway that strips the byte. Just like turns servers,
>> this reduces the user experience by adding latency and jitter and is
>> expensive to run due to the bandwidth required.
>=20
> How is this different from implicit multiplexing and talking to =
devices
> that don't support it?

Sorry, I should have been clear that I think the multiplexing has to be =
negotiated with SDP=20=

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> Timing is a real problem. If we want to do multiplexing in RTCWEB 1.0 we
> need to pick what is ready and available. We can't go around proposing
> new things that are complicated. Changing the SSRC field semantics in
> RTP/RTCP is a complicated thing.

[BA] We need to ask ourselves "Does this really need to be in the critical =
path for RTCWEB 1.0"? =20
As I understand it=2C this would be a negotiated feature=2C and presumably =
all implementations
would be required to support RTP/RTCP as defined in RFC 3550 and suitable p=
rofiles.=20
So from my perspective=2C I'd prefer to see this handled as an extension on=
 its own path=2C not a=20
mandatory-to-implement feature in 1.0.  =20


> Isn't that gateway going to be present anyway in most case. I think the
> ICE connectivity checks are one thing that will force a gateway in many
> cases. In the cases it isn't needed=2C then you can skip the multiplexing
> in that case to reach the legacy.

[BA] Just because we're likely to require a gateway doesn't mean we should
heap more and more things on the camel's back.  We're already talking about
ICE=2C shims=2C transcoding=2C changes to RFC 3550....  that's a lot of fun=
ctionality
for the "gateway".  So I agree with Jonathan about the disadvantages of a s=
him
(or DCCP for that matter).=20


> The big difference is that we either have a well reviewed and by the
> community and IESG approved specification in DCCP. Or we create
> something our selves that needs to go through all that review and will
> be even less tested than DCCP is.

[BA] Deployment counts=2C too.  How widely deployed is DCCP for realtime us=
es?=20


 		 	   		  =

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<body class=3D'hmmessage'><div dir=3D'ltr'>
<br><div>&gt=3B Timing is a real problem. If we want to do multiplexing in =
RTCWEB 1.0 we<br>&gt=3B need to pick what is ready and available. We can't =
go around proposing<br>&gt=3B new things that are complicated. Changing the=
 SSRC field semantics in<br>&gt=3B RTP/RTCP is a complicated thing.<br><br>=
[BA] We need to ask ourselves "Does this really need to be in the critical =
path for RTCWEB 1.0"?&nbsp=3B <br>As I understand it=2C this would be a neg=
otiated feature=2C and presumably all implementations<br>would be required =
to support RTP/RTCP as defined in RFC 3550 and suitable profiles. <br>So fr=
om my perspective=2C I'd prefer to see this handled as an extension on its =
own path=2C not a <br>mandatory-to-implement feature in 1.0.&nbsp=3B&nbsp=
=3B <br><br><br>&gt=3B Isn't that gateway going to be present anyway in mos=
t case. I think the<br>&gt=3B ICE connectivity checks are one thing that wi=
ll force a gateway in many<br>&gt=3B cases. In the cases it isn't needed=2C=
 then you can skip the multiplexing<br>&gt=3B in that case to reach the leg=
acy.<br><br>[BA] Just because we're likely to require a gateway doesn't mea=
n we should<br>heap more and more things on the camel's back.&nbsp=3B We're=
 already talking about<br>ICE=2C shims=2C transcoding=2C changes to RFC 355=
0....&nbsp=3B that's a lot of functionality<br>for the "gateway".&nbsp=3B S=
o I agree with Jonathan about the disadvantages of a shim<br>(or DCCP for t=
hat matter). <br><br><br>&gt=3B The big difference is that we either have a=
 well reviewed and by the<br>&gt=3B community and IESG approved specificati=
on in DCCP. Or we create<br>&gt=3B something our selves that needs to go th=
rough all that review and will<br>&gt=3B be even less tested than DCCP is.<=
br><br>[BA] Deployment counts=2C too.&nbsp=3B How widely deployed is DCCP f=
or realtime uses? <br><br><br></div> 		 	   		  </div></body>
</html>=

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> Ahh=2C I think we have come to the key issues here - perhaps I have not b=
een explaining myself very well. So in your experience roughly how long doe=
s it take to set up a for two scenarios using ICE as is today.=20
>=20
> First scenario: I have two devices that want to set up a single audio str=
eam and they each have the candidates from: local v4 IP address=2C v4 stun=
=2C v4 turn=2C local v6=2C turn v6. Obviously they could have a lot more bu=
t that seems like a reasonable starting point.=20
>=20
> Second scenario: same as first address wise but instead of a single audio=
 stream they want to set up a single audio stream  to a conference bridge p=
lus 7 video streams for the video from the 5 people on the bridge plus a pr=
esentation stream and stream of video from active speaker. I don't really c=
are much about the scenario other than there are 8 streams being set up. Bu=
t this type of scenario is becoming very common for multi party video chat =
as it allows you to see perhaps small versions of everyone plus a large ver=
sion of active speaker.=20
>=20
> My experience is the answer to the first scenario is not as quick as you =
would like and the answer to how long the second takes is about 8 times lon=
ger than the first one. You might do a bit better than that depending on ho=
w clever your implementation is but it still a lot longer.=20

[BA] Yes=2C that is correct.  However at the moment I'd characterize that a=
s a "high quality problem"=2C since it isn't even clear that the current WH=
ATWG API can handle that case=2C due to glare and ICE timing issues. =20
 		 	   		  =

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<br><div>&gt=3B Ahh=2C I think we have come to the key issues here - perhap=
s I have not been explaining myself very well. So in your experience roughl=
y how long does it take to set up a for two scenarios using ICE as is today=
. <br>&gt=3B <br>&gt=3B First scenario: I have two devices that want to set=
 up a single audio stream and they each have the candidates from: local v4 =
IP address=2C v4 stun=2C v4 turn=2C local v6=2C turn v6. Obviously they cou=
ld have a lot more but that seems like a reasonable starting point. <br>&gt=
=3B <br>&gt=3B Second scenario: same as first address wise but instead of a=
 single audio stream they want to set up a single audio stream  to a confer=
ence bridge plus 7 video streams for the video from the 5 people on the bri=
dge plus a presentation stream and stream of video from active speaker. I d=
on't really care much about the scenario other than there are 8 streams bei=
ng set up. But this type of scenario is becoming very common for multi part=
y video chat as it allows you to see perhaps small versions of everyone plu=
s a large version of active speaker. <br>&gt=3B <br>&gt=3B My experience is=
 the answer to the first scenario is not as quick as you would like and the=
 answer to how long the second takes is about 8 times longer than the first=
 one. You might do a bit better than that depending on how clever your impl=
ementation is but it still a lot longer. <br><br>[BA] Yes=2C that is correc=
t.&nbsp=3B However at the moment I'd characterize that as a "high quality p=
roblem"=2C since it isn't even clear that the current WHATWG API can handle=
 that case=2C due to glare and ICE timing issues.&nbsp=3B <br></div> 		 	  =
 		  </div></body>
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Subject: Re: [rtcweb] To multiplex or not!
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On Jul 20, 2011, at 7:24 , Bernard Aboba wrote:

>=20
> > Ahh, I think we have come to the key issues here - perhaps I have =
not been explaining myself very well. So in your experience roughly how =
long does it take to set up a for two scenarios using ICE as is today.=20=

> >=20
> > First scenario: I have two devices that want to set up a single =
audio stream and they each have the candidates from: local v4 IP =
address, v4 stun, v4 turn, local v6, turn v6. Obviously they could have =
a lot more but that seems like a reasonable starting point.=20
> >=20
> > Second scenario: same as first address wise but instead of a single =
audio stream they want to set up a single audio stream to a conference =
bridge plus 7 video streams for the video from the 5 people on the =
bridge plus a presentation stream and stream of video from active =
speaker. I don't really care much about the scenario other than there =
are 8 streams being set up. But this type of scenario is becoming very =
common for multi party video chat as it allows you to see perhaps small =
versions of everyone plus a large version of active speaker.=20
> >=20
> > My experience is the answer to the first scenario is not as quick as =
you would like and the answer to how long the second takes is about 8 =
times longer than the first one. You might do a bit better than that =
depending on how clever your implementation is but it still a lot =
longer.=20
>=20
> [BA] Yes, that is correct.  However at the moment I'd characterize =
that as a "high quality problem", since it isn't even clear that the =
current WHATWG API can handle that case, due to glare and ICE timing =
issues. =20

Agree - but I think we need to fix that regardless of what we decide =
with this.
=20


From dzonatas@gmail.com  Wed Jul 20 07:53:54 2011
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From: Dzonatas Sol <dzonatas@gmail.com>
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Subject: Re: [rtcweb] To multiplex or not!
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I found the answer through the continued discussion in other messages 
sent to this WG list.

There is no semantics between spectral modes and multiplex modes.

In fact, news yesterday reveals "they're getting it" now, so I may not 
seem so much like a pest...

"A team of researchers at the Oak Ridge National Laboratory have found 
that it's hydrogen rather than carbon which plays a fundamental part in 
creating a uniform crystalline form."

Read more: 
http://www.techeye.net/science/graphene-discovery-points-to-large-scale-production#ixzz1SelXjqyn

Yes, we have been thinking in light ("spectral" crystal ICs) and 
hydrogen economy for years (stateful vs stateless protocols), and 
realized many do not, were stuck, or haven't yet realized this much at 
this level because their trained to think in unlimited resources. 
Practicality and duplication is near-term now.

...well, a resource-pest around standards to determine sustainable 
technology. Real-time and dates are always an issue with assets.

If the above still doesn't make think, realize that generally hydrogen 
has no shape ("nobody"/nothing holds it), therefore considered 
unreliable for transport, until now...

Thanks.

On 07/19/2011 08:30 AM, Dzonatas Sol wrote:
> On 07/19/2011 07:28 AM, Magnus Westerlund wrote:
>> a) MUST be sent as Individual flows for each component.
>>
>> b) MUST be multiplexed into a single transport flow.
>>
>> c) SHOULD be multiplexed into a single transport flow, but the RTCWEB
>> peer MUST be able to send them as individual flows.
>>
>
> One clarification please, if there are known ranges of A as one 
> spectrum, does this WG still consider that multiplexed? IPv6, for 
> example, allows us to multiplex in a TCP-less way, simply by 
> fulfillment of the flows to more than one individual address. Those 
> addresses could constitute one spectrum in a stateful manner, or not.
>


-- 
--- http://twitter.com/Dzonatas_Sol ---
Web Development, Software Engineering
Ag-Biotech, Virtual Reality, Consultant


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Subject: Re: [rtcweb] realiable data service
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On 7/20/2011 5:43 AM, Stefan Håkansson LK wrote:
> On 2011-07-19 19:09, Serge Lachapelle wrote:
>> There are many very established ways to transfer files from a web
>> browser. Dropbox, Google Docs, yousendit, mobileme, not counting the
>> media specific ways, such as Flickr, Spotify, Youtube, etc... All
>> offering great APIs for uploads, ACLs, etc...
>>
>> Seems like re-inventing the wheel.
> I tend to agree. What the data channel of rtcweb could provide is 
> shorter latency - but to me that does not seem relevant for a file 
> transfer.

For some use-cases that's important (shorter on average, not guaranteed) 
- for example games.  Others like the Google Wave demo Tim posted 
(https://babyis60.wordpress.com/2009/10/24/my-astricon-googlewave-ibook-skype-demo/) 
would be painful (and hurt usability) to implement using drop-box like 
transfers.

For other cases (file xfer), yes you can use those - but the problem is 
that when you want to transfer to the person you're chatting with, you 
would need to use another service, agree on it, deal with the 
semantics/accounts/limits/etc for that service, etc.  A horrible user 
experience.  Even if it was "hidden" by the application, it would be 
problematic and subject to breakage continually.

-- 
Randell Jesup
randell-ietf@jesup.org


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From: Justin Uberti <juberti@google.com>
Date: Wed, 20 Jul 2011 11:37:12 -0400
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Subject: Re: [rtcweb] PseudoTCP implementation (Re: realiable data service)
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On Wed, Jul 20, 2011 at 9:15 AM, Harald Alvestrand <harald@alvestrand.no>wrote:

> On 07/20/11 03:13, Cullen Jennings wrote:
>
>> On Jul 18, 2011, at 15:11 , Randell Jesup wrote:
>>
>>  All that said - yes, there are complexity issues to consider, which was
>>> why I was suggesting leveraging
>>> an existing tunneled protocol like SCTP or even TCP-over-UDP.
>>>
>> I agree with bunch of what you are saying but the previous several times
>> I've seen the use SCTP over UDP conversation, it usually ends about the
>> point the we ask for a user land implementation. Whatever we do more or less
>> has to be user land or already exist in the OS that people run browsers on.
>>
>> If someone has a user land implementation of SCTP or TCP, I imagine that
>> might change the outcome a bit from previous times. I have not looked at the
>> TCP over UPD library Justin mentioned but, at casual glance, I noticed is
>> around 400 lines of code which is very small, which is cool. But given the
>> lines of code in say the BSD TCP stack, it does make me wonder what's
>> missing. That said, I don't think we need something with the perforce of the
>> BSD TCP stack as long as what is there is TCP friendly and robust.
>>
> I believe this is the current URL:
>
> http://code.google.com/p/**libjingle/source/browse/trunk/**
> talk/session/tunnel/**pseudotcpchannel.cc<http://code.google.com/p/libjingle/source/browse/trunk/talk/session/tunnel/pseudotcpchannel.cc>(500 or so lines).
>
> The real protocol implementation is here:
> http://code.google.com/p/**libjingle/source/browse/trunk/**
> talk/p2p/base/pseudotcp.cc<http://code.google.com/p/libjingle/source/browse/trunk/talk/p2p/base/pseudotcp.cc>
> which is another 1000 lines (not counting the various .h files).
>
> We're not THAT good at writing compact code :-)
>

Yes, I was going to send a similar email. There are some
deficiencies/optimizations for the fact it runs over ICE-UDP - doesn't do
path MTU discovery, no checksum, no OOB data - but it is based on TCP and
implements slow start and other congestion avoidance mechanisms, and is
actively used by a number of applications where reliable transfer is
important, such as file transfer and remote desktop applications.


>
>>
>>
>>
>> ______________________________**_________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/**listinfo/rtcweb<https://www.ietf.org/mailman/listinfo/rtcweb>
>>
>>
> ______________________________**_________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/**listinfo/rtcweb<https://www.ietf.org/mailman/listinfo/rtcweb>
>

--000e0cd56f26c0b79004a8820279
Content-Type: text/html; charset=UTF-8
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<br><br><div class=3D"gmail_quote">On Wed, Jul 20, 2011 at 9:15 AM, Harald =
Alvestrand <span dir=3D"ltr">&lt;<a href=3D"mailto:harald@alvestrand.no">ha=
rald@alvestrand.no</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quot=
e" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"=
>

On 07/20/11 03:13, Cullen Jennings wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
On Jul 18, 2011, at 15:11 , Randell Jesup wrote:<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
All that said - yes, there are complexity issues to consider, which was why=
 I was suggesting leveraging<br>
an existing tunneled protocol like SCTP or even TCP-over-UDP.<br>
</blockquote>
I agree with bunch of what you are saying but the previous several times I&=
#39;ve seen the use SCTP over UDP conversation, it usually ends about the p=
oint the we ask for a user land implementation. Whatever we do more or less=
 has to be user land or already exist in the OS that people run browsers on=
.<br>


<br>
If someone has a user land implementation of SCTP or TCP, I imagine that mi=
ght change the outcome a bit from previous times. I have not looked at the =
TCP over UPD library Justin mentioned but, at casual glance, I noticed is a=
round 400 lines of code which is very small, which is cool. But given the l=
ines of code in say the BSD TCP stack, it does make me wonder what&#39;s mi=
ssing. That said, I don&#39;t think we need something with the perforce of =
the BSD TCP stack as long as what is there is TCP friendly and robust.<br>


</blockquote>
I believe this is the current URL:<br>
<br>
<a href=3D"http://code.google.com/p/libjingle/source/browse/trunk/talk/sess=
ion/tunnel/pseudotcpchannel.cc" target=3D"_blank">http://code.google.com/p/=
<u></u>libjingle/source/browse/trunk/<u></u>talk/session/tunnel/<u></u>pseu=
dotcpchannel.cc</a> (500 or so lines).<br>


<br>
The real protocol implementation is here:<br>
<a href=3D"http://code.google.com/p/libjingle/source/browse/trunk/talk/p2p/=
base/pseudotcp.cc" target=3D"_blank">http://code.google.com/p/<u></u>libjin=
gle/source/browse/trunk/<u></u>talk/p2p/base/pseudotcp.cc</a><br>
which is another 1000 lines (not counting the various .h files).<br>
<br>
We&#39;re not THAT good at writing compact code :-)<br></blockquote><div><b=
r></div><div>Yes, I was going to send a similar email. There are some defic=
iencies/optimizations for the fact it runs over ICE-UDP - doesn&#39;t do pa=
th MTU discovery, no checksum, no OOB data - but it is based on TCP and imp=
lements slow start and other congestion avoidance mechanisms, and is active=
ly used by a number of applications where reliable transfer is important, s=
uch as file transfer and remote desktop applications.</div>

<div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8=
ex;border-left:1px #ccc solid;padding-left:1ex;">
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<br>
<br>
<br>
<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>
<br>
</blockquote>
<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>

--000e0cd56f26c0b79004a8820279--

From john.elwell@siemens-enterprise.com  Wed Jul 20 08:38:36 2011
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Wed, 20 Jul 2011 17:38:30 +0200
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=20

> -----Original Message-----
> From: rtcweb-bounces@ietf.org=20
> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Magnus Westerlund
> Sent: 19 July 2011 15:28
> To: rtcweb@ietf.org
> Subject: [rtcweb] To multiplex or not!
>=20
> Hi,
>=20
> This email is as an individual contributor.
>=20
> I want to get started on the discussion of the Multiplexing of the
> various protocols over single lower layer transport flow,=20
> such as a UDP
> flow. I will attempt to split up the questions into different emails.
>=20
> The first question I think is reasonably easy to get answered, but I
> think it is time we determine if my belief in the answer is=20
> correct or not.
>=20
> The traffic between two RTCWEB peers from the various components, such
> as RTP sessions, datagram service:
>=20
> a) MUST be sent as Individual flows for each component.
>=20
> b) MUST be multiplexed into a single transport flow.
>=20
> c) SHOULD be multiplexed into a single transport flow, but the RTCWEB
> peer MUST be able to send them as individual flows.
[JRE] As soon as we start talking about what an RTCWEB peer MUST support, t=
he question arises whether this applies to the browser, the script or both.=
 For example, a browser might implement each PeerConnection object over sep=
arate transports, but if the script chooses to use a single PeerConnection =
object for both audio and video it might not be interoperable with a peer t=
hat doesn't support multiplexing.

So whilst I agree that picking something along the lines of c) might be pos=
sible, we would need to define more precisely what we mean before coming to=
 such an agreement.

John

John Elwell
Tel: +44 1908 817801 (office and mobile)
Email: john.elwell@siemens-enterprise.com
http://www.siemens-enterprise.com/uk/

Siemens Enterprise Communications Limited.
Registered office: Brickhill Street, Willen Lake, Milton Keynes, MK15 0DJ.
Registered No: 5903714, England.

Siemens Enterprise Communications Limited is a Trademark Licensee of Siemen=
s AG.



>=20
> I would love if people can indicate their choice or preferences.
>=20
> I personally prefer A as it it is simplest in all aspect=20
> except the NAT
> traversal.
> - It allows for flow based QoS.
> - It is the what the implementation that exist mostly do
> - Signaling protocols that exist support it, no extra functionality
> - People are used to the concept
> - It minimizes the difference to legacy.
>=20
> Thus it is the quickest road to define something with the least formal
> push back and concern over maturity of any solution.
>=20
> The downside with B and C is that we do have to solve the multiplexing
> and get an agreement that gets through all the hurdles.
>=20
> Of these two opens I do prefer C.  Although it results in the extra
> complexities of having both alternatives, it will give us both a
> fallback, flow based QoS and better legacy support.
>=20
> Now it is your time to make your opinion heard!
>=20
> Cheers
>=20
> Magnus Westerlund
>=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
> =

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Subject: Re: [rtcweb] To multiplex or not!
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On 07/20/11 17:38, Elwell, John wrote:
>
>
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org
>> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Magnus Westerlund
>> Sent: 19 July 2011 15:28
>> To: rtcweb@ietf.org
>> Subject: [rtcweb] To multiplex or not!
>>
>> Hi,
>>
>> This email is as an individual contributor.
>>
>> I want to get started on the discussion of the Multiplexing of the
>> various protocols over single lower layer transport flow,
>> such as a UDP
>> flow. I will attempt to split up the questions into different emails.
>>
>> The first question I think is reasonably easy to get answered, but I
>> think it is time we determine if my belief in the answer is
>> correct or not.
>>
>> The traffic between two RTCWEB peers from the various components, such=

>> as RTP sessions, datagram service:
>>
>> a) MUST be sent as Individual flows for each component.
>>
>> b) MUST be multiplexed into a single transport flow.
>>
>> c) SHOULD be multiplexed into a single transport flow, but the RTCWEB
>> peer MUST be able to send them as individual flows.
> [JRE] As soon as we start talking about what an RTCWEB peer MUST suppor=
t, the question arises whether this applies to the browser, the script or=
 both. For example, a browser might implement each PeerConnection object =
over separate transports, but if the script chooses to use a single PeerC=
onnection object for both audio and video it might not be interoperable w=
ith a peer that doesn't support multiplexing.
It's not at all obvious to me how we map PeerConnection objects to RTP=20
sessions. It might be that there can only be a single PeerConnection=20
object to a specific remote peer, and that the PeerConnection object=20
decides internally how to map media streams to RTP sessions (for=20
instance). Or indeed where that mapping ought to be specified.

In the IETF, we may want to concentrate on what the packets should be on =

the wire; in the W3C, we may want to concentrate on making the API=20
powerful enough and easy enough to use - sooner or later, all the pieces =

have to match up, though.

> So whilst I agree that picking something along the lines of c) might be=
 possible, we would need to define more precisely what we mean before com=
ing to such an agreement.
>
> John
>
> John Elwell
> Tel: +44 1908 817801 (office and mobile)
> Email: john.elwell@siemens-enterprise.com
> http://www.siemens-enterprise.com/uk/
>
> Siemens Enterprise Communications Limited.
> Registered office: Brickhill Street, Willen Lake, Milton Keynes, MK15 0=
DJ.
> Registered No: 5903714, England.
>
> Siemens Enterprise Communications Limited is a Trademark Licensee of Si=
emens AG.
>
>
>
>> I would love if people can indicate their choice or preferences.
>>
>> I personally prefer A as it it is simplest in all aspect
>> except the NAT
>> traversal.
>> - It allows for flow based QoS.
>> - It is the what the implementation that exist mostly do
>> - Signaling protocols that exist support it, no extra functionality
>> - People are used to the concept
>> - It minimizes the difference to legacy.
>>
>> Thus it is the quickest road to define something with the least formal=

>> push back and concern over maturity of any solution.
>>
>> The downside with B and C is that we do have to solve the multiplexing=

>> and get an agreement that gets through all the hurdles.
>>
>> Of these two opens I do prefer C.  Although it results in the extra
>> complexities of having both alternatives, it will give us both a
>> fallback, flow based QoS and better legacy support.
>>
>> Now it is your time to make your opinion heard!
>>
>> Cheers
>>
>> Magnus Westerlund
>>
>> ----------------------------------------------------------------------=

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



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Subject: Re: [rtcweb] To multiplex or not!
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> > c) SHOULD be multiplexed into a single transport flow=2C but the RTCWEB
> > peer MUST be able to send them as individual flows.
> [JRE] As soon as we start talking about what an RTCWEB peer MUST support=
=2C the question arises whether this applies to the browser=2C the script o=
r both. For example=2C a browser might implement each PeerConnection object=
 over separate transports=2C but if the script chooses to use a single Peer=
Connection object for both audio and video it might not be interoperable wi=
th a peer that doesn't support multiplexing.
>=20
> So whilst I agree that picking something along the lines of c) might be p=
ossible=2C we would need to define more precisely what we mean before comin=
g to such an agreement.

[BA] .MUST support sending as individual flows makes sense to me=2C but mak=
ing normative statements about multiplexing onto a single transport doesn't=
.   If both sides support multiplexing into a single transport flow=2C then=
 it can be negotiated and used.=20
 		 	   		  =

--_0e6491f5-f097-4c25-a02d-2326213cd974_
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<html>
<head>
<style><!--
.hmmessage P
{
margin:0px=3B
padding:0px
}
body.hmmessage
{
font-size: 10pt=3B
font-family:Tahoma
}
--></style>
</head>
<body class=3D'hmmessage'><div dir=3D'ltr'>
<br><div>&gt=3B &gt=3B c) SHOULD be multiplexed into a single transport flo=
w=2C but the RTCWEB<br>&gt=3B &gt=3B peer MUST be able to send them as indi=
vidual flows.<br>&gt=3B [JRE] As soon as we start talking about what an RTC=
WEB peer MUST support=2C the question arises whether this applies to the br=
owser=2C the script or both. For example=2C a browser might implement each =
PeerConnection object over separate transports=2C but if the script chooses=
 to use a single PeerConnection object for both audio and video it might no=
t be interoperable with a peer that doesn't support multiplexing.<br>&gt=3B=
 <br>&gt=3B So whilst I agree that picking something along the lines of c) =
might be possible=2C we would need to define more precisely what we mean be=
fore coming to such an agreement.<br><br>[BA] .MUST support sending as indi=
vidual flows makes sense to me=2C but making normative statements about mul=
tiplexing onto a single transport doesn't.&nbsp=3B&nbsp=3B If both sides su=
pport multiplexing into a single transport flow=2C then it can be negotiate=
d and used. <br></div> 		 	   		  </div></body>
</html>=

--_0e6491f5-f097-4c25-a02d-2326213cd974_--

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Subject: [rtcweb] DSCP and media
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Many home NATs support DSCP based QoS and it does help. I think that the =
IETF should recommend that some default DSCP for audio and video from =
the browser as well as suggest there should be an API to set the DSCP =
for different media steam.=20

Cullen <with my individual contributor hat on>


From dzonatas@gmail.com  Wed Jul 20 11:34:53 2011
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On 07/20/2011 10:59 AM, Cullen Jennings wrote:
> Many home NATs support DSCP based QoS and it does help. I think that the IETF should recommend that some default DSCP for audio and video from the browser as well as suggest there should be an API to set the DSCP for different media steam.
>
> Cullen<with my individual contributor hat on>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>    

XML entities with DNA-like name sequences, I knew there was some use for 
these non-aspects. If DSCP is t.co or G.co aware, then 
&co.G.sequence...; is possible for speedy QoS. I haven't found any 
reason for AOT and JIT namespaces to conflict in such entities. It's 
just not as automated as some desire, yet could be universally 
coincidental. No need to make these "far" synopsis any harder for 
internationalization of such namespaces.

Glad ICE continues to support direct IPv4 override, hypervisors still 
need this mode for games.

-- 
--- http://twitter.com/Dzonatas_Sol ---
Web Development, Software Engineering
Ag-Biotech, Virtual Reality, Consultant


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References: <CAMKM2LzpVcS9jjXjfffuXy+YQmjZXbdaSJBp+O22nLd4N2KAvg@mail.gmail.com> <CA4AFBFA.1C4B5%henry.sinnreich@gmail.com> <CAOJ7v-3R0PeUSdVZ0n7AE08J=UjYMJqJ+4-Vkbj7w0qs0u=Rgw@mail.gmail.com> <4E25B2BA.8000004@jesup.org> <4E25B893.6010200@jitsi.org> <CAMKM2Lz_sgrmHVpuuGfyukmxdO-+qaWjyOhQzU6vTSDaAwytxQ@mail.gmail.com> <4E26A354.2020306@ericsson.com> <4E26F460.3080007@jesup.org>
From: Serge Lachapelle <sergel@google.com>
Date: Wed, 20 Jul 2011 21:11:17 +0200
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>
>
> For other cases (file xfer), yes you can use those - but the problem is
> that when you want to transfer to the person you're chatting with, you would
> need to use another service, agree on it, deal with the
> semantics/accounts/limits/etc for that service, etc.  A horrible user
> experience.  Even if it was "hidden" by the application, it would be
> problematic and subject to breakage continually.


If I am building a 1:1 or hangout style application connecting my users, I
would simply integrate it or add the functionality in the web UI I am
building. This is very easy to do with web services these days.

/Serge

--000e0cd348baf86d5504a885006f
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<div class=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"margi=
n:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"><br>
For other cases (file xfer), yes you can use those - but the problem is tha=
t when you want to transfer to the person you&#39;re chatting with, you wou=
ld need to use another service, agree on it, deal with the semantics/accoun=
ts/limits/etc for that service, etc. =A0A horrible user experience. =A0Even=
 if it was &quot;hidden&quot; by the application, it would be problematic a=
nd subject to breakage continually.</blockquote>

<div><br></div><div>If I am building a 1:1 or hangout style application con=
necting my users, I would simply integrate it or add the functionality in t=
he web UI I am building. This is very easy to do with web services these da=
ys.</div>

<div><br></div><div>/Serge</div></div>

--000e0cd348baf86d5504a885006f--

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From: Justin Uberti <juberti@google.com>
Date: Wed, 20 Jul 2011 15:39:04 -0400
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Can you go into more detail about what these NATs do with the DSCP markings?

I'm not convinced we want to expose a generic API for this (too easy to
misuse), but I think it definitely makes sense for implementations to set
appropriate markings on audio and video packets, including different levels
on individual video packets when using hierarchical prediction structures.

On Wed, Jul 20, 2011 at 1:59 PM, Cullen Jennings <fluffy@cisco.com> wrote:

>
> Many home NATs support DSCP based QoS and it does help. I think that the
> IETF should recommend that some default DSCP for audio and video from the
> browser as well as suggest there should be an API to set the DSCP for
> different media steam.
>
> Cullen <with my individual contributor hat on>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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Can you go into more detail about what these NATs do with the DSCP markings=
?<div><br></div><div>I&#39;m not convinced we want to expose a generic API =
for this (too easy to misuse), but I think it definitely makes sense for im=
plementations to set appropriate markings on audio and video packets, inclu=
ding different levels on individual video packets when using hierarchical p=
rediction structures.<br>

<br><div class=3D"gmail_quote">On Wed, Jul 20, 2011 at 1:59 PM, Cullen Jenn=
ings <span dir=3D"ltr">&lt;<a href=3D"mailto:fluffy@cisco.com">fluffy@cisco=
.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"ma=
rgin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">

<br>
Many home NATs support DSCP based QoS and it does help. I think that the IE=
TF should recommend that some default DSCP for audio and video from the bro=
wser as well as suggest there should be an API to set the DSCP for differen=
t media steam.<br>


<br>
Cullen &lt;with my individual contributor hat on&gt;<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>

--00235447193cbacfd004a885633a--

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From: Justin Uberti <juberti@google.com>
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On Wed, Jul 20, 2011 at 10:18 AM, Bernard Aboba
<bernard_aboba@hotmail.com>wrote:

>
> > Timing is a real problem. If we want to do multiplexing in RTCWEB 1.0 we
> > need to pick what is ready and available. We can't go around proposing
> > new things that are complicated. Changing the SSRC field semantics in
> > RTP/RTCP is a complicated thing.
>
> [BA] We need to ask ourselves "Does this really need to be in the critical
> path for RTCWEB 1.0"?
> As I understand it, this would be a negotiated feature, and presumably all
> implementations
> would be required to support RTP/RTCP as defined in RFC 3550 and suitable
> profiles.
> So from my perspective, I'd prefer to see this handled as an extension on
> its own path, not a
> mandatory-to-implement feature in 1.0.
>

This is where I've ended up on this topic. We can easily multiplex multiple
RTP sources (of the same type) over a single RTP session using SSRC. We can
also mux RTCP over the RTP session using RTCP mux. So, for an arbitrary
video call, we have just 2 RTP sessions/NAT bindings.

Is it worth going the extra mile to get down to 1 in v1.0, given the lack of
consensus that exists right now? Is there even a compelling argument to do
so?

>
>
> > Isn't that gateway going to be present anyway in most case. I think the
> > ICE connectivity checks are one thing that will force a gateway in many
> > cases. In the cases it isn't needed, then you can skip the multiplexing
> > in that case to reach the legacy.
>
> [BA] Just because we're likely to require a gateway doesn't mean we should
> heap more and more things on the camel's back.  We're already talking about
> ICE, shims, transcoding, changes to RFC 3550....  that's a lot of
> functionality
> for the "gateway".  So I agree with Jonathan about the disadvantages of a
> shim
> (or DCCP for that matter).
>
>
> > The big difference is that we either have a well reviewed and by the
> > community and IESG approved specification in DCCP. Or we create
> > something our selves that needs to go through all that review and will
> > be even less tested than DCCP is.
>
> [BA] Deployment counts, too.  How widely deployed is DCCP for realtime
> uses?
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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<br><br><div class=3D"gmail_quote">On Wed, Jul 20, 2011 at 10:18 AM, Bernar=
d Aboba <span dir=3D"ltr">&lt;<a href=3D"mailto:bernard_aboba@hotmail.com" =
target=3D"_blank">bernard_aboba@hotmail.com</a>&gt;</span> wrote:<br><block=
quote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc=
 solid;padding-left:1ex">






<div><div dir=3D"ltr">
<br><div><div>&gt; Timing is a real problem. If we want to do multiplexing =
in RTCWEB 1.0 we<br>&gt; need to pick what is ready and available. We can&#=
39;t go around proposing<br>&gt; new things that are complicated. Changing =
the SSRC field semantics in<br>


&gt; RTP/RTCP is a complicated thing.<br><br></div>[BA] We need to ask ours=
elves &quot;Does this really need to be in the critical path for RTCWEB 1.0=
&quot;?=C2=A0 <br>As I understand it, this would be a negotiated feature, a=
nd presumably all implementations<br>


would be required to support RTP/RTCP as defined in RFC 3550 and suitable p=
rofiles. <br>So from my perspective, I&#39;d prefer to see this handled as =
an extension on its own path, not a <br>mandatory-to-implement feature in 1=
.0.=C2=A0=C2=A0 <br>


</div></div></div></blockquote><div><br></div><div>This is where I&#39;ve e=
nded up on this topic. We can easily multiplex multiple RTP sources (of the=
 same type) over a single RTP session using SSRC. We can also mux RTCP over=
 the RTP session using RTCP mux. So, for an arbitrary video call, we have j=
ust 2 RTP sessions/NAT bindings.=C2=A0</div>


<div><br></div><div>Is it worth going the extra mile to get down to 1 in v1=
.0, given the lack of consensus that exists right now? Is there even a comp=
elling argument to do so?</div><blockquote class=3D"gmail_quote" style=3D"m=
argin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">


<div><div dir=3D"ltr"><div><div><br><br>&gt; Isn&#39;t that gateway going t=
o be present anyway in most case. I think the<br>&gt; ICE connectivity chec=
ks are one thing that will force a gateway in many<br>&gt; cases. In the ca=
ses it isn&#39;t needed, then you can skip the multiplexing<br>


&gt; in that case to reach the legacy.<br><br></div>[BA] Just because we&#3=
9;re likely to require a gateway doesn&#39;t mean we should<br>heap more an=
d more things on the camel&#39;s back.=C2=A0 We&#39;re already talking abou=
t<br>


ICE, shims, transcoding, changes to RFC 3550....=C2=A0 that&#39;s a lot of =
functionality<br>for the &quot;gateway&quot;.=C2=A0 So I agree with Jonatha=
n about the disadvantages of a shim<br>(or DCCP for that matter). <br><div>
<br><br>&gt; The big difference is that we either have a well reviewed and =
by the<br>&gt; community and IESG approved specification in DCCP. Or we cre=
ate<br>&gt; something our selves that needs to go through all that review a=
nd will<br>


&gt; be even less tested than DCCP is.<br><br></div>[BA] Deployment counts,=
 too.=C2=A0 How widely deployed is DCCP for realtime uses? <br><br><br></di=
v> 		 	   		  </div></div>
<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br>

--0016364ec992144f4704a8857e3b--

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On Jul 20, 2011, at 12:46 PM, Justin Uberti wrote:

>=20
> This is where I've ended up on this topic. We can easily multiplex =
multiple RTP sources (of the same type) over a single RTP session using =
SSRC. We can also mux RTCP over the RTP session using RTCP mux. So, for =
an arbitrary video call, we have just 2 RTP sessions/NAT bindings.=20
>=20
> Is it worth going the extra mile to get down to 1 in v1.0, given the =
lack of consensus that exists right now? Is there even a compelling =
argument to do so?

Yes and yes.

I really can't understand why, if we can multiplex 3 totally different =
types of video streams over the same RTP session using SSRC (with wildly =
different bit-rates, inter-packet times, etc.) we can't also multiplex =
audio and video. Not a single argument that has been presented has =
convinced me, and getting from 2 to 1 is a *50%* reduction in NAT port =
utilization. (And a significant reduction in the number of "strange" =
failure modes, where one traversal worked and the other didn't.)

Matthew Kaufman


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From: Justin Uberti <juberti@google.com>
Date: Wed, 20 Jul 2011 15:59:18 -0400
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--00235447193c16cea904a885acff
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On Wed, Jul 20, 2011 at 3:51 PM, Matthew Kaufman
<matthew.kaufman@skype.net>wrote:

>
> On Jul 20, 2011, at 12:46 PM, Justin Uberti wrote:
>
> >
> > This is where I've ended up on this topic. We can easily multiplex
> multiple RTP sources (of the same type) over a single RTP session using
> SSRC. We can also mux RTCP over the RTP session using RTCP mux. So, for an
> arbitrary video call, we have just 2 RTP sessions/NAT bindings.
> >
> > Is it worth going the extra mile to get down to 1 in v1.0, given the lack
> of consensus that exists right now? Is there even a compelling argument to
> do so?
>
> Yes and yes.
>
> I really can't understand why, if we can multiplex 3 totally different
> types of video streams over the same RTP session using SSRC (with wildly
> different bit-rates, inter-packet times, etc.) we can't also multiplex audio
> and video. Not a single argument that has been presented has convinced me,
> and getting from 2 to 1 is a *50%* reduction in NAT port utilization. (And a
> significant reduction in the number of "strange" failure modes, where one
> traversal worked and the other didn't.)
>

Clearly we can, but we need to do additional work to get there. This work
will take time and delay v1.0, so we need a pretty convincing rationale to
do so.

I concur that solving strange failure modes is significant. But with regard
to NAT port utilization, this ignores the utilization by the browser for
HTTP; 50% reduction seems possible in theory only. (Here I am essentially
restating 2.2.2 from
https://datatracker.ietf.org/doc/draft-perkins-rtcweb-rtp-usage/?include_text=1,
which I haven't yet seen a clear response to.)





> Matthew Kaufman
>
>

--00235447193c16cea904a885acff
Content-Type: text/html; charset=UTF-8
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<br><br><div class=3D"gmail_quote">On Wed, Jul 20, 2011 at 3:51 PM, Matthew=
 Kaufman <span dir=3D"ltr">&lt;<a href=3D"mailto:matthew.kaufman@skype.net"=
 target=3D"_blank">matthew.kaufman@skype.net</a>&gt;</span> wrote:<br><bloc=
kquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #cc=
c solid;padding-left:1ex">


<div><br>
On Jul 20, 2011, at 12:46 PM, Justin Uberti wrote:<br>
<br>
&gt;<br>
&gt; This is where I&#39;ve ended up on this topic. We can easily multiplex=
 multiple RTP sources (of the same type) over a single RTP session using SS=
RC. We can also mux RTCP over the RTP session using RTCP mux. So, for an ar=
bitrary video call, we have just 2 RTP sessions/NAT bindings.<br>



&gt;<br>
&gt; Is it worth going the extra mile to get down to 1 in v1.0, given the l=
ack of consensus that exists right now? Is there even a compelling argument=
 to do so?<br>
<br>
</div>Yes and yes.<br>
<br>
I really can&#39;t understand why, if we can multiplex 3 totally different =
types of video streams over the same RTP session using SSRC (with wildly di=
fferent bit-rates, inter-packet times, etc.) we can&#39;t also multiplex au=
dio and video. Not a single argument that has been presented has convinced =
me, and getting from 2 to 1 is a *50%* reduction in NAT port utilization. (=
And a significant reduction in the number of &quot;strange&quot; failure mo=
des, where one traversal worked and the other didn&#39;t.)<br>

</blockquote><div><br></div><div>Clearly we can, but we need to do addition=
al work to get there. This work will take time and delay v1.0, so we need a=
 pretty convincing rationale to do so.</div><div><br></div><div>I concur th=
at solving strange failure modes is significant. But with regard to NAT por=
t utilization, this ignores the utilization by the browser for HTTP; 50% re=
duction seems possible in theory only. (Here I am essentially restating 2.2=
.2 from=C2=A0<a href=3D"https://datatracker.ietf.org/doc/draft-perkins-rtcw=
eb-rtp-usage/?include_text=3D1">https://datatracker.ietf.org/doc/draft-perk=
ins-rtcweb-rtp-usage/?include_text=3D1</a>, which I haven&#39;t yet seen a =
clear response to.)</div>

<div><br></div><div><br></div><div><br></div><div><br></div><blockquote cla=
ss=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;pa=
dding-left:1ex">

<font color=3D"#888888"><br>
Matthew Kaufman<br>
<br>
</font></blockquote></div><br>

--00235447193c16cea904a885acff--

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From: Serge Lachapelle <sergel@google.com>
Date: Wed, 20 Jul 2011 22:00:32 +0200
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Just to put the file transfer debate to bed from my end, I am not against
reliable channels, in fact, a ton of innovation can come from it when made
available to browsers.

I just want to avoid any re-engineering of what browsers offer today as much
as possible.

/Serge

On Wed, Jul 20, 2011 at 21:11, Serge Lachapelle <sergel@google.com> wrote:

>
>> For other cases (file xfer), yes you can use those - but the problem is
>> that when you want to transfer to the person you're chatting with, you would
>> need to use another service, agree on it, deal with the
>> semantics/accounts/limits/etc for that service, etc.  A horrible user
>> experience.  Even if it was "hidden" by the application, it would be
>> problematic and subject to breakage continually.
>
>
> If I am building a 1:1 or hangout style application connecting my users, I
> would simply integrate it or add the functionality in the web UI I am
> building. This is very easy to do with web services these days.
>
> /Serge
>



-- 
Serge Lachapelle | Product Manager | sergel@google.com | +46 732 01 22 32
Google Sweden AB | Kungsbron 2, SE-111 22 Stockholm | Org. nr. 556656-6880

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Just to put the file transfer debate to bed from my end, I am not against r=
eliable channels, in fact, a ton of innovation can come from it when made a=
vailable to browsers.<div><br></div><div>I just want to avoid any re-engine=
ering of what browsers offer today as much as possible.<div>

<div><br></div><div>/Serge<br><br><div class=3D"gmail_quote">On Wed, Jul 20=
, 2011 at 21:11, Serge Lachapelle <span dir=3D"ltr">&lt;<a href=3D"mailto:s=
ergel@google.com">sergel@google.com</a>&gt;</span> wrote:<br><blockquote cl=
ass=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;p=
adding-left:1ex;">

<div class=3D"gmail_quote"><div class=3D"im"><blockquote class=3D"gmail_quo=
te" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"=
><br>
For other cases (file xfer), yes you can use those - but the problem is tha=
t when you want to transfer to the person you&#39;re chatting with, you wou=
ld need to use another service, agree on it, deal with the semantics/accoun=
ts/limits/etc for that service, etc. =A0A horrible user experience. =A0Even=
 if it was &quot;hidden&quot; by the application, it would be problematic a=
nd subject to breakage continually.</blockquote>


<div><br></div></div><div>If I am building a 1:1 or hangout style applicati=
on connecting my users, I would simply integrate it or add the functionalit=
y in the web UI I am building. This is very easy to do with web services th=
ese days.</div>


<div><br></div><font color=3D"#888888"><div>/Serge</div></font></div>
</blockquote></div><br><br clear=3D"all"><br>-- <br><meta charset=3D"utf-8"=
><div><meta charset=3D"utf-8"><div style=3D"line-height: 1.5em; padding-top=
: 10px; margin-top: 10px; color: rgb(85, 85, 85); font-family: sans-serif; =
font-size: small; ">

<span style=3D"border-top-width: 2px; border-right-width: 0px; border-botto=
m-width: 0px; border-left-width: 0px; border-top-style: solid; border-right=
-style: solid; border-bottom-style: solid; border-left-style: solid; border=
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bottom-color: rgb(213, 15, 37); border-left-color: rgb(213, 15, 37); paddin=
g-top: 2px; margin-top: 2px; ">Serge Lachapelle=A0|</span><span style=3D"bo=
rder-top-width: 2px; border-right-width: 0px; border-bottom-width: 0px; bor=
der-left-width: 0px; border-top-style: solid; border-right-style: solid; bo=
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gb(51, 105, 232); border-left-color: rgb(51, 105, 232); padding-top: 2px; m=
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dth: 2px; border-right-width: 0px; border-bottom-width: 0px; border-left-wi=
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8, 178, 17); padding-top: 2px; margin-top: 2px; ">=A0+46 732 01 22 32</span=
></div>

</div><font class=3D"Apple-style-span" color=3D"#999999" size=3D"1">Google =
Sweden AB | Kungsbron 2, SE-111 22 Stockholm | Org. nr. 556656-6880=A0</fon=
t><br><br><font class=3D"Apple-style-span" color=3D"#999999" size=3D"1">App=
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Subject: Re: [rtcweb] realiable data service
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Agreed.=20

To make this a bit more concrete.  From your perspective=2C what is the pre=
ferred mechanism for file transfer in XMPP=2C and what would it take to sup=
port this in a browser?=20

From: sergel@google.com
Date: Wed=2C 20 Jul 2011 22:00:32 +0200
To: randell1@jesup.org
CC: rtcweb@ietf.org
Subject: Re: [rtcweb] realiable data service

Just to put the file transfer debate to bed from my end=2C I am not against=
 reliable channels=2C in fact=2C a ton of innovation can come from it when =
made available to browsers.
I just want to avoid any re-engineering of what browsers offer today as muc=
h as possible.


/Serge

On Wed=2C Jul 20=2C 2011 at 21:11=2C Serge Lachapelle <sergel@google.com> w=
rote:




For other cases (file xfer)=2C yes you can use those - but the problem is t=
hat when you want to transfer to the person you're chatting with=2C you wou=
ld need to use another service=2C agree on it=2C deal with the semantics/ac=
counts/limits/etc for that service=2C etc.  A horrible user experience.  Ev=
en if it was "hidden" by the application=2C it would be problematic and sub=
ject to breakage continually.



If I am building a 1:1 or hangout style application connecting my users=2C =
I would simply integrate it or add the functionality in the web UI I am bui=
lding. This is very easy to do with web services these days.



/Serge


--=20


Serge Lachapelle | Product Manager | sergel@google.com | +46 732 01 22 32

Google Sweden AB | Kungsbron 2=2C SE-111 22 Stockholm | Org. nr. 556656-688=
0=20

Apparently=2C this footer is required in Europe. Apologies. This email may =
be confidential or privileged.  If you received this communication by mista=
ke=2C please don't forward it to anyone else=2Cplease erase all copies and =
attachments=2C and please let me know that it went to the wrong person.  Th=
anks.





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

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<body class=3D'hmmessage'><div dir=3D'ltr'>
Agreed. <br><br>To make this a bit more concrete.&nbsp=3B From your perspec=
tive=2C what is the preferred mechanism for file transfer in XMPP=2C and wh=
at would it take to support this in a browser? <br><br><div><hr id=3D"stopS=
pelling">From: sergel@google.com<br>Date: Wed=2C 20 Jul 2011 22:00:32 +0200=
<br>To: randell1@jesup.org<br>CC: rtcweb@ietf.org<br>Subject: Re: [rtcweb] =
realiable data service<br><br>Just to put the file transfer debate to bed f=
rom my end=2C I am not against reliable channels=2C in fact=2C a ton of inn=
ovation can come from it when made available to browsers.<div><br></div><di=
v>I just want to avoid any re-engineering of what browsers offer today as m=
uch as possible.<div>

<div><br></div><div>/Serge<br><br><div class=3D"ecxgmail_quote">On Wed=2C J=
ul 20=2C 2011 at 21:11=2C Serge Lachapelle <span dir=3D"ltr">&lt=3B<a href=
=3D"mailto:sergel@google.com">sergel@google.com</a>&gt=3B</span> wrote:<br>=
<blockquote class=3D"ecxgmail_quote" style=3D"border-left:1px #ccc solid=3B=
padding-left:1ex">

<div class=3D"ecxgmail_quote"><div class=3D"ecxim"><blockquote class=3D"ecx=
gmail_quote" style=3D"border-left:1px #ccc solid=3Bpadding-left:1ex"><br>
For other cases (file xfer)=2C yes you can use those - but the problem is t=
hat when you want to transfer to the person you're chatting with=2C you wou=
ld need to use another service=2C agree on it=2C deal with the semantics/ac=
counts/limits/etc for that service=2C etc. &nbsp=3BA horrible user experien=
ce. &nbsp=3BEven if it was "hidden" by the application=2C it would be probl=
ematic and subject to breakage continually.</blockquote>


<div><br></div></div><div>If I am building a 1:1 or hangout style applicati=
on connecting my users=2C I would simply integrate it or add the functional=
ity in the web UI I am building. This is very easy to do with web services =
these days.</div>


<div><br></div><font color=3D"#888888"><div>/Serge</div></font></div>
</blockquote></div><br><br clear=3D"all"><br>-- <br><div><div style=3D"line=
-height:1.5em=3Bpadding-top:10px=3Bcolor:rgb(85=2C 85=2C 85)=3Bfont-family:=
sans-serif=3Bfont-size:small">

<span style=3D"border-top-width:2px=3Bborder-right-width:0px=3Bborder-botto=
m-width:0px=3Bborder-left-width:0px=3Bborder-top-style:solid=3Bborder-right=
-style:solid=3Bborder-bottom-style:solid=3Bborder-left-style:solid=3Bborder=
-top-color:rgb(213=2C 15=2C 37)=3Bborder-right-color:rgb(213=2C 15=2C 37)=
=3Bborder-bottom-color:rgb(213=2C 15=2C 37)=3Bborder-left-color:rgb(213=2C =
15=2C 37)=3Bpadding-top:2px">Serge Lachapelle&nbsp=3B|</span><span style=3D=
"border-top-width:2px=3Bborder-right-width:0px=3Bborder-bottom-width:0px=3B=
border-left-width:0px=3Bborder-top-style:solid=3Bborder-right-style:solid=
=3Bborder-bottom-style:solid=3Bborder-left-style:solid=3Bborder-top-color:r=
gb(51=2C 105=2C 232)=3Bborder-right-color:rgb(51=2C 105=2C 232)=3Bborder-bo=
ttom-color:rgb(51=2C 105=2C 232)=3Bborder-left-color:rgb(51=2C 105=2C 232)=
=3Bpadding-top:2px">&nbsp=3BProduct Manager&nbsp=3B|</span><span style=3D"b=
order-top-width:2px=3Bborder-right-width:0px=3Bborder-bottom-width:0px=3Bbo=
rder-left-width:0px=3Bborder-top-style:solid=3Bborder-right-style:solid=3Bb=
order-bottom-style:solid=3Bborder-left-style:solid=3Bborder-top-color:rgb(0=
=2C 153=2C 57)=3Bborder-right-color:rgb(0=2C 153=2C 57)=3Bborder-bottom-col=
or:rgb(0=2C 153=2C 57)=3Bborder-left-color:rgb(0=2C 153=2C 57)=3Bpadding-to=
p:2px">&nbsp=3B<a href=3D"mailto:sergel@google.com">sergel@google.com</a>&n=
bsp=3B|</span><span style=3D"border-top-width:2px=3Bborder-right-width:0px=
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olor:rgb(238=2C 178=2C 17)=3Bpadding-top:2px">&nbsp=3B+46 732 01 22 32</spa=
n></div>

</div><font class=3D"ecxApple-style-span" color=3D"#999999" size=3D"1">Goog=
le Sweden AB | Kungsbron 2=2C SE-111 22 Stockholm | Org. nr. 556656-6880&nb=
sp=3B</font><br><br><font class=3D"ecxApple-style-span" color=3D"#999999" s=
ize=3D"1">Apparently=2C this footer is required in Europe. Apologies. This =
email may be confidential or privileged. &nbsp=3BIf you received this commu=
nication by mistake=2C please don't forward it to anyone else=2Cplease eras=
e all copies and attachments=2C and please let me know that it went to the =
wrong person. &nbsp=3BThanks.</font><br>


</div></div></div>
<br>_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb</div> 		 	   		  </div></body>
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Subject: Re: [rtcweb] DSCP and media
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I think we're talking about things like support of WFA WMM.  Not a NAT func=
tion per se.=20

Some info on how this is implemented in the OS is available here.=20

Whether this actually accomplishes anything in practice is a subject of deb=
ate.=20

From: juberti@google.com
Date: Wed=2C 20 Jul 2011 15:39:04 -0400
To: fluffy@cisco.com
CC: rtcweb@ietf.org
Subject: Re: [rtcweb] DSCP and media

Can you go into more detail about what these NATs do with the DSCP markings=
?
I'm not convinced we want to expose a generic API for this (too easy to mis=
use)=2C but I think it definitely makes sense for implementations to set ap=
propriate markings on audio and video packets=2C including different levels=
 on individual video packets when using hierarchical prediction structures.



On Wed=2C Jul 20=2C 2011 at 1:59 PM=2C Cullen Jennings <fluffy@cisco.com> w=
rote:




Many home NATs support DSCP based QoS and it does help. I think that the IE=
TF should recommend that some default DSCP for audio and video from the bro=
wser as well as suggest there should be an API to set the DSCP for differen=
t media steam.





Cullen <with my individual contributor hat on>



_______________________________________________

rtcweb mailing list

rtcweb@ietf.org

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




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

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<body class=3D'hmmessage'><div dir=3D'ltr'>
I think we're talking about things like support of WFA WMM.&nbsp=3B Not a N=
AT function per se. <br><br>Some info on how this is implemented in the OS =
is available <a href=3D"http://blogs.msdn.com/b/wndp/archive/2006/06/28/650=
363.aspx">here</a>. <br><br>Whether this actually accomplishes anything in =
practice<a href=3D"http://www.smallnetbuilder.com/wireless/wireless-feature=
s/30837-does-wi-fi-multimedia-wmm-really-do-anything-part-3?start=3D1"> is =
a subject of debate</a>. <br><br><div><hr id=3D"stopSpelling">From: juberti=
@google.com<br>Date: Wed=2C 20 Jul 2011 15:39:04 -0400<br>To: fluffy@cisco.=
com<br>CC: rtcweb@ietf.org<br>Subject: Re: [rtcweb] DSCP and media<br><br>C=
an you go into more detail about what these NATs do with the DSCP markings?=
<div><br></div><div>I'm not convinced we want to expose a generic API for t=
his (too easy to misuse)=2C but I think it definitely makes sense for imple=
mentations to set appropriate markings on audio and video packets=2C includ=
ing different levels on individual video packets when using hierarchical pr=
ediction structures.<br>

<br><div class=3D"ecxgmail_quote">On Wed=2C Jul 20=2C 2011 at 1:59 PM=2C Cu=
llen Jennings <span dir=3D"ltr">&lt=3B<a href=3D"mailto:fluffy@cisco.com">f=
luffy@cisco.com</a>&gt=3B</span> wrote:<br><blockquote class=3D"ecxgmail_qu=
ote" style=3D"border-left:1px #ccc solid=3Bpadding-left:1ex">

<br>
Many home NATs support DSCP based QoS and it does help. I think that the IE=
TF should recommend that some default DSCP for audio and video from the bro=
wser as well as suggest there should be an API to set the DSCP for differen=
t media steam.<br>


<br>
Cullen &lt=3Bwith my individual contributor hat on&gt=3B<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>
<br>_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb</div> 		 	   		  </div></body>
</html>=

--_af88799b-417d-40e8-95f9-e104fdc23fb8_--

From Bala@vidyo.com  Wed Jul 20 13:57:41 2011
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From: Bala Pitchandi <Bala@vidyo.com>
To: Matthew Kaufman <matthew.kaufman@skype.net>, Justin Uberti <juberti@google.com>
Date: Wed, 20 Jul 2011 16:57:27 -0400
Thread-Topic: [rtcweb] How to multiplex between peers
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Subject: Re: [rtcweb] How to multiplex between peers
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+1 for sending audio & video (and any other type of media, for that matter)=
 on a single RTP session. With RTCP mux, that means all media can be sent o=
n a single UDP port.=20

-- Bala


-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Matthew Kaufman
Sent: Wednesday, July 20, 2011 3:51 PM
To: Justin Uberti
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] How to multiplex between peers


On Jul 20, 2011, at 12:46 PM, Justin Uberti wrote:

>=20
> This is where I've ended up on this topic. We can easily multiplex multip=
le RTP sources (of the same type) over a single RTP session using SSRC. We =
can also mux RTCP over the RTP session using RTCP mux. So, for an arbitrary=
 video call, we have just 2 RTP sessions/NAT bindings.=20
>=20
> Is it worth going the extra mile to get down to 1 in v1.0, given the lack=
 of consensus that exists right now? Is there even a compelling argument to=
 do so?

Yes and yes.

I really can't understand why, if we can multiplex 3 totally different type=
s of video streams over the same RTP session using SSRC (with wildly differ=
ent bit-rates, inter-packet times, etc.) we can't also multiplex audio and =
video. Not a single argument that has been presented has convinced me, and =
getting from 2 to 1 is a *50%* reduction in NAT port utilization. (And a si=
gnificant reduction in the number of "strange" failure modes, where one tra=
versal worked and the other didn't.)

Matthew Kaufman

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

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--00235447193c625e9d04a886a7b2
Content-Type: text/plain; charset=UTF-8

OK, that's what I suspected.

Ensuring that this is handled properly by the RTCWEB platform can only help
things.

On Wed, Jul 20, 2011 at 4:43 PM, Bernard Aboba <bernard_aboba@hotmail.com>wrote:

>  I think we're talking about things like support of WFA WMM.  Not a NAT
> function per se.
>
> Some info on how this is implemented in the OS is available here<http://blogs.msdn.com/b/wndp/archive/2006/06/28/650363.aspx>.
>
>
> Whether this actually accomplishes anything in practice is a subject of
> debate<http://www.smallnetbuilder.com/wireless/wireless-features/30837-does-wi-fi-multimedia-wmm-really-do-anything-part-3?start=1>.
>
>
> ------------------------------
> From: juberti@google.com
> Date: Wed, 20 Jul 2011 15:39:04 -0400
> To: fluffy@cisco.com
> CC: rtcweb@ietf.org
> Subject: Re: [rtcweb] DSCP and media
>
>
> Can you go into more detail about what these NATs do with the DSCP
> markings?
>
> I'm not convinced we want to expose a generic API for this (too easy to
> misuse), but I think it definitely makes sense for implementations to set
> appropriate markings on audio and video packets, including different levels
> on individual video packets when using hierarchical prediction structures.
>
> On Wed, Jul 20, 2011 at 1:59 PM, Cullen Jennings <fluffy@cisco.com> wrote:
>
>
> Many home NATs support DSCP based QoS and it does help. I think that the
> IETF should recommend that some default DSCP for audio and video from the
> browser as well as suggest there should be an API to set the DSCP for
> different media steam.
>
> Cullen <with my individual contributor hat on>
>
> _______________________________________________
> 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
>

--00235447193c625e9d04a886a7b2
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OK, that&#39;s what I suspected.<div><br></div><div>Ensuring that this is h=
andled properly by the RTCWEB platform can only help things.<br><br><div cl=
ass=3D"gmail_quote">On Wed, Jul 20, 2011 at 4:43 PM, Bernard Aboba <span di=
r=3D"ltr">&lt;<a href=3D"mailto:bernard_aboba@hotmail.com">bernard_aboba@ho=
tmail.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex;">



<div><div dir=3D"ltr">
I think we&#39;re talking about things like support of WFA WMM.=C2=A0 Not a=
 NAT function per se. <br><br>Some info on how this is implemented in the O=
S is available <a href=3D"http://blogs.msdn.com/b/wndp/archive/2006/06/28/6=
50363.aspx" target=3D"_blank">here</a>. <br>

<br>Whether this actually accomplishes anything in practice<a href=3D"http:=
//www.smallnetbuilder.com/wireless/wireless-features/30837-does-wi-fi-multi=
media-wmm-really-do-anything-part-3?start=3D1" target=3D"_blank"> is a subj=
ect of debate</a>. <br>

<br><div><hr>From: <a href=3D"mailto:juberti@google.com" target=3D"_blank">=
juberti@google.com</a><br>Date: Wed, 20 Jul 2011 15:39:04 -0400<br>To: <a h=
ref=3D"mailto:fluffy@cisco.com" target=3D"_blank">fluffy@cisco.com</a><br>C=
C: <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a>=
<br>

Subject: Re: [rtcweb] DSCP and media<div><div></div><div class=3D"h5"><br><=
br>Can you go into more detail about what these NATs do with the DSCP marki=
ngs?<div><br></div><div>I&#39;m not convinced we want to expose a generic A=
PI for this (too easy to misuse), but I think it definitely makes sense for=
 implementations to set appropriate markings on audio and video packets, in=
cluding different levels on individual video packets when using hierarchica=
l prediction structures.<br>



<br><div>On Wed, Jul 20, 2011 at 1:59 PM, Cullen Jennings <span dir=3D"ltr"=
>&lt;<a href=3D"mailto:fluffy@cisco.com" target=3D"_blank">fluffy@cisco.com=
</a>&gt;</span> wrote:<br><blockquote style=3D"border-left:1px #ccc solid;p=
adding-left:1ex">



<br>
Many home NATs support DSCP based QoS and it does help. I think that the IE=
TF should recommend that some default DSCP for audio and video from the bro=
wser as well as suggest there should be an API to set the DSCP for differen=
t media steam.<br>




<br>
Cullen &lt;with my individual contributor hat on&gt;<br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div><br></div>
<br></div></div>_______________________________________________
rtcweb mailing list
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a></div> 		 	   		  </div></d=
iv>
</blockquote></div><br></div>

--00235447193c625e9d04a886a7b2--

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On Wed, Jul 20, 2011 at 3:59 PM, Justin Uberti <juberti@google.com> wrote:

>
>
> On Wed, Jul 20, 2011 at 3:51 PM, Matthew Kaufman <
> matthew.kaufman@skype.net> wrote:
>
>>
>> On Jul 20, 2011, at 12:46 PM, Justin Uberti wrote:
>>
>> >
>> > This is where I've ended up on this topic. We can easily multiplex
>> multiple RTP sources (of the same type) over a single RTP session using
>> SSRC. We can also mux RTCP over the RTP session using RTCP mux. So, for an
>> arbitrary video call, we have just 2 RTP sessions/NAT bindings.
>> >
>> > Is it worth going the extra mile to get down to 1 in v1.0, given the
>> lack of consensus that exists right now? Is there even a compelling argument
>> to do so?
>>
>> Yes and yes.
>>
>> I really can't understand why, if we can multiplex 3 totally different
>> types of video streams over the same RTP session using SSRC (with wildly
>> different bit-rates, inter-packet times, etc.) we can't also multiplex audio
>> and video. Not a single argument that has been presented has convinced me,
>> and getting from 2 to 1 is a *50%* reduction in NAT port utilization. (And a
>> significant reduction in the number of "strange" failure modes, where one
>> traversal worked and the other didn't.)
>>
>
> Clearly we can, but we need to do additional work to get there. This work
> will take time and delay v1.0, so we need a pretty convincing rationale to
> do so.
>
> I concur that solving strange failure modes is significant. But with regard
> to NAT port utilization, this ignores the utilization by the browser for
> HTTP; 50% reduction seems possible in theory only. (Here I am essentially
> restating 2.2.2 from
> https://datatracker.ietf.org/doc/draft-perkins-rtcweb-rtp-usage/?include_text=1,
> which I haven't yet seen a clear response to.)
>

Also: when falling back to send media over TCP/WebSockets, a lost packet may
cause a momentary audio dropout as the transport stalls while the packet is
retransmitted. I expect that this will occur more frequently if video and
audio are combined on the same TCP connection, if for no other reason than
the much higher packet rate.


>
>
>
>
>> Matthew Kaufman
>>
>>
>

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<br><br><div class=3D"gmail_quote">On Wed, Jul 20, 2011 at 3:59 PM, Justin =
Uberti <span dir=3D"ltr">&lt;<a href=3D"mailto:juberti@google.com">juberti@=
google.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=
=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">

<br><br><div class=3D"gmail_quote"><div><div></div><div class=3D"h5">On Wed=
, Jul 20, 2011 at 3:51 PM, Matthew Kaufman <span dir=3D"ltr">&lt;<a href=3D=
"mailto:matthew.kaufman@skype.net" target=3D"_blank">matthew.kaufman@skype.=
net</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">

<div><br>
On Jul 20, 2011, at 12:46 PM, Justin Uberti wrote:<br>
<br>
&gt;<br>
&gt; This is where I&#39;ve ended up on this topic. We can easily multiplex=
 multiple RTP sources (of the same type) over a single RTP session using SS=
RC. We can also mux RTCP over the RTP session using RTCP mux. So, for an ar=
bitrary video call, we have just 2 RTP sessions/NAT bindings.<br>




&gt;<br>
&gt; Is it worth going the extra mile to get down to 1 in v1.0, given the l=
ack of consensus that exists right now? Is there even a compelling argument=
 to do so?<br>
<br>
</div>Yes and yes.<br>
<br>
I really can&#39;t understand why, if we can multiplex 3 totally different =
types of video streams over the same RTP session using SSRC (with wildly di=
fferent bit-rates, inter-packet times, etc.) we can&#39;t also multiplex au=
dio and video. Not a single argument that has been presented has convinced =
me, and getting from 2 to 1 is a *50%* reduction in NAT port utilization. (=
And a significant reduction in the number of &quot;strange&quot; failure mo=
des, where one traversal worked and the other didn&#39;t.)<br>


</blockquote><div><br></div></div></div><div>Clearly we can, but we need to=
 do additional work to get there. This work will take time and delay v1.0, =
so we need a pretty convincing rationale to do so.</div><div><br></div>

<div>I concur that solving strange failure modes is significant. But with r=
egard to NAT port utilization, this ignores the utilization by the browser =
for HTTP; 50% reduction seems possible in theory only. (Here I am essential=
ly restating 2.2.2 from=C2=A0<a href=3D"https://datatracker.ietf.org/doc/dr=
aft-perkins-rtcweb-rtp-usage/?include_text=3D1" target=3D"_blank">https://d=
atatracker.ietf.org/doc/draft-perkins-rtcweb-rtp-usage/?include_text=3D1</a=
>, which I haven&#39;t yet seen a clear response to.)</div>

</div></blockquote><div><br></div><div>Also: when falling back to send medi=
a over TCP/WebSockets, a lost packet may cause a momentary audio dropout as=
 the transport stalls while the packet is retransmitted. I expect that this=
 will occur more frequently if video and audio are combined on the same TCP=
 connection, if for no other reason than the much higher packet rate.</div>

<div><br></div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex=
;border-left:1px #ccc solid;padding-left:1ex;"><div class=3D"gmail_quote">
<div><br></div><div><br></div><div><br></div><div><br></div><blockquote cla=
ss=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;pa=
dding-left:1ex">

<font color=3D"#888888"><br>
Matthew Kaufman<br>
<br>
</font></blockquote></div><br>
</blockquote></div><br>

--00235429d1d4aedf5704a886bce7--

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Another +1 for multiplexing everything together. My only worry is that the
base spec for RTCP doesn't mandate that SSRC is included in every RTCP
message type, and thus you could end up with the multiplexing layer needing
to understand every  RTCP messaged used for video (TMMBR, TMMBN, etc)

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Another +1 for multiplexing everything together. My only worry is that the =
base spec for RTCP doesn&#39;t mandate that SSRC is included in every RTCP =
message type, and thus you could end up with the multiplexing layer needing=
 to understand every =A0RTCP messaged used for video (TMMBR, TMMBN, etc)<br=
 clear=3D"all">

<div><br></div>

--20cf3040e40e34fd0e04a886be8e--

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Subject: Re: [rtcweb] DSCP and media
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As Bernard points out there is the qos marking for the wireless networks =
and local networks but what I am mostly interested in is the device =
(typically an NAT) that is sequencing the packets on to the broadband =
uplink from my home to my ISP. Lots of NATs support some simple set of =
priorities and send the higher priority packets ahead of the lower =
priority. My view is this was largely done for the online gaming people =
but in many cases it does help voip. Many of the voip devices do take =
advantage of it - there is pretty much no downside for setting the DSCP. =
The cases where I see it helping are things like the following: I have a =
128Kbps uplink, I am trying to send an 80Kbps audio stream, and at the =
same time I email a 25 MB powerpoint file. With QoS, typically the audio =
gets it's 80k and email gets the rest. WIthout QOS, the audio gets =
packet loss.  It helps in cases where the uplink is the congested part =
of the network. It does not help if your downlink is the problem. Keep =
in mind most people have much faster downlink than uplink. It's easy to =
run your own experiments and try this out - just get a nat with support =
and turn it on and off and see how things go. It also helps with =
wireless but I have not done a lot of experimenting with that - my =
wireless is so much faster than my broadband that it has not been a =
problem but I heard rumors that it can help file transfer over the lan =
not impact voip - I just don't know as much about this sort of use case.=20=


To your point about too risky to expose a generic API. I'm on the fence. =
You could be right but it's sort of hard to decide why if it too risking =
to let some JS send an packet with higher priority, that it is OK to let =
is send normal priority ones. It would be interesting to talk about the =
risk, and what we might do to mitigate them. One point of view is that =
rate shaping and policing of packets can never be done by the endpoint =
and needs to be done by network router and switches. The people that buy =
into this don't see it as any worse that the JS can set the DSCP than =
the fact that a think app on my notebook computer can set the DSCP. =20

The other issue is that it is not clear what values you should set the =
DSCP to. If you read the appropriate RFCs, you will come to the =
conclusion that they are pretty confused on the topic. If you look at =
the values being used by Cisco, the mapping from 802.11, and Microsoft, =
you would notice they are not the same and there may be some uh "gaming =
the system" going on. One of the things this WG could do would be to =
come up with some clear guidance on what values to use for browsers and =
when to use them.=20



On Jul 20, 2011, at 12:39 , Justin Uberti wrote:

> Can you go into more detail about what these NATs do with the DSCP =
markings?
>=20
> I'm not convinced we want to expose a generic API for this (too easy =
to misuse), but I think it definitely makes sense for implementations to =
set appropriate markings on audio and video packets, including different =
levels on individual video packets when using hierarchical prediction =
structures.
>=20
> On Wed, Jul 20, 2011 at 1:59 PM, Cullen Jennings <fluffy@cisco.com> =
wrote:
>=20
> Many home NATs support DSCP based QoS and it does help. I think that =
the IETF should recommend that some default DSCP for audio and video =
from the browser as well as suggest there should be an API to set the =
DSCP for different media steam.
>=20
> Cullen <with my individual contributor hat on>
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


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



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As I recall=2C there was some conflict between the DSCP mappings defined wi=
thin WMM and those recommended in RFC 4594.  There was some discussion of h=
ow to fix this in WFA and DLNA.  Fred would probably know the current statu=
s. AFAIK this is still an issue.=20

> From: fluffy@cisco.com
> Date: Wed=2C 20 Jul 2011 14:25:16 -0700
> To: juberti@google.com
> CC: rtcweb@ietf.org
> Subject: Re: [rtcweb] DSCP and media
>=20
>=20
> As Bernard points out there is the qos marking for the wireless networks =
and local networks but what I am mostly interested in is the device (typica=
lly an NAT) that is sequencing the packets on to the broadband uplink from =
my home to my ISP. Lots of NATs support some simple set of priorities and s=
end the higher priority packets ahead of the lower priority. My view is thi=
s was largely done for the online gaming people but in many cases it does h=
elp voip. Many of the voip devices do take advantage of it - there is prett=
y much no downside for setting the DSCP. The cases where I see it helping a=
re things like the following: I have a 128Kbps uplink=2C I am trying to sen=
d an 80Kbps audio stream=2C and at the same time I email a 25 MB powerpoint=
 file. With QoS=2C typically the audio gets it's 80k and email gets the res=
t. WIthout QOS=2C the audio gets packet loss.  It helps in cases where the =
uplink is the congested part of the network. It does not help if your downl=
ink is the problem. Keep=20
>  in mind most people have much faster downlink than uplink. It's easy to =
run your own experiments and try this out - just get a nat with support and=
 turn it on and off and see how things go. It also helps with wireless but =
I have not done a lot of experimenting with that - my wireless is so much f=
aster than my broadband that it has not been a problem but I heard rumors t=
hat it can help file transfer over the lan not impact voip - I just don't k=
now as much about this sort of use case.=20
>=20
> To your point about too risky to expose a generic API. I'm on the fence. =
You could be right but it's sort of hard to decide why if it too risking to=
 let some JS send an packet with higher priority=2C that it is OK to let is=
 send normal priority ones. It would be interesting to talk about the risk=
=2C and what we might do to mitigate them. One point of view is that rate s=
haping and policing of packets can never be done by the endpoint and needs =
to be done by network router and switches. The people that buy into this do=
n't see it as any worse that the JS can set the DSCP than the fact that a t=
hink app on my notebook computer can set the DSCP. =20
>=20
> The other issue is that it is not clear what values you should set the DS=
CP to. If you read the appropriate RFCs=2C you will come to the conclusion =
that they are pretty confused on the topic. If you look at the values being=
 used by Cisco=2C the mapping from 802.11=2C and Microsoft=2C you would not=
ice they are not the same and there may be some uh "gaming the system" goin=
g on. One of the things this WG could do would be to come up with some clea=
r guidance on what values to use for browsers and when to use them.=20
>=20
>=20
>=20
> On Jul 20=2C 2011=2C at 12:39 =2C Justin Uberti wrote:
>=20
> > Can you go into more detail about what these NATs do with the DSCP mark=
ings?
> >=20
> > I'm not convinced we want to expose a generic API for this (too easy to=
 misuse)=2C but I think it definitely makes sense for implementations to se=
t appropriate markings on audio and video packets=2C including different le=
vels on individual video packets when using hierarchical prediction structu=
res.
> >=20
> > On Wed=2C Jul 20=2C 2011 at 1:59 PM=2C Cullen Jennings <fluffy@cisco.co=
m> wrote:
> >=20
> > Many home NATs support DSCP based QoS and it does help. I think that th=
e IETF should recommend that some default DSCP for audio and video from the=
 browser as well as suggest there should be an API to set the DSCP for diff=
erent media steam.
> >=20
> > Cullen <with my individual contributor hat on>
> >=20
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
> >=20
>=20
>=20
> Cullen Jennings
> For corporate legal information go to:
> http://www.cisco.com/web/about/doing_business/legal/cri/index.html
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
 		 	   		  =

--_889ff884-2858-4d74-9f4a-e5952f036ed9_
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Content-Transfer-Encoding: quoted-printable

<html>
<head>
<style><!--
.hmmessage P
{
margin:0px=3B
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}
body.hmmessage
{
font-size: 10pt=3B
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</head>
<body class=3D'hmmessage'><div dir=3D'ltr'>
As I recall=2C there was some conflict between the DSCP mappings defined wi=
thin WMM and those recommended in RFC 4594.&nbsp=3B There was some discussi=
on of how to fix this in WFA and DLNA.&nbsp=3B Fred would probably know the=
 current status. AFAIK this is still an issue. <br><br><div>&gt=3B From: fl=
uffy@cisco.com<br>&gt=3B Date: Wed=2C 20 Jul 2011 14:25:16 -0700<br>&gt=3B =
To: juberti@google.com<br>&gt=3B CC: rtcweb@ietf.org<br>&gt=3B Subject: Re:=
 [rtcweb] DSCP and media<br>&gt=3B <br>&gt=3B <br>&gt=3B As Bernard points =
out there is the qos marking for the wireless networks and local networks b=
ut what I am mostly interested in is the device (typically an NAT) that is =
sequencing the packets on to the broadband uplink from my home to my ISP. L=
ots of NATs support some simple set of priorities and send the higher prior=
ity packets ahead of the lower priority. My view is this was largely done f=
or the online gaming people but in many cases it does help voip. Many of th=
e voip devices do take advantage of it - there is pretty much no downside f=
or setting the DSCP. The cases where I see it helping are things like the f=
ollowing: I have a 128Kbps uplink=2C I am trying to send an 80Kbps audio st=
ream=2C and at the same time I email a 25 MB powerpoint file. With QoS=2C t=
ypically the audio gets it's 80k and email gets the rest. WIthout QOS=2C th=
e audio gets packet loss.  It helps in cases where the uplink is the conges=
ted part of the network. It does not help if your downlink is the problem. =
Keep <br>&gt=3B  in mind most people have much faster downlink than uplink.=
 It's easy to run your own experiments and try this out - just get a nat wi=
th support and turn it on and off and see how things go. It also helps with=
 wireless but I have not done a lot of experimenting with that - my wireles=
s is so much faster than my broadband that it has not been a problem but I =
heard rumors that it can help file transfer over the lan not impact voip - =
I just don't know as much about this sort of use case. <br>&gt=3B <br>&gt=
=3B To your point about too risky to expose a generic API. I'm on the fence=
. You could be right but it's sort of hard to decide why if it too risking =
to let some JS send an packet with higher priority=2C that it is OK to let =
is send normal priority ones. It would be interesting to talk about the ris=
k=2C and what we might do to mitigate them. One point of view is that rate =
shaping and policing of packets can never be done by the endpoint and needs=
 to be done by network router and switches. The people that buy into this d=
on't see it as any worse that the JS can set the DSCP than the fact that a =
think app on my notebook computer can set the DSCP.  <br>&gt=3B <br>&gt=3B =
The other issue is that it is not clear what values you should set the DSCP=
 to. If you read the appropriate RFCs=2C you will come to the conclusion th=
at they are pretty confused on the topic. If you look at the values being u=
sed by Cisco=2C the mapping from 802.11=2C and Microsoft=2C you would notic=
e they are not the same and there may be some uh "gaming the system" going =
on. One of the things this WG could do would be to come up with some clear =
guidance on what values to use for browsers and when to use them. <br>&gt=
=3B <br>&gt=3B <br>&gt=3B <br>&gt=3B On Jul 20=2C 2011=2C at 12:39 =2C Just=
in Uberti wrote:<br>&gt=3B <br>&gt=3B &gt=3B Can you go into more detail ab=
out what these NATs do with the DSCP markings?<br>&gt=3B &gt=3B <br>&gt=3B =
&gt=3B I'm not convinced we want to expose a generic API for this (too easy=
 to misuse)=2C but I think it definitely makes sense for implementations to=
 set appropriate markings on audio and video packets=2C including different=
 levels on individual video packets when using hierarchical prediction stru=
ctures.<br>&gt=3B &gt=3B <br>&gt=3B &gt=3B On Wed=2C Jul 20=2C 2011 at 1:59=
 PM=2C Cullen Jennings &lt=3Bfluffy@cisco.com&gt=3B wrote:<br>&gt=3B &gt=3B=
 <br>&gt=3B &gt=3B Many home NATs support DSCP based QoS and it does help. =
I think that the IETF should recommend that some default DSCP for audio and=
 video from the browser as well as suggest there should be an API to set th=
e DSCP for different media steam.<br>&gt=3B &gt=3B <br>&gt=3B &gt=3B Cullen=
 &lt=3Bwith my individual contributor hat on&gt=3B<br>&gt=3B &gt=3B <br>&gt=
=3B &gt=3B _______________________________________________<br>&gt=3B &gt=3B=
 rtcweb mailing list<br>&gt=3B &gt=3B rtcweb@ietf.org<br>&gt=3B &gt=3B http=
s://www.ietf.org/mailman/listinfo/rtcweb<br>&gt=3B &gt=3B <br>&gt=3B <br>&g=
t=3B <br>&gt=3B Cullen Jennings<br>&gt=3B For corporate legal information g=
o to:<br>&gt=3B http://www.cisco.com/web/about/doing_business/legal/cri/ind=
ex.html<br>&gt=3B <br>&gt=3B <br>&gt=3B ___________________________________=
____________<br>&gt=3B rtcweb mailing list<br>&gt=3B rtcweb@ietf.org<br>&gt=
=3B https://www.ietf.org/mailman/listinfo/rtcweb<br></div> 		 	   		  </div=
></body>
</html>=

--_889ff884-2858-4d74-9f4a-e5952f036ed9_--

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On Jul 20, 2011, at 14:37 , Bernard Aboba wrote:

> As I recall, there was some conflict between the DSCP mappings defined =
within WMM and those recommended in RFC 4594.  There was some discussion =
of how to fix this in WFA and DLNA.  Fred would probably know the =
current status. AFAIK this is still an issue.=20

Fred, James - do you guys know status of mapping DSCP to WMM ?



From csp@csperkins.org  Wed Jul 20 15:21:08 2011
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From: Colin Perkins <csp@csperkins.org>
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To: Cullen Jennings <fluffy@cisco.com>
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Subject: Re: [rtcweb] To multiplex or not!
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On 20 Jul 2011, at 15:11, Cullen Jennings wrote:
> On Jul 20, 2011, at 4:08 , Emil Ivov wrote:
>> =D0=9D=D0=B0 20.07.11 02:05, Cullen Jennings =D0=BD=D0=B0=D0=BF=D0=B8=D1=
=81=D0=B0:
>>> I'd like to see it that devices MUST support multiplexing and I'd =
probably be in favor of also MUST support non multiplexed flows. The =
multiplexing is a big advantage for any device that needs to deal with =
NATs so I suspect many "legacy" devices would be moving to this even if =
there was no need to deal with the browser like devices. I also think =
that allowing this type of multiplexing will make firewall traversal =
easier where a device can configure a firewall to open just one =
specified port and use it for RTP.
>>=20
>> I don't really agree with this part. Devices would still need to =
handle candidates that come from  STUN, TURN, UPnP, host IPv4, host =
IPv6, and potentially VPN, MIPv6 and other tunnelling mechanisms. =
Whether or not this only happens for a single component/stream doesn't =
really simplify code does it?
>>=20
>> According to my personal experience with our work on ice4j.org, =
managing multiple checklists was far less resource consuming than =
implementing TURN tunnels and persmissions, STUN multiplexing, =
synchronisation and even getting pacing right.
>=20
> Ahh, I think we have come to the key issues here - perhaps I have not =
been explaining myself very well. So in your experience roughly how long =
does it take to set up a for two scenarios using ICE as is today.=20
>=20
> First scenario: I have two devices that want to set up a single audio =
stream and they each have the candidates from: local v4 IP address, v4 =
stun, v4 turn, local v6, turn v6. Obviously they could have a lot more =
but that seems like a reasonable starting point.=20
>=20
> Second scenario: same as first address wise but instead of a single =
audio stream they want to set up a single audio stream  to a conference =
bridge plus 7 video streams for the video from the 5 people on the =
bridge plus a presentation stream and stream of video from active =
speaker. I don't really care much about the scenario other than there =
are 8 streams being set up. But this type of scenario is becoming very =
common for multi party video chat as it allows you to see perhaps small =
versions of everyone plus a large version of active speaker.=20
>=20
> My experience is the answer to the first scenario is not as quick as =
you would like and the answer to how long the second takes is about 8 =
times longer than the first one. You might do a bit better than that =
depending on how clever your implementation is but it still a lot =
longer.=20

I may be misunderstanding, but why would this take 8 times as long to =
set up? Wouldn't you implement it as one UDP flow for the RTP audio, =
plus one UDP flow for the RTP video (with the 7 video streams =
multiplexed by SSRC, since RTP has always allowed multiparty flows of =
the same media type in a single RTP session)? This might take twice as =
long to setup as a single audio flow in the worst case, but not eight =
times as long.

Cheers,
Colin


From csp@csperkins.org  Wed Jul 20 15:24:55 2011
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From: Colin Perkins <csp@csperkins.org>
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Subject: Re: [rtcweb] To multiplex or not!
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If you're not doing network-based QoS to prioritise certain flows, then =
this is requires two bidirectional UDP flows: one for the audio and one =
for video, using the standard RTP mechanisms (which is what I understand =
Magnus' option A to be).

Colin


On 20 Jul 2011, at 10:49, Harald Alvestrand wrote:
> Magnus, I cannot parse your list of choices.
>=20
> In the case where you have
>=20
> Sender A
> Recipient B
> Three video streams going from A to B
> Three audio streams going from A to B
> One video stream going from B to A
> One audio stream going from B to A
>=20
> with the application not requiring the network to treat the packets =
any differently, and with the application using the CNAME mechanism to =
match audio streams with corresponding video streams
>=20
> it is completely unclear to me whether your choice A) requires 8 =
transport flows, 4 transport flows or 2 transport flows, and it is =
completely unclear to me what C) actually means. (I assume "transport =
flow" =3D "RTP sessions" here)
>=20
> Please - can you clarify what your question is intended to mean?
>=20
>                 Harald
>=20
>=20
> On 07/19/11 16:28, Magnus Westerlund wrote:
>> Hi,
>>=20
>> This email is as an individual contributor.
>>=20
>> I want to get started on the discussion of the Multiplexing of the
>> various protocols over single lower layer transport flow, such as a =
UDP
>> flow. I will attempt to split up the questions into different emails.
>>=20
>> The first question I think is reasonably easy to get answered, but I
>> think it is time we determine if my belief in the answer is correct =
or not.
>>=20
>> The traffic between two RTCWEB peers from the various components, =
such
>> as RTP sessions, datagram service:
>>=20
>> a) MUST be sent as Individual flows for each component.
>>=20
>> b) MUST be multiplexed into a single transport flow.
>>=20
>> c) SHOULD be multiplexed into a single transport flow, but the RTCWEB
>> peer MUST be able to send them as individual flows.
>>=20
>> I would love if people can indicate their choice or preferences.
>>=20
>> I personally prefer A as it it is simplest in all aspect except the =
NAT
>> traversal.
>> - It allows for flow based QoS.
>> - It is the what the implementation that exist mostly do
>> - Signaling protocols that exist support it, no extra functionality
>> - People are used to the concept
>> - It minimizes the difference to legacy.
>>=20
>> Thus it is the quickest road to define something with the least =
formal
>> push back and concern over maturity of any solution.
>>=20
>> The downside with B and C is that we do have to solve the =
multiplexing
>> and get an agreement that gets through all the hurdles.
>>=20
>> Of these two opens I do prefer C.  Although it results in the extra
>> complexities of having both alternatives, it will give us both a
>> fallback, flow based QoS and better legacy support.
>>=20
>> Now it is your time to make your opinion heard!
>>=20
>> Cheers
>>=20
>> Magnus Westerlund
>>=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
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb



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




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From: Cullen Jennings <fluffy@cisco.com>
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To: Fred Baker <fred@cisco.com>
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Cc: James M Polk <jmpolk@cisco.com>, rtcweb@ietf.org
Subject: Re: [rtcweb] DSCP and media
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On Jul 20, 2011, at 15:13 , Fred Baker wrote:

>=20
> On Jul 20, 2011, at 2:39 PM, Cullen Jennings wrote:
>=20
>>=20
>> On Jul 20, 2011, at 14:37 , Bernard Aboba wrote:
>>=20
>>> As I recall, there was some conflict between the DSCP mappings =
defined within WMM and those recommended in RFC 4594.  There was some =
discussion of how to fix this in WFA and DLNA.  Fred would probably know =
the current status. AFAIK this is still an issue.=20
>>=20
>> Fred, James - do you guys know status of mapping DSCP to WMM ?
>=20
> Define "WMM"?

The "Wireless Multimedia Extensions from Wi-FI. It's been awhile since I =
looked at this but I seem to recall that 802.11e had four levels of =
priority, something like voice, video, background, and best effort. And =
802.1p had 8 levels, and some confusion on status of mapping between =
them and DSCP.=20

=20=

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To: Bala Pitchandi <Bala@vidyo.com>
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Except that it doesn't work in all cases: please read the discussion in =
draft-perkins-rtpweb-rtp-usage-02. You may argue that you don't think =
the breakage is important, but there *is* breakage in doing this, and it =
is not backwards compatible.

Colin


On 20 Jul 2011, at 21:57, Bala Pitchandi wrote:
> +1 for sending audio & video (and any other type of media, for that =
matter) on a single RTP session. With RTCP mux, that means all media can =
be sent on a single UDP port.=20
>=20
> -- Bala
>=20
>=20
> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On =
Behalf Of Matthew Kaufman
> Sent: Wednesday, July 20, 2011 3:51 PM
> To: Justin Uberti
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] How to multiplex between peers
>=20
>=20
> On Jul 20, 2011, at 12:46 PM, Justin Uberti wrote:
>=20
>>=20
>> This is where I've ended up on this topic. We can easily multiplex =
multiple RTP sources (of the same type) over a single RTP session using =
SSRC. We can also mux RTCP over the RTP session using RTCP mux. So, for =
an arbitrary video call, we have just 2 RTP sessions/NAT bindings.=20
>>=20
>> Is it worth going the extra mile to get down to 1 in v1.0, given the =
lack of consensus that exists right now? Is there even a compelling =
argument to do so?
>=20
> Yes and yes.
>=20
> I really can't understand why, if we can multiplex 3 totally different =
types of video streams over the same RTP session using SSRC (with wildly =
different bit-rates, inter-packet times, etc.) we can't also multiplex =
audio and video. Not a single argument that has been presented has =
convinced me, and getting from 2 to 1 is a *50%* reduction in NAT port =
utilization. (And a significant reduction in the number of "strange" =
failure modes, where one traversal worked and the other didn't.)
>=20
> Matthew Kaufman
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb



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




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References: <4E259EAD.3060505@ericsson.com> <FAE78F7C-8C51-41C4-B3D7-6497396E12A5@cisco.com> <4E26C5CF.1080007@ericsson.com> <BLU152-W54BE1A03753680FF0094C4934C0@phx.gbl> <CAOJ7v-2kwiCipJSHmNT9GuGJJzEjPV-X00TLnf-LwbsJ1ADwDw@mail.gmail.com> <896BDC4C-849C-4553-89C8-7EFEF9FFEC6B@skype.net> <38DF8F00ABAB77498A75469448CB081B3AE69BEBC4@BE235.mail.lan> <CAB+e8F6dbTEDDorimVgKuFq5EAXGyy6FFKXC7JQ3=qieqnOnpQ@mail.gmail.com>
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On 20 Jul 2011, at 22:15, Aron Rosenberg wrote:
> Another +1 for multiplexing everything together. My only worry is that =
the base spec for RTCP doesn't mandate that SSRC is included in every =
RTCP message type, and thus you could end up with the multiplexing layer =
needing to understand every  RTCP messaged used for video (TMMBR, TMMBN, =
etc)

...which is another reason why this doesn't work in all cases, in =
addition to the reasons we document in draft-perkins-rtcweb-rtp-usage-02


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




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<html><head></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
"><div><div>On 20 Jul 2011, at 22:15, Aron Rosenberg =
wrote:</div><blockquote type=3D"cite">Another +1 for multiplexing =
everything together. My only worry is that the base spec for RTCP =
doesn't mandate that SSRC is included in every RTCP message type, and =
thus you could end up with the multiplexing layer needing to understand =
every &nbsp;RTCP messaged used for video (TMMBR, TMMBN, etc)<br =
clear=3D"all">

</blockquote><br></div><div>...which is another reason why this doesn't =
work in all cases, in addition to the reasons we document in =
draft-perkins-rtcweb-rtp-usage-02</div><div><span =
class=3D"Apple-style-span" style=3D"border-collapse: separate; color: =
rgb(0, 0, 0); font-family: Arial; font-size: 9px; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; =
line-height: normal; orphans: 2; text-align: auto; text-indent: 0px; =
text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; =
-webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0; "><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; color: rgb(0, 0, 0); font-family: =
'Lucida Sans Typewriter'; font-size: 9px; 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; "><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; -webkit-border-horizontal-spacing: =
0px; -webkit-border-vertical-spacing: 0px; color: rgb(0, 0, 0); =
font-family: 'Lucida Sans Typewriter'; font-size: 9px; font-style: =
normal; font-variant: normal; font-weight: normal; letter-spacing: =
normal; line-height: normal; -webkit-text-decorations-in-effect: none; =
text-indent: 0px; -webkit-text-size-adjust: auto; text-transform: none; =
orphans: 2; white-space: normal; widows: 2; word-spacing: 0px; "><span =
class=3D"Apple-style-span" style=3D"border-collapse: separate; =
-webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; color: rgb(0, 0, 0); font-family: 'Lucida Sans Typewriter'; =
font-size: 9px; font-style: normal; font-variant: normal; font-weight: =
normal; letter-spacing: normal; line-height: normal; =
-webkit-text-decorations-in-effect: none; text-indent: 0px; =
-webkit-text-size-adjust: auto; text-transform: none; orphans: 2; =
white-space: normal; widows: 2; word-spacing: 0px; "><span =
class=3D"Apple-style-span" style=3D"border-collapse: separate; =
-webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; color: rgb(0, 0, 0); font-family: 'Lucida Sans Typewriter'; =
font-size: 9px; font-style: normal; font-variant: normal; font-weight: =
normal; letter-spacing: normal; line-height: normal; =
-webkit-text-decorations-in-effect: none; text-indent: 0px; =
-webkit-text-size-adjust: auto; text-transform: none; orphans: 2; =
white-space: normal; widows: 2; word-spacing: 0px; "><div><br =
class=3D"Apple-interchange-newline"><br =
class=3D"khtml-block-placeholder"></div><div>--&nbsp;</div><div></div><div=
>Colin Perkins</div><div><a =
href=3D"http://csperkins.org/">http://csperkins.org/</a></div></span></spa=
n></span><br class=3D"Apple-interchange-newline"></span><br =
class=3D"Apple-interchange-newline"></span>
</div>
<br></body></html>=

--Apple-Mail-11-33951676--

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The specific issue was the mapping between IEEE 802.1p and DSCP included in=
 the WMM appendix.  This wasn't included in IEEE 802.11e (thankfully).  The=
 appendix was included prior to publication of RFC 4594=2C and was incompat=
ible with it.  As I recall=2C the WMM mapping was as follows:

PJLIB Traffic Type     IP DSCP  WMM  802.1p =20
 BEST_EFFORT  0x00     BE (Bulk Effort)    0   =20
 BACKGROUND   0x08     BK (Bulk)     2   =20
 VIDEO        0x28     VI (Video)      5   =20
 VOICE        0x30     VO (Voice)      6   =20
 CONTROL      0x38     VO (Voice)      7   =20



> Subject: Re: [rtcweb] DSCP and media
> From: fluffy@cisco.com
> Date: Wed=2C 20 Jul 2011 15:25:49 -0700
> CC: jmpolk@cisco.com=3B juberti@google.com=3B bernard_aboba@hotmail.com=
=3B rtcweb@ietf.org
> To: fred@cisco.com
>=20
>=20
> On Jul 20=2C 2011=2C at 15:13 =2C Fred Baker wrote:
>=20
> >=20
> > On Jul 20=2C 2011=2C at 2:39 PM=2C Cullen Jennings wrote:
> >=20
> >>=20
> >> On Jul 20=2C 2011=2C at 14:37 =2C Bernard Aboba wrote:
> >>=20
> >>> As I recall=2C there was some conflict between the DSCP mappings defi=
ned within WMM and those recommended in RFC 4594.  There was some discussio=
n of how to fix this in WFA and DLNA.  Fred would probably know the current=
 status. AFAIK this is still an issue.=20
> >>=20
> >> Fred=2C James - do you guys know status of mapping DSCP to WMM ?
> >=20
> > Define "WMM"?
>=20
> The "Wireless Multimedia Extensions from Wi-FI. It's been awhile since I =
looked at this but I seem to recall that 802.11e had four levels of priorit=
y=2C something like voice=2C video=2C background=2C and best effort. And 80=
2.1p had 8 levels=2C and some confusion on status of mapping between them a=
nd DSCP.=20
>=20
> =20
 		 	   		  =

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<html>
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<body class=3D'hmmessage'><div dir=3D'ltr'>
The specific issue was the mapping between IEEE 802.1p and DSCP included in=
 the WMM appendix.&nbsp=3B This wasn't included in IEEE 802.11e (thankfully=
).&nbsp=3B The appendix was included prior to publication of RFC 4594=2C an=
d was incompatible with it.&nbsp=3B As I recall=2C the WMM mapping was as f=
ollows:<br><br><table class=3D"wiki"><tbody><tr><td>PJLIB Traffic Type    <=
/td><td> IP <span class=3D"searchword1">DSCP</span> </td><td> <span class=
=3D"searchword0">WMM</span> </td><td> 802.1p =20
</td></tr><tr><td> BEST_EFFORT </td><td> 0x00    </td><td> BE (Bulk Effort)=
 </td><td>   0   =20
</td></tr><tr><td> BACKGROUND  </td><td> 0x08    </td><td> BK (Bulk)  </td>=
<td>   2   =20
</td></tr><tr><td> VIDEO       </td><td> 0x28    </td><td> VI (Video)   </t=
d><td>   5   =20
</td></tr><tr><td> VOICE       </td><td> 0x30    </td><td> VO (Voice)   </t=
d><td>   6   =20
</td></tr><tr><td> CONTROL     </td><td> 0x38    </td><td> VO (Voice)   </t=
d><td>   7   =20
</td></tr></tbody></table><br><br><br><div>&gt=3B Subject: Re: [rtcweb] DSC=
P and media<br>&gt=3B From: fluffy@cisco.com<br>&gt=3B Date: Wed=2C 20 Jul =
2011 15:25:49 -0700<br>&gt=3B CC: jmpolk@cisco.com=3B juberti@google.com=3B=
 bernard_aboba@hotmail.com=3B rtcweb@ietf.org<br>&gt=3B To: fred@cisco.com<=
br>&gt=3B <br>&gt=3B <br>&gt=3B On Jul 20=2C 2011=2C at 15:13 =2C Fred Bake=
r wrote:<br>&gt=3B <br>&gt=3B &gt=3B <br>&gt=3B &gt=3B On Jul 20=2C 2011=2C=
 at 2:39 PM=2C Cullen Jennings wrote:<br>&gt=3B &gt=3B <br>&gt=3B &gt=3B&gt=
=3B <br>&gt=3B &gt=3B&gt=3B On Jul 20=2C 2011=2C at 14:37 =2C Bernard Aboba=
 wrote:<br>&gt=3B &gt=3B&gt=3B <br>&gt=3B &gt=3B&gt=3B&gt=3B As I recall=2C=
 there was some conflict between the DSCP mappings defined within WMM and t=
hose recommended in RFC 4594.  There was some discussion of how to fix this=
 in WFA and DLNA.  Fred would probably know the current status. AFAIK this =
is still an issue. <br>&gt=3B &gt=3B&gt=3B <br>&gt=3B &gt=3B&gt=3B Fred=2C =
James - do you guys know status of mapping DSCP to WMM ?<br>&gt=3B &gt=3B <=
br>&gt=3B &gt=3B Define "WMM"?<br>&gt=3B <br>&gt=3B The "Wireless Multimedi=
a Extensions from Wi-FI. It's been awhile since I looked at this but I seem=
 to recall that 802.11e had four levels of priority=2C something like voice=
=2C video=2C background=2C and best effort. And 802.1p had 8 levels=2C and =
some confusion on status of mapping between them and DSCP. <br>&gt=3B <br>&=
gt=3B  <br></div> 		 	   		  </div></body>
</html>=

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From csp@csperkins.org  Wed Jul 20 15:39:05 2011
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Subject: Re: [rtcweb] To multiplex or not!
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My preference is for option A: send each RTP session on a separate lower =
layer flow (e.g., a separate UDP flow). Yes, this means that your =
typical multiparty audio/video conference uses two UDP ports rather than =
1, but I do not believe this is a significant problem, since most =
browsers open far more than 2 TCP ports simultaneously when fetching web =
resources.

That said, I understand the desire to reduce ICE negotiation times. If =
this is a real concern (and I'm not convinced it should be, since I =
don't see that this adds more than 1 RTT to the negotiation in the usual =
case), then some means of explicitly multiplexing RTP sessions might be =
useful to develop. I do not believe SSRC-based multiplexing of RTP =
sessions works though: we would need to develop a multiplexing mechanism =
that conforms to RTP semantics, or get AVTCORE to agree to a change in =
RTP with all the backwards compatibility issues that raises.

Colin



On 19 Jul 2011, at 15:28, Magnus Westerlund wrote:
> Hi,
>=20
> This email is as an individual contributor.
>=20
> I want to get started on the discussion of the Multiplexing of the
> various protocols over single lower layer transport flow, such as a =
UDP
> flow. I will attempt to split up the questions into different emails.
>=20
> The first question I think is reasonably easy to get answered, but I
> think it is time we determine if my belief in the answer is correct or =
not.
>=20
> The traffic between two RTCWEB peers from the various components, such
> as RTP sessions, datagram service:
>=20
> a) MUST be sent as Individual flows for each component.
>=20
> b) MUST be multiplexed into a single transport flow.
>=20
> c) SHOULD be multiplexed into a single transport flow, but the RTCWEB
> peer MUST be able to send them as individual flows.
>=20
> I would love if people can indicate their choice or preferences.
>=20
> I personally prefer A as it it is simplest in all aspect except the =
NAT
> traversal.
> - It allows for flow based QoS.
> - It is the what the implementation that exist mostly do
> - Signaling protocols that exist support it, no extra functionality
> - People are used to the concept
> - It minimizes the difference to legacy.
>=20
> Thus it is the quickest road to define something with the least formal
> push back and concern over maturity of any solution.
>=20
> The downside with B and C is that we do have to solve the multiplexing
> and get an agreement that gets through all the hurdles.
>=20
> Of these two opens I do prefer C.  Although it results in the extra
> complexities of having both alternatives, it will give us both a
> fallback, flow based QoS and better legacy support.
>=20
> Now it is your time to make your opinion heard!
>=20
> Cheers
>=20
> Magnus Westerlund
>=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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On 7/20/2011 3:31 PM, Colin Perkins wrote:
> On 20 Jul 2011, at 22:15, Aron Rosenberg wrote:
>> Another +1 for multiplexing everything together. My only worry is 
>> that the base spec for RTCP doesn't mandate that SSRC is included in 
>> every RTCP message type, and thus you could end up with the 
>> multiplexing layer needing to understand every  RTCP messaged used 
>> for video (TMMBR, TMMBN, etc)
>
> ...which is another reason why this doesn't work in all cases, in 
> addition to the reasons we document in draft-perkins-rtcweb-rtp-usage-02
>

So let me get this straight... RTP and RTCP is broken and inflexible and 
therefore we must use it, and use it in the broken and inflexible way?

If anything this is an argument for ignoring RTP and RTCP and doing 
something entirely new that is actually appropriate for what we're 
trying to build, not living with crap just because there's an RFC for it.

Matthew Kaufman

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On Jul 20, 2011, at 3:39 PM, Colin Perkins wrote:

> we would need to develop a multiplexing mechanism that conforms to RTP =
semantics, or get AVTCORE to agree to a change in RTP with all the =
backwards compatibility issues that raises.

Or decide that RTP is old and broken and not use it at all.

Matthew Kaufman=

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On 20 Jul 2011, at 23:38, Matthew Kaufman wrote:
> On 7/20/2011 3:31 PM, Colin Perkins wrote:
>> On 20 Jul 2011, at 22:15, Aron Rosenberg wrote:
>>> Another +1 for multiplexing everything together. My only worry is =
that the base spec for RTCP doesn't mandate that SSRC is included in =
every RTCP message type, and thus you could end up with the multiplexing =
layer needing to understand every  RTCP messaged used for video (TMMBR, =
TMMBN, etc)
>>=20
>> ...which is another reason why this doesn't work in all cases, in =
addition to the reasons we document in draft-perkins-rtcweb-rtp-usage-02
>>=20
>=20
> So let me get this straight... RTP and RTCP is broken and inflexible =
and therefore we must use it, and use it in the broken and inflexible =
way?
>=20
> If anything this is an argument for ignoring RTP and RTCP and doing =
something entirely new that is actually appropriate for what we're =
trying to build, not living with crap just because there's an RFC for =
it.


You're free to re-invent that wheel, if you wish. However, if you want =
to use RTP for RTCWeb, then you need to accept that doing so has both =
some advantages and some disadvantages. One of those disadvantages, for =
this application domain, is that it requires an explicit demultiplexing =
layer for distinct RTP sessions. Whether that demultiplexing layer is =
UDP, or some yet-to-be defined shim layer is a question for this working =
group, but something is needed if RTP is used.

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




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On Jul 20, 2011, at 3:48 PM, Colin Perkins wrote:

>=20
>=20
> You're free to re-invent that wheel, if you wish. However, if you want =
to use RTP for RTCWeb, then you need to accept that doing so has both =
some advantages and some disadvantages.

I believe we've managed to demonstrate that the disadvantages are =
sufficient to argue against using RTP for RTCWeb. Too bad, because it =
seems otherwise like such a good choice.

Matthew Kaufman


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On Jul 20, 2011, at 3:31 PM, Colin Perkins wrote:

> Except that it doesn't work in all cases: please read the discussion =
in draft-perkins-rtpweb-rtp-usage-02. You may argue that you don't think =
the breakage is important, but there *is* breakage in doing this, and it =
is not backwards compatible.
>=20

It doesn't need to be "backwards compatible" in order for two RTCWEB =
endpoints to communicate. And the muxing can be disabled in order to =
talk to a legacy endpoint.

Could we confine the claims about breakage to only ones that are NOT =
backwards compatibility issues?

Matthew Kaufman


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> On 20 Jul 2011, at 21:57, Bala Pitchandi wrote:
>> +1 for sending audio & video (and any other type of media, for that matt=
er) on a single RTP session. With RTCP mux, that means all media can be sen=
t on a single UDP port.=20

On Jul 20, 2011, at 6:31 PM, Colin Perkins wrote:

> Except that it doesn't work in all cases: please read the discussion in d=
raft-perkins-rtpweb-rtp-usage-02. You may argue that you don't think the br=
eakage is important, but there *is* breakage in doing this, and it is not b=
ackwards compatible.


Hi, Colin,

I've read draft-perkins-rtcweb-rtp-usage-02, and I still don't understand w=
hat you're claiming SSRC-muxing would break.

Running audio and video (and FEC, and RTX, and simulcast video, and Real-Ti=
me Text, and etc. etc.) in a single RTP session does, not, as far as I've b=
een able to tell (after a lot of thought), break anything in RTP.

To be sure, it's not backward compatible with RTP endpoints that assume a s=
ingle source per session.  It would also be hard to retrofit SDP to describ=
e and negotiate it.  And for some use cases it requires some source-level s=
ignaling, as in RFC 5576.  But the RTP model, per se, is perfectly amenable=
 to it, as far as I can tell.

I do, very much, dislike the proposal in draft-rosenberg-rtcweb-rtpmux of e=
ncoding a magic number into the RTP SSRC -- I think that *does* break a fai=
r amount of the RTP model.  But this isn't an essential feature of that pro=
posal, as far as I can tell; it's just a convenience for deep packet inspec=
tors.  (Deep packet inspectors already have to use heuristics to distinguis=
h audio vs. video RTP flows today, if they can't see signaling; and those t=
hat *can* see signaling can just use payload types to distinguish media typ=
es in the mux'ed case.)

--
Jonathan Lennox
jonathan@vidyo.com



From henry.sinnreich@gmail.com  Wed Jul 20 17:41:44 2011
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+1

> If anything this is an argument for ignoring RTP and RTCP and doing
> something entirely new that is actually appropriate for what we're
> trying to build, not living with crap just because there's an RFC for it.

Not to forget disposing of ancient SDP as well.
Use standard metadata instead, since it is equally usable for all apps, not
only for RTC apps.

Thanks, Henry


On 7/20/11 5:38 PM, "Matthew Kaufman" <matthew.kaufman@skype.net> wrote:

> On 7/20/2011 3:31 PM, Colin Perkins wrote:
>> On 20 Jul 2011, at 22:15, Aron Rosenberg wrote:
>>> Another +1 for multiplexing everything together. My only worry is
>>> that the base spec for RTCP doesn't mandate that SSRC is included in
>>> every RTCP message type, and thus you could end up with the
>>> multiplexing layer needing to understand every  RTCP messaged used
>>> for video (TMMBR, TMMBN, etc)
>> 
>> ...which is another reason why this doesn't work in all cases, in
>> addition to the reasons we document in draft-perkins-rtcweb-rtp-usage-02
>> 
> 
> So let me get this straight... RTP and RTCP is broken and inflexible and
> therefore we must use it, and use it in the broken and inflexible way?
> 
> If anything this is an argument for ignoring RTP and RTCP and doing
> something entirely new that is actually appropriate for what we're
> trying to build, not living with crap just because there's an RFC for it.
> 
> Matthew Kaufman
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb



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On 07/20/2011 03:50 PM, Matthew Kaufman wrote:
> On Jul 20, 2011, at 3:48 PM, Colin Perkins wrote:
>
>    
>>
>> You're free to re-invent that wheel, if you wish. However, if you want to use RTP for RTCWeb, then you need to accept that doing so has both some advantages and some disadvantages.
>>      
> I believe we've managed to demonstrate that the disadvantages are sufficient to argue against using RTP for RTCWeb. Too bad, because it seems otherwise like such a good choice.
>
>    

If we digress to rhinophytophilia use-cases, which the step-by-step 
process exploits good flows in breath analyzers, in RTP sense, then I 
think this would be the focus to weigh advantages and disadvantages. Not 
everybody breaths syllables into the net online, imagine some terrified 
already of such analytics. Maybe someday we can predict this kind of 
discussion and prepare for the colossal mix of schemes, yet for now 
there is the coined bit of "no a/s/l". Sane? Certain safety.

-- 
--- http://twitter.com/Dzonatas_Sol ---
Web Development, Software Engineering
Ag-Biotech, Virtual Reality, Consultant


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On 7/20/11 2:37 PM, Bernard Aboba wrote:
> Agreed.
>
> To make this a bit more concrete. From your perspective, what is the
> preferred mechanism for file transfer in XMPP, and what would it take to
> support this in a browser?

The great thing about file transfer mechanisms in XMPP is that there are 
so many to choose from. :)

The most reliable mechanism is "in-band bytestreams", a simple chunking 
method that uses the XMPP channel itself. That's not the fastest method 
but it's the most reliable. Clearly it would be faster to set up a 
direct connection, or if that fails because of NATs to use a relay. We 
have a technology for that, too, but we don't have a lot of relays 
deployed on the network. Then there's the pseudo-tcp stuff that Google 
uses in Google Talk. Or you can upload the file somewhere and just send 
a URL (which makes a lot of sense if you're using BOSH instead of the 
traditional TCP connection method).

I'm not yet convinced that any of those methods is quite right for 
RTCWEB, but I haven't studied the problem intensively.

Peter

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Maybe, at some point.

At the moment though I am not convinced that this is the right answer.  
It always looks easier to re-design everything, but this approach more 
often than not ends up in a standard (and implementations) full of 
bugs.  Looking at what has transpired in Transport (now RAI) in the past 
15 years, one would easily find examples.

I think the decision on whether to ignore backward compatibility ought 
to be made when there is an agreement on core use cases and it is 
understood that none of them would be broken by lack of backward 
compatibility.

(It is not at all that I am against re-design on principle.  Quite the 
opposite!  But I believe that re-design is the research issue; here, we 
write a standard, and a standard needs to take into account a plethora 
of issues that the research phase may and should ignore.)

Igor

On 7/20/2011 7:15 PM, Matthew Kaufman wrote:
> ...
>
> Could we confine the claims about breakage to only ones that are NOT backwards compatibility issues?
>
> ...

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From: Matthew Kaufman <matthew.kaufman@skype.net>
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] How to multiplex between peers
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On 7/20/2011 8:34 PM, Igor Faynberg wrote:
>
> I think the decision on whether to ignore backward compatibility ought 
> to be made when there is an agreement on core use cases and it is 
> understood that none of them would be broken by lack of backward 
> compatibility.

It is trivial to demonstrate that an RTCWEB endpoint talking to another 
RTCWEB endpoint has no issues with backward compatibility.

The deceptive conflation of (possible) real issues with using SSRC as a 
multiplexing point for A+V over RTP for RTCWEB to RTCWEB and the class 
of issues that are ONLY relevant when talking RTCWEB to something that 
requires backwards compatibility is taking us farther, rather than 
closer, to a resolution on this issue. Once we agree to RTCWEB to Legacy 
will not use SSRC for multiplexing, there is no need, whatsoever, to 
bring up the potential backwards compatibility issues again.

Matthew Kaufman

From ron.even.tlv@gmail.com  Wed Jul 20 22:12:40 2011
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Aron Rosenberg'" <arosenberg@logitech.com>, <rtcweb@ietf.org>
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Date: Thu, 21 Jul 2011 08:09:54 +0300
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Hi,

I am not sure about your claim. The SSRC of the media sender is always there
, otherwise how can RTCP report or request action on a media stream. It is
true that for some of the RTCP messages like TMMBR it is inside the FCI part
of the message but it is there. As for understanding all RTCP messages, it
is not required today and the support is negotiated in SDP. 

For video streams I assume that RTCweb will mandate the support of controls
like FIR and TMBBR.

 

Regards

Roni Even

 

From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of
Aron Rosenberg
Sent: Thursday, July 21, 2011 12:16 AM
To: rtcweb@ietf.org
Subject: Re: [rtcweb] How to multiplex between peers

 

Another +1 for multiplexing everything together. My only worry is that the
base spec for RTCP doesn't mandate that SSRC is included in every RTCP
message type, and thus you could end up with the multiplexing layer needing
to understand every  RTCP messaged used for video (TMMBR, TMMBN, etc)


 


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style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Hi,<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>I am not sure about your claim. The SSRC of the media sender is =
always there , otherwise how can RTCP report or request action on a =
media stream. It is true that for some of the RTCP messages like TMMBR =
it is inside the FCI part of the message but it is there. As for =
understanding all RTCP messages, it is not required today and the =
support is negotiated in SDP. <o:p></o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>For video streams I assume that RTCweb will mandate the support of =
controls like FIR and TMBBR.<o:p></o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Regards<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Roni Even<o:p></o:p></span></p><p class=3DMsoNormal><span =
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1.0pt;padding:3.0pt 0in 0in 0in'><p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> =
rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] <b>On Behalf Of =
</b>Aron Rosenberg<br><b>Sent:</b> Thursday, July 21, 2011 12:16 =
AM<br><b>To:</b> rtcweb@ietf.org<br><b>Subject:</b> Re: [rtcweb] How to =
multiplex between peers<o:p></o:p></span></p></div></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>Another +1 =
for multiplexing everything together. My only worry is that the base =
spec for RTCP doesn't mandate that SSRC is included in every RTCP =
message type, and thus you could end up with the multiplexing layer =
needing to understand every &nbsp;RTCP messaged used for video (TMMBR, =
TMMBN, etc)<br clear=3Dall><o:p></o:p></p><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div></div></div></body></html>
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On 2011-07-20 16:24, Bernard Aboba wrote:
>

>  > My experience is the answer to the first scenario is not as quick as
> you would like and the answer to how long the second takes is about 8
> times longer than the first one. You might do a bit better than that
> depending on how clever your implementation is but it still a lot longer.
(I think Colin concluded it would be 2 not 8)
>
> [BA] Yes, that is correct. However at the moment I'd characterize that
> as a "high quality problem", since it isn't even clear that the current
> WHATWG API can handle that case, due to glare and ICE timing issues.
Not sure what you're meaning. We've made an experimental implementation, 
and do not see glare or ICE timing issues.


From csp@csperkins.org  Wed Jul 20 23:52:02 2011
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References: <4E259EAD.3060505@ericsson.com> <FAE78F7C-8C51-41C4-B3D7-6497396E12A5@cisco.com> <4E26C5CF.1080007@ericsson.com> <BLU152-W54BE1A03753680FF0094C4934C0@phx.gbl> <CAOJ7v-2kwiCipJSHmNT9GuGJJzEjPV-X00TLnf-LwbsJ1ADwDw@mail.gmail.com> <896BDC4C-849C-4553-89C8-7EFEF9FFEC6B@skype.net> <38DF8F00ABAB77498A75469448CB081B3AE69BEBC4@BE235.mail.lan> <DB9A2414-CE23-4F6E-811D-DEEBAF6E0280@csperkins.org> <42CD13B5-C0DB-457B-9131-6081DD795832@vidyo.com>
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On 21 Jul 2011, at 00:24, Jonathan Lennox wrote:
...
> I've read draft-perkins-rtcweb-rtp-usage-02, and I still don't =
understand what you're claiming SSRC-muxing would break.
>=20
> Running audio and video (and FEC, and RTX, and simulcast video, and =
Real-Time Text, and etc. etc.) in a single RTP session does, not, as far =
as I've been able to tell (after a lot of thought), break anything in =
RTP.
>=20
> To be sure, it's not backward compatible with RTP endpoints that =
assume a single source per session.  It would also be hard to retrofit =
SDP to describe and negotiate it.  And for some use cases it requires =
some source-level signaling, as in RFC 5576.  But the RTP model, per se, =
is perfectly amenable to it, as far as I can tell.


Section 2.2.1 of draft-perkins-rtcweb-rtp-usage-02 lists several =
problems that occur when multiplexing RTP sessions onto a single =
lower-level flow.=20

There are clearly some limitations caused: separate endpoints, quality =
of service, limitations with mixers and translators, retransmission, =
FEC, sampling group membership, and so on. There is also some breakage =
when several sessions are multiplexed: RTCP is problematic with several =
clock rates per session, SSRC allocation is broken, and the RTCP =
reporting interval and timing rules are broken.

You may not think some of these are important. Some can be avoided by =
the use of appropriate signalling. Some can be mitigated by allowing =
multiple different media types in an RTP session, or by splitting the =
SSRC space. I'm not arguing that there aren't ways to fix this problem, =
or ignore it in some cases, but they cause various degrees of change to =
RTP, and need to be carefully evaluated by AVTCORE if they're to be =
done.=20

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




From stewe@stewe.org  Thu Jul 21 00:43:20 2011
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From: Stephan Wenger <stewe@stewe.org>
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Hi Colin,

On 7.20.2011 23:51 , "Colin Perkins" <csp@csperkins.org> wrote:

>On 21 Jul 2011, at 00:24, Jonathan Lennox wrote:
>...
>> I've read draft-perkins-rtcweb-rtp-usage-02, and I still don't
>>understand what you're claiming SSRC-muxing would break.
>> 
>> Running audio and video (and FEC, and RTX, and simulcast video, and
>>Real-Time Text, and etc. etc.) in a single RTP session does, not, as far
>>as I've been able to tell (after a lot of thought), break anything in
>>RTP.
>> 
>> To be sure, it's not backward compatible with RTP endpoints that assume
>>a single source per session.  It would also be hard to retrofit SDP to
>>describe and negotiate it.  And for some use cases it requires some
>>source-level signaling, as in RFC 5576.  But the RTP model, per se, is
>>perfectly amenable to it, as far as I can tell.
>
>
>Section 2.2.1 of draft-perkins-rtcweb-rtp-usage-02 lists several problems
>that occur when multiplexing RTP sessions onto a single lower-level flow.
>
>There are clearly some limitations caused: separate endpoints, quality of
>service, limitations with mixers and translators, retransmission, FEC,
>sampling group membership, and so on. There is also some breakage when
>several sessions are multiplexed: RTCP is problematic with several clock
>rates per session, SSRC allocation is broken, and the RTCP reporting
>interval and timing rules are broken.
>
>You may not think some of these are important. Some can be avoided by the
>use of appropriate signalling. Some can be mitigated by allowing multiple
>different media types in an RTP session, or by splitting the SSRC space.
>I'm not arguing that there aren't ways to fix this problem, or ignore it
>in some cases, but they cause various degrees of change to RTP, and need
>to be carefully evaluated by AVTCORE if they're to be done.

Rtcweb is working towards what I would call a "system"
specification--something, the IETF (or at least RAI) is not usually
involved in.  Outside the IETF, it's not uncommon that "system"
specifications effectively modify an underlying base specification--RTP in
this case.  Such modifications can as easily remove functionality deemed
unnecessary as add new one.

Speaking about session multiplexing as an example, I would guess that a
lot of people here are implicitly arguing removal of functionality of RTP
not needed for the application--namely multicast support.  SSRC
(re-)allocation) may be broken, but since there should not be any need for
it (no multicast), who cares?  RTCP reporting intervals and timing rules
are broken: again, who cares if the RTP/RTCP communication relationship is
either endpoint-endpoint or endpoint-MCU (and RTCP is not being forwarded
to all endpoints)?  RTCP-Terminating MCU and not Mixer in RFC 5117
namespace?  We will not see network flooding or something with RTCP RRs in
point-to-point RTP communication relationships... I have not thought about
RTCP's issues with multiple clock rates, but those difficulties are
probably not insurmountable, either, because, while there may be multiple
clock rates, those ought to be a finite, and small, set.

Now to my key point: you argue these things need to be "carefully
evaluated" in AVTCORE.
IMO, while it may be desirable for AVTCORE experts to chip in (as they do
on this very thread already:-), it is by no means necessary.  Neither
formally, not practically.  And certainly AVTCORE should not assume a
"blocking" position.  This puts IETF system level specification at a
disadvantage to system level specification work elsewhere (where it
happens on a daily basis, see ITU, see 3GPP), and where no one asks for
AVTCORE's permission to re-use RTP data structures without re-using all of
RTP.  Tinkering with the multicast functionally of RTP is not new, there
is a lot of implementation experience in the field, and for once we can
indeed rely on "running code" answers.

I advocate pragmatism.
(And, heretic as I am, in the long term we should think about RTPv3,
dumping multicast.  Have I said this before? :-)

Regards,
Stephan

P.s.: I haven't though much about shim layers or other techniques that
multiplex RTP sessions on a single low level flow.  If any of these ideas
were implemented only to preserve RTP functionalities not needed for the
application in question, then I would argue that being a silly idea.

> 
>
>-- 
>Colin Perkins
>http://csperkins.org/
>
>
>
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb



From csp@csperkins.org  Thu Jul 21 01:25:09 2011
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On 21 Jul 2011, at 08:43, Stephan Wenger wrote:
> On 7.20.2011 23:51 , "Colin Perkins" <csp@csperkins.org> wrote:
>> On 21 Jul 2011, at 00:24, Jonathan Lennox wrote:
>> ...
> Rtcweb is working towards what I would call a "system" =
specification--something, the IETF (or at least RAI) is not usually =
involved in.  Outside the IETF, it's not uncommon that "system" =
specifications effectively modify an underlying base specification--RTP =
in this case.  Such modifications can as easily remove functionality =
deemed unnecessary as add new one.
>=20
> Speaking about session multiplexing as an example, I would guess that =
a lot of people here are implicitly arguing removal of functionality of =
RTP not needed for the application--namely multicast support.  SSRC =
(re-)allocation) may be broken, but since there should not be any need =
for it (no multicast), who cares?  RTCP reporting intervals and timing =
rules are broken: again, who cares if the RTP/RTCP communication =
relationship is either endpoint-endpoint or endpoint-MCU (and RTCP is =
not being forwarded to all endpoints)?  RTCP-Terminating MCU and not =
Mixer in RFC 5117 namespace?  We will not see network flooding or =
something with RTCP RRs in point-to-point RTP communication =
relationships... I have not thought about RTCP's issues with multiple =
clock rates, but those difficulties are probably not insurmountable, =
either, because, while there may be multiple clock rates, those ought to =
be a finite, and small, set.

I don't think anyone is arguing that there difficulties cannot be =
solved. Of course they can. What's being argued is that the result is =
different to RTP, is not backwards compatible, and is difficult to =
gateway into a standard RTP environment.=20

Some of us believe that there's benefit to being able to reuse the RTP =
infrastructure, and to easily interoperate with existing systems. I do =
not believe the benefit of saving a small number of UDP ports, in a =
browser environment that regularly uses dozens of TCP ports, outweighs =
the loss of compatibility with other systems.

> Now to my key point: you argue these things need to be "carefully =
evaluated" in AVTCORE. IMO, while it may be desirable for AVTCORE =
experts to chip in (as they do on this very thread already:-), it is by =
no means necessary.  Neither formally, not practically.  And certainly =
AVTCORE should not assume a "blocking" position.  This puts IETF system =
level specification at a disadvantage to system level specification work =
elsewhere (where it happens on a daily basis, see ITU, see 3GPP), and =
where no one asks for AVTCORE's permission to re-use RTP data structures =
without re-using all of RTP.  Tinkering with the multicast functionally =
of RTP is not new, there is a lot of implementation experience in the =
field, and for once we can indeed rely on "running code" answers.
>=20
> I advocate pragmatism. (And, heretic as I am, in the long term we =
should think about RTPv3, dumping multicast.  Have I said this before? =
:-)

You have, of course. Many people have. What's not clear to be is how you =
would remove multicast support from RTP without also removing multiparty =
support? I see lots of RTP features that are there to support group =
communication, very few that are specific to multicast.

And, besides, I can certainly envisage scenarios where one might want a =
browser-based RTP implementation to watch the multicast IPTV flows that =
ISPs are now offering.=20

Colin=

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On 07/21/11 02:41, Henry Sinnreich wrote:
> +1
>
>> If anything this is an argument for ignoring RTP and RTCP and doing
>> something entirely new that is actually appropriate for what we're
>> trying to build, not living with crap just because there's an RFC for it.
> Not to forget disposing of ancient SDP as well.
> Use standard metadata instead, since it is equally usable for all apps, not
> only for RTC apps.
Henry, which standard?
http://www.xkcd.com/927/ applies to metadata too, I think.

> Thanks, Henry
>
>
> On 7/20/11 5:38 PM, "Matthew Kaufman"<matthew.kaufman@skype.net>  wrote:
>
>> On 7/20/2011 3:31 PM, Colin Perkins wrote:
>>> On 20 Jul 2011, at 22:15, Aron Rosenberg wrote:
>>>> Another +1 for multiplexing everything together. My only worry is
>>>> that the base spec for RTCP doesn't mandate that SSRC is included in
>>>> every RTCP message type, and thus you could end up with the
>>>> multiplexing layer needing to understand every  RTCP messaged used
>>>> for video (TMMBR, TMMBN, etc)
>>> ...which is another reason why this doesn't work in all cases, in
>>> addition to the reasons we document in draft-perkins-rtcweb-rtp-usage-02
>>>
>> So let me get this straight... RTP and RTCP is broken and inflexible and
>> therefore we must use it, and use it in the broken and inflexible way?
>>
>> If anything this is an argument for ignoring RTP and RTCP and doing
>> something entirely new that is actually appropriate for what we're
>> trying to build, not living with crap just because there's an RFC for it.
>>
>> Matthew Kaufman
>> _______________________________________________
>> 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  Thu Jul 21 02:18:41 2011
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Hi Colin,

On 7.21.2011 01:25 , "Colin Perkins" <csp@csperkins.org> wrote:

>[...]
>
>> Now to my key point: you argue these things need to be "carefully
>>evaluated" in AVTCORE. IMO, while it may be desirable for AVTCORE
>>experts to chip in (as they do on this very thread already:-), it is by
>>no means necessary.  Neither formally, not practically.  And certainly
>>AVTCORE should not assume a "blocking" position.  This puts IETF system
>>level specification at a disadvantage to system level specification work
>>elsewhere (where it happens on a daily basis, see ITU, see 3GPP), and
>>where no one asks for AVTCORE's permission to re-use RTP data structures
>>without re-using all of RTP.  Tinkering with the multicast functionally
>>of RTP is not new, there is a lot of implementation experience in the
>>field, and for once we can indeed rely on "running code" answers.
>> 
>> I advocate pragmatism. (And, heretic as I am, in the long term we
>>should think about RTPv3, dumping multicast.  Have I said this before?
>>:-)
>
>You have, of course. Many people have. What's not clear to be is how you
>would remove multicast support from RTP without also removing multiparty
>support? I see lots of RTP features that are there to support group
>communication, very few that are specific to multicast.

Remove multiparty support, then.  Terminology glitch on my side.

>
>And, besides, I can certainly envisage scenarios where one might want a
>browser-based RTP implementation to watch the multicast IPTV flows that
>ISPs are now offering.

There is no reason why RTPv2 and RTPv3 couldn't co-exist, in a browser or
elsewhere.  They do, today, if you have a videoconferencing software and
an IPTV software on the same box.

It would be stupid to deprecate RTPv2 including multiparty support; no one
I know is arguing in favor of that.

Problem is, AVT (and, admittedly, the industry at large) did not act on
the RTPv3 idea.  Instead, application spec profiling continued with many
(dozens?  I lost count a long time ago) of non-interoperabile solutions
for the optimized-for-point-to-point RTP space.  (To a certain extent that
even happens in the multiparty case with different IPTV standards.) We are
about to create one more P2P RTP flavor here.  That's sad, but (I fear)
now unavoidable.  From what I gather, people don't think we can wait half
a decade for an RTPv3 proposal to make it through an AVT-to-be-named WG.

Stephan

> 
>
>Colin



From keith.drage@alcatel-lucent.com  Thu Jul 21 03:44:49 2011
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: Stephan Wenger <stewe@stewe.org>, Colin Perkins <csp@csperkins.org>, Jonathan Lennox <jonathan@vidyo.com>
Date: Thu, 21 Jul 2011 12:44:38 +0200
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This discussion seems to be hovering in some sort of middle ground.

Either 1)

It has not decided what transport to use for real time data, in which case =
it should document the requirements for such a transport protocol, and then=
 evaluate whether existing solutions meet those requirements, versus develo=
ping an entirely new protocol.

Or 2)

I has decided that RTP should be used, in which case it should identify the=
 shortcomings (i.e. a set of requirements on how it needs to be further dev=
eloped) and send these to the AVTCORE working group.

Neither of those seem to be happening at the moment.

Regards

Keith

> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
> Of Stephan Wenger
> Sent: 21 July 2011 08:43
> To: Colin Perkins; Jonathan Lennox
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] How to multiplex between peers
>=20
> Hi Colin,
>=20
> On 7.20.2011 23:51 , "Colin Perkins" <csp@csperkins.org> wrote:
>=20
> >On 21 Jul 2011, at 00:24, Jonathan Lennox wrote:
> >...
> >> I've read draft-perkins-rtcweb-rtp-usage-02, and I still don't
> >>understand what you're claiming SSRC-muxing would break.
> >>
> >> Running audio and video (and FEC, and RTX, and simulcast video, and
> >>Real-Time Text, and etc. etc.) in a single RTP session does, not, as fa=
r
> >>as I've been able to tell (after a lot of thought), break anything in
> >>RTP.
> >>
> >> To be sure, it's not backward compatible with RTP endpoints that assum=
e
> >>a single source per session.  It would also be hard to retrofit SDP to
> >>describe and negotiate it.  And for some use cases it requires some
> >>source-level signaling, as in RFC 5576.  But the RTP model, per se, is
> >>perfectly amenable to it, as far as I can tell.
> >
> >
> >Section 2.2.1 of draft-perkins-rtcweb-rtp-usage-02 lists several problem=
s
> >that occur when multiplexing RTP sessions onto a single lower-level flow=
.
> >
> >There are clearly some limitations caused: separate endpoints, quality o=
f
> >service, limitations with mixers and translators, retransmission, FEC,
> >sampling group membership, and so on. There is also some breakage when
> >several sessions are multiplexed: RTCP is problematic with several clock
> >rates per session, SSRC allocation is broken, and the RTCP reporting
> >interval and timing rules are broken.
> >
> >You may not think some of these are important. Some can be avoided by th=
e
> >use of appropriate signalling. Some can be mitigated by allowing multipl=
e
> >different media types in an RTP session, or by splitting the SSRC space.
> >I'm not arguing that there aren't ways to fix this problem, or ignore it
> >in some cases, but they cause various degrees of change to RTP, and need
> >to be carefully evaluated by AVTCORE if they're to be done.
>=20
> Rtcweb is working towards what I would call a "system"
> specification--something, the IETF (or at least RAI) is not usually
> involved in.  Outside the IETF, it's not uncommon that "system"
> specifications effectively modify an underlying base specification--RTP i=
n
> this case.  Such modifications can as easily remove functionality deemed
> unnecessary as add new one.
>=20
> Speaking about session multiplexing as an example, I would guess that a
> lot of people here are implicitly arguing removal of functionality of RTP
> not needed for the application--namely multicast support.  SSRC
> (re-)allocation) may be broken, but since there should not be any need fo=
r
> it (no multicast), who cares?  RTCP reporting intervals and timing rules
> are broken: again, who cares if the RTP/RTCP communication relationship i=
s
> either endpoint-endpoint or endpoint-MCU (and RTCP is not being forwarded
> to all endpoints)?  RTCP-Terminating MCU and not Mixer in RFC 5117
> namespace?  We will not see network flooding or something with RTCP RRs i=
n
> point-to-point RTP communication relationships... I have not thought abou=
t
> RTCP's issues with multiple clock rates, but those difficulties are
> probably not insurmountable, either, because, while there may be multiple
> clock rates, those ought to be a finite, and small, set.
>=20
> Now to my key point: you argue these things need to be "carefully
> evaluated" in AVTCORE.
> IMO, while it may be desirable for AVTCORE experts to chip in (as they do
> on this very thread already:-), it is by no means necessary.  Neither
> formally, not practically.  And certainly AVTCORE should not assume a
> "blocking" position.  This puts IETF system level specification at a
> disadvantage to system level specification work elsewhere (where it
> happens on a daily basis, see ITU, see 3GPP), and where no one asks for
> AVTCORE's permission to re-use RTP data structures without re-using all o=
f
> RTP.  Tinkering with the multicast functionally of RTP is not new, there
> is a lot of implementation experience in the field, and for once we can
> indeed rely on "running code" answers.
>=20
> I advocate pragmatism.
> (And, heretic as I am, in the long term we should think about RTPv3,
> dumping multicast.  Have I said this before? :-)
>=20
> Regards,
> Stephan
>=20
> P.s.: I haven't though much about shim layers or other techniques that
> multiplex RTP sessions on a single low level flow.  If any of these ideas
> were implemented only to preserve RTP functionalities not needed for the
> application in question, then I would argue that being a silly idea.
>=20
> >
> >
> >--
> >Colin Perkins
> >http://csperkins.org/
> >
> >
> >
> >_______________________________________________
> >rtcweb mailing list
> >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 harald@alvestrand.no  Thu Jul 21 04:05:33 2011
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On 07/21/11 12:44, DRAGE, Keith (Keith) wrote:
> This discussion seems to be hovering in some sort of middle ground.
>
> Either 1)
>
> It has not decided what transport to use for real time data, in which case it should document the requirements for such a transport protocol, and then evaluate whether existing solutions meet those requirements, versus developing an entirely new protocol.
>
> Or 2)
>
> I has decided that RTP should be used, in which case it should identify the shortcomings (i.e. a set of requirements on how it needs to be further developed) and send these to the AVTCORE working group.
Don't forget

3)

It has decided that RTP should be used, with no changes and no new 
extensions, and is going through the ritual of swearing at the resulting 
shortcomings while figuring out which parts of RTP it can say "we don't 
need them, so it's OK to do things that mean we can't use them" to.

If this group gets lined up behind an 1-year dependency on another group 
before it can deliver its initial work products, this group is going to 
be irrelevant.

               Harald


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Agreed. 3 options. I'd rephrase the third as package only what is available=
 amongst existing protocol, and document what it doesn't do as a result.

But the current discussion is still in middle ground between all of them wi=
th no clear direction forward as to which it wants to do.

Regards

Keith

> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
> Of Harald Alvestrand
> Sent: 21 July 2011 12:06
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] How to multiplex between peers
>=20
> On 07/21/11 12:44, DRAGE, Keith (Keith) wrote:
> > This discussion seems to be hovering in some sort of middle ground.
> >
> > Either 1)
> >
> > It has not decided what transport to use for real time data, in which
> case it should document the requirements for such a transport protocol,
> and then evaluate whether existing solutions meet those requirements,
> versus developing an entirely new protocol.
> >
> > Or 2)
> >
> > I has decided that RTP should be used, in which case it should identify
> the shortcomings (i.e. a set of requirements on how it needs to be furthe=
r
> developed) and send these to the AVTCORE working group.
> Don't forget
>=20
> 3)
>=20
> It has decided that RTP should be used, with no changes and no new
> extensions, and is going through the ritual of swearing at the resulting
> shortcomings while figuring out which parts of RTP it can say "we don't
> need them, so it's OK to do things that mean we can't use them" to.
>=20
> If this group gets lined up behind an 1-year dependency on another group
> before it can deliver its initial work products, this group is going to
> be irrelevant.
>=20
>                Harald
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From dzonatas@gmail.com  Thu Jul 21 07:28:14 2011
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Yes, and note that the issue experienced is more wide-spread than this 
WG. This is clearly evident when protocols attempt to sustain stateful 
connections and virtualization later proves false security. Keep this in 
mind before thinking to toss RTP and UDP.

I can't even uphold SSL as sustainable anymore.

On 07/21/2011 04:56 AM, DRAGE, Keith (Keith) wrote:
> Agreed. 3 options. I'd rephrase the third as package only what is available amongst existing protocol, and document what it doesn't do as a result.
>
> But the current discussion is still in middle ground between all of them with no clear direction forward as to which it wants to do.
>
> Regards
>
> Keith
>
>    
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
>> Of Harald Alvestrand
>> Sent: 21 July 2011 12:06
>> To: rtcweb@ietf.org
>> Subject: Re: [rtcweb] How to multiplex between peers
>>
>> On 07/21/11 12:44, DRAGE, Keith (Keith) wrote:
>>      
>>> This discussion seems to be hovering in some sort of middle ground.
>>>
>>> Either 1)
>>>
>>> It has not decided what transport to use for real time data, in which
>>>        
>> case it should document the requirements for such a transport protocol,
>> and then evaluate whether existing solutions meet those requirements,
>> versus developing an entirely new protocol.
>>      
>>> Or 2)
>>>
>>> I has decided that RTP should be used, in which case it should identify
>>>        
>> the shortcomings (i.e. a set of requirements on how it needs to be further
>> developed) and send these to the AVTCORE working group.
>> Don't forget
>>
>> 3)
>>
>> It has decided that RTP should be used, with no changes and no new
>> extensions, and is going through the ritual of swearing at the resulting
>> shortcomings while figuring out which parts of RTP it can say "we don't
>> need them, so it's OK to do things that mean we can't use them" to.
>>
>> If this group gets lined up behind an 1-year dependency on another group
>> before it can deliver its initial work products, this group is going to
>> be irrelevant.
>>
>>                 Harald
>>
>> _______________________________________________
>> 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
>
>    


-- 
--- http://twitter.com/Dzonatas_Sol ---
Web Development, Software Engineering
Ag-Biotech, Virtual Reality, Consultant


From henry.sinnreich@gmail.com  Thu Jul 21 08:36:36 2011
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From: Henry Sinnreich <henry.sinnreich@gmail.com>
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> Henry, which standard?

Thanks for the good question Harald!

A good start for metadata standards to replace SDP and thus applicable
across any web apps is

http://dublincore.org/metadata-basics/

Such metadata is actually implemented in successful RAI products, but I
would rather not go there on this post.

Henry


On 7/21/11 4:16 AM, "Harald Alvestrand" <harald@alvestrand.no> wrote:

> On 07/21/11 02:41, Henry Sinnreich wrote:
>> +1
>> 
>>> If anything this is an argument for ignoring RTP and RTCP and doing
>>> something entirely new that is actually appropriate for what we're
>>> trying to build, not living with crap just because there's an RFC for it.
>> Not to forget disposing of ancient SDP as well.
>> Use standard metadata instead, since it is equally usable for all apps, not
>> only for RTC apps.
> Henry, which standard?
> http://www.xkcd.com/927/ applies to metadata too, I think.
> 
>> Thanks, Henry
>> 
>> 
>> On 7/20/11 5:38 PM, "Matthew Kaufman"<matthew.kaufman@skype.net>  wrote:
>> 
>>> On 7/20/2011 3:31 PM, Colin Perkins wrote:
>>>> On 20 Jul 2011, at 22:15, Aron Rosenberg wrote:
>>>>> Another +1 for multiplexing everything together. My only worry is
>>>>> that the base spec for RTCP doesn't mandate that SSRC is included in
>>>>> every RTCP message type, and thus you could end up with the
>>>>> multiplexing layer needing to understand every  RTCP messaged used
>>>>> for video (TMMBR, TMMBN, etc)
>>>> ...which is another reason why this doesn't work in all cases, in
>>>> addition to the reasons we document in draft-perkins-rtcweb-rtp-usage-02
>>>> 
>>> So let me get this straight... RTP and RTCP is broken and inflexible and
>>> therefore we must use it, and use it in the broken and inflexible way?
>>> 
>>> If anything this is an argument for ignoring RTP and RTCP and doing
>>> something entirely new that is actually appropriate for what we're
>>> trying to build, not living with crap just because there's an RFC for it.
>>> 
>>> Matthew Kaufman
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>> 
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>> 
> 
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb



From jonathan@vidyo.com  Thu Jul 21 08:59:36 2011
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From: Jonathan Lennox <jonathan@vidyo.com>
To: Colin Perkins <csp@csperkins.org>
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On Jul 21, 2011, at 2:51 AM, Colin Perkins wrote:

> On 21 Jul 2011, at 00:24, Jonathan Lennox wrote:
> ...
>> I've read draft-perkins-rtcweb-rtp-usage-02, and I still don't understan=
d what you're claiming SSRC-muxing would break.
>>=20
>> Running audio and video (and FEC, and RTX, and simulcast video, and Real=
-Time Text, and etc. etc.) in a single RTP session does, not, as far as I'v=
e been able to tell (after a lot of thought), break anything in RTP.
>>=20
>> To be sure, it's not backward compatible with RTP endpoints that assume =
a single source per session.  It would also be hard to retrofit SDP to desc=
ribe and negotiate it.  And for some use cases it requires some source-leve=
l signaling, as in RFC 5576.  But the RTP model, per se, is perfectly amena=
ble to it, as far as I can tell.
>=20
>=20
> Section 2.2.1 of draft-perkins-rtcweb-rtp-usage-02 lists several problems=
 that occur when multiplexing RTP sessions onto a single lower-level flow.=
=20
>=20
> There are clearly some limitations caused: separate endpoints, quality of=
 service, limitations with mixers and translators, retransmission, FEC, sam=
pling group membership, and so on. There is also some breakage when several=
 sessions are multiplexed: RTCP is problematic with several clock rates per=
 session, SSRC allocation is broken, and the RTCP reporting interval and ti=
ming rules are broken.
>=20
> You may not think some of these are important. Some can be avoided by the=
 use of appropriate signalling. Some can be mitigated by allowing multiple =
different media types in an RTP session, or by splitting the SSRC space. I'=
m not arguing that there aren't ways to fix this problem, or ignore it in s=
ome cases, but they cause various degrees of change to RTP, and need to be =
carefully evaluated by AVTCORE if they're to be done.=20

I absolutely agree that multiplexing several RTP sessions into a single low=
er-level flow could break things, but that's not what I'm advocating.  Inst=
ead, I'm only advocating relaxing the rule that every source in a *single* =
RTP session -- a session which follows all the RFC 3550 rules -- must have =
the same top-level media type.

Since RTP was defined to allow up to 2^32-1 distinct sources in a session, =
all the rules of RFC 3550 seem to follow naturally, as do other RTP-family =
specs except those whose designers forgot to take RTP's group-communication=
 nature into account.

It may be useful to relax *more* rules, for the topological cases that Step=
han mentioned (endpoint-endpoint or RTCP-terminating-MCU) -- particularly s=
ince a lot of RTP's (or specifically, RTCP's) existing timing rules don't m=
ake a lot of sense in scenarios where you have widely asymmetric bandwidths=
.  This might need to formally be a new RTP profile, though I'd rather avoi=
d that if possible.  But the starting point should be, I think, assuming a =
single RTP session with arbitrarily many distinct sources, and see if the R=
FC 3550 rules lead us somewhere bad *in that scenario*.

--
Jonathan Lennox
jonathan@vidyo.com



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Chair,

I claimed yesterday that I wouldn't post here today, this date, unless there was something included to spend at-least one trillion worth of transient dollars.

Found one.

[...QoS...]

On 07/20/2011 02:25 PM, Cullen Jennings wrote:
> The other issue is that it is not clear what values you should set the DSCP to. If you read the appropriate RFCs, you will come to the conclusion that they are pretty confused on the topic. If you look at the values being used by Cisco, the mapping from 802.11, and Microsoft, you would notice they are not the same and there may be some uh "gaming the system" going on. One of the things this WG could do would be to come up with some clear guidance on what values to use for browsers and when to use them.
>    


Silverlight/Moonlight crashes or hangs in Crome/book. This means the 
"browser based run-time tests" are unfair. I could finger if I was a 
judge, yet I am clearly not one.

I can define "stupid" simply by [A-Z]+pyramidal-shape found, and I have 
already declared that elsewhere for good causes. If judges want to use 
graphene or the silver pen is their choice on the minutes, yet I seen 
why some question anti-truth/anti-trust when that choice is already 
tainted. I doubt anything like "auto-jury" is any game, yet still looks 
stupid like "Gaming the system."

Tax the debt makes sense to me. Any clue? Or two?

Filed: Friday.

-- 
--- http://twitter.com/Dzonatas_Sol ---
Web Development, Software Engineering
Ag-Biotech, Virtual Reality, Consultant


From magnus.westerlund@ericsson.com  Sat Jul 23 03:58:35 2011
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Cullen,

I mostly agree with Colin here. But looking at the advanced use case
this might be considered, and a bit on the control hierachy around it I
would say that this is either 2, 3 or 4 RTP Sessions. It is primarily
about how you would write your application and what type of RTP
conference mixer you have and what behavior it has.

For an application that has over the top labeling of the streams and
where the RTP mixer also can handle these streams appropriate it can be
only two RTP sessions. One for audio and one for video using SSRC
multiplexing to mux all the video streams.

In case you have your 5 participants being composited together in the
mixer and delivered as one stream, or some other behavior in the central
node you likely need to have one RTP session for that behavior. The
Active speaker and the presentation could simply be relayed as equal
SSRC multiplexed streams in one RTP session. Thus giving us one audio
and two video.

In worst case from a session perspective, the application writer has
decided that it is simplest to put map the stream semantics to the RTP
sessions, thus creating one Video for conference participants, one for
the presentation, and one for the active speaker(presenter) video. Thus
resulting in four streams.

There is two main things affecting the RTP session usage in an
particular application. The need of the application itself, and the
capabilities of any central node to perform the behavior desired by the
application.

Yes, I know that RTP Sessions appear a difficult concept. But, I would
recommend that everyone remembers that:

- An application can contain any number of RTP sessions.
- An RTP session can carry multiple media streams of the same type
- Multiple RTP sessions for the same media type can be used to separate
streams of different usages/behavior/processing in the applications or
any central node.

Cheers

Magnus

On 2011-07-21 00:21, Colin Perkins wrote:
> On 20 Jul 2011, at 15:11, Cullen Jennings wrote:

>> Second scenario: same as first address wise but instead of a single
>> audio stream they want to set up a single audio stream  to a
>> conference bridge plus 7 video streams for the video from the 5
>> people on the bridge plus a presentation stream and stream of video
>> from active speaker. I don't really care much about the scenario
>> other than there are 8 streams being set up. But this type of
>> scenario is becoming very common for multi party video chat as it
>> allows you to see perhaps small versions of everyone plus a large
>> version of active speaker.
>> 
>> My experience is the answer to the first scenario is not as quick
>> as you would like and the answer to how long the second takes is
>> about 8 times longer than the first one. You might do a bit better
>> than that depending on how clever your implementation is but it
>> still a lot longer.
> 
> I may be misunderstanding, but why would this take 8 times as long to
> set up? Wouldn't you implement it as one UDP flow for the RTP audio,
> plus one UDP flow for the RTP video (with the 7 video streams
> multiplexed by SSRC, since RTP has always allowed multiparty flows of
> the same media type in a single RTP session)? This might take twice
> as long to setup as a single audio flow in the worst case, but not
> eight times as long.
> 
-- 

Magnus Westerlund

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On 2011-07-21 11:01, Stephan Wenger wrote:

> Remove multiparty support, then.  Terminology glitch on my side.

So you argue for removing multi-party support the fact that RTCWEB have
clear multi-party use cases in our use case document? Even you make
arguments in your layered coding pitch using functions which is based on
the multi-party functions of RTP. So I don't see how that match up.

Cheers

Magnus Westerlund

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On 2011-07-21 00:38, Matthew Kaufman wrote:
> On 7/20/2011 3:31 PM, Colin Perkins wrote:
>> On 20 Jul 2011, at 22:15, Aron Rosenberg wrote:

> 
> So let me get this straight... RTP and RTCP is broken and inflexible and 
> therefore we must use it, and use it in the broken and inflexible way?

And Matthew we haven't argued against attempting to fix the big issue
you see with it, namely that it requires a lower layer to separate RTP
sessions. We just don't see it happing quickly or in the way you
proposed. But I think we are going to very seriously consider attempting
to do something about the short coming in AVTCORE.

> 
> If anything this is an argument for ignoring RTP and RTCP and doing 
> something entirely new that is actually appropriate for what we're 
> trying to build, not living with crap just because there's an RFC for it.

I don't think we have an realistic alternative of doing something new. I
would love to develop a new real-time media transport protocol. I just
think that it will take 3 years before we are close to having something
published. I also fear that it might suffer from second system syndrome
and be hellishly complex.

I also don't think it helps the people that want to gateway to their
legacy systems.

Do you have any other well defined media transport alternative for us to
use?

Cheers

Magnus Westerlund

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On Jul 20, 2011, at 8:37 , Justin Uberti wrote:

>=20
>=20
> On Wed, Jul 20, 2011 at 9:15 AM, Harald Alvestrand =
<harald@alvestrand.no> wrote:
> On 07/20/11 03:13, Cullen Jennings wrote:
> On Jul 18, 2011, at 15:11 , Randell Jesup wrote:
>=20
> All that said - yes, there are complexity issues to consider, which =
was why I was suggesting leveraging
> an existing tunneled protocol like SCTP or even TCP-over-UDP.
> I agree with bunch of what you are saying but the previous several =
times I've seen the use SCTP over UDP conversation, it usually ends =
about the point the we ask for a user land implementation. Whatever we =
do more or less has to be user land or already exist in the OS that =
people run browsers on.
>=20
> If someone has a user land implementation of SCTP or TCP, I imagine =
that might change the outcome a bit from previous times. I have not =
looked at the TCP over UPD library Justin mentioned but, at casual =
glance, I noticed is around 400 lines of code which is very small, which =
is cool. But given the lines of code in say the BSD TCP stack, it does =
make me wonder what's missing. That said, I don't think we need =
something with the perforce of the BSD TCP stack as long as what is =
there is TCP friendly and robust.
> I believe this is the current URL:
>=20
> =
http://code.google.com/p/libjingle/source/browse/trunk/talk/session/tunnel=
/pseudotcpchannel.cc (500 or so lines).
>=20
> The real protocol implementation is here:
> =
http://code.google.com/p/libjingle/source/browse/trunk/talk/p2p/base/pseud=
otcp.cc
> which is another 1000 lines (not counting the various .h files).
>=20
> We're not THAT good at writing compact code :-)
>=20
> Yes, I was going to send a similar email. There are some =
deficiencies/optimizations for the fact it runs over ICE-UDP - doesn't =
do path MTU discovery, no checksum, no OOB data - but it is based on TCP =
and implements slow start and other congestion avoidance mechanisms, and =
is actively used by a number of applications where reliable transfer is =
important, such as file transfer and remote desktop applications.
> =20
>=20

It would be really interesting to see someone write a draft that subsets =
TCP to make it=20

1) still be congestion safe and TCP friendly=20

2) be valid TCP from point of view nats and firewalls

3) match an implementation like the one you have here.=20

Most our use cases do not need OOB data. A subset protocol that was =
missing something like path MTU discovery might be slower than full =
blown modern TCP but again, most out use cases would still work just =
fine. As much as the IETF hates sub setting protocols, having something =
concrete and in hand like this would be a pretty useful thing. As well =
as being useful for the wwebrtc work, something like this would be =
useful for very constrained systems such very cheap sensor networks. I'm =
seeing lots of really minimal subset of TCP being done for these very =
low end system.=20



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References: <4E0832FE.7010401@ericsson.com> <4E1DC07B.7000807@ericsson.com> <D1BE71E1-4F3B-474E-8A28-AA53CE6B684E@cisco.com> <CA+9kkMCJiE+bfEqZzOBo46aXVH-H2sehHh6UJv3tVdJKGjaokQ@mail.gmail.com> <49CD37FC-7951-45A0-84C4-A443F8B151F3@cisco.com> <4E24AF7C.4080604@jesup.org> <CB57E808-FB8D-41D7-90C6-0EA1051629A8@cisco.com> <4E26D4D6.3020602@alvestrand.no> <CAOJ7v-3RLhaxOWyNJ6p4nE7QyFxOCAeM7xoENPvNhYhoDmyKOw@mail.gmail.com> <87CA4E28-DF95-4A03-9809-BC1B2077B4B1@cisco.com>
To: Cullen Jennings <fluffy@cisco.com>
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On 23 Jul 2011, at 03:33, Cullen Jennings wrote:

>=20
> On Jul 20, 2011, at 8:37 , Justin Uberti wrote:
>>=20
>> Yes, I was going to send a similar email. There are some =
deficiencies/optimizations for the fact it runs over ICE-UDP - doesn't =
do path MTU discovery, no checksum, no OOB data - but it is based on TCP =
and implements slow start and other congestion avoidance mechanisms, and =
is actively used by a number of applications where reliable transfer is =
important, such as file transfer and remote desktop applications.
>>=20
>>=20
>=20
> It would be really interesting to see someone write a draft that =
subsets TCP to make it=20
>=20

I just want to flag that TCP isn't an especially good fit here, =
datagrams delivered in events are a very useful
abstraction in JavaScript, which doesn't really do streams of data very =
well.=20

I know we can do datagrams over TCP,  but
it seems wasteful to de-packetize in the TCP layer, just to try and =
re-packetize it a couple of layers further up.

If there is a reliable datagram protocol as opposed to a reliable =
stream, I'd go for the datagram every time.

Tim.


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<html><head></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
"><br><div><div>On 23 Jul 2011, at 03:33, Cullen Jennings =
wrote:</div><br class=3D"Apple-interchange-newline"><blockquote =
type=3D"cite"><div><br>On Jul 20, 2011, at 8:37 , Justin Uberti =
wrote:<br><blockquote type=3D"cite"><font class=3D"Apple-style-span" =
color=3D"#1855d0"><br></font></blockquote><blockquote type=3D"cite">Yes, =
I was going to send a similar email. There are some =
deficiencies/optimizations for the fact it runs over ICE-UDP - doesn't =
do path MTU discovery, no checksum, no OOB data - but it is based on TCP =
and implements slow start and other congestion avoidance mechanisms, and =
is actively used by a number of applications where reliable transfer is =
important, such as file transfer and remote desktop =
applications.<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote =
type=3D"cite"><br></blockquote><br>It would be really interesting to see =
someone write a draft that subsets TCP to make it =
<br><br></div></blockquote><br></div><div>I just want to flag that TCP =
isn't an especially good fit here, datagrams delivered in events are a =
very useful</div><div>abstraction in JavaScript, which doesn't really do =
streams of data very well.&nbsp;</div><div><br></div><div>I know we can =
do datagrams over TCP, &nbsp;but</div><div>it seems wasteful to =
de-packetize in the TCP layer, just to try and re-packetize it a couple =
of layers further up.</div><div><br></div><div>If there is a reliable =
datagram protocol as opposed to a reliable stream, I'd go for the =
datagram every =
time.</div><div><br></div><div>Tim.</div><br></body></html>=

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From emil@sip-communicator.org  Sat Jul 23 08:26:37 2011
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=D0=9D=D0=B0 23.07.11 04:33, Cullen Jennings =D0=BD=D0=B0=D0=BF=D0=B8=D1=81=
=D0=B0:
> It would be really interesting to see someone write a draft that subset=
s TCP to make it=20
>=20
> 1) still be congestion safe and TCP friendly=20
>=20
> 2) be valid TCP from point of view nats and firewalls

Why does it need to be valid TCP? What NATs and firewalls will see is
valid UDP which is the whole point, isn't it?

The fact that the protocol carried in the datagrams is TCP-like only
matters to the application.

Emil

--
http://jitsi.org


From fd@w3.org  Sat Jul 23 08:52:22 2011
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Subject: [rtcweb] Logistics info on W3C WebRTC meeting - Saturday 23 July
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Hi,

For those of you who were planning to attend the W3C Web Real-Time Communications Working Group meeting this afternoon, here are latest logistics info.

Location: Quebec City Convention Center (same location as IETF meeting)
Room: room 2103 (seems like finding the room is not trivial, allow for enough time!)
Time: from 14:00 to 18:00
Agenda: http://www.w3.org/2011/04/webrtc/wiki/July_23_2011
IRC: #webrtc channel on irc.w3.org, port 6665
(a Web interface is available at: http://irc.w3.org )

Francois Daoust,
W3C Staff Contact for the W3C Web RTC WG.

From stewe@stewe.org  Sat Jul 23 14:13:13 2011
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From: Stephan Wenger <stewe@stewe.org>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Hi Magnus,

Not sure you got the context right here.  The "removal of multiparty
support" was related to my suggestion for a feature list of what I called
RTPv3, which would need to co-exist with RTPv2 (that includes multiparty
support).

That said, if I'm not completely mistaken, you can implement the
multiparty use cases using the "RTCP terminating MCU" topology of RFC
5117.  This is common implementation practice in video conferencing.  If
you do so, the multiparty support of RTPv2 is neither exercised nor needed
in a multiparty video conference.

Stephan

On 7.22.2011 06:17 , "Magnus Westerlund" <magnus.westerlund@ericsson.com>
wrote:

>On 2011-07-21 11:01, Stephan Wenger wrote:
>
>> Remove multiparty support, then.  Terminology glitch on my side.
>
>So you argue for removing multi-party support the fact that RTCWEB have
>clear multi-party use cases in our use case document? Even you make
>arguments in your layered coding pitch using functions which is based on
>the multi-party functions of RTP. So I don't see how that match up.
>
>Cheers
>
>Magnus Westerlund
>
>----------------------------------------------------------------------
>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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From: Justin Uberti <juberti@google.com>
Date: Sun, 24 Jul 2011 07:40:25 -0700
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On Sat, Jul 23, 2011 at 5:50 AM, Tim Panton <tim@phonefromhere.com> wrote:

>
> On 23 Jul 2011, at 03:33, Cullen Jennings wrote:
>
>
> On Jul 20, 2011, at 8:37 , Justin Uberti wrote:
>
>
> Yes, I was going to send a similar email. There are some
> deficiencies/optimizations for the fact it runs over ICE-UDP - doesn't do
> path MTU discovery, no checksum, no OOB data - but it is based on TCP and
> implements slow start and other congestion avoidance mechanisms, and is
> actively used by a number of applications where reliable transfer is
> important, such as file transfer and remote desktop applications.
>
>
>
>
> It would be really interesting to see someone write a draft that subsets
> TCP to make it
>
>
> I just want to flag that TCP isn't an especially good fit here, datagrams
> delivered in events are a very useful
> abstraction in JavaScript, which doesn't really do streams of data very
> well.
>
> I know we can do datagrams over TCP,  but
> it seems wasteful to de-packetize in the TCP layer, just to try and
> re-packetize it a couple of layers further up.
>
> If there is a reliable datagram protocol as opposed to a reliable stream,
> I'd go for the datagram every time.
>

Yes, the API here will be datagram-oriented, just like WebSockets. Ideally,
the API for this would match WebSockets exactly, so that code in higher
layers can be agnostic about exactly how the data is being sent.

Even as a "datagram" service, I think TCP is still the best fit. The fact
that it is well-understood and has countless working implementations is far
more significant than any cost of framing (which will likely be trivial).

>
> Tim.
>
>

--0022150485071e97f604a8d1af32
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<br><br><div class=3D"gmail_quote">On Sat, Jul 23, 2011 at 5:50 AM, Tim Pan=
ton <span dir=3D"ltr">&lt;<a href=3D"mailto:tim@phonefromhere.com">tim@phon=
efromhere.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" st=
yle=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">

<div style=3D"word-wrap:break-word"><br><div><div class=3D"im"><div>On 23 J=
ul 2011, at 03:33, Cullen Jennings wrote:</div><br></div><blockquote type=
=3D"cite"><div><div class=3D"im"><br>On Jul 20, 2011, at 8:37 , Justin Uber=
ti wrote:<br>

<blockquote type=3D"cite"><font color=3D"#1855d0"><br></font></blockquote><=
/div><div class=3D"im"><blockquote type=3D"cite">Yes, I was going to send a=
 similar email. There are some deficiencies/optimizations for the fact it r=
uns over ICE-UDP - doesn&#39;t do path MTU discovery, no checksum, no OOB d=
ata - but it is based on TCP and implements slow start and other congestion=
 avoidance mechanisms, and is actively used by a number of applications whe=
re reliable transfer is important, such as file transfer and remote desktop=
 applications.<br>

</blockquote><blockquote type=3D"cite"><br></blockquote><blockquote type=3D=
"cite"><br></blockquote><br>It would be really interesting to see someone w=
rite a draft that subsets TCP to make it <br><br></div></div></blockquote>
<br>
</div><div>I just want to flag that TCP isn&#39;t an especially good fit he=
re, datagrams delivered in events are a very useful</div><div>abstraction i=
n JavaScript, which doesn&#39;t really do streams of data very well.=C2=A0<=
/div>

<div><br></div><div>I know we can do datagrams over TCP, =C2=A0but</div><di=
v>it seems wasteful to de-packetize in the TCP layer, just to try and re-pa=
cketize it a couple of layers further up.</div><div><br></div><div>If there=
 is a reliable datagram protocol as opposed to a reliable stream, I&#39;d g=
o for the datagram every time.</div>

</div></blockquote><div><br></div><div>Yes, the API here will be datagram-o=
riented, just like WebSockets. Ideally, the API for this would match WebSoc=
kets exactly, so that code in higher layers can be agnostic about exactly h=
ow the data is being sent.</div>

<div><br></div><div>Even as a &quot;datagram&quot; service, I think TCP is =
still the best fit. The fact that it is well-understood and has countless w=
orking implementations is far more significant than any cost of framing (wh=
ich will likely be trivial).</div>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex;"><div style=3D"word-wrap:break-word"><div><b=
r></div><font color=3D"#888888"><div>Tim.</div><br></font></div></blockquot=
e></div>

<br>

--0022150485071e97f604a8d1af32--

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To: Emil Ivov <emcho@jitsi.org>
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agree on the high level API you want something datagram oriented... I =
was just thinking about the congestion control / reliability approach of =
TCP and fine if a reliable , in order, datagram is put on top of that.=20=


inline ...

On Jul 23, 2011, at 11:26 , Emil Ivov wrote:

> =D0=9D=D0=B0 23.07.11 04:33, Cullen Jennings =D0=BD=D0=B0=D0=BF=D0=B8=D1=
=81=D0=B0:
>> It would be really interesting to see someone write a draft that =
subsets TCP to make it=20
>>=20
>> 1) still be congestion safe and TCP friendly=20
>>=20
>> 2) be valid TCP from point of view nats and firewalls
>=20
> Why does it need to be valid TCP? What NATs and firewalls will see is
> valid UDP which is the whole point, isn't it?

I should not have put nats you are right - I'm so used to typing nats =
and fws :-) On the firewalls that do protocol inspection, they may very =
well want to be able to parse this layer and even the things inside it. =
I suspect that code that is there is already valid TCP from the point of =
view of firewalls so I doubt this requirement really changes anything =
one way or the other.=20

>=20
> The fact that the protocol carried in the datagrams is TCP-like only
> matters to the application.

Sort of, lots of firewalls are configured not to allow UDP packets to =
pass unless they understand what is in them.

>=20
> Emil
>=20
> --
> http://jitsi.org
>=20


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



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From: Justin Uberti <juberti@google.com>
Date: Sun, 24 Jul 2011 08:03:12 -0700
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On Fri, Jul 22, 2011 at 7:33 PM, Cullen Jennings <fluffy@cisco.com> wrote:

>
> On Jul 20, 2011, at 8:37 , Justin Uberti wrote:
>
> >
> >
> > On Wed, Jul 20, 2011 at 9:15 AM, Harald Alvestrand <harald@alvestrand.no>
> wrote:
> > On 07/20/11 03:13, Cullen Jennings wrote:
> > On Jul 18, 2011, at 15:11 , Randell Jesup wrote:
> >
> > All that said - yes, there are complexity issues to consider, which was
> why I was suggesting leveraging
> > an existing tunneled protocol like SCTP or even TCP-over-UDP.
> > I agree with bunch of what you are saying but the previous several times
> I've seen the use SCTP over UDP conversation, it usually ends about the
> point the we ask for a user land implementation. Whatever we do more or less
> has to be user land or already exist in the OS that people run browsers on.
> >
> > If someone has a user land implementation of SCTP or TCP, I imagine that
> might change the outcome a bit from previous times. I have not looked at the
> TCP over UPD library Justin mentioned but, at casual glance, I noticed is
> around 400 lines of code which is very small, which is cool. But given the
> lines of code in say the BSD TCP stack, it does make me wonder what's
> missing. That said, I don't think we need something with the perforce of the
> BSD TCP stack as long as what is there is TCP friendly and robust.
> > I believe this is the current URL:
> >
> >
> http://code.google.com/p/libjingle/source/browse/trunk/talk/session/tunnel/pseudotcpchannel.cc(500 or so lines).
> >
> > The real protocol implementation is here:
> >
> http://code.google.com/p/libjingle/source/browse/trunk/talk/p2p/base/pseudotcp.cc
> > which is another 1000 lines (not counting the various .h files).
> >
> > We're not THAT good at writing compact code :-)
> >
> > Yes, I was going to send a similar email. There are some
> deficiencies/optimizations for the fact it runs over ICE-UDP - doesn't do
> path MTU discovery, no checksum, no OOB data - but it is based on TCP and
> implements slow start and other congestion avoidance mechanisms, and is
> actively used by a number of applications where reliable transfer is
> important, such as file transfer and remote desktop applications.
> >
> >
>
> It would be really interesting to see someone write a draft that subsets
> TCP to make it
>
> 1) still be congestion safe and TCP friendly
>
> 2) be valid TCP from point of view nats and firewalls
>
> 3) match an implementation like the one you have here.
>
> Most our use cases do not need OOB data. A subset protocol that was missing
> something like path MTU discovery might be slower than full blown modern TCP
> but again, most out use cases would still work just fine. As much as the
> IETF hates sub setting protocols, having something concrete and in hand like
> this would be a pretty useful thing. As well as being useful for the wwebrtc
> work, something like this would be useful for very constrained systems such
> very cheap sensor networks. I'm seeing lots of really minimal subset of TCP
> being done for these very low end system.
>

Interesting. From the webrtc context, the requirements are a little
different - e.g. when we're running as TCP/UDP, we don't need the port or
checksum, as those are handled by the UDP layer.

So the goal would become two specs:
(1) minimal subset of TCP
(2) adaptation of #1 to run over UDP instead of IP

Who can speak to what requirements/RFCs should constitute #1? We could
certainly make a proposal based on what we've implemented, but if it's going
to be used more broadly, I'm guessing a LOT of folks will care about these
details.

For #2, in addition to the work we've done with what we call PseudoTcp, I've
noticed these two other drafts on TCP/UDP, which are very similar to
PseudoTcp. The first gives a very good treatment (in Section 8) of why a
PseudoTcp-like approach is the best way for reliable p2p data transfer.

http://tools.ietf.org/html/draft-baset-tsvwg-tcp-over-udp-01
http://tools.ietf.org/html/<http://tools.ietf.org/html/draft-denis-udp-transport-00>
draft-denis-udp-transport-00<http://tools.ietf.org/html/draft-denis-udp-transport-00>

--000e0cd4b4228901e704a8d200d9
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<br><br><div class=3D"gmail_quote">On Fri, Jul 22, 2011 at 7:33 PM, Cullen =
Jennings <span dir=3D"ltr">&lt;<a href=3D"mailto:fluffy@cisco.com">fluffy@c=
isco.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=
=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">

<div class=3D"im"><br>
On Jul 20, 2011, at 8:37 , Justin Uberti wrote:<br>
<br>
&gt;<br>
&gt;<br>
&gt; On Wed, Jul 20, 2011 at 9:15 AM, Harald Alvestrand &lt;<a href=3D"mail=
to:harald@alvestrand.no">harald@alvestrand.no</a>&gt; wrote:<br>
&gt; On 07/20/11 03:13, Cullen Jennings wrote:<br>
&gt; On Jul 18, 2011, at 15:11 , Randell Jesup wrote:<br>
&gt;<br>
&gt; All that said - yes, there are complexity issues to consider, which wa=
s why I was suggesting leveraging<br>
&gt; an existing tunneled protocol like SCTP or even TCP-over-UDP.<br>
&gt; I agree with bunch of what you are saying but the previous several tim=
es I&#39;ve seen the use SCTP over UDP conversation, it usually ends about =
the point the we ask for a user land implementation. Whatever we do more or=
 less has to be user land or already exist in the OS that people run browse=
rs on.<br>


&gt;<br>
&gt; If someone has a user land implementation of SCTP or TCP, I imagine th=
at might change the outcome a bit from previous times. I have not looked at=
 the TCP over UPD library Justin mentioned but, at casual glance, I noticed=
 is around 400 lines of code which is very small, which is cool. But given =
the lines of code in say the BSD TCP stack, it does make me wonder what&#39=
;s missing. That said, I don&#39;t think we need something with the perforc=
e of the BSD TCP stack as long as what is there is TCP friendly and robust.=
<br>


&gt; I believe this is the current URL:<br>
&gt;<br>
&gt; <a href=3D"http://code.google.com/p/libjingle/source/browse/trunk/talk=
/session/tunnel/pseudotcpchannel.cc" target=3D"_blank">http://code.google.c=
om/p/libjingle/source/browse/trunk/talk/session/tunnel/pseudotcpchannel.cc<=
/a> (500 or so lines).<br>


&gt;<br>
&gt; The real protocol implementation is here:<br>
&gt; <a href=3D"http://code.google.com/p/libjingle/source/browse/trunk/talk=
/p2p/base/pseudotcp.cc" target=3D"_blank">http://code.google.com/p/libjingl=
e/source/browse/trunk/talk/p2p/base/pseudotcp.cc</a><br>
&gt; which is another 1000 lines (not counting the various .h files).<br>
&gt;<br>
&gt; We&#39;re not THAT good at writing compact code :-)<br>
&gt;<br>
&gt; Yes, I was going to send a similar email. There are some deficiencies/=
optimizations for the fact it runs over ICE-UDP - doesn&#39;t do path MTU d=
iscovery, no checksum, no OOB data - but it is based on TCP and implements =
slow start and other congestion avoidance mechanisms, and is actively used =
by a number of applications where reliable transfer is important, such as f=
ile transfer and remote desktop applications.<br>


&gt;<br>
&gt;<br>
<br>
</div>It would be really interesting to see someone write a draft that subs=
ets TCP to make it<br>
<br>
1) still be congestion safe and TCP friendly<br>
<br>
2) be valid TCP from point of view nats and firewalls<br>
<br>
3) match an implementation like the one you have here.<br>
<br>
Most our use cases do not need OOB data. A subset protocol that was missing=
 something like path MTU discovery might be slower than full blown modern T=
CP but again, most out use cases would still work just fine. As much as the=
 IETF hates sub setting protocols, having something concrete and in hand li=
ke this would be a pretty useful thing. As well as being useful for the wwe=
brtc work, something like this would be useful for very constrained systems=
 such very cheap sensor networks. I&#39;m seeing lots of really minimal sub=
set of TCP being done for these very low end system.<br>

</blockquote><div><br></div><div>Interesting. From the webrtc context, the =
requirements are a little different - e.g. when we&#39;re running as TCP/UD=
P, we don&#39;t need the port or checksum, as those are handled by the UDP =
layer.=C2=A0</div>

<div><br></div><div>So the goal would become two specs:</div><div>(1) minim=
al subset of TCP</div><div>(2) adaptation of #1 to run over UDP instead of =
IP</div><div><br></div><div>Who can speak to what requirements/RFCs should =
constitute #1? We could certainly make a proposal based on what we&#39;ve i=
mplemented, but if it&#39;s going to be used more broadly, I&#39;m guessing=
 a LOT of folks will care about these details.</div>

<div><br></div><div>For #2, in addition to the work we&#39;ve done with wha=
t we call PseudoTcp, I&#39;ve noticed these two other drafts on TCP/UDP, wh=
ich are very similar to PseudoTcp. The first gives a very good treatment (i=
n Section 8) of why a PseudoTcp-like approach is the best way for reliable =
p2p data transfer.</div>

<div><span class=3D"Apple-style-span" style=3D"border-collapse: collapse; c=
olor: rgb(80, 0, 80); font-family: arial, sans-serif; font-size: 13px; back=
ground-color: rgb(255, 255, 255);"><br><a href=3D"http://tools.ietf.org/htm=
l/draft-baset-tsvwg-tcp-over-udp-01" target=3D"_blank" style=3D"color: rgb(=
51, 51, 204); ">http://tools.ietf.org/html/draft-baset-tsvwg-<span class=3D=
"il" style=3D"background-image: initial; background-attachment: initial; ba=
ckground-origin: initial; background-clip: initial; color: rgb(34, 34, 34);=
 ">tcp</span>-over-udp-01</a><br>

<a href=3D"http://tools.ietf.org/html/draft-denis-udp-transport-00" target=
=3D"_blank" style=3D"color: rgb(51, 51, 204); ">http://tools.ietf.org/html/=
</a></span><span class=3D"Apple-style-span" style=3D"border-collapse: colla=
pse; color: rgb(80, 0, 80); font-family: arial, sans-serif; font-size: 13px=
; background-color: rgb(255, 255, 255);"><a href=3D"http://tools.ietf.org/h=
tml/draft-denis-udp-transport-00" target=3D"_blank" style=3D"color: rgb(51,=
 51, 204); ">draft-denis-udp-transport-00</a><br>

<br></span></div></div>

--000e0cd4b4228901e704a8d200d9--

From harald@alvestrand.no  Sun Jul 24 08:21:01 2011
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Subject: Re: [rtcweb] PseudoTCP implementation (Re: realiable data service)
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On 07/22/11 22:33, Cullen Jennings wrote:
> On Jul 20, 2011, at 8:37 , Justin Uberti wrote:
>
>>
>> On Wed, Jul 20, 2011 at 9:15 AM, Harald Alvestrand<harald@alvestrand.no>  wrote:
>> On 07/20/11 03:13, Cullen Jennings wrote:
>> On Jul 18, 2011, at 15:11 , Randell Jesup wrote:
>>
>> All that said - yes, there are complexity issues to consider, which was why I was suggesting leveraging
>> an existing tunneled protocol like SCTP or even TCP-over-UDP.
>> I agree with bunch of what you are saying but the previous several times I've seen the use SCTP over UDP conversation, it usually ends about the point the we ask for a user land implementation. Whatever we do more or less has to be user land or already exist in the OS that people run browsers on.
>>
>> If someone has a user land implementation of SCTP or TCP, I imagine that might change the outcome a bit from previous times. I have not looked at the TCP over UPD library Justin mentioned but, at casual glance, I noticed is around 400 lines of code which is very small, which is cool. But given the lines of code in say the BSD TCP stack, it does make me wonder what's missing. That said, I don't think we need something with the perforce of the BSD TCP stack as long as what is there is TCP friendly and robust.
>> I believe this is the current URL:
>>
>> http://code.google.com/p/libjingle/source/browse/trunk/talk/session/tunnel/pseudotcpchannel.cc (500 or so lines).
>>
>> The real protocol implementation is here:
>> http://code.google.com/p/libjingle/source/browse/trunk/talk/p2p/base/pseudotcp.cc
>> which is another 1000 lines (not counting the various .h files).
>>
>> We're not THAT good at writing compact code :-)
>>
>> Yes, I was going to send a similar email. There are some deficiencies/optimizations for the fact it runs over ICE-UDP - doesn't do path MTU discovery, no checksum, no OOB data - but it is based on TCP and implements slow start and other congestion avoidance mechanisms, and is actively used by a number of applications where reliable transfer is important, such as file transfer and remote desktop applications.
>>
>>
> It would be really interesting to see someone write a draft that subsets TCP to make it
>
> 1) still be congestion safe and TCP friendly
>
> 2) be valid TCP from point of view nats and firewalls
>
> 3) match an implementation like the one you have here.
>
> Most our use cases do not need OOB data. A subset protocol that was missing something like path MTU discovery might be slower than full blown modern TCP but again, most out use cases would still work just fine. As much as the IETF hates sub setting protocols, having something concrete and in hand like this would be a pretty useful thing. As well as being useful for the wwebrtc work, something like this would be useful for very constrained systems such very cheap sensor networks. I'm seeing lots of really minimal subset of TCP being done for these very low end system.
ObNitpick: TCP doesn't have OOB data.

What it has is an "urgent data downstream" pointer, which the Unix 
implementors used to pick the byte pointed to out of the data stream and 
deliver it "urgently" as if it was OOB data.
I exchanged mail with Jon Postel about this in 1984, and he confirmed 
that if a TCP protocol stack gets 2 TCP segments with different URG 
pointers, a TCP implementation can merge them and only deliver the 
latest URG pointer - which means that delivery of the OOB data is 
unreliable if you have more than 1 in flight at any one time; you might 
get the byte inband, and you might get it out of band.

This conflict of viewpoints between the protocol design and the API 
design has, I believe, rendered the TCP URG pointer useless for all 
practical purposes.

Returning to more relevant stuff: I think we need to handle PMTU issues 
somehow. This may be hard.

                   Harald



From juberti@google.com  Sun Jul 24 08:27:33 2011
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From: Justin Uberti <juberti@google.com>
Date: Sun, 24 Jul 2011 08:27:09 -0700
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Subject: Re: [rtcweb] PseudoTCP implementation (Re: realiable data service)
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--0016369c8dda2b629a04a8d25699
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On Sun, Jul 24, 2011 at 8:20 AM, Harald Alvestrand <harald@alvestrand.no>wrote:

> On 07/22/11 22:33, Cullen Jennings wrote:
>
>> On Jul 20, 2011, at 8:37 , Justin Uberti wrote:
>>
>>
>>> On Wed, Jul 20, 2011 at 9:15 AM, Harald Alvestrand<harald@alvestrand.**
>>> no <harald@alvestrand.no>>  wrote:
>>> On 07/20/11 03:13, Cullen Jennings wrote:
>>> On Jul 18, 2011, at 15:11 , Randell Jesup wrote:
>>>
>>> All that said - yes, there are complexity issues to consider, which was
>>> why I was suggesting leveraging
>>> an existing tunneled protocol like SCTP or even TCP-over-UDP.
>>> I agree with bunch of what you are saying but the previous several times
>>> I've seen the use SCTP over UDP conversation, it usually ends about the
>>> point the we ask for a user land implementation. Whatever we do more or less
>>> has to be user land or already exist in the OS that people run browsers on.
>>>
>>> If someone has a user land implementation of SCTP or TCP, I imagine that
>>> might change the outcome a bit from previous times. I have not looked at the
>>> TCP over UPD library Justin mentioned but, at casual glance, I noticed is
>>> around 400 lines of code which is very small, which is cool. But given the
>>> lines of code in say the BSD TCP stack, it does make me wonder what's
>>> missing. That said, I don't think we need something with the perforce of the
>>> BSD TCP stack as long as what is there is TCP friendly and robust.
>>> I believe this is the current URL:
>>>
>>> http://code.google.com/p/**libjingle/source/browse/trunk/**
>>> talk/session/tunnel/**pseudotcpchannel.cc<http://code.google.com/p/libjingle/source/browse/trunk/talk/session/tunnel/pseudotcpchannel.cc>(500 or so lines).
>>>
>>> The real protocol implementation is here:
>>> http://code.google.com/p/**libjingle/source/browse/trunk/**
>>> talk/p2p/base/pseudotcp.cc<http://code.google.com/p/libjingle/source/browse/trunk/talk/p2p/base/pseudotcp.cc>
>>> which is another 1000 lines (not counting the various .h files).
>>>
>>> We're not THAT good at writing compact code :-)
>>>
>>> Yes, I was going to send a similar email. There are some
>>> deficiencies/optimizations for the fact it runs over ICE-UDP - doesn't do
>>> path MTU discovery, no checksum, no OOB data - but it is based on TCP and
>>> implements slow start and other congestion avoidance mechanisms, and is
>>> actively used by a number of applications where reliable transfer is
>>> important, such as file transfer and remote desktop applications.
>>>
>>>
>>>  It would be really interesting to see someone write a draft that subsets
>> TCP to make it
>>
>> 1) still be congestion safe and TCP friendly
>>
>> 2) be valid TCP from point of view nats and firewalls
>>
>> 3) match an implementation like the one you have here.
>>
>> Most our use cases do not need OOB data. A subset protocol that was
>> missing something like path MTU discovery might be slower than full blown
>> modern TCP but again, most out use cases would still work just fine. As much
>> as the IETF hates sub setting protocols, having something concrete and in
>> hand like this would be a pretty useful thing. As well as being useful for
>> the wwebrtc work, something like this would be useful for very constrained
>> systems such very cheap sensor networks. I'm seeing lots of really minimal
>> subset of TCP being done for these very low end system.
>>
> ObNitpick: TCP doesn't have OOB data.
>
> What it has is an "urgent data downstream" pointer, which the Unix
> implementors used to pick the byte pointed to out of the data stream and
> deliver it "urgently" as if it was OOB data.
> I exchanged mail with Jon Postel about this in 1984, and he confirmed that
> if a TCP protocol stack gets 2 TCP segments with different URG pointers, a
> TCP implementation can merge them and only deliver the latest URG pointer -
> which means that delivery of the OOB data is unreliable if you have more
> than 1 in flight at any one time; you might get the byte inband, and you
> might get it out of band.
>
> This conflict of viewpoints between the protocol design and the API design
> has, I believe, rendered the TCP URG pointer useless for all practical
> purposes.
>
> Returning to more relevant stuff: I think we need to handle PMTU issues
> somehow. This may be hard.
>

The easy way may be to do what we're likely to do for RTP data: assume a MTU
of 1280 (IPv6 minimum MTU) and packetize as such, leaving PMTU discovery as
a future optimization.


>
>                  Harald
>
>
>

--0016369c8dda2b629a04a8d25699
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<br><br><div class=3D"gmail_quote">On Sun, Jul 24, 2011 at 8:20 AM, Harald =
Alvestrand <span dir=3D"ltr">&lt;<a href=3D"mailto:harald@alvestrand.no">ha=
rald@alvestrand.no</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quot=
e" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"=
>

<div><div></div><div class=3D"h5">On 07/22/11 22:33, Cullen Jennings wrote:=
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
On Jul 20, 2011, at 8:37 , Justin Uberti wrote:<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<br>
On Wed, Jul 20, 2011 at 9:15 AM, Harald Alvestrand&lt;<a href=3D"mailto:har=
ald@alvestrand.no" target=3D"_blank">harald@alvestrand.<u></u>no</a>&gt; =
=C2=A0wrote:<br>
On 07/20/11 03:13, Cullen Jennings wrote:<br>
On Jul 18, 2011, at 15:11 , Randell Jesup wrote:<br>
<br>
All that said - yes, there are complexity issues to consider, which was why=
 I was suggesting leveraging<br>
an existing tunneled protocol like SCTP or even TCP-over-UDP.<br>
I agree with bunch of what you are saying but the previous several times I&=
#39;ve seen the use SCTP over UDP conversation, it usually ends about the p=
oint the we ask for a user land implementation. Whatever we do more or less=
 has to be user land or already exist in the OS that people run browsers on=
.<br>


<br>
If someone has a user land implementation of SCTP or TCP, I imagine that mi=
ght change the outcome a bit from previous times. I have not looked at the =
TCP over UPD library Justin mentioned but, at casual glance, I noticed is a=
round 400 lines of code which is very small, which is cool. But given the l=
ines of code in say the BSD TCP stack, it does make me wonder what&#39;s mi=
ssing. That said, I don&#39;t think we need something with the perforce of =
the BSD TCP stack as long as what is there is TCP friendly and robust.<br>


I believe this is the current URL:<br>
<br>
<a href=3D"http://code.google.com/p/libjingle/source/browse/trunk/talk/sess=
ion/tunnel/pseudotcpchannel.cc" target=3D"_blank">http://code.google.com/p/=
<u></u>libjingle/source/browse/trunk/<u></u>talk/session/tunnel/<u></u>pseu=
dotcpchannel.cc</a> (500 or so lines).<br>


<br>
The real protocol implementation is here:<br>
<a href=3D"http://code.google.com/p/libjingle/source/browse/trunk/talk/p2p/=
base/pseudotcp.cc" target=3D"_blank">http://code.google.com/p/<u></u>libjin=
gle/source/browse/trunk/<u></u>talk/p2p/base/pseudotcp.cc</a><br>
which is another 1000 lines (not counting the various .h files).<br>
<br>
We&#39;re not THAT good at writing compact code :-)<br>
<br>
Yes, I was going to send a similar email. There are some deficiencies/optim=
izations for the fact it runs over ICE-UDP - doesn&#39;t do path MTU discov=
ery, no checksum, no OOB data - but it is based on TCP and implements slow =
start and other congestion avoidance mechanisms, and is actively used by a =
number of applications where reliable transfer is important, such as file t=
ransfer and remote desktop applications.<br>


<br>
<br>
</blockquote>
It would be really interesting to see someone write a draft that subsets TC=
P to make it<br>
<br>
1) still be congestion safe and TCP friendly<br>
<br>
2) be valid TCP from point of view nats and firewalls<br>
<br>
3) match an implementation like the one you have here.<br>
<br>
Most our use cases do not need OOB data. A subset protocol that was missing=
 something like path MTU discovery might be slower than full blown modern T=
CP but again, most out use cases would still work just fine. As much as the=
 IETF hates sub setting protocols, having something concrete and in hand li=
ke this would be a pretty useful thing. As well as being useful for the wwe=
brtc work, something like this would be useful for very constrained systems=
 such very cheap sensor networks. I&#39;m seeing lots of really minimal sub=
set of TCP being done for these very low end system.<br>


</blockquote></div></div>
ObNitpick: TCP doesn&#39;t have OOB data.<br>
<br>
What it has is an &quot;urgent data downstream&quot; pointer, which the Uni=
x implementors used to pick the byte pointed to out of the data stream and =
deliver it &quot;urgently&quot; as if it was OOB data.<br>
I exchanged mail with Jon Postel about this in 1984, and he confirmed that =
if a TCP protocol stack gets 2 TCP segments with different URG pointers, a =
TCP implementation can merge them and only deliver the latest URG pointer -=
 which means that delivery of the OOB data is unreliable if you have more t=
han 1 in flight at any one time; you might get the byte inband, and you mig=
ht get it out of band.<br>


<br>
This conflict of viewpoints between the protocol design and the API design =
has, I believe, rendered the TCP URG pointer useless for all practical purp=
oses.<br>
<br>
Returning to more relevant stuff: I think we need to handle PMTU issues som=
ehow. This may be hard.<br></blockquote><div><br></div><div>The easy way ma=
y be to do what we&#39;re likely to do for RTP data: assume a MTU of 1280 (=
IPv6 minimum MTU) and packetize as such, leaving PMTU discovery as a future=
 optimization.</div>

<div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8=
ex;border-left:1px #ccc solid;padding-left:1ex;"><font color=3D"#888888">
<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0Harald<br>
<br>
<br>
</font></blockquote></div><br>

--0016369c8dda2b629a04a8d25699--

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I've been looking for technical reasons and/or hard requirements that =
would argue against doing RTP multiplexing as proposed in this draft.

So far my conclusion is that the RTP specifications recommend against =
this, but do not prohibit it.

Supporting evidence:
RFC 3550 Section 2.2 notes that multiplexing A&V together may be a =
problem if you want to receive one and not together (presumably in the =
multicast case) and refers to Section 5.2 for more support.

RFC 3550 Section 5.2 says "Separate audio and video streams SHOULD NOT =
be carried in a single RTP session and demultiplexed based on the =
payload type or SSRC fields." and provides five reasons. Note the =
"SHOULD NOT" as opposed to "MUST NOT".

The same section says that the five reasons apply to multiplexing on =
payload type, only the last two apply to multiplexing on SSRC. These =
remaining reasons are:

   4. An RTP mixer would not be able to combine interleaved streams of
      incompatible media into one stream.

Not relevant to the RTCWEB peer-to-peer case as the browser is not an =
RTP mixer. Also not relevant to any RTCWEB use cases that I can =
identify. And, if an RTP mixer at one end it can simply not agree to use =
multiplexing.

   5. Carrying multiple media in one RTP session precludes: the use of
      different network paths or network resource allocations if
      appropriate;

Appears to not be relevant to the RTCWEB case where a unicast path will =
be negotiated using ICE.

      reception of a subset of the media if desired, for
      example just audio if video would exceed the available bandwidth;

Appears to not be relevant to anything but the multicast cases. Any =
unicast case, including RTCWEB, can simply not have that component of =
the stream sent to it.

      and receiver implementations that use separate processes for the
      different media, whereas using separate RTP sessions permits
      either single- or multiple-process implementations.

Appears to not be relevant to the RTCWEB use case. Additionally, if it =
is necessary for there to be separate processes or even devices at the =
far end this can be solved by again simply not agreeing to use =
multiplexing.

In all my reading today I have not been able to find anything more =
concrete than the "SHOULD NOT" in section 5.2 of RFC3550. PLEASE follow =
up if you are aware of any other relevant specifications that would =
argue against using SSRC to multiplex audio and video streams over a =
single RTP session between a pair of compatible endpoints that agree to =
do so.

Matthew Kaufman



=20=

From randell1@jesup.org  Sun Jul 24 17:16:10 2011
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On 7/24/2011 4:44 PM, Matthew Kaufman wrote:
> RFC 3550 Section 5.2 says "Separate audio and video streams SHOULD NOT be carried in a single RTP session and demultiplexed based on the payload type or SSRC fields." and provides five reasons. Note the "SHOULD NOT" as opposed to "MUST NOT".
>
> The same section says that the five reasons apply to multiplexing on payload type, only the last two apply to multiplexing on SSRC. These remaining reasons are:
>
>     4. An RTP mixer would not be able to combine interleaved streams of
>        incompatible media into one stream.
>
> Not relevant to the RTCWEB peer-to-peer case as the browser is not an RTP mixer. Also not relevant to any RTCWEB use cases that I can identify. And, if an RTP mixer at one end it can simply not agree to use multiplexing.

Well, there's no reason a browser using rtcweb couldn't be acting as a 
mixer for sessions with multiple other rtcweb clients/browsers.  
However, a) it could disable multiplexing (if we provide a control for 
that), and b) I'm not sure that I agree with the reasoning in 4) above.  
Perhaps someone could explain why that would be a problem (not having 
played with mixers much, I can't think of a reason off the top of my head).

>     5. Carrying multiple media in one RTP session precludes: the use of
>        different network paths or network resource allocations if
>        appropriate;
>
> Appears to not be relevant to the RTCWEB case where a unicast path will be negotiated using ICE.

Well, 5) is relevant, though you can decide you don't care.  For example 
the concern raised about network-applied QOS to different streams 
from/to a client.  The different network paths isn't important here.

>        reception of a subset of the media if desired, for
>        example just audio if video would exceed the available bandwidth;
>
> Appears to not be relevant to anything but the multicast cases. Any unicast case, including RTCWEB, can simply not have that component of the stream sent to it.

Right, and no bandwidth will be saved unless you stop them from sending 
the stream.  Given offer/answer/etc this issue is moot.

>        and receiver implementations that use separate processes for the
>        different media, whereas using separate RTP sessions permits
>        either single- or multiple-process implementations.
>
> Appears to not be relevant to the RTCWEB use case. Additionally, if it is necessary for there to be separate processes or even devices at the far end this can be solved by again simply not agreeing to use multiplexing.

Agreed, not relevant to our case.

> In all my reading today I have not been able to find anything more concrete than the "SHOULD NOT" in section 5.2 of RFC3550. PLEASE follow up if you are aware of any other relevant specifications that would argue against using SSRC to multiplex audio and video streams over a single RTP session between a pair of compatible endpoints that agree to do so.

People certainly have done it, and people have also (though perhaps 
inadvisedly) had clients that would have several sessions with different 
clients terminating at the same port -- though people having done it 
doesn't mean it's good.

-- 
Randell Jesup
randell-ietf@jesup.org


From magnus.westerlund@ericsson.com  Mon Jul 25 04:53:37 2011
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On 2011-07-23 17:12, Stephan Wenger wrote:
> Hi Magnus,
> 
> Not sure you got the context right here.  The "removal of multiparty
> support" was related to my suggestion for a feature list of what I called
> RTPv3, which would need to co-exist with RTPv2 (that includes multiparty
> support).

Ok, I misunderstood that.

> 
> That said, if I'm not completely mistaken, you can implement the
> multiparty use cases using the "RTCP terminating MCU" topology of RFC
> 5117.  This is common implementation practice in video conferencing.  If
> you do so, the multiparty support of RTPv2 is neither exercised nor needed
> in a multiparty video conference.

First of all if you do that you loose functionality as discussed in RFC
5117. For example the indication of who actually sent a packet
originally or contribute to the mix can't be provided unless you let the
SSRC or CSRC plus CNAMES be forwarded across the Mixer/MCU.

In addition it forces the central unit into one particular
implementation, instead of allowing to use what it most appropriate.

An RTP mixer when you want something that has functionality and
application enhancing behavior and the cost is more complexity.

Or you use an relay which simply copies a packet from one entity to all
the others. That is the simplest central node you can build and it will
have no media processing at all.

Also, in any gateway case to existing legacy you don't want to have to
implement additional gatewaying functions just because one want to use
non standard RTP.

For me it is clear that ensuring that RTCWEBs RTP usage is vanilla will
help us avoid a lot of problems down the road.

If we are going to multiplex this mechanism needs to have a behavior
that doesn't change the core behaviors of RTP. If we can do that the
multiplexing becomes much less a hot topic.


Cheers

Magnus Westerlund

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


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Subject: [rtcweb] Reasons not to multiplex audio with video (Re: Fwd: I-D Action: draft-rosenberg-rtcweb-rtpmux-00.txt)
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On 07/24/11 16:44, Matthew Kaufman wrote:
> In all my reading today I have not been able to find anything more concrete than the "SHOULD NOT" in section 5.2 of RFC3550. PLEASE follow up if you are aware of any other relevant specifications that would argue against using SSRC to multiplex audio and video streams over a single RTP session between a pair of compatible endpoints that agree to do so.
I have found *one* reason not mentioned in the draft:

An RTP session with both "audio" and "video" media types cannot be 
represented by an SDP description, since SDP ties RTP sessions 1-1 to 
the "m" line of the description, which contains the top-level type, and 
the codec descriptions omit the top-level type in their codec naming.

I've said elsewhere that I consider this to be a design mistake for 
entirely different reasons, but this debate has reinforced my opinion 
that it is.

                      Harald



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Subject: Re: [rtcweb] Reasons not to multiplex audio with video (Re: Fwd: I-D Action: draft-rosenberg-rtcweb-rtpmux-00.txt)
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=D0=9D=D0=B0 25.07.11 14:06, Harald Alvestrand =D0=BD=D0=B0=D0=BF=D0=B8=D1=
=81=D0=B0:
> On 07/24/11 16:44, Matthew Kaufman wrote:
>> In all my reading today I have not been able to find anything more con=
crete than the "SHOULD NOT" in section 5.2 of RFC3550. PLEASE follow up i=
f you are aware of any other relevant specifications that would argue aga=
inst using SSRC to multiplex audio and video streams over a single RTP se=
ssion between a pair of compatible endpoints that agree to do so.
> I have found *one* reason not mentioned in the draft:
>=20
> An RTP session with both "audio" and "video" media types cannot be=20
> represented by an SDP description, since SDP ties RTP sessions 1-1 to=20
> the "m" line of the description, which contains the top-level type, and=
=20
> the codec descriptions omit the top-level type in their codec naming.

I am not sure I understand this. Why would one need to use the same "m"
line? What would be the consequences of using two different "m" lines?

Personally, I am still not convinced that multiplexing brings
significant improvement over the status quo but if we accept it then it
definitely needs to keep separate "m" lines for tons of reasons.

Emil


--=20
http://jitsi.org


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On 07/25/11 08:17, Emil Ivov wrote:
> На 25.07.11 14:06, Harald Alvestrand написа:
>> On 07/24/11 16:44, Matthew Kaufman wrote:
>>> In all my reading today I have not been able to find anything more concrete than the "SHOULD NOT" in section 5.2 of RFC3550. PLEASE follow up if you are aware of any other relevant specifications that would argue against using SSRC to multiplex audio and video streams over a single RTP session between a pair of compatible endpoints that agree to do so.
>> I have found *one* reason not mentioned in the draft:
>>
>> An RTP session with both "audio" and "video" media types cannot be
>> represented by an SDP description, since SDP ties RTP sessions 1-1 to
>> the "m" line of the description, which contains the top-level type, and
>> the codec descriptions omit the top-level type in their codec naming.
> I am not sure I understand this. Why would one need to use the same "m"
> line? What would be the consequences of using two different "m" lines?
>
> Personally, I am still not convinced that multiplexing brings
> significant improvement over the status quo but if we accept it then it
> definitely needs to keep separate "m" lines for tons of reasons.
RFC 4566 section 5:

    An SDP session description consists of a session-level section
    followed by zero or more media-level sections.  The session-level
    part starts with a "v=" line and continues to the first media-level
    section.  Each media-level section starts with an "m=" line and
    continues to the next media-level section or end of the whole session
    description.

....

5.14.  Media Descriptions ("m=")

       m=<media> <port> <proto> <fmt> ...

    A session description may contain a number of media descriptions.
    Each media description starts with an "m=" field and is terminated by
    either the next "m=" field or by the end of the session description.
    A media field has several sub-fields:

<media> is the media type.  Currently defined media are "audio",
       "video", "text", "application", and "message", although this list
       may be extended in the future (see Section 8).

<port> is the transport port to which the media stream is sent.

In other words: an "m" line is tied to a port number, and (from RTP) the 
port number identifies the RTP session. The ICE parameters can override 
the port number, but they're carried within the "m" block, so they're 
tied to a single "m" line too.

I do not know what would happen if you used the same port number and ICE 
parameters with two "m" lines.


From randell1@jesup.org  Mon Jul 25 05:32:23 2011
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Subject: Re: [rtcweb] Reasons not to multiplex audio with video (Re: Fwd: I-D Action: draft-rosenberg-rtcweb-rtpmux-00.txt)
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On 7/25/2011 8:17 AM, Emil Ivov wrote:
> На 25.07.11 14:06, Harald Alvestrand написа:
>> On 07/24/11 16:44, Matthew Kaufman wrote:
>>> In all my reading today I have not been able to find anything more concrete than the "SHOULD NOT" in section 5.2 of RFC3550. PLEASE follow up if you are aware of any other relevant specifications that would argue against using SSRC to multiplex audio and video streams over a single RTP session between a pair of compatible endpoints that agree to do so.
>> I have found *one* reason not mentioned in the draft:
>>
>> An RTP session with both "audio" and "video" media types cannot be
>> represented by an SDP description, since SDP ties RTP sessions 1-1 to
>> the "m" line of the description, which contains the top-level type, and
>> the codec descriptions omit the top-level type in their codec naming.
> I am not sure I understand this. Why would one need to use the same "m"
> line? What would be the consequences of using two different "m" lines?
>
> Personally, I am still not convinced that multiplexing brings
> significant improvement over the status quo but if we accept it then it
> definitely needs to keep separate "m" lines for tons of reasons.

I've been assuming the method of indicating multiplexing would be two m= 
lines with
the same c= destination and the same port specified.  This is a 
mechanism that has
been discussed before on the AVT list and/or in drafts (and I'm sure has 
also been
done before).  It does require avoiding using the same payload values in 
the two m= lines,
but beyond that it's fairly straightforward.  It also allows for the 
answerer to clearly reject a
media type explicitly, instead of an implicit rejection by removing the 
payloads of a type.

-- 
Randell Jesup
randell-ietf@jesup.org


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Section 2.2 of draft-perkins-rtcweb-rtp-usage-02 lists several reasons =
why this sort of multiplexing is problematic. I would suggest, again, =
that you read that draft.=20

Colin



On 24 Jul 2011, at 16:44, Matthew Kaufman wrote:
> I've been looking for technical reasons and/or hard requirements that =
would argue against doing RTP multiplexing as proposed in this draft.
>=20
> So far my conclusion is that the RTP specifications recommend against =
this, but do not prohibit it.
>=20
> Supporting evidence:
> RFC 3550 Section 2.2 notes that multiplexing A&V together may be a =
problem if you want to receive one and not together (presumably in the =
multicast case) and refers to Section 5.2 for more support.
>=20
> RFC 3550 Section 5.2 says "Separate audio and video streams SHOULD NOT =
be carried in a single RTP session and demultiplexed based on the =
payload type or SSRC fields." and provides five reasons. Note the =
"SHOULD NOT" as opposed to "MUST NOT".
>=20
> The same section says that the five reasons apply to multiplexing on =
payload type, only the last two apply to multiplexing on SSRC. These =
remaining reasons are:
>=20
>   4. An RTP mixer would not be able to combine interleaved streams of
>      incompatible media into one stream.
>=20
> Not relevant to the RTCWEB peer-to-peer case as the browser is not an =
RTP mixer. Also not relevant to any RTCWEB use cases that I can =
identify. And, if an RTP mixer at one end it can simply not agree to use =
multiplexing.
>=20
>   5. Carrying multiple media in one RTP session precludes: the use of
>      different network paths or network resource allocations if
>      appropriate;
>=20
> Appears to not be relevant to the RTCWEB case where a unicast path =
will be negotiated using ICE.
>=20
>      reception of a subset of the media if desired, for
>      example just audio if video would exceed the available bandwidth;
>=20
> Appears to not be relevant to anything but the multicast cases. Any =
unicast case, including RTCWEB, can simply not have that component of =
the stream sent to it.
>=20
>      and receiver implementations that use separate processes for the
>      different media, whereas using separate RTP sessions permits
>      either single- or multiple-process implementations.
>=20
> Appears to not be relevant to the RTCWEB use case. Additionally, if it =
is necessary for there to be separate processes or even devices at the =
far end this can be solved by again simply not agreeing to use =
multiplexing.
>=20
> In all my reading today I have not been able to find anything more =
concrete than the "SHOULD NOT" in section 5.2 of RFC3550. PLEASE follow =
up if you are aware of any other relevant specifications that would =
argue against using SSRC to multiplex audio and video streams over a =
single RTP session between a pair of compatible endpoints that agree to =
do so.
>=20
> Matthew Kaufman
>=20
>=20
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb



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




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Subject: Re: [rtcweb] Reasons not to multiplex audio with video (Re: Fwd: I-D Action: draft-rosenberg-rtcweb-rtpmux-00.txt)
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=D0=9D=D0=B0 25.07.11 14:30, Harald Alvestrand =D0=BD=D0=B0=D0=BF=D0=B8=D1=
=81=D0=B0:
> On 07/25/11 08:17, Emil Ivov wrote:
>> =D0=9D=D0=B0 25.07.11 14:06, Harald Alvestrand =D0=BD=D0=B0=D0=BF=D0=B8=
=D1=81=D0=B0:
>>> On 07/24/11 16:44, Matthew Kaufman wrote:
>>>> In all my reading today I have not been able to find anything more c=
oncrete than the "SHOULD NOT" in section 5.2 of RFC3550. PLEASE follow up=
 if you are aware of any other relevant specifications that would argue a=
gainst using SSRC to multiplex audio and video streams over a single RTP =
session between a pair of compatible endpoints that agree to do so.
>>> I have found *one* reason not mentioned in the draft:
>>>
>>> An RTP session with both "audio" and "video" media types cannot be
>>> represented by an SDP description, since SDP ties RTP sessions 1-1 to=

>>> the "m" line of the description, which contains the top-level type, a=
nd
>>> the codec descriptions omit the top-level type in their codec naming.=

>> I am not sure I understand this. Why would one need to use the same "m=
"
>> line? What would be the consequences of using two different "m" lines?=

>>
>> Personally, I am still not convinced that multiplexing brings
>> significant improvement over the status quo but if we accept it then i=
t
>> definitely needs to keep separate "m" lines for tons of reasons.
> RFC 4566 section 5:
>=20
>     An SDP session description consists of a session-level section
>     followed by zero or more media-level sections.  The session-level
>     part starts with a "v=3D" line and continues to the first media-lev=
el
>     section.  Each media-level section starts with an "m=3D" line and
>     continues to the next media-level section or end of the whole sessi=
on
>     description.
>=20
> ....
>=20
> 5.14.  Media Descriptions ("m=3D")
>=20
>        m=3D<media> <port> <proto> <fmt> ...
>=20
>     A session description may contain a number of media descriptions.
>     Each media description starts with an "m=3D" field and is terminate=
d by
>     either the next "m=3D" field or by the end of the session descripti=
on.
>     A media field has several sub-fields:
>=20
> <media> is the media type.  Currently defined media are "audio",
>        "video", "text", "application", and "message", although this lis=
t
>        may be extended in the future (see Section 8).
>=20
> <port> is the transport port to which the media stream is sent.
>=20
> In other words: an "m" line is tied to a port number, and (from RTP) th=
e=20
> port number identifies the RTP session. The ICE parameters can override=
=20
> the port number, but they're carried within the "m" block, so they're=20
> tied to a single "m" line too.

OK, but this simply brings the discussion back to whether or not we can
have audio and video on the same port. I still don't understand how it
adds extra constraints and why two m-lines with the same port would be
prohibited by SDP itself.

> I do not know what would happen if you used the same port number and IC=
E=20
> parameters with two "m" lines.

Aha! Right!

ICE itself would be able to handle this based on the user credentials.
However, having two sets of ICE candidates in both media descriptions
would simply trigger ICE processing for both of them.

Given that the whole idea behind multiplexing is to simplify ICE
signalling, I suppose this kind of kills it indeed.

Emil

--=20
http://jitsi.org


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On 24 Jul 2011, at 16:44, Matthew Kaufman wrote:
> I've been looking for technical reasons and/or hard requirements that =
would argue against doing RTP multiplexing as proposed in this draft.
>=20
> So far my conclusion is that the RTP specifications recommend against =
this, but do not prohibit it.
>=20
> Supporting evidence:
> RFC 3550 Section 2.2 notes that multiplexing A&V together may be a =
problem if you want to receive one and not together (presumably in the =
multicast case) and refers to Section 5.2 for more support.
>=20
> RFC 3550 Section 5.2 says "Separate audio and video streams SHOULD NOT =
be carried in a single RTP session and demultiplexed based on the =
payload type or SSRC fields." and provides five reasons. Note the =
"SHOULD NOT" as opposed to "MUST NOT".
>=20
> The same section says that the five reasons apply to multiplexing on =
payload type, only the last two apply to multiplexing on SSRC. These =
remaining reasons are:
>=20
>   4. An RTP mixer would not be able to combine interleaved streams of
>      incompatible media into one stream.
>=20
> Not relevant to the RTCWEB peer-to-peer case as the browser is not an =
RTP mixer. Also not relevant to any RTCWEB use cases that I can =
identify. And, if an RTP mixer at one end it can simply not agree to use =
multiplexing.

This would work if you are directly talking to a mixer. It precludes the =
cases where you are talking indirectly to a mixer via a gateway, unless =
you define additional signalling to determine that there is a mixer on =
the remote side of the gateway.=20

>   5. Carrying multiple media in one RTP session precludes: the use of
>      different network paths or network resource allocations if
>      appropriate;
>=20
> Appears to not be relevant to the RTCWEB case where a unicast path =
will be negotiated using ICE.

An RTCWeb client using ICE could negotiate different network paths for =
the audio and video flows, and could use network-based QoS mechanisms, =
if flows were sent separately. Those mechanisms work just fine in =
unicast.

>      reception of a subset of the media if desired, for
>      example just audio if video would exceed the available bandwidth;
>=20
> Appears to not be relevant to anything but the multicast cases. Any =
unicast case, including RTCWEB, can simply not have that component of =
the stream sent to it.

Relevant for proxies which do in-network adaptation.

>      and receiver implementations that use separate processes for the
>      different media, whereas using separate RTP sessions permits
>      either single- or multiple-process implementations.
>=20
> Appears to not be relevant to the RTCWEB use case. Additionally, if it =
is necessary for there to be separate processes or even devices at the =
far end this can be solved by again simply not agreeing to use =
multiplexing.
>=20
> In all my reading today I have not been able to find anything more =
concrete than the "SHOULD NOT" in section 5.2 of RFC3550. PLEASE follow =
up if you are aware of any other relevant specifications that would =
argue against using SSRC to multiplex audio and video streams over a =
single RTP session between a pair of compatible endpoints that agree to =
do so.


   Timing and Sequence Number Space:  An RTP SSRC is defined to identify
      a single timing and sequence number space.  Interleaving multiple
      payload types would require different timing spaces if the media
      clock rates differ and would require different sequence number
      spaces to tell which payload type suffered packet loss.  Using
      multiple clock rates in a single RTP session is problematic, as
      discussed in [I-D.ietf-avtext-multiple-clock-rates].  This can be
      avoided by partitioning the SSRC space between the two sessions,
      but that causes other problems as discussed below.

   RTCP Reception Reports:  RTCP sender reports and receiver reports can
      only describe one timing and sequence number space per SSRC, and
      do not carry a payload type field.  Multiplexing sessions based on
      the payload type breaks RTCP.  This can be avoided by partitioning
      the SSRC space between the two sessions, but that causes other
      problems as discussed below.

   Scalability:  RTP was built with media scalability in consideration.
      The simplest way of achieving separation between different
      scalability layers is placing them in different RTP sessions, and
      using the same SSRC and CNAME in each session to bind them
      together.  This is most commonly done in multicast, and not
      particularly applicable to RTC-Web, but gatewaying of such a
      session would then require more alterations and likely stateful
      translation.

   RTP Retransmission in Session Multiplexing mode:  RTP Retransmission
      [RFC4588] does have a mode for session multiplexing.  This would
      not be the main mode used in RTC-Web, but for interoperability and
      reduced cost in translation support for different RTP Sessions are
      beneficial.

   Forward Error Correction:  The "An RTP Payload Format for Generic
      Forward Error Correction" [RFC2733] and its update [RFC5109] can
      only be used on media formats that produce RTP packets that are
      smaller than half the MTU if the FEC flow and media flow being
      protected are to be sent in the same RTP session, this is due to
      "RTP Payload for Redundant Audio Data" [RFC2198].  This is because
      the SSRC value of the original flow is recovered from the FEC
      packets SSRC field.  So for anything that desires to use these
      format with RTP payloads that are close to MTU needs to put the
      FEC data in a separate RTP session compared to the original
      transmissions.  The usage of this type of FEC data has not been
      decided on in RTCWEB.

   SSRC Allocation and Collision:  The SSRC identifier is a random 32-
      bit number that is required to be globally unique within an RTP
      session, and that is reallocated to a new random value if an SSRC
      collision occurs between participants.  If two or more RTP
      sessions share a transport layer flow, there is no guarantee that
      their choice of SSRC values will be distinct, and there is no way
      in standard RTP to signal which SSRC values are used by which RTP
      session.  RTP is explicitly a group-based communication protocol,
      and new participants can join an RTP session at any time; these
      new participants may chose SSRC values that conflict with the SSRC
      values used in any of the multiplexed RTP sessions.  This problem
      can be avoided by partitioning the SSRC space, and signalling how
      the space is to be subdivided, but this is not backwards
      compatible with any existing RTP system.  In addition, subdividing
      the SSRC space makes it difficult to gateway between multiplexed
      RTP sessions and standard RTP sessions: the standard sessions may
      use parts of the SSRC space reserved in the multiplexed RTP
      sessions, requiring the gateway to rewrite RTCP packets, as well
      as the SSRC and CSRC list in RTP data packets.  Rewriting RTCP is
      a difficult task, especially when one considers extensions such as
      RTCP XR.

   Conflicting RTCP Report Types:  The extension mechanisms used in RTCP
      depend on separation of RTP sessions for different media types.
      For example, the RTCP Extended Report block for VoIP is suitable
      for conversational audio, but clearly not useful for Video.  This
      may cause unusable or unwanted reports to be generated for some
      streams, wasting capacity and confusing monitoring systems.  While
      this is problem may be unlikely for VoIP reports, it may be an
      issue for the more detailed media agnostic reports which are
      sometimes be used for different media types.  Also, this makes the
      implementation of RTCP more complex, since partitioning the SSRC
      space by media type needs not only to be one the media processing
      side, but also on the RTCP reporting

   RTCP Reporting and Scheduling:  The RTCP reporting interval and its
      packet scheduling will be affected if several RTP sessions are
      multiplexed onto the same transport layer flow.  The reporting
      interval is determined by the session bandwidth, and the reporting
      interval chosen for a high-rate video session will be different to
      the interval chosen by a low-rate VoIP session.  If such sessions
      are multiplexed, then participants in one session will see the
      SSRC values of the other session.  This will cause them to
      overestimate the number of participants in the session by a factor
      of two, thus doubling their RTCP reporting interval, and making
      their feedback less timely.  In the worst case, when an RTP
      session with very low RTCP bandwidth is multiplexed with an RTP
      session with high RTCP bandwidth, this may cause repeated RTCP
      timer reconsideration, leading to the members of the low bandwidth
      session timing out.  Participants in an RTP session configured
      with high bandwidth (and short RTCP reporting interval) will see
      RTCP reports from participants in the low bandwidth session much
      less often than expected, potentially causing them to repeatedly
      timeout and re-create state for those participants.  The split of
      RTCP bandwidth between senders and receivers (where at least 25%
      of the RTCP bandwidth is allocated to senders) will be disrupted
      if a session with few senders (e.g., a VoIP session) is
      multiplexed with a session with many senders (e.g., a video
      session).  These issues can be resolved if the partition of the
      SSRC is signalled, but this is not backwards compatible with any
      existing RTP system.  The partition would require re-implementing
      large part of the RTCP processing to take the individual sessions
      into account.

   Sampling Group Membership:  The mechanism defined in RFC2762 to
      sample the group membership, allowing participants to keep less
      state, assumes a single flat 32-bit SSRC space, and breaks if the
      SSRC space is shared between several RTP sessions.


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




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Thread-Topic: [rtcweb] Reasons not to multiplex audio with video (Re: Fwd: I-D Action: draft-rosenberg-rtcweb-rtpmux-00.txt)
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References: <4E123C54.10405@jdrosen.net><8785C0A3-31E5-44D7-8557-3BEEE4F95E3D@skype.net><4E2D5C5D.6060402@alvestrand.no> <4E2D5ED5.5060706@jitsi.org> <4E2D620F.5010003@jesup.org>
From: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
To: "Randell Jesup" <randell1@jesup.org>, <rtcweb@ietf.org>
X-OriginalArrivalTime: 25 Jul 2011 12:57:37.0933 (UTC) FILETIME=[6E2F13D0:01CC4ACA]
Subject: Re: [rtcweb] Reasons not to multiplex audio with video (Re: Fwd: I-D Action: draft-rosenberg-rtcweb-rtpmux-00.txt)
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Subject: Re: [rtcweb] Reasons not to multiplex audio with video (Re: Fwd: I-D Action: draft-rosenberg-rtcweb-rtpmux-00.txt)
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On Jul 25, 2011, at 8:06 AM, Harald Alvestrand wrote:

> On 07/24/11 16:44, Matthew Kaufman wrote:
>> In all my reading today I have not been able to find anything more =
concrete than the "SHOULD NOT" in section 5.2 of RFC3550. PLEASE follow =
up if you are aware of any other relevant specifications that would =
argue against using SSRC to multiplex audio and video streams over a =
single RTP session between a pair of compatible endpoints that agree to =
do so.
> I have found *one* reason not mentioned in the draft:
>=20
> An RTP session with both "audio" and "video" media types cannot be =
represented by an SDP description, since SDP ties RTP sessions 1-1 to =
the "m" line of the description, which contains the top-level type, and =
the codec descriptions omit the top-level type in their codec naming.
>=20
> I've said elsewhere that I consider this to be a design mistake for =
entirely different reasons, but this debate has reinforced my opinion =
that it is.

Good point, but perhaps it just argues for either A) not using SDP or B) =
extending SDP if we use it for RTCWEB (we probably need to do that =
anyway if we use SDP, given all the discussion about "hints" that we =
want to send from one end to the other)

Matthew Kaufman=

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Thread-Topic: [rtcweb] Reasons not to multiplex audio with video (Re: Fwd:I-D Action: draft-rosenberg-rtcweb-rtpmux-00.txt)
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From: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
To: "Matthew Kaufman" <matthew.kaufman@skype.net>, "Harald Alvestrand" <harald@alvestrand.no>
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Reasons not to multiplex audio with video (Re: Fwd:I-D Action: draft-rosenberg-rtcweb-rtpmux-00.txt)
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c) Using SDP for legacy interop, but extending STUN, by defining a
(comprehension optional) STUN attribute that indicates it is a
multiplexed media flow and that the SSRC space has been partitioned to
enable DPI. Since rtcweb apps are expected to use STUN, it should be
trivial for them to support this STUN attribute. In addition, the
software in on-path routers would have to be upgraded anyway to support
this king of DPI, so they could also be enhanced to detect STUN and look
for this attribute.

One another benefit to indicating this in STUN is that even if both
rtcweb endpoints said they supported this mechanism via SDP, the routers
where QoS, traffic engineering etc are applied to media packets many not
be there in the signaling path to understand that it is a multiplexed
media flow using the same port, and in particular that the SSRC space
has been partitioned as proposed in draft-rosenberg-rtcweb-rtpmux to
enable them to perform DPI and figure out the media type (audio/video
etc).

Muthu

|-----Original Message-----
|From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
Behalf Of Matthew Kaufman
|Sent: Monday, July 25, 2011 6:35 PM
|To: Harald Alvestrand
|Cc: rtcweb@ietf.org
|Subject: Re: [rtcweb] Reasons not to multiplex audio with video (Re:
Fwd:I-D Action: draft-rosenberg-
|rtcweb-rtpmux-00.txt)
|
|
|On Jul 25, 2011, at 8:06 AM, Harald Alvestrand wrote:
|
|> On 07/24/11 16:44, Matthew Kaufman wrote:
|>> In all my reading today I have not been able to find anything more
concrete than the "SHOULD NOT"
|in section 5.2 of RFC3550. PLEASE follow up if you are aware of any
other relevant specifications that
|would argue against using SSRC to multiplex audio and video streams
over a single RTP session between
|a pair of compatible endpoints that agree to do so.
|> I have found *one* reason not mentioned in the draft:
|>
|> An RTP session with both "audio" and "video" media types cannot be
represented by an SDP
|description, since SDP ties RTP sessions 1-1 to the "m" line of the
description, which contains the
|top-level type, and the codec descriptions omit the top-level type in
their codec naming.
|>
|> I've said elsewhere that I consider this to be a design mistake for
entirely different reasons, but
|this debate has reinforced my opinion that it is.
|
|Good point, but perhaps it just argues for either A) not using SDP or
B) extending SDP if we use it
|for RTCWEB (we probably need to do that anyway if we use SDP, given all
the discussion about "hints"
|that we want to send from one end to the other)
|
|Matthew Kaufman
|_______________________________________________
|rtcweb mailing list
|rtcweb@ietf.org
|https://www.ietf.org/mailman/listinfo/rtcweb

From harald@alvestrand.no  Mon Jul 25 06:50:20 2011
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Reasons not to multiplex audio with video (Re: Fwd:I-D Action: draft-rosenberg-rtcweb-rtpmux-00.txt)
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On 07/25/11 09:35, Muthu Arul Mozhi Perumal (mperumal) wrote:
> c) Using SDP for legacy interop, but extending STUN, by defining a
> (comprehension optional) STUN attribute that indicates it is a
> multiplexed media flow and that the SSRC space has been partitioned to
> enable DPI.
I think that's a layering violation. STUN has no references to RTP; 
there is no assumption in STUN that the content of the flow will be RTP. 
I think we should keep it that way.
>   Since rtcweb apps are expected to use STUN, it should be
> trivial for them to support this STUN attribute. In addition, the
> software in on-path routers would have to be upgraded anyway to support
> this king of DPI, so they could also be enhanced to detect STUN and look
> for this attribute.
>
> One another benefit to indicating this in STUN is that even if both
> rtcweb endpoints said they supported this mechanism via SDP, the routers
> where QoS, traffic engineering etc are applied to media packets many not
> be there in the signaling path to understand that it is a multiplexed
> media flow using the same port, and in particular that the SSRC space
> has been partitioned as proposed in draft-rosenberg-rtcweb-rtpmux to
> enable them to perform DPI and figure out the media type (audio/video
> etc).
>
> Muthu
>
> |-----Original Message-----
> |From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Matthew Kaufman
> |Sent: Monday, July 25, 2011 6:35 PM
> |To: Harald Alvestrand
> |Cc: rtcweb@ietf.org
> |Subject: Re: [rtcweb] Reasons not to multiplex audio with video (Re:
> Fwd:I-D Action: draft-rosenberg-
> |rtcweb-rtpmux-00.txt)
> |
> |
> |On Jul 25, 2011, at 8:06 AM, Harald Alvestrand wrote:
> |
> |>  On 07/24/11 16:44, Matthew Kaufman wrote:
> |>>  In all my reading today I have not been able to find anything more
> concrete than the "SHOULD NOT"
> |in section 5.2 of RFC3550. PLEASE follow up if you are aware of any
> other relevant specifications that
> |would argue against using SSRC to multiplex audio and video streams
> over a single RTP session between
> |a pair of compatible endpoints that agree to do so.
> |>  I have found *one* reason not mentioned in the draft:
> |>
> |>  An RTP session with both "audio" and "video" media types cannot be
> represented by an SDP
> |description, since SDP ties RTP sessions 1-1 to the "m" line of the
> description, which contains the
> |top-level type, and the codec descriptions omit the top-level type in
> their codec naming.
> |>
> |>  I've said elsewhere that I consider this to be a design mistake for
> entirely different reasons, but
> |this debate has reinforced my opinion that it is.
> |
> |Good point, but perhaps it just argues for either A) not using SDP or
> B) extending SDP if we use it
> |for RTCWEB (we probably need to do that anyway if we use SDP, given all
> the discussion about "hints"
> |that we want to send from one end to the other)
> |
> |Matthew Kaufman
> |_______________________________________________
> |rtcweb mailing list
> |rtcweb@ietf.org
> |https://www.ietf.org/mailman/listinfo/rtcweb
>


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Subject: Re: [rtcweb] Reasons not to multiplex audio with video (Re: Fwd:I-D Action: draft-rosenberg-rtcweb-rtpmux-00.txt)
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On Jul 25, 2011, at 9:35 AM, Muthu Arul Mozhi Perumal (mperumal) wrote:

> c) Using SDP for legacy interop, but extending STUN, by defining a
> (comprehension optional) STUN attribute that indicates it is a
> multiplexed media flow and that the SSRC space has been partitioned to
> enable DPI. Since rtcweb apps are expected to use STUN, it should be
> trivial for them to support this STUN attribute. In addition, the
> software in on-path routers would have to be upgraded anyway to =
support
> this king of DPI, so they could also be enhanced to detect STUN and =
look
> for this attribute.

Agree with Harald that this is a layering violation, but I would not be =
averse to putting a hint in STUN for this, or indeed a hint in STUN (as =
a non-mandatory attribute) saying that we're doing other RTCWEB things, =
for devices that can only see the RTP/RTCP/STUN packets.

That'd get yet another WG into the mix, of course.

Matthew Kaufman=

From jonathan@vidyo.com  Mon Jul 25 10:34:53 2011
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From: Jonathan Lennox <jonathan@vidyo.com>
To: Colin Perkins <csp@csperkins.org>
Date: Mon, 25 Jul 2011 13:34:46 -0400
Thread-Topic: [rtcweb] Fwd: I-D Action: draft-rosenberg-rtcweb-rtpmux-00.txt
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On Jul 25, 2011, at 8:50 AM, Colin Perkins wrote:

> On 24 Jul 2011, at 16:44, Matthew Kaufman wrote:
>>  4. An RTP mixer would not be able to combine interleaved streams of
>>     incompatible media into one stream.
>>=20
>> Not relevant to the RTCWEB peer-to-peer case as the browser is not an RT=
P mixer. Also not relevant to any RTCWEB use cases that I can identify. And=
, if an RTP mixer at one end it can simply not agree to use multiplexing.
>=20
> This would work if you are directly talking to a mixer. It precludes the =
cases where you are talking indirectly to a mixer via a gateway, unless you=
 define additional signalling to determine that there is a mixer on the rem=
ote side of the gateway.

Clearly, the gateway which is talking to the mixer needs to perform signali=
ng negotiation with it, so it understands the RTP sessions at all, and so i=
t will know if the mixer understands muxing or not.

>>  5. Carrying multiple media in one RTP session precludes: the use of
>>     different network paths or network resource allocations if
>>     appropriate;
>>=20
>> Appears to not be relevant to the RTCWEB case where a unicast path will =
be negotiated using ICE.
>=20
> An RTCWeb client using ICE could negotiate different network paths for th=
e audio and video flows, and could use network-based QoS mechanisms, if flo=
ws were sent separately. Those mechanisms work just fine in unicast.

Yes, it could use these mechanisms, if they're appropriate.  If it's not ap=
propriate, it wouldn't need to.

>> In all my reading today I have not been able to find anything more concr=
ete than the "SHOULD NOT" in section 5.2 of RFC3550. PLEASE follow up if yo=
u are aware of any other relevant specifications that would argue against u=
sing SSRC to multiplex audio and video streams over a single RTP session be=
tween a pair of compatible endpoints that agree to do so.

[Quote from Colin's draft omitted]

A lot of these are responses to the ideas of doing PT muxing, or muxing sev=
eral distinct RTP sessions onto a single transport flow.  They don't seem, =
mostly, to be addressing muxing RTP audio and video sources onto a single (=
proper) RTP session, i.e. with a single SSRC space, single (global) RTCP ba=
ndwidth value, single value for RTCP timing intervals, etc.

It's clearly the case that audio's share of 0.05 * (audio bw + video bw) is=
 going to be much greater than 0.05 * (audio bw), so you're going to be sen=
ding audio RTCP a lot more frequently with mux'ed media than you would in a=
n audio-only RTP session, whereas you'll send video RTCP slightly less freq=
uently than a video-only session.  But this isn't, as far as I can tell, a =
problem, especially if you set a sensible value for trr_int (assuming AVPF)=
.

In SSRC-muxing, RTX and FEC require a way of binding together original and =
repair streams, but RFC 5576 already defines a mechanism (ssrc-group) for d=
oing this in SDP, and Magnus's SRCNAME proposal could alternately do it in =
RTCP.

I admit to not understanding the XRBlocks well enough to say whether it exp=
ects to send VoIP feedback reports for every source in a session that it th=
inks is audio, but presumably a monitoring device that's monitoring a VoIP =
RTP session already needs to be told whether the session it's monitoring is=
 audio, video, or other, and would be able to (or have to) ignore a mux'd s=
ession.

--
Jonathan Lennox
jonathan@vidyo.com



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From: Jonathan Lennox <jonathan@vidyo.com>
To: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
Date: Mon, 25 Jul 2011 13:54:14 -0400
Thread-Topic: [rtcweb] Reasons not to multiplex audio with video (Re: Fwd: I-D Action: draft-rosenberg-rtcweb-rtpmux-00.txt)
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Subject: Re: [rtcweb] Reasons not to multiplex audio with video (Re: Fwd:	I-D Action: draft-rosenberg-rtcweb-rtpmux-00.txt)
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On Jul 25, 2011, at 8:57 AM, Muthu Arul Mozhi Perumal (mperumal) wrote:

> A few options to indicate this multiplexing in SDP and being backward com=
patible with legacy implementations:


Unfortunately, I think the "correct" way to do this would be SDP media capa=
bility negotiation -- you'd offer an alternative of either two m=3D lines (=
one audio and one video), or else a single, mux'd m=3D line.  Anything else=
 means that your SDP is, effectively, lying about what it's actually doing,=
 and I'm really worried about how that would interact with other SDP extens=
ions.

I'm certainly not crazy about this suggestion, though, because SDP Capabili=
ty Negotiation is pretty ugly, and requires an additional offer/answer exch=
ange to converge.

For RTCWeb proper, the need not to have worry about compatibility with lega=
cy implementations means that much of this could potentially be avoided.

--
Jonathan Lennox
jonathan@vidyo.com



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Hi Magnus,


On 7.25.2011 03:30 , "Magnus Westerlund" <magnus.westerlund@ericsson.com>
wrote:

>[...]
>
>>
>> That said, if I'm not completely mistaken, you can implement the
>> multiparty use cases using the "RTCP terminating MCU" topology of RFC
>> 5117.  This is common implementation practice in video conferencing.  If
>> you do so, the multiparty support of RTPv2 is neither exercised nor
>>needed
>> in a multiparty video conference.
>
>First of all if you do that you loose functionality as discussed in RFC
>5117. For example the indication of who actually sent a packet
>originally or contribute to the mix can't be provided unless you let the
>SSRC or CSRC plus CNAMES be forwarded across the Mixer/MCU.

Correct.  So what?  There is such a thing as a signaling plane.  In all
practical commercial systems I'm aware of, the signaling plane information
is the ONLY information that is used for the identification of
participants (including mixed participants).  Unfortunately, this
information is most often conveyed using nonstandard means.  No one in
video conferencing that I'm aware of relies on the SSRC/CSRC + CNAME
mechanism.  Let me know if I'm wrong.

I'm not saying that this design choice is necessarily the best possible
for rtcweb.  But I'm claiming that the practical implementation experience
speaks in favor of it.  So the burden of proof of requiring its use (which
may break other aspects that have been identified by many here as
critical, such as minimizing the number of ports), should be on you, not
on me. =20

>In addition it forces the central unit into one particular
>implementation, instead of allowing to use what it most appropriate.
>
>An RTP mixer when you want something that has functionality and
>application enhancing behavior and the cost is more complexity.
>
>Or you use an relay which simply copies a packet from one entity to all
>the others. That is the simplest central node you can build and it will
>have no media processing at all.

Sorry, I wasn't able to follow this train of thought.  If this was related
to implementation complexity of media plane signaling of mixed media
streams vs. signaling plane signaling of similar information, I would
argue that the majority of the implementers, so far, appear to have sided
with signaling plane rather than media plane=8A  I'm sure that they had a
reason. =20

>
>Also, in any gateway case to existing legacy you don't want to have to
>implement additional gatewaying functions just because one want to use
>non standard RTP.

That's also my understanding.  Is it strong enough an argument, given the
(many, I believe) other issues a gateway may need to worry about?

>
>For me it is clear that ensuring that RTCWEBs RTP usage is vanilla will
>help us avoid a lot of problems down the road.

I'm arguing the same thing as you, but we define "vanilla" differently.
You seem to define it as "use of RTP as specified in both language and
intent of the standard, and following the 1990 generation of understanding
of the standard".  I define it "use RTP in the way it has been implemented
in the industry relevant here (video conferencing industry) over the last
decade or so."

>
>If we are going to multiplex this mechanism needs to have a behavior
>that doesn't change the core behaviors of RTP. If we can do that the
>multiplexing becomes much less a hot topic.
>

Again, we agree, perhaps except for the interpretation of "core behavior".
 In my view, "core behavior" does not include mechanisms such as CC, CSRC,
as they are AFAIK  not practiced in most real-world systems.  It may
include, though, include mechanisms such as SSRC multiplexing, as this
mechanism IS deployed in millions of systems, with no known adverse
effects on the operation of the Internet as a whole, or the RTPish things
that are part of the Internet.

Cheers,
Stephan
=20

>
>Cheers
>
>Magnus Westerlund
>
>----------------------------------------------------------------------
>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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On Jul 25, 2011, at 8:06 AM, Harald Alvestrand wrote:

> An RTP session with both "audio" and "video" media types cannot be
> represented by an SDP description, since SDP ties RTP sessions 1-1 to the "m"
> line of the description, which contains the top-level type, and the codec
> descriptions omit the top-level type in their codec naming.
> 
> I've said elsewhere that I consider this to be a design mistake for entirely
> different reasons, but this debate has reinforced my opinion that it is.

This about sums it up why both SDP and RTP from VoIP of the early 1990's is
just not adequate. There are plenty of better solutions, such as metadata
standards for the app description data to replace stale SDP and app level
framing over UDP for RT media to replace RTP (a protocol build with media
mixing intermediaries/servers in mind).

Solutions without both SDP and RTP are widely deployed in dominant Internet
and Web RT communications products, albeit proprietary. And we want a
standard instead.

Avoiding telephony style VoIP and telecom multimedia usage scenarios and
their associated protocols is required to produce a successful standard for
webrtc, much preferable to the dominant proprietary products of today,
excellent as they may be.

Back to the drawing board?
It is not too late.

Thanks, Henry





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Subject: [rtcweb] Support for websockets
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Hi all

By when can we expect WebRTC code (Google code) to support Websockets?

Regards
Ranjit

From ekr@rtfm.com  Tue Jul 26 03:34:01 2011
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From: Eric Rescorla <ekr@rtfm.com>
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Subject: [rtcweb] Retransmit: Summary of Alternatives for media keying
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Sorry, sent to wrong list so some of you may not have it.

Hi Folks,

This is very late, but here is my attempted summary of the discussion
about COMSEC choices at the interim. I'm not saying that this captures
everyone's position, but the following positions are the ones that in
my view have significant levels of support.


Alternative A: [DTLS MUST, NO SDES, NO RTP]

   An RTC-Web client MUST support DTLS-SRTP and ONLY DTLS-SRTP for
   media, without support for either SRTP with supplied keying
   material (SDES-style) AND plain RTP. DTLS-SRTP provides for
   end-to-end key negotation between the two RTCWEB clients. The
   client MUST support the SRTP_AES128_CM_HMAC_SHA1_80 protection
   profile and the DTLS cipher suite
   TLS_DHE_RSA_WITH_AES_128_CBC_SHA. Note that this requirement
   differs from the current TLS default, TLS_RSA_WITH_AES_128_CBC_SHA
   in that it mandates support for Diffie-Hellman key exchange in
   order to provide Perfect Forward Secrecy.

   An RTCWEB client MUST provide its user with the ability to to see
   keying material information sufficient to allow indepent
   verification of their peer's identity.  (REF
   draft-kaufman-rtcweb-security-ui).

   The primary drawback of this alternative is the lack of backwards
   compatibility with devices and software that only support plain
   RTP, but the requirement for a handshake makes interoperation with
   these devices not completely trivial anyway.



Alternative B: [DTLS-SRTP MUST, SDES MAY, RTP MUST]

   An RTC-Web client MUST support DTLS-SRTP for media. DTLS-SRTP
   provides for end-to-end key negotation between the two RTCWEB
   clients.  The client MUST support the SRTP_AES128_CM_HMAC_SHA1_80
   protection profile and the DTLS cipher suite
   TLS_DHE_RSA_WITH_AES_128_CBC_SHA. Note that this requirement
   differs from the current TLS default, TLS_RSA_WITH_AES_128_CBC_SHA
   in that it mandates support for Diffie-Hellman key exchange in
   order to provide Perfect Forward Secrecy. This MUST be the default
   mode of operation.

   An RTCWEB client MAY support SRTP with the keying material supplied
   via the signaling channel with the SRTP_AES128_CM_HMAC_SHA1_80
   protection profile.  In the case of a web browser client, the
   keying material should be supplied via a Javascript API.
   DTLS-SRTP, with its end-to-end keying and authentication capability
   is preferred over SDES-style [RFC4568] keying.  However, the
   additional API overhead required to add support for a way to force
   a particular key is low.  In addition, once plain RTP is to be
   supported the arguments against the lower security level provided
   by SDES-style keying are no longer relevant.  Also there are a
   small number of potential use cases where interoperability with
   existing SDES-keyed software or devices may be achieved if the
   RTCWEB endpoint supports this mode of keying.

   An RTCWEB client MUST support RTP.  This provides no privacy but
   maximizes interoperability.  Note that SRTP with a Null cipher has
   equivalent security but does not meet the interoperability
   requirement.  Plain RTP provides no protection for the media, and
   so is discouraged as a mode of operation for RTCWEB.  However,
   support for RTP is required in order to provide interoperability
   with legacy RTP devices and software that do not support
   encryption.  In addition, some use cases such as high-volume PSTN
   or PBX gateways within an organization may scale more readily
   without the overhead of media encryption.

   An RTCWEB client MUST provide its user with the ability to know
   whether or not the media they are sending is protected by encryption
   and with the ability to see keying material information sufficient
   to allow indepent verification of their peer's identity.
   (REF draft-kaufman-rtcweb-security-ui). Note that this user
   interface element is much more critical---and hence much more
   problematic than with alternative A. If DTLS-SRTP is always
   used, then the user knows what security mechanisms are provided.
   As soon as multiple alternatives with widely varying security
   (or no security in the case of RTP) are provided, then users
   need to actually verify that the security level is satisfactory,
   which is inherently problematic given typical user behavior.

I expect to leave time for discussion of this during my slot tomorrow.
I look forward to a good discussion.

Best,
-Ekr

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Subject: Re: [rtcweb] Support for websockets
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On 07/26/11 01:28, Avasarala, Ranjit wrote:
> Hi all
>
> By when can we expect WebRTC code (Google code) to support Websockets?
Implementation issues with the Google WebRTC code should be addressed to 
the Google WebRTC implementation support mailing list: 
discuss-webrtc@googlegroups.com

In this forum, it may be appropriate to describe exactly how you see 
Websockets fitting in with the RTCWEB specification suite; so far, it's 
seemed to be not very closely related.
> Regards
> Ranjit
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


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    On 07/26/11 01:28, Avasarala, Ranjit wrote:
    <blockquote
cite="mid:9F9278CB1892FB48BF35CB5CC3992478A5395C720E@HKGMBOXPRD01.polycom.com"
      type="cite">
      <pre wrap="">Hi all

By when can we expect WebRTC code (Google code) to support Websockets?
</pre>
    </blockquote>
    Implementation issues with the Google WebRTC code should be
    addressed to the Google WebRTC implementation support mailing list:
    discuss-webrtc@googl<wbr>egroups.com<br>
    <br>
    In this forum, it may be appropriate to describe exactly how you see
    Websockets fitting in with the RTCWEB specification suite; so far,
    it's seemed to be not very closely related.<span
      class="Apple-style-span" style="color: rgb(119, 119, 119);
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cite="mid:9F9278CB1892FB48BF35CB5CC3992478A5395C720E@HKGMBOXPRD01.polycom.com"
      type="cite">
      <pre wrap="">
Regards
Ranjit
_______________________________________________
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>
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From bernard_aboba@hotmail.com  Tue Jul 26 06:18:35 2011
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Subject: Re: [rtcweb] Recordings from last interim meeting
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[BA] Do we have minutes of the meeting available?

Cullen Jennings said:

Here are the recordings from the last interim meeting. The first half of th=
e 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 wi=
ll 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

 		 	   		  =

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<body class=3D'hmmessage'><div dir=3D'ltr'>
[BA] Do we have minutes of the meeting available?<br><br><pre>Cullen Jennin=
gs said:<br><br>Here are the recordings from the last interim meeting. The =
first half of the meeting before the break is in the "PartA" files while th=
e 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 wi=
ndows users will probably do better with the .mp4 files.

<a rel=3D"nofollow" href=3D"http://dl.dropbox.com/u/17089001/IETF-RTCWeb-80=
.5-PartA.mov">http://dl.dropbox.com/u/17089001/IETF-RTCWeb-80.5-PartA.mov</=
a>
<a rel=3D"nofollow" href=3D"http://dl.dropbox.com/u/17089001/IETF-RTCWeb-80=
.5-PartA.mp4">http://dl.dropbox.com/u/17089001/IETF-RTCWeb-80.5-PartA.mp4</=
a>

<a rel=3D"nofollow" href=3D"http://dl.dropbox.com/u/17089001/IETF-RTCWeb-80=
.5-partB.mov">http://dl.dropbox.com/u/17089001/IETF-RTCWeb-80.5-partB.mov</=
a>
<a rel=3D"nofollow" href=3D"http://dl.dropbox.com/u/17089001/IETF-RTCWeb-80=
.5-partB.mp4">http://dl.dropbox.com/u/17089001/IETF-RTCWeb-80.5-partB.mp4</=
a></pre><br><div><br></div> 		 	   		  </div></body>
</html>=

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From: Ted Hardie <ted.ietf@gmail.com>
To: Bernard Aboba <bernard_aboba@hotmail.com>
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You should find them in the proceedings.  See:

http://www.ietf.org/proceedings/80/interim/rtcweb-interim-2011-06-08.txt

<http://www.ietf.org/proceedings/80/interim/rtcweb-interim-2011-06-08.txt>
regards,

Ted

On Tue, Jul 26, 2011 at 9:18 AM, Bernard Aboba <bernard_aboba@hotmail.com>w=
rote:

>  [BA] Do we have minutes of the meeting available?
>
> Cullen Jennings said:
>
> 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 wi=
ll probably do better with the .mp4 files.
> http://dl.dropbox.com/u/17089001/IETF-RTCWeb-80.5-PartA.movhttp://dl.drop=
box.com/u/17089001/IETF-RTCWeb-80.5-PartA.mp4
> http://dl.dropbox.com/u/17089001/IETF-RTCWeb-80.5-partB.movhttp://dl.drop=
box.com/u/17089001/IETF-RTCWeb-80.5-partB.mp4
>
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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You should find them in the proceedings. =A0See:<div><br></div><div><a href=
=3D"http://www.ietf.org/proceedings/80/interim/rtcweb-interim-2011-06-08.tx=
t">http://www.ietf.org/proceedings/80/interim/rtcweb-interim-2011-06-08.txt=
</a></div>
<div><br></div><div><a href=3D"http://www.ietf.org/proceedings/80/interim/r=
tcweb-interim-2011-06-08.txt"></a>regards,</div><div><br></div><div>Ted<br>=
<br><div class=3D"gmail_quote">On Tue, Jul 26, 2011 at 9:18 AM, Bernard Abo=
ba <span dir=3D"ltr">&lt;<a href=3D"mailto:bernard_aboba@hotmail.com" targe=
t=3D"_blank">bernard_aboba@hotmail.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">



<div><div dir=3D"ltr">
[BA] Do we have minutes of the meeting available?<br><br><pre>Cullen Jennin=
gs said:<br><br>Here are the recordings from the last interim meeting. The =
first half of the meeting before the break is in the &quot;PartA&quot; file=
s while the second half of the meeting after the break is in the &quot;Part=
B&quot; 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=
.

<a rel=3D"nofollow" href=3D"http://dl.dropbox.com/u/17089001/IETF-RTCWeb-80=
.5-PartA.mov" target=3D"_blank">http://dl.dropbox.com/u/17089001/IETF-RTCWe=
b-80.5-PartA.mov</a>
<a rel=3D"nofollow" href=3D"http://dl.dropbox.com/u/17089001/IETF-RTCWeb-80=
.5-PartA.mp4" target=3D"_blank">http://dl.dropbox.com/u/17089001/IETF-RTCWe=
b-80.5-PartA.mp4</a>

<a rel=3D"nofollow" href=3D"http://dl.dropbox.com/u/17089001/IETF-RTCWeb-80=
.5-partB.mov" target=3D"_blank">http://dl.dropbox.com/u/17089001/IETF-RTCWe=
b-80.5-partB.mov</a>
<a rel=3D"nofollow" href=3D"http://dl.dropbox.com/u/17089001/IETF-RTCWeb-80=
.5-partB.mp4" target=3D"_blank">http://dl.dropbox.com/u/17089001/IETF-RTCWe=
b-80.5-partB.mp4</a></pre><br><div><br></div> 		 	   		  </div></div>
<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br>
</div>

--20cf303f671ae787f304a8f8edf8--

From D.Malas@cablelabs.com  Tue Jul 26 06:58:05 2011
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From: Daryl Malas <D.Malas@cablelabs.com>
To: Eric Rescorla <ekr@rtfm.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Tue, 26 Jul 2011 07:58:01 -0600
Thread-Topic: [rtcweb] Retransmit: Summary of Alternatives for media keying
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Ekr et al,

I support Alternative B, and the reason is the support for plain RTP.  My
first reaction was Alternative A, and someone would be foolish to use
plain RTP over the Internet (unless they live in Hollywood and love to
have their phone calls recorded and played back on TMZ).  I think some
enterprises will want this level of support for interoperability with
their "older" IP-PBX/Media Gateway, and they will require their users to
use a VPN for using the web application within the enterprise (assuming a
enterprise web application for remote users), hence encrypting the media
anyway.  This will save them on additional costs they don't want to incur
for encryption/decryption.

Regards,

Daryl



On 7/26/11 6:33 AM, "Eric Rescorla" <ekr@rtfm.com> wrote:

>Sorry, sent to wrong list so some of you may not have it.
>
>Hi Folks,
>
>This is very late, but here is my attempted summary of the discussion
>about COMSEC choices at the interim. I'm not saying that this captures
>everyone's position, but the following positions are the ones that in
>my view have significant levels of support.
>
>
>Alternative A: [DTLS MUST, NO SDES, NO RTP]
>
>   An RTC-Web client MUST support DTLS-SRTP and ONLY DTLS-SRTP for
>   media, without support for either SRTP with supplied keying
>   material (SDES-style) AND plain RTP. DTLS-SRTP provides for
>   end-to-end key negotation between the two RTCWEB clients. The
>   client MUST support the SRTP_AES128_CM_HMAC_SHA1_80 protection
>   profile and the DTLS cipher suite
>   TLS_DHE_RSA_WITH_AES_128_CBC_SHA. Note that this requirement
>   differs from the current TLS default, TLS_RSA_WITH_AES_128_CBC_SHA
>   in that it mandates support for Diffie-Hellman key exchange in
>   order to provide Perfect Forward Secrecy.
>
>   An RTCWEB client MUST provide its user with the ability to to see
>   keying material information sufficient to allow indepent
>   verification of their peer's identity.  (REF
>   draft-kaufman-rtcweb-security-ui).
>
>   The primary drawback of this alternative is the lack of backwards
>   compatibility with devices and software that only support plain
>   RTP, but the requirement for a handshake makes interoperation with
>   these devices not completely trivial anyway.
>
>
>
>Alternative B: [DTLS-SRTP MUST, SDES MAY, RTP MUST]
>
>   An RTC-Web client MUST support DTLS-SRTP for media. DTLS-SRTP
>   provides for end-to-end key negotation between the two RTCWEB
>   clients.  The client MUST support the SRTP_AES128_CM_HMAC_SHA1_80
>   protection profile and the DTLS cipher suite
>   TLS_DHE_RSA_WITH_AES_128_CBC_SHA. Note that this requirement
>   differs from the current TLS default, TLS_RSA_WITH_AES_128_CBC_SHA
>   in that it mandates support for Diffie-Hellman key exchange in
>   order to provide Perfect Forward Secrecy. This MUST be the default
>   mode of operation.
>
>   An RTCWEB client MAY support SRTP with the keying material supplied
>   via the signaling channel with the SRTP_AES128_CM_HMAC_SHA1_80
>   protection profile.  In the case of a web browser client, the
>   keying material should be supplied via a Javascript API.
>   DTLS-SRTP, with its end-to-end keying and authentication capability
>   is preferred over SDES-style [RFC4568] keying.  However, the
>   additional API overhead required to add support for a way to force
>   a particular key is low.  In addition, once plain RTP is to be
>   supported the arguments against the lower security level provided
>   by SDES-style keying are no longer relevant.  Also there are a
>   small number of potential use cases where interoperability with
>   existing SDES-keyed software or devices may be achieved if the
>   RTCWEB endpoint supports this mode of keying.
>
>   An RTCWEB client MUST support RTP.  This provides no privacy but
>   maximizes interoperability.  Note that SRTP with a Null cipher has
>   equivalent security but does not meet the interoperability
>   requirement.  Plain RTP provides no protection for the media, and
>   so is discouraged as a mode of operation for RTCWEB.  However,
>   support for RTP is required in order to provide interoperability
>   with legacy RTP devices and software that do not support
>   encryption.  In addition, some use cases such as high-volume PSTN
>   or PBX gateways within an organization may scale more readily
>   without the overhead of media encryption.
>
>   An RTCWEB client MUST provide its user with the ability to know
>   whether or not the media they are sending is protected by encryption
>   and with the ability to see keying material information sufficient
>   to allow indepent verification of their peer's identity.
>   (REF draft-kaufman-rtcweb-security-ui). Note that this user
>   interface element is much more critical---and hence much more
>   problematic than with alternative A. If DTLS-SRTP is always
>   used, then the user knows what security mechanisms are provided.
>   As soon as multiple alternatives with widely varying security
>   (or no security in the case of RTP) are provided, then users
>   need to actually verify that the security level is satisfactory,
>   which is inherently problematic given typical user behavior.
>
>I expect to leave time for discussion of this during my slot tomorrow.
>I look forward to a good discussion.
>
>Best,
>-Ekr
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb


From magnus.westerlund@ericsson.com  Tue Jul 26 07:53:42 2011
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Hi,

All meeting material for today's session is now available:
https://datatracker.ietf.org/meeting/81/materials.html

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Frgatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
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Hi all,=0A=0Aa virtual room has been reserved on the Meetecho system. =0A=
=0AAccess to the on-line session (including audio and video streams) will=
 be available from 12:50 at:=0Ahttp://www.meetecho.com/ietf81/rtcweb=0A=0A=
The Meetecho session automatically logs you into the standard IETF jabber=
 room. So, from there, you can have an integrated experience involving al=
l media and allowing you to interact with the room.=0A=0AA tutorial of in=
teractivity features of the tool can be found at:=0Ahttp://www.meetecho.c=
om/ietf81/tutorials=0A=0AFor further information you can contact us at ie=
tf-support@meetecho.com.=0A=0ACheers,=0Athe Meetecho Team


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Thread-Topic: [rtcweb] Reasons not to multiplex audio with video (Re: Fwd:I-D Action: draft-rosenberg-rtcweb-rtpmux-00.txt)
Thread-Index: AcxK2Z4HiCeprPBrSK+zmam2QTyx4QAzFDSw
References: <4E123C54.10405@jdrosen.net><8785C0A3-31E5-44D7-8557-3BEEE4F95E3D@skype.net><4E2D5C5D.6060402@alvestrand.no> <AA1D0F89-32EC-48DE-B5B5-F856F72349DF@skype.net> <1D062974A4845E4D8A343C653804920205F72BF7@XMB-BGL-414.cisco.com> <D02EC89D-3500-4D7F-978F-BB8828EB76EC@skype.net>
From: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
To: "Matthew Kaufman" <matthew.kaufman@skype.net>
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Reasons not to multiplex audio with video (Re: Fwd:I-D Action: draft-rosenberg-rtcweb-rtpmux-00.txt)
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First refinement:
Define a new SDP attribute "a=3Dmuxed-candidate" that is similar to the
"a=3Dcandidate" attribute defined by ICE, except that the same
muxed-candidate can appear under multiple media stream descriptions in
SDP. The muxed-candidate identifies the transport address where the
RTCWEB endpoint would like to receive all these (multiplexed) media
streams. Non-RTCWEB endpoints would ignore this attribute.

Define a new comprehension-optional STUN attribute MUXED-CANDIDATE
(similar to the USE-CANDIDATE attribute). This attribute has no content
(the Length field of the attribute is zero) -- it serves as a flag.  It
has an attribute value of 0x802C (the first unassigned value from the
first half of the comprehension-optional range). It can be present in
the STUN connectivity check(s) sent when ICE concludes (i.e those STUN
Binding requests that contain the USE-CANDIDATE attribute) and serves as
an indication to routers and firewalls that any media stream that flows
between the selected pair of candidates is a multiplexed media stream --
for RTP it means that the SSRC space has been partitioned as described
in draft-rosenberg-rtcweb-rtpmux to enable DPI.

IMO, this isn't a layer violation per se, since we aren't carrying any
RTP stuff in STUN, rather just providing an indication for a particular
RTP usage.

Once issue, though -- comprehension-optional STUN attribute aren't
protected by STUN MESSAGE-INTEGRITY. So, MUXED-CANDIDATE would have to
protect itself or a new comprehension-optional STUN needs to be defined
that protects other comprehension-optional STUN attributes.

Muthu

|-----Original Message-----
|From: Matthew Kaufman [mailto:matthew.kaufman@skype.net]
|Sent: Monday, July 25, 2011 8:16 PM
|To: Muthu Arul Mozhi Perumal (mperumal)
|Cc: Harald Alvestrand; rtcweb@ietf.org
|Subject: Re: [rtcweb] Reasons not to multiplex audio with video (Re:
Fwd:I-D Action: draft-rosenberg-
|rtcweb-rtpmux-00.txt)
|
|
|On Jul 25, 2011, at 9:35 AM, Muthu Arul Mozhi Perumal (mperumal) wrote:
|
|> c) Using SDP for legacy interop, but extending STUN, by defining a
|> (comprehension optional) STUN attribute that indicates it is a
|> multiplexed media flow and that the SSRC space has been partitioned
to
|> enable DPI. Since rtcweb apps are expected to use STUN, it should be
|> trivial for them to support this STUN attribute. In addition, the
|> software in on-path routers would have to be upgraded anyway to
support
|> this king of DPI, so they could also be enhanced to detect STUN and
look
|> for this attribute.
|
|Agree with Harald that this is a layering violation, but I would not be
averse to putting a hint in
|STUN for this, or indeed a hint in STUN (as a non-mandatory attribute)
saying that we're doing other
|RTCWEB things, for devices that can only see the RTP/RTCP/STUN packets.
|
|That'd get yet another WG into the mix, of course.
|
|Matthew Kaufman

From matthew.kaufman@skype.net  Tue Jul 26 10:49:37 2011
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From: Matthew Kaufman <matthew.kaufman@skype.net>
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Subject: [rtcweb] NAT / firewall use cases
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We have something in there with a NAT... but I don't think we have a use =
case that covers the situation where one or both users is behind a NAT =
or firewall that blocks UDP entirely (which fortunately also covers some =
of the NAT cases where ICE doesn't succeed). This would require media =
over TCP... something that I believe some of the implementations have =
already tackled in different ways, but there's no use case and no drafts =
proposing which solution to choose either.

Matthew Kaufman=

From ron.even.tlv@gmail.com  Tue Jul 26 10:51:04 2011
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From: "Roni Even" <ron.even.tlv@gmail.com>
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Hi,

For other IETF WG use cases:

 

The CLUE WG is working on use-cases for high quality video conferencing
(telepresence) it may be interesting to the use cases in RTCweb

http://tools.ietf.org/html/draft-ietf-clue-telepresence-use-cases-01

 

On conferencing requirements and scenarios done in SIPPING and XCON WGs are
in http://tools.ietf.org/html/rfc4245 and http://tools.ietf.org/html/rfc4597


 

Roni Even 


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<html xmlns:v=3D"urn:schemas-microsoft-com:vml" =
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</o:shapelayout></xml><![endif]--></head><body lang=3DEN-US link=3Dblue =
vlink=3Dpurple><div class=3DWordSection1><p =
class=3DMsoNormal>Hi,<o:p></o:p></p><p class=3DMsoNormal>For other IETF =
WG use cases:<o:p></o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal>The CLUE WG is working on use-cases for high quality =
video conferencing (telepresence) it may be interesting to the use cases =
in RTCweb<o:p></o:p></p><p class=3DMsoNormal><a =
href=3D"http://tools.ietf.org/html/draft-ietf-clue-telepresence-use-cases=
-01">http://tools.ietf.org/html/draft-ietf-clue-telepresence-use-cases-01=
</a><o:p></o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal>On conferencing requirements and scenarios done in =
SIPPING and XCON WGs are in <a =
href=3D"http://tools.ietf.org/html/rfc4245">http://tools.ietf.org/html/rf=
c4245</a> and <a =
href=3D"http://tools.ietf.org/html/rfc4597">http://tools.ietf.org/html/rf=
c4597</a> <o:p></o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal>Roni Even <o:p></o:p></p></div></body></html>
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From bernard_aboba@hotmail.com  Tue Jul 26 11:24:25 2011
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From: Emil Ivov <emcho@jitsi.org>
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+1

Emil

-- sent from my mobile
On Jul 26, 2011 8:24 PM, "Bernard Aboba" <bernard_aboba@hotmail.com> wrote:
>
> This is indeed a real use case.
>
> Matthew Kaufman <matthew.kaufman@skype.net> wrote:
>
> >We have something in there with a NAT... but I don't think we have a use
case that covers the situation where one or both users is behind a NAT or
firewall that blocks UDP entirely (which fortunately also covers some of the
NAT cases where ICE doesn't succeed). This would require media over TCP...
something that I believe some of the implementations have already tackled in
different ways, but there's no use case and no drafts proposing which
solution to choose either.
> >
> >Matthew Kaufman
> >_______________________________________________
> >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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<p>+1</p>
<p>Emil</p>
<p>-- sent from my mobile<br>
On Jul 26, 2011 8:24 PM, &quot;Bernard Aboba&quot; &lt;<a href=3D"mailto:be=
rnard_aboba@hotmail.com">bernard_aboba@hotmail.com</a>&gt; wrote:<br>
&gt;<br>
&gt; This is indeed a real use case.<br>
&gt;<br>
&gt; Matthew Kaufman &lt;<a href=3D"mailto:matthew.kaufman@skype.net">matth=
ew.kaufman@skype.net</a>&gt; wrote:<br>
&gt;<br>
&gt; &gt;We have something in there with a NAT... but I don&#39;t think we =
have a use case that covers the situation where one or both users is behind=
 a NAT or firewall that blocks UDP entirely (which fortunately also covers =
some of the NAT cases where ICE doesn&#39;t succeed). This would require me=
dia over TCP... something that I believe some of the implementations have a=
lready tackled in different ways, but there&#39;s no use case and no drafts=
 proposing which solution to choose either.<br>

&gt; &gt;<br>
&gt; &gt;Matthew Kaufman<br>
&gt; &gt;_______________________________________________<br>
&gt; &gt;rtcweb mailing list<br>
&gt; &gt;<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; &gt;<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://w=
ww.ietf.org/mailman/listinfo/rtcweb</a><br>
&gt; &gt;<br>
&gt; _______________________________________________<br>
&gt; rtcweb mailing list<br>
&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.i=
etf.org/mailman/listinfo/rtcweb</a><br>
</p>

--bcaec547c74b4c3d9a04a8fd25dd--

From john.elwell@siemens-enterprise.com  Tue Jul 26 13:14:33 2011
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To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Tue, 26 Jul 2011 22:14:29 +0200
Thread-Topic: Possible new use case
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(Not sure whether this should go to the W3C list,  but trying the IETF list=
 to start with)

I raised this during the WEBRTC session on Saturday.

Does anyone have any interest in adding a use case concerning recording of =
real-time media at a remote recorder? I noticed something on local recordin=
g in the proposed APIs, but remote recording might have other impacts, e.g.=
,
- taking a stream from a capture device (mic, camera) and sending it not on=
ly to the remote browser but also to a remote recorder;
- taking a stream from the remote browser and sending it to a remote record=
er (as well as rendering locally);
- possibly mixing the two streams (in case of audio) and sending as a singl=
e mixed stream to the recording device.

John


John Elwell
Tel: +44 1908 817801 (office and mobile)
Email: john.elwell@siemens-enterprise.com
http://www.siemens-enterprise.com/uk/

Siemens Enterprise Communications Limited.
Registered office: Brickhill Street, Willen Lake, Milton Keynes, MK15 0DJ.
Registered No: 5903714, England.

Siemens Enterprise Communications Limited is a Trademark Licensee of Siemen=
s AG.
 =

From rsf@ns.live555.com  Tue Jul 26 13:55:46 2011
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To: rtcweb@ietf.org
From: Ross Finlayson <finlayson@live555.com>
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Subject: Re: [rtcweb] Reasons (not?) to multiplex audio with video
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When I first started thinking about the 'multiplexing' question, I 
tended to support Colin's point-of-view.  Carrying audio and video on 
separate ports gives a system more flexibility, is compatible with 
existing implementations (including mine :-), and is in line with 
long-standing assumptions about the use of the SSRC, RTCP etc.

But over time I have become increasingly sympathetic to Jonathan's 
arguments - especially regarding NAT traversal - in particular, setup 
time, use of resources (# of ports), and failure modes.  We have to 
assume that many, if not most users of RTCWEB systems will be on 
unsophisticated, heavily NATed IPv4 home (or hotel, or coffee shop 
etc.) networks.  As IPv4 addresses become scarcer, this situation 
will likely only get worse.

However, as someone noted (in the context of "security") during 
today's RTCWEB session, we're not the first people to be designing a 
(hopefully) widely-deployed consumer-oriented peer-to-peer A/V chat 
system.  There are several existing systems out there, so it might be 
instructive to look at how they address the 'media multiplexing' 
issue - in particular:

1/ Apple's "FaceTime".  OK, this is not browser-based (although with 
a bit of work it probably could have been), but it does use RTP.  Is 
it using separate ports for audio and video (and two more for RTCP)? 
If so, has NAT traversal been a problem, and would it have been 
beneficial to have been able to reduce the number of ports used?  (Is 
Dave Singer on this mailing list?)

2/ Facebook's new video chat.  OK, this uses Skype, and thus 
presumably not RTP (at least, not now).  But does this multiplex 
audio+video on a single port?  (And if so, did this help motivate 
Jonathan's point-of-view?)

3/ Google Plus "Hangouts".  Similar question to 1/.

4/ Any others (whether RTP-based or not)?

-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/

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On Jul 26, 2011, at 4:55 PM, Ross Finlayson wrote:
>=20
> 2/ Facebook's new video chat.  OK, this uses Skype, and thus =
presumably not RTP (at least, not now).  But does this multiplex =
audio+video on a single port?

Yes.

>  (And if so, did this help motivate Jonathan's point-of-view?)

I can't answer this.

>=20
>=20
> 4/ Any others (whether RTP-based or not)?

All systems that use Flash Player's RTMFP for peer-to-peer audio/video =
communication (an early example was Chatroulette) multiplex audio+video =
on a single port *and* in fact use that single port to talk to all the =
other participants in the multi-party cases. (So for instance, when you =
"next" someone in Chatroulette, your local address+port are the same as =
they were for the last guy.)

Matthew Kaufman


--Apple-Mail-20-548917979
Content-Transfer-Encoding: quoted-printable
Content-Type: text/html;
	charset=us-ascii

<html><head></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
"><br><div><div>On Jul 26, 2011, at 4:55 PM, Ross Finlayson =
wrote:</div><blockquote type=3D"cite"><div><font =
class=3D"Apple-style-span" color=3D"#000000"><br></font>2/ Facebook's =
new video chat. &nbsp;OK, this uses Skype, and thus presumably not RTP =
(at least, not now). &nbsp;But does this multiplex audio+video on a =
single =
port?</div></blockquote><div><br></div>Yes.</div><div><br><blockquote =
type=3D"cite"><div> &nbsp;(And if so, did this help motivate Jonathan's =
point-of-view?)<br></div></blockquote><div><br></div>I can't answer =
this.</div><div><br><blockquote type=3D"cite"><div><br><br>4/ Any others =
(whether RTP-based or not)?<br></div></blockquote><div><br></div>All =
systems that use Flash Player's RTMFP for peer-to-peer audio/video =
communication (an early example was Chatroulette) multiplex audio+video =
on a single port *and* in fact use that single port to talk to all the =
other participants in the multi-party cases. (So for instance, when you =
"next" someone in Chatroulette, your local address+port are the same as =
they were for the last guy.)</div><div><br></div><div>Matthew =
Kaufman</div><div><br></div></body></html>=

--Apple-Mail-20-548917979--

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On 07/26/11 16:55, Ross Finlayson wrote:
>
> 3/ Google Plus "Hangouts".  Similar question to 1/.
Google+ Hangouts uses SRTP, with one port for video and one port for 
audio. Many audio and video channels (multiplexed by SSRC) in the 
server->client direction.

We multiplex RTCP onto the RTP port as specified in the relevant RFC.

                       Harald


From Bala@vidyo.com  Tue Jul 26 14:48:34 2011
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From: Bala Pitchandi <Bala@vidyo.com>
To: Ross Finlayson <finlayson@live555.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Tue, 26 Jul 2011 17:48:30 -0400
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> 1/ Apple's "FaceTime".  OK, this is not browser-based (although with a bi=
t of work it probably could have been), but it does use RTP.  Is it using s=
eparate ports for audio and video (and two more for RTCP)?=20
> If so, has NAT traversal been a problem, and would it have been beneficia=
l to have been able to reduce the number of ports used?  (Is Dave Singer on=
 this mailing list?)

I believe FaceTime multiplexes audio & video on to the same port. In fact, =
it even multiplexes the signaling on to the same port. See: http://blog.roy=
chowdhury.org/2010/06/25/facetime-on-iphone-4-vanilla-unencrypted-stun-and-=
sip/

-- Bala


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From: Aron Rosenberg <arosenberg@logitech.com>
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--20cf305b12824b60bd04a900083f
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Facetime muxes everything onto the same port using RTP SSRC and signals that
in SDP using an a=rtpID attribute inside each m= line. Each m= line has the
same <port> value.

-Aron

On Tue, Jul 26, 2011 at 2:48 PM, Bala Pitchandi <Bala@vidyo.com> wrote:

> > 1/ Apple's "FaceTime".  OK, this is not browser-based (although with a
> bit of work it probably could have been), but it does use RTP.  Is it using
> separate ports for audio and video (and two more for RTCP)?
> > If so, has NAT traversal been a problem, and would it have been
> beneficial to have been able to reduce the number of ports used?  (Is Dave
> Singer on this mailing list?)
>
> I believe FaceTime multiplexes audio & video on to the same port. In fact,
> it even multiplexes the signaling on to the same port. See:
> http://blog.roychowdhury.org/2010/06/25/facetime-on-iphone-4-vanilla-unencrypted-stun-and-sip/
>
> -- Bala
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--20cf305b12824b60bd04a900083f
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Facetime muxes everything onto the same port using RTP SSRC and signals tha=
t in SDP using an a=3DrtpID attribute inside each m=3D line. Each m=3D line=
 has the same &lt;port&gt; value.<br clear=3D"all"><div><br></div><div>-Aro=
n</div>

<br><div class=3D"gmail_quote">On Tue, Jul 26, 2011 at 2:48 PM, Bala Pitcha=
ndi <span dir=3D"ltr">&lt;<a href=3D"mailto:Bala@vidyo.com">Bala@vidyo.com<=
/a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:=
0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">

<div class=3D"im">&gt; 1/ Apple&#39;s &quot;FaceTime&quot;. =A0OK, this is =
not browser-based (although with a bit of work it probably could have been)=
, but it does use RTP. =A0Is it using separate ports for audio and video (a=
nd two more for RTCP)?<br>


&gt; If so, has NAT traversal been a problem, and would it have been benefi=
cial to have been able to reduce the number of ports used? =A0(Is Dave Sing=
er on this mailing list?)<br>
<br>
</div>I believe FaceTime multiplexes audio &amp; video on to the same port.=
 In fact, it even multiplexes the signaling on to the same port. See: <a hr=
ef=3D"http://blog.roychowdhury.org/2010/06/25/facetime-on-iphone-4-vanilla-=
unencrypted-stun-and-sip/" target=3D"_blank">http://blog.roychowdhury.org/2=
010/06/25/facetime-on-iphone-4-vanilla-unencrypted-stun-and-sip/</a><br>


<font color=3D"#888888"><br>
-- Bala<br>
</font><div><div></div><div class=3D"h5"><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>

--20cf305b12824b60bd04a900083f--

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Subject: [rtcweb] E911 Use Case
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--_008a9b44-395c-4cfb-86eb-dcb73b1344eb_
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I thought I would put forward a potential E911 use case for consideration b=
y the group:

* Emergency Access for the disabled.   An individual who is speech and/or h=
earing impaired and is fluent in ASL needs to make an emergency call.   The=
 call will support location conveyance as well as video (to accommodate ASL=
 interpretation) and an audio stream (which might be only one-way) and will=
 terminate on an ESInet conferencing system as described in NENA i3 stage 3=
.  The conferencing system will automatically bridge in an ASL interpreter =
who will provide an audio stream and the mixed audio as well as video strea=
m will be provided to the dispatcher=2C along with the caller location info=
rmation.=20


 		 	   		  =

--_008a9b44-395c-4cfb-86eb-dcb73b1344eb_
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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'>
I thought I would put forward a potential E911 use case for consideration b=
y the group:<br><br>* Emergency Access for the disabled.&nbsp=3B&nbsp=3B An=
 individual who is speech and/or hearing impaired and is fluent in ASL need=
s to make an emergency call.&nbsp=3B&nbsp=3B The call will support location=
 conveyance as well as video (to accommodate ASL interpretation) and an aud=
io stream (which might be only one-way) and will terminate on an ESInet con=
ferencing system as described in NENA i3 stage 3.&nbsp=3B The conferencing =
system will automatically bridge in an ASL interpreter who will provide an =
audio stream and the mixed audio as well as video stream will be provided t=
o the dispatcher=2C along with the caller location information. <br><br><br=
> 		 	   		  </div></body>
</html>=

--_008a9b44-395c-4cfb-86eb-dcb73b1344eb_--

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Dear all,the full recording (synchronized video, audio, slides and jabber=
 room) of yesterday's RTCWEB session is available at the following URL:ht=
tp://www.meetecho.com/ietf81/recordingsFor the chair(s): please feel free=
 to put the link to the recording in the minutes, if you think this might=
 be useful.In case of problems with the playout, just drop an e-mail to i=
etf-support@meetecho.com.Cheers,the Meetecho Team

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=0A<div class=3D"xam_msg_class">=0A<div style=3D"font: normal 13px Arial;=
 color:rgb(0, 0, 0);">Dear all,<br><br>the full recording (synchronized v=
ideo, audio, slides and jabber room) <br>of yesterday's RTCWEB session is=
 available at the following URL:<br><br>http://www.meetecho.com/ietf81/re=
cordings<br><br>For the chair(s): please feel free to put the link to the=
 recording in the minutes, if you think this might be useful.<br><br>In c=
ase of problems with the playout, just drop an e-mail to <br>ietf-support=
@meetecho.com.<br><br>Cheers,<br>the Meetecho Team<br></div>=0A</div>=0A

--_=__=_XaM3_.1311779425.2A.747026.42.27022.52.42.007.1309547597--


From ted.ietf@gmail.com  Wed Jul 27 13:22:19 2011
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Subject: [rtcweb] Updated agenda for meeting 2
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We've updated the agenda for meeting two; it is uploaded and included
below for your convenience.

thanks,

Ted, Cullen, Magnus


Agenda Bash & Status Report (Chairs) - 10 min

Liaison Report from W3C WebRTC -(WebRTC Chairs) - 10 min
   - The W3C will have just had a meeting the Saturday before

Implementation Experience (Tim, Cary, Harald, Stefan)- 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
   - 3 minute updates from four groups

Use Cases (Stefan) - 20 min
   draft-ietf-rtcweb-use-cases-and-requirements

Architecture and System Overview (Harald) -  20 min
   draft-ietf-rtcweb-overview

Security (Eric) -  20 min
   draft-rescorla-rtcweb-security
   draft-kaufman-rtcweb-security-ui
   draft-johnston-rtcweb-media-privacy


Start of Session two (Chairs) - 5 min

Multiplexing
   * discussion updated during the week
 Jonathan Lennox and JDRosenberg 20 minutes

Negotiation & signaling (Cullen) - 20 min
  draft-cbran-rtcweb-negotiation

Media other than multiplexing  (Colin) - 20 min
   draft-perkins-rtcweb-rtp-usage
   draft-cbran-rtcweb-media

 NAT (Matthew/Cary) - 15 min
   draft-kaufman-rtcweb-traversal
   draft-cbran-rtcweb-nat

Datagram transports (Cary/Matthew) 20 min
   draft-cbran-rtcweb-data
   draft-kaufman-rtp-compatible-data

Layered Codecs (Stephan) - 10 min
   draft-wenger-rtcweb-layered-codec

Media processing & codec (Cary) - 10 min
   draft-cbran-rtcweb-codec
   * focus on requirements for codecs

From HKaplan@acmepacket.com  Wed Jul 27 23:17:47 2011
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Thu, 28 Jul 2011 02:17:42 -0400
Thread-Topic: Retransmit: Summary of Alternatives for media keying  
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Subject: Re: [rtcweb] Retransmit: Summary of Alternatives for media keying
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Well gee that doesn't sound like too hard a choice set to choose from, thou=
gh I guess it depends on whether there's a requirement/desire to interop wi=
th existing VoIP devices and the PSTN, or not.

If we don't expect/desire RTCWEB clients to communicate with existing VoIP =
devices or PSTN, then sure we can start with a clean slate.  In fact, you c=
ould also create a whole new version of SIP for inter-server, since there's=
 no need to use SIP v2.0 between RTCWEB domains and we've learned things we=
 could have done better, and for a pure inter-domain server-to-server signa=
ling protocol it could be a lot simpler.=20

But anyway... with regards to your alternatives below, you seem to imply th=
at using DTLS-SRTP is by itself "secure", such that the UI could display a =
lock-icon for example.  I was under the impression that DTLS-SRTP provided =
no guarantee of anything without the humans on both ends verifying they're =
using the same keying material/SAS/etc.  Is that not the case? (okay yes it=
 requires being a MitM to subvert, but that's not a high enough bar to clai=
m anything)

Assuming that's true, and given how few humans would actually know to perfo=
rm manual verification of dtls-srtp, I think this topic isn't as cut-and-dr=
ied as choosing between "secure" vs. "insecure".  Both alternatives are cap=
able of being secure or insecure, and it's up to the human to do something =
to figure it out in either case.  There is no real alternative which is inh=
erently "secure", afaict.

-hadriel


Recently EKR wrote:
> This is very late, but here is my attempted summary of the discussion
> about COMSEC choices at the interim. I'm not saying that this captures
> everyone's position, but the following positions are the ones that in
> my view have significant levels of support.
>=20
>=20
> Alternative A: [DTLS MUST, NO SDES, NO RTP]
>=20
>    An RTC-Web client MUST support DTLS-SRTP and ONLY DTLS-SRTP for
>    media, without support for either SRTP with supplied keying
>    material (SDES-style) AND plain RTP. DTLS-SRTP provides for
>    end-to-end key negotation between the two RTCWEB clients. The
>    client MUST support the SRTP_AES128_CM_HMAC_SHA1_80 protection
>    profile and the DTLS cipher suite
>    TLS_DHE_RSA_WITH_AES_128_CBC_SHA. Note that this requirement
>    differs from the current TLS default, TLS_RSA_WITH_AES_128_CBC_SHA
>    in that it mandates support for Diffie-Hellman key exchange in
>    order to provide Perfect Forward Secrecy.
>=20
>    An RTCWEB client MUST provide its user with the ability to to see
>    keying material information sufficient to allow indepent
>    verification of their peer's identity.  (REF
>    draft-kaufman-rtcweb-security-ui).
>=20
>    The primary drawback of this alternative is the lack of backwards
>    compatibility with devices and software that only support plain
>    RTP, but the requirement for a handshake makes interoperation with
>    these devices not completely trivial anyway.
>=20
>=20
>=20
> Alternative B: [DTLS-SRTP MUST, SDES MAY, RTP MUST]
>=20
>    An RTC-Web client MUST support DTLS-SRTP for media. DTLS-SRTP
>    provides for end-to-end key negotation between the two RTCWEB
>    clients.  The client MUST support the SRTP_AES128_CM_HMAC_SHA1_80
>    protection profile and the DTLS cipher suite
>    TLS_DHE_RSA_WITH_AES_128_CBC_SHA. Note that this requirement
>    differs from the current TLS default, TLS_RSA_WITH_AES_128_CBC_SHA
>    in that it mandates support for Diffie-Hellman key exchange in
>    order to provide Perfect Forward Secrecy. This MUST be the default
>    mode of operation.
>=20
>    An RTCWEB client MAY support SRTP with the keying material supplied
>    via the signaling channel with the SRTP_AES128_CM_HMAC_SHA1_80
>    protection profile.  In the case of a web browser client, the
>    keying material should be supplied via a Javascript API.
>    DTLS-SRTP, with its end-to-end keying and authentication capability
>    is preferred over SDES-style [RFC4568] keying.  However, the
>    additional API overhead required to add support for a way to force
>    a particular key is low.  In addition, once plain RTP is to be
>    supported the arguments against the lower security level provided
>    by SDES-style keying are no longer relevant.  Also there are a
>    small number of potential use cases where interoperability with
>    existing SDES-keyed software or devices may be achieved if the
>    RTCWEB endpoint supports this mode of keying.
>=20
>    An RTCWEB client MUST support RTP.  This provides no privacy but
>    maximizes interoperability.  Note that SRTP with a Null cipher has
>    equivalent security but does not meet the interoperability
>    requirement.  Plain RTP provides no protection for the media, and
>    so is discouraged as a mode of operation for RTCWEB.  However,
>    support for RTP is required in order to provide interoperability
>    with legacy RTP devices and software that do not support
>    encryption.  In addition, some use cases such as high-volume PSTN
>    or PBX gateways within an organization may scale more readily
>    without the overhead of media encryption.
>=20
>    An RTCWEB client MUST provide its user with the ability to know
>    whether or not the media they are sending is protected by encryption
>    and with the ability to see keying material information sufficient
>    to allow indepent verification of their peer's identity.
>    (REF draft-kaufman-rtcweb-security-ui). Note that this user
>    interface element is much more critical---and hence much more
>    problematic than with alternative A. If DTLS-SRTP is always
>    used, then the user knows what security mechanisms are provided.
>    As soon as multiple alternatives with widely varying security
>    (or no security in the case of RTP) are provided, then users
>    need to actually verify that the security level is satisfactory,
>    which is inherently problematic given typical user behavior.
>=20
> I expect to leave time for discussion of this during my slot tomorrow.
> I look forward to a good discussion.
>=20
> Best,
> -Ekr
>=20


From HKaplan@acmepacket.com  Thu Jul 28 00:52:27 2011
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Thu, 28 Jul 2011 03:52:20 -0400
Thread-Topic: Strawman for how to prevent voice-hammer without ICE
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Subject: [rtcweb] Strawman for how to prevent voice-hammer without ICE
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Howdy,
With regard to mandating ICE, such that an RTCWEB browser cannot send RTP w=
ithout doing ICE successfully first... is that restriction purely to preven=
t the voice-hammer attacks?  If so, then it's unfortunate because obviously=
 it would seriously reduce interop with non-RTCWEB VoIP devices.  But I thi=
nk there's a way to prevent the hammer attack without using ICE, and withou=
t requiring legacy VoIP devices to change whatsoever.

One way would be to use the receipt of RTP as an indicator the far-end expe=
cts to receive it from you.  So have the browser generate RTP/RTCP packets,=
 at a relatively slow rate, for a short duration (e.g., use the same rate/r=
etransmit timers of STUN connectivity checks in ICE).  If the browser recei=
ves RTP/RTCP packets, then the far-end expected to receive them as well and=
 the browser can do normal RTP from then on.

If this was a hammer attack, this doesn't generate any more load on the tar=
get than ICE, since ICE would have sent X number of STUN packets for Y time=
 as well, and I'm suggesting the X and Y be the same values for the initial=
 RTP packets during the "check" phase.

The major weakness of this approach is that a malicious web-server trying t=
o get your browser to do the voice hammer could send RTP to your browser, s=
ince it knows the address/ports of both sides, codec payload types, etc. (i=
.e., it can spoof being the attack target to make your browser think it's o=
k to do normal RTP)  But we could probably play games with RTCP SR/RR or ev=
en just require continued RTP receipt to send RTP, in order to mitigate thi=
s weakness or make it of low value to exploit.

Does anyone else care about interop-ing with legacy non-RTCWEB voip devices=
?  I checked draft-ietf-rtcweb-use-cases-and-requirements-01 and I don't se=
e it, so I'm not sure.

-hadriel


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Agree that the "clean slate" approach is not very appealing.  "Since we req=
uire ICE=2C we can also require X=2C Y=2C Z" is flawed logic=2C not only fr=
om a deployment/interoperability point of view=2C but also from a regulator=
y point of view.  The definitions of interconnected/non-interconnected VOIP=
 and the associated regulatory requirements do not have a browser exemption=
.=20

> From: HKaplan@acmepacket.com
> To: rtcweb@ietf.org
> Date: Thu=2C 28 Jul 2011 02:17:42 -0400
> Subject: Re: [rtcweb] Retransmit: Summary of Alternatives for media keyin=
g
>=20
>=20
> Well gee that doesn't sound like too hard a choice set to choose from=2C =
though I guess it depends on whether there's a requirement/desire to intero=
p with existing VoIP devices and the PSTN=2C or not.
>=20
> If we don't expect/desire RTCWEB clients to communicate with existing VoI=
P devices or PSTN=2C then sure we can start with a clean slate.  In fact=2C=
 you could also create a whole new version of SIP for inter-server=2C since=
 there's no need to use SIP v2.0 between RTCWEB domains and we've learned t=
hings we could have done better=2C and for a pure inter-domain server-to-se=
rver signaling protocol it could be a lot simpler.=20
>=20
> But anyway... with regards to your alternatives below=2C you seem to impl=
y that using DTLS-SRTP is by itself "secure"=2C such that the UI could disp=
lay a lock-icon for example.  I was under the impression that DTLS-SRTP pro=
vided no guarantee of anything without the humans on both ends verifying th=
ey're using the same keying material/SAS/etc.  Is that not the case? (okay =
yes it requires being a MitM to subvert=2C but that's not a high enough bar=
 to claim anything)
>=20
> Assuming that's true=2C and given how few humans would actually know to p=
erform manual verification of dtls-srtp=2C I think this topic isn't as cut-=
and-dried as choosing between "secure" vs. "insecure".  Both alternatives a=
re capable of being secure or insecure=2C and it's up to the human to do so=
mething to figure it out in either case.  There is no real alternative whic=
h is inherently "secure"=2C afaict.
>=20
> -hadriel
>=20
>=20
> Recently EKR wrote:
> > This is very late=2C but here is my attempted summary of the discussion
> > about COMSEC choices at the interim. I'm not saying that this captures
> > everyone's position=2C but the following positions are the ones that in
> > my view have significant levels of support.
> >=20
> >=20
> > Alternative A: [DTLS MUST=2C NO SDES=2C NO RTP]
> >=20
> >    An RTC-Web client MUST support DTLS-SRTP and ONLY DTLS-SRTP for
> >    media=2C without support for either SRTP with supplied keying
> >    material (SDES-style) AND plain RTP. DTLS-SRTP provides for
> >    end-to-end key negotation between the two RTCWEB clients. The
> >    client MUST support the SRTP_AES128_CM_HMAC_SHA1_80 protection
> >    profile and the DTLS cipher suite
> >    TLS_DHE_RSA_WITH_AES_128_CBC_SHA. Note that this requirement
> >    differs from the current TLS default=2C TLS_RSA_WITH_AES_128_CBC_SHA
> >    in that it mandates support for Diffie-Hellman key exchange in
> >    order to provide Perfect Forward Secrecy.
> >=20
> >    An RTCWEB client MUST provide its user with the ability to to see
> >    keying material information sufficient to allow indepent
> >    verification of their peer's identity.  (REF
> >    draft-kaufman-rtcweb-security-ui).
> >=20
> >    The primary drawback of this alternative is the lack of backwards
> >    compatibility with devices and software that only support plain
> >    RTP=2C but the requirement for a handshake makes interoperation with
> >    these devices not completely trivial anyway.
> >=20
> >=20
> >=20
> > Alternative B: [DTLS-SRTP MUST=2C SDES MAY=2C RTP MUST]
> >=20
> >    An RTC-Web client MUST support DTLS-SRTP for media. DTLS-SRTP
> >    provides for end-to-end key negotation between the two RTCWEB
> >    clients.  The client MUST support the SRTP_AES128_CM_HMAC_SHA1_80
> >    protection profile and the DTLS cipher suite
> >    TLS_DHE_RSA_WITH_AES_128_CBC_SHA. Note that this requirement
> >    differs from the current TLS default=2C TLS_RSA_WITH_AES_128_CBC_SHA
> >    in that it mandates support for Diffie-Hellman key exchange in
> >    order to provide Perfect Forward Secrecy. This MUST be the default
> >    mode of operation.
> >=20
> >    An RTCWEB client MAY support SRTP with the keying material supplied
> >    via the signaling channel with the SRTP_AES128_CM_HMAC_SHA1_80
> >    protection profile.  In the case of a web browser client=2C the
> >    keying material should be supplied via a Javascript API.
> >    DTLS-SRTP=2C with its end-to-end keying and authentication capabilit=
y
> >    is preferred over SDES-style [RFC4568] keying.  However=2C the
> >    additional API overhead required to add support for a way to force
> >    a particular key is low.  In addition=2C once plain RTP is to be
> >    supported the arguments against the lower security level provided
> >    by SDES-style keying are no longer relevant.  Also there are a
> >    small number of potential use cases where interoperability with
> >    existing SDES-keyed software or devices may be achieved if the
> >    RTCWEB endpoint supports this mode of keying.
> >=20
> >    An RTCWEB client MUST support RTP.  This provides no privacy but
> >    maximizes interoperability.  Note that SRTP with a Null cipher has
> >    equivalent security but does not meet the interoperability
> >    requirement.  Plain RTP provides no protection for the media=2C and
> >    so is discouraged as a mode of operation for RTCWEB.  However=2C
> >    support for RTP is required in order to provide interoperability
> >    with legacy RTP devices and software that do not support
> >    encryption.  In addition=2C some use cases such as high-volume PSTN
> >    or PBX gateways within an organization may scale more readily
> >    without the overhead of media encryption.
> >=20
> >    An RTCWEB client MUST provide its user with the ability to know
> >    whether or not the media they are sending is protected by encryption
> >    and with the ability to see keying material information sufficient
> >    to allow indepent verification of their peer's identity.
> >    (REF draft-kaufman-rtcweb-security-ui). Note that this user
> >    interface element is much more critical---and hence much more
> >    problematic than with alternative A. If DTLS-SRTP is always
> >    used=2C then the user knows what security mechanisms are provided.
> >    As soon as multiple alternatives with widely varying security
> >    (or no security in the case of RTP) are provided=2C then users
> >    need to actually verify that the security level is satisfactory=2C
> >    which is inherently problematic given typical user behavior.
> >=20
> > I expect to leave time for discussion of this during my slot tomorrow.
> > I look forward to a good discussion.
> >=20
> > Best=2C
> > -Ekr
> >=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
 		 	   		  =

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<body class=3D'hmmessage'><div dir=3D'ltr'>
Agree that the "clean slate" approach is not very appealing.&nbsp=3B "Since=
 we require ICE=2C we can also require X=2C Y=2C Z" is flawed logic=2C not =
only from a deployment/interoperability point of view=2C but also from a re=
gulatory point of view.&nbsp=3B The definitions of interconnected/non-inter=
connected VOIP and the associated regulatory requirements do not have a bro=
wser exemption. <br><br><div>&gt=3B From: HKaplan@acmepacket.com<br>&gt=3B =
To: rtcweb@ietf.org<br>&gt=3B Date: Thu=2C 28 Jul 2011 02:17:42 -0400<br>&g=
t=3B Subject: Re: [rtcweb] Retransmit: Summary of Alternatives for media ke=
ying<br>&gt=3B <br>&gt=3B <br>&gt=3B Well gee that doesn't sound like too h=
ard a choice set to choose from=2C though I guess it depends on whether the=
re's a requirement/desire to interop with existing VoIP devices and the PST=
N=2C or not.<br>&gt=3B <br>&gt=3B If we don't expect/desire RTCWEB clients =
to communicate with existing VoIP devices or PSTN=2C then sure we can start=
 with a clean slate.  In fact=2C you could also create a whole new version =
of SIP for inter-server=2C since there's no need to use SIP v2.0 between RT=
CWEB domains and we've learned things we could have done better=2C and for =
a pure inter-domain server-to-server signaling protocol it could be a lot s=
impler. <br>&gt=3B <br>&gt=3B But anyway... with regards to your alternativ=
es below=2C you seem to imply that using DTLS-SRTP is by itself "secure"=2C=
 such that the UI could display a lock-icon for example.  I was under the i=
mpression that DTLS-SRTP provided no guarantee of anything without the huma=
ns on both ends verifying they're using the same keying material/SAS/etc.  =
Is that not the case? (okay yes it requires being a MitM to subvert=2C but =
that's not a high enough bar to claim anything)<br>&gt=3B <br>&gt=3B Assumi=
ng that's true=2C and given how few humans would actually know to perform m=
anual verification of dtls-srtp=2C I think this topic isn't as cut-and-drie=
d as choosing between "secure" vs. "insecure".  Both alternatives are capab=
le of being secure or insecure=2C and it's up to the human to do something =
to figure it out in either case.  There is no real alternative which is inh=
erently "secure"=2C afaict.<br>&gt=3B <br>&gt=3B -hadriel<br>&gt=3B <br>&gt=
=3B <br>&gt=3B Recently EKR wrote:<br>&gt=3B &gt=3B This is very late=2C bu=
t here is my attempted summary of the discussion<br>&gt=3B &gt=3B about COM=
SEC choices at the interim. I'm not saying that this captures<br>&gt=3B &gt=
=3B everyone's position=2C but the following positions are the ones that in=
<br>&gt=3B &gt=3B my view have significant levels of support.<br>&gt=3B &gt=
=3B <br>&gt=3B &gt=3B <br>&gt=3B &gt=3B Alternative A: [DTLS MUST=2C NO SDE=
S=2C NO RTP]<br>&gt=3B &gt=3B <br>&gt=3B &gt=3B    An RTC-Web client MUST s=
upport DTLS-SRTP and ONLY DTLS-SRTP for<br>&gt=3B &gt=3B    media=2C withou=
t support for either SRTP with supplied keying<br>&gt=3B &gt=3B    material=
 (SDES-style) AND plain RTP. DTLS-SRTP provides for<br>&gt=3B &gt=3B    end=
-to-end key negotation between the two RTCWEB clients. The<br>&gt=3B &gt=3B=
    client MUST support the SRTP_AES128_CM_HMAC_SHA1_80 protection<br>&gt=
=3B &gt=3B    profile and the DTLS cipher suite<br>&gt=3B &gt=3B    TLS_DHE=
_RSA_WITH_AES_128_CBC_SHA. Note that this requirement<br>&gt=3B &gt=3B    d=
iffers from the current TLS default=2C TLS_RSA_WITH_AES_128_CBC_SHA<br>&gt=
=3B &gt=3B    in that it mandates support for Diffie-Hellman key exchange i=
n<br>&gt=3B &gt=3B    order to provide Perfect Forward Secrecy.<br>&gt=3B &=
gt=3B <br>&gt=3B &gt=3B    An RTCWEB client MUST provide its user with the =
ability to to see<br>&gt=3B &gt=3B    keying material information sufficien=
t to allow indepent<br>&gt=3B &gt=3B    verification of their peer's identi=
ty.  (REF<br>&gt=3B &gt=3B    draft-kaufman-rtcweb-security-ui).<br>&gt=3B =
&gt=3B <br>&gt=3B &gt=3B    The primary drawback of this alternative is the=
 lack of backwards<br>&gt=3B &gt=3B    compatibility with devices and softw=
are that only support plain<br>&gt=3B &gt=3B    RTP=2C but the requirement =
for a handshake makes interoperation with<br>&gt=3B &gt=3B    these devices=
 not completely trivial anyway.<br>&gt=3B &gt=3B <br>&gt=3B &gt=3B <br>&gt=
=3B &gt=3B <br>&gt=3B &gt=3B Alternative B: [DTLS-SRTP MUST=2C SDES MAY=2C =
RTP MUST]<br>&gt=3B &gt=3B <br>&gt=3B &gt=3B    An RTC-Web client MUST supp=
ort DTLS-SRTP for media. DTLS-SRTP<br>&gt=3B &gt=3B    provides for end-to-=
end key negotation between the two RTCWEB<br>&gt=3B &gt=3B    clients.  The=
 client MUST support the SRTP_AES128_CM_HMAC_SHA1_80<br>&gt=3B &gt=3B    pr=
otection profile and the DTLS cipher suite<br>&gt=3B &gt=3B    TLS_DHE_RSA_=
WITH_AES_128_CBC_SHA. Note that this requirement<br>&gt=3B &gt=3B    differ=
s from the current TLS default=2C TLS_RSA_WITH_AES_128_CBC_SHA<br>&gt=3B &g=
t=3B    in that it mandates support for Diffie-Hellman key exchange in<br>&=
gt=3B &gt=3B    order to provide Perfect Forward Secrecy. This MUST be the =
default<br>&gt=3B &gt=3B    mode of operation.<br>&gt=3B &gt=3B <br>&gt=3B =
&gt=3B    An RTCWEB client MAY support SRTP with the keying material suppli=
ed<br>&gt=3B &gt=3B    via the signaling channel with the SRTP_AES128_CM_HM=
AC_SHA1_80<br>&gt=3B &gt=3B    protection profile.  In the case of a web br=
owser client=2C the<br>&gt=3B &gt=3B    keying material should be supplied =
via a Javascript API.<br>&gt=3B &gt=3B    DTLS-SRTP=2C with its end-to-end =
keying and authentication capability<br>&gt=3B &gt=3B    is preferred over =
SDES-style [RFC4568] keying.  However=2C the<br>&gt=3B &gt=3B    additional=
 API overhead required to add support for a way to force<br>&gt=3B &gt=3B  =
  a particular key is low.  In addition=2C once plain RTP is to be<br>&gt=
=3B &gt=3B    supported the arguments against the lower security level prov=
ided<br>&gt=3B &gt=3B    by SDES-style keying are no longer relevant.  Also=
 there are a<br>&gt=3B &gt=3B    small number of potential use cases where =
interoperability with<br>&gt=3B &gt=3B    existing SDES-keyed software or d=
evices may be achieved if the<br>&gt=3B &gt=3B    RTCWEB endpoint supports =
this mode of keying.<br>&gt=3B &gt=3B <br>&gt=3B &gt=3B    An RTCWEB client=
 MUST support RTP.  This provides no privacy but<br>&gt=3B &gt=3B    maximi=
zes interoperability.  Note that SRTP with a Null cipher has<br>&gt=3B &gt=
=3B    equivalent security but does not meet the interoperability<br>&gt=3B=
 &gt=3B    requirement.  Plain RTP provides no protection for the media=2C =
and<br>&gt=3B &gt=3B    so is discouraged as a mode of operation for RTCWEB=
.  However=2C<br>&gt=3B &gt=3B    support for RTP is required in order to p=
rovide interoperability<br>&gt=3B &gt=3B    with legacy RTP devices and sof=
tware that do not support<br>&gt=3B &gt=3B    encryption.  In addition=2C s=
ome use cases such as high-volume PSTN<br>&gt=3B &gt=3B    or PBX gateways =
within an organization may scale more readily<br>&gt=3B &gt=3B    without t=
he overhead of media encryption.<br>&gt=3B &gt=3B <br>&gt=3B &gt=3B    An R=
TCWEB client MUST provide its user with the ability to know<br>&gt=3B &gt=
=3B    whether or not the media they are sending is protected by encryption=
<br>&gt=3B &gt=3B    and with the ability to see keying material informatio=
n sufficient<br>&gt=3B &gt=3B    to allow indepent verification of their pe=
er's identity.<br>&gt=3B &gt=3B    (REF draft-kaufman-rtcweb-security-ui). =
Note that this user<br>&gt=3B &gt=3B    interface element is much more crit=
ical---and hence much more<br>&gt=3B &gt=3B    problematic than with altern=
ative A. If DTLS-SRTP is always<br>&gt=3B &gt=3B    used=2C then the user k=
nows what security mechanisms are provided.<br>&gt=3B &gt=3B    As soon as =
multiple alternatives with widely varying security<br>&gt=3B &gt=3B    (or =
no security in the case of RTP) are provided=2C then users<br>&gt=3B &gt=3B=
    need to actually verify that the security level is satisfactory=2C<br>&=
gt=3B &gt=3B    which is inherently problematic given typical user behavior=
.<br>&gt=3B &gt=3B <br>&gt=3B &gt=3B I expect to leave time for discussion =
of this during my slot tomorrow.<br>&gt=3B &gt=3B I look forward to a good =
discussion.<br>&gt=3B &gt=3B <br>&gt=3B &gt=3B Best=2C<br>&gt=3B &gt=3B -Ek=
r<br>&gt=3B &gt=3B <br>&gt=3B <br>&gt=3B __________________________________=
_____________<br>&gt=3B rtcweb mailing list<br>&gt=3B rtcweb@ietf.org<br>&g=
t=3B https://www.ietf.org/mailman/listinfo/rtcweb<br></div> 		 	   		  </di=
v></body>
</html>=

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To: Hadriel Kaplan <HKaplan@acmepacket.com>
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Subject: Re: [rtcweb] Strawman for how to prevent voice-hammer without ICE
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On Jul 28, 2011, at 3:52 AM, Hadriel Kaplan wrote:

> Howdy,
> With regard to mandating ICE, such that an RTCWEB browser cannot send =
RTP without doing ICE successfully first... is that restriction purely =
to prevent the voice-hammer attacks?  If so, then it's unfortunate =
because obviously it would seriously reduce interop with non-RTCWEB VoIP =
devices.

Agree that it limits interoperability.

>  But I think there's a way to prevent the hammer attack without using =
ICE, and without requiring legacy VoIP devices to change whatsoever.
>=20
> One way would be to use the receipt of RTP as an indicator the far-end =
expects to receive it from you.  So have the browser generate RTP/RTCP =
packets, at a relatively slow rate, for a short duration (e.g., use the =
same rate/retransmit timers of STUN connectivity checks in ICE).  If the =
browser receives RTP/RTCP packets, then the far-end expected to receive =
them as well and the browser can do normal RTP from then on.

This is not as safe. There are devices/software that do things upon =
receipt of RTP packets (like play them out through loudspeakers, or =
initiate ringing of the softphone), and more important RTP packets are =
not carefully designed to avoid imitating other types of traffic such as =
SNMP, whereas STUN packets contain a relatively large header with a =
magic number that makes them much less likely to be misinterpreted by =
other protocols.

Additionally the STUN connectivity check used for ICE contains =
short-term credentials and a transaction ID that is unknown to the =
signaling layer (or Javascript) that ensure that you are in fact =
conducting a pairwise negotiation with the far end that you are thinking =
of. With RTP/RTCP packets alone you can create attacks both from the =
browser (by sending RTP/RTCP and using something else to spoof enough =
replies that the test passes) or on the browser (by sending it RTP from =
somewhere else).

And of course ICE, or something like it, is required anyway for doing =
NAT traversal.


>=20
> If this was a hammer attack, this doesn't generate any more load on =
the target than ICE, since ICE would have sent X number of STUN packets =
for Y time as well, and I'm suggesting the X and Y be the same values =
for the initial RTP packets during the "check" phase.

Problem is that it isn't just load. Another class of attack is sending =
packets to things like SNMP servers behind a firewall. No matter how =
slow the rate, if one of the packets causes bad things to happen then =
you have a security problem.

>=20
> The major weakness of this approach is that a malicious web-server =
trying to get your browser to do the voice hammer could send RTP to your =
browser, since it knows the address/ports of both sides, codec payload =
types, etc. (i.e., it can spoof being the attack target to make your =
browser think it's ok to do normal RTP)

Indeed. See above.

>  But we could probably play games with RTCP SR/RR or even just require =
continued RTP receipt to send RTP, in order to mitigate this weakness or =
make it of low value to exploit.

I don't believe this is the case. There's so many cases where one side =
sends A+V but the other side sends audio only, and sending HD-video-rate =
traffic to an unsuspecting endpoint is unacceptable.

This is even worse if we allow arbitrary datagrams with a relatively =
short header, as these will be even easier to craft into attacks on =
other protocols if there isn't confirmed consent.

>=20
> Does anyone else care about interop-ing with legacy non-RTCWEB voip =
devices?  I checked draft-ietf-rtcweb-use-cases-and-requirements-01 and =
I don't see it, so I'm not sure.

I care, but I think that the legacy devices are going to need a way to =
consent. At one of our first meetings prior to forming the WG I came up =
with additional terminology to describe devices which are legacy and =
cannot be upgraded vs. devices which are legacy but which can be =
upgraded sufficiently to add the ICE handshake... we should probably =
bring that back for convenience.

Matthew Kaufman


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On Jul 28, 2011, at 2:17 AM, Hadriel Kaplan wrote:

>=20
> Well gee that doesn't sound like too hard a choice set to choose from, =
though I guess it depends on whether there's a requirement/desire to =
interop with existing VoIP devices and the PSTN, or not.

There is a requirement to interop with existing VoIP devices and the =
PSTN, however other requirements (like ICE for consent to send media) =
will impose additional requirements that limit which existing VoIP =
devices are or can be made compatible.

>=20
> If we don't expect/desire RTCWEB clients to communicate with existing =
VoIP devices or PSTN, then sure we can start with a clean slate.

There is a desire for the media to be transported in RTP for a variety =
of reasons, including the ability to easily terminate the traffic in =
PSTN gateways, reuse of existing RTP stacks for building both browsers =
and interoperable devices, and middlebox awareness of what the traffic =
might be.

>  In fact, you could also create a whole new version of SIP for =
inter-server, since there's no need to use SIP v2.0 between RTCWEB =
domains and we've learned things we could have done better, and for a =
pure inter-domain server-to-server signaling protocol it could be a lot =
simpler.=20
>=20

You could. In fact, for inter-domain federation you are probably free to =
use whatever you and the other domain agree to use.

However, again for software reuse reasons, you're likely to pick SIP.

> But anyway... with regards to your alternatives below, you seem to =
imply that using DTLS-SRTP is by itself "secure", such that the UI could =
display a lock-icon for example.  I was under the impression that =
DTLS-SRTP provided no guarantee of anything without the humans on both =
ends verifying they're using the same keying material/SAS/etc.  Is that =
not the case? (okay yes it requires being a MitM to subvert, but that's =
not a high enough bar to claim anything)

Plain RTP - can be intercepted by both MitM attackers and passive =
observers. No perfect forward secrecy (in fact, no secrecy at all).=20

SRTP w/SDES-style keying - both MitM attackers and passive observers can =
record the traffic but not play it out until they obtain key. The key is =
provided by the server and is likely recorded in server logs. Protection =
of the key on the wire requires HTTPS or similar signaling. Impossible =
to provide perfect forward secrecy.

DTLS-SRTP (or any other end-to-end keying) - assuming the use of DH key =
agreement, both MitM attackers and passive observers can record the =
traffic, however ONLY MitM attackers are able to participate in the key =
agreement algorithm, and so ONLY MitM attackers would ever be able to =
play out the contents. The key agreement occurs between the two =
endpoints (assuming no MitM attacker, of course) and is not recorded =
anywhere else and is not available to the signaling server. Provides =
perfect forward secrecy. MitM attacks be avoided if you trust the =
signaling service and exchange fingerprints between the two ends for =
authentication OR if you use a user interface that lets you examine the =
fingerprints and verify them out-of-band OR if you use additional =
software that you trust (along with a 3rd party that you trust) to =
examine the fingerprints automatically and alert you in the case of =
mismatch.

All of these statements of course only apply between you and whatever is =
terminating the session. If it is a gateway to another protocol, the =
security provided stops at that point and is only continued if the far =
end of the gateway is also secure *and* you trust both that security and =
the gateway.



>=20
> Assuming that's true, and given how few humans would actually know to =
perform manual verification of dtls-srtp, I think this topic isn't as =
cut-and-dried as choosing between "secure" vs. "insecure".  Both =
alternatives are capable of being secure or insecure, and it's up to the =
human to do something to figure it out in either case.  There is no real =
alternative which is inherently "secure", afaict.

There are gradations of security that are provided by default, depending =
on the choices made.

If the browser is able to send plain RTP at all, then you need a way for =
the user to tell if it is sending plain RTP or encrypted RTP. If it can =
only send encrypted RTP, then this requirement goes away. For some =
threat models (for instance passive observers watching your unencrypted =
wifi traffic), any encryption at all is better than plain RTP. For =
others (passive observers that can record the traffic and also have =
future access to the server logs), encryption + end-to-end keying is far =
superior to any of the other choices.

But, like all security arguments, you are correct. It is never black and =
white, so you can never say "this is secure". But you can certainly say =
"this ISN'T secure" for some of them (like plain RTP, or even the =
ability for the browser to switch you from SRTP to RTP without making it =
clear to the user).

Think of the threat models you are likely to experience yourself... then =
tell me which level of protection YOU want.

Matthew Kaufman


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> But anyway... with regards to your alternatives below, you=20
> seem to imply that using DTLS-SRTP is by itself "secure",=20
> such that the UI could display a lock-icon for example.  I=20
> was under the impression that DTLS-SRTP provided no guarantee=20
> of anything without the humans on both ends verifying they're=20
> using the same keying material/SAS/etc.  Is that not the=20
> case? (okay yes it requires being a MitM to subvert, but=20
> that's not a high enough bar to claim anything)
>=20
> Assuming that's true, and given how few humans would actually=20
> know to perform manual verification of dtls-srtp, I think=20
> this topic isn't as cut-and-dried as choosing between=20
> "secure" vs. "insecure".  Both alternatives are capable of=20
> being secure or insecure, and it's up to the human to do=20
> something to figure it out in either case.  There is no real=20
> alternative which is inherently "secure", afaict.
[JRE] I agree with Hadriel's concern. DTLS-SRTP itself, if using only self-=
signed certs at the endpoints, requires other means (signalling path or hum=
an-assisted) to  provide strong guarantees that you are talking to the pers=
on or organization you expect. For SIP, the (theoretical) solution is to se=
nd a fingerprint of the certificate and have that bound to authenticated id=
entity in accordance with RFC 4474, although not feasible in most deployed =
SIP environments. Although RTC-Web is not specifying signalling, clearly it=
 needs to say something about the problem, and can't just indicate to the u=
ser that communication is secure just because DTLS-SRTP is used.

Secondly, if interworking with legacy SIP, where DTLS-SRTP (or whatever we =
select) is terminated at a gateway, there is a real problem ensuring that t=
he user knows the call is not necessarily secure end-to-end. Perhaps this a=
rgues towards option B, since if DTLS-SRTP is not used from browser to gate=
way, at least the browser should know to indicate a lesser level of securit=
y to the user.

The whole issue of the browser knowing what it can tell the user concerning=
 the security posture of a communication is a really difficult problem.

John

John Elwell
Tel: +44 1908 817801 (office and mobile)
Email: john.elwell@siemens-enterprise.com
http://www.siemens-enterprise.com/uk/

Siemens Enterprise Communications Limited.
Registered office: Brickhill Street, Willen Lake, Milton Keynes, MK15 0DJ.
Registered No: 5903714, England.

Siemens Enterprise Communications Limited is a Trademark Licensee of Siemen=
s AG.


>=20
> -hadriel
>=20
>=20
> Recently EKR wrote:
> > This is very late, but here is my attempted summary of the=20
> discussion
> > about COMSEC choices at the interim. I'm not saying that=20
> this captures
> > everyone's position, but the following positions are the=20
> ones that in
> > my view have significant levels of support.
> >=20
> >=20
> > Alternative A: [DTLS MUST, NO SDES, NO RTP]
> >=20
> >    An RTC-Web client MUST support DTLS-SRTP and ONLY DTLS-SRTP for
> >    media, without support for either SRTP with supplied keying
> >    material (SDES-style) AND plain RTP. DTLS-SRTP provides for
> >    end-to-end key negotation between the two RTCWEB clients. The
> >    client MUST support the SRTP_AES128_CM_HMAC_SHA1_80 protection
> >    profile and the DTLS cipher suite
> >    TLS_DHE_RSA_WITH_AES_128_CBC_SHA. Note that this requirement
> >    differs from the current TLS default,=20
> TLS_RSA_WITH_AES_128_CBC_SHA
> >    in that it mandates support for Diffie-Hellman key exchange in
> >    order to provide Perfect Forward Secrecy.
> >=20
> >    An RTCWEB client MUST provide its user with the ability to to see
> >    keying material information sufficient to allow indepent
> >    verification of their peer's identity.  (REF
> >    draft-kaufman-rtcweb-security-ui).
> >=20
> >    The primary drawback of this alternative is the lack of backwards
> >    compatibility with devices and software that only support plain
> >    RTP, but the requirement for a handshake makes=20
> interoperation with
> >    these devices not completely trivial anyway.
> >=20
> >=20
> >=20
> > Alternative B: [DTLS-SRTP MUST, SDES MAY, RTP MUST]
> >=20
> >    An RTC-Web client MUST support DTLS-SRTP for media. DTLS-SRTP
> >    provides for end-to-end key negotation between the two RTCWEB
> >    clients.  The client MUST support the SRTP_AES128_CM_HMAC_SHA1_80
> >    protection profile and the DTLS cipher suite
> >    TLS_DHE_RSA_WITH_AES_128_CBC_SHA. Note that this requirement
> >    differs from the current TLS default,=20
> TLS_RSA_WITH_AES_128_CBC_SHA
> >    in that it mandates support for Diffie-Hellman key exchange in
> >    order to provide Perfect Forward Secrecy. This MUST be=20
> the default
> >    mode of operation.
> >=20
> >    An RTCWEB client MAY support SRTP with the keying=20
> material supplied
> >    via the signaling channel with the SRTP_AES128_CM_HMAC_SHA1_80
> >    protection profile.  In the case of a web browser client, the
> >    keying material should be supplied via a Javascript API.
> >    DTLS-SRTP, with its end-to-end keying and authentication=20
> capability
> >    is preferred over SDES-style [RFC4568] keying.  However, the
> >    additional API overhead required to add support for a=20
> way to force
> >    a particular key is low.  In addition, once plain RTP is to be
> >    supported the arguments against the lower security level provided
> >    by SDES-style keying are no longer relevant.  Also there are a
> >    small number of potential use cases where interoperability with
> >    existing SDES-keyed software or devices may be achieved if the
> >    RTCWEB endpoint supports this mode of keying.
> >=20
> >    An RTCWEB client MUST support RTP.  This provides no privacy but
> >    maximizes interoperability.  Note that SRTP with a Null=20
> cipher has
> >    equivalent security but does not meet the interoperability
> >    requirement.  Plain RTP provides no protection for the media, and
> >    so is discouraged as a mode of operation for RTCWEB.  However,
> >    support for RTP is required in order to provide interoperability
> >    with legacy RTP devices and software that do not support
> >    encryption.  In addition, some use cases such as high-volume PSTN
> >    or PBX gateways within an organization may scale more readily
> >    without the overhead of media encryption.
> >=20
> >    An RTCWEB client MUST provide its user with the ability to know
> >    whether or not the media they are sending is protected=20
> by encryption
> >    and with the ability to see keying material information=20
> sufficient
> >    to allow indepent verification of their peer's identity.
> >    (REF draft-kaufman-rtcweb-security-ui). Note that this user
> >    interface element is much more critical---and hence much more
> >    problematic than with alternative A. If DTLS-SRTP is always
> >    used, then the user knows what security mechanisms are provided.
> >    As soon as multiple alternatives with widely varying security
> >    (or no security in the case of RTP) are provided, then users
> >    need to actually verify that the security level is satisfactory,
> >    which is inherently problematic given typical user behavior.
> >=20
> > I expect to leave time for discussion of this during my=20
> slot tomorrow.
> > I look forward to a good discussion.
> >=20
> > Best,
> > -Ekr
> >=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> =

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Subject: Re: [rtcweb] Strawman for how to prevent voice-hammer without ICE
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=20

> -----Original Message-----
> From: rtcweb-bounces@ietf.org=20
> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Hadriel Kaplan
> Sent: 28 July 2011 08:52
> To: rtcweb@ietf.org
> Subject: [rtcweb] Strawman for how to prevent voice-hammer without ICE
>=20
> Howdy,
> With regard to mandating ICE, such that an RTCWEB browser=20
> cannot send RTP without doing ICE successfully first... is=20
> that restriction purely to prevent the voice-hammer attacks? =20
> If so, then it's unfortunate because obviously it would=20
> seriously reduce interop with non-RTCWEB VoIP devices.  But I=20
> think there's a way to prevent the hammer attack without=20
> using ICE, and without requiring legacy VoIP devices to=20
> change whatsoever.
>=20
> One way would be to use the receipt of RTP as an indicator=20
> the far-end expects to receive it from you.  So have the=20
> browser generate RTP/RTCP packets, at a relatively slow rate,=20
> for a short duration (e.g., use the same rate/retransmit=20
> timers of STUN connectivity checks in ICE).  If the browser=20
> receives RTP/RTCP packets, then the far-end expected to=20
> receive them as well and the browser can do normal RTP from then on.
>=20
> If this was a hammer attack, this doesn't generate any more=20
> load on the target than ICE, since ICE would have sent X=20
> number of STUN packets for Y time as well, and I'm suggesting=20
> the X and Y be the same values for the initial RTP packets=20
> during the "check" phase.
>=20
> The major weakness of this approach is that a malicious=20
> web-server trying to get your browser to do the voice hammer=20
> could send RTP to your browser, since it knows the=20
> address/ports of both sides, codec payload types, etc. (i.e.,=20
> it can spoof being the attack target to make your browser=20
> think it's ok to do normal RTP)  But we could probably play=20
> games with RTCP SR/RR or even just require continued RTP=20
> receipt to send RTP, in order to mitigate this weakness or=20
> make it of low value to exploit.
[JRE] The problem with requiring continued RTP receipt (or even ANY RTP rec=
eipt) is sessions where RTP is one-way, e.g., when mic muted, camera off. W=
ith many asymmetric possibilities coming out of the CLUE work, this could b=
e a problem, although it perhaps goes away if everything is multiplexed ont=
o the same port. It depends whether we can expect all legacy devices to beh=
ave in a way that would make this work.

John

John Elwell
Tel: +44 1908 817801 (office and mobile)
Email: john.elwell@siemens-enterprise.com
http://www.siemens-enterprise.com/uk/

Siemens Enterprise Communications Limited.
Registered office: Brickhill Street, Willen Lake, Milton Keynes, MK15 0DJ.
Registered No: 5903714, England.

Siemens Enterprise Communications Limited is a Trademark Licensee of Siemen=
s AG.

>=20
> Does anyone else care about interop-ing with legacy=20
> non-RTCWEB voip devices?  I checked=20
> draft-ietf-rtcweb-use-cases-and-requirements-01 and I don't=20
> see it, so I'm not sure.
>=20
> -hadriel
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> =

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Subject: Re: [rtcweb] Retransmit: Summary of Alternatives for media keying
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On Jul 28, 2011, at 8:04 AM, Elwell, John wrote:

> [JRE] I agree with Hadriel's concern. DTLS-SRTP itself, if using only =
self-signed certs at the endpoints, requires other means (signalling =
path or human-assisted) to  provide strong guarantees that you are =
talking to the person or organization you expect.

Agreed.

However there are weaker guarantees of properties that are certainly =
favorable as compared to, for instance, plain RTP or even server-keyed =
SRTP. (See my previous email.)

> For SIP, the (theoretical) solution is to send a fingerprint of the =
certificate and have that bound to authenticated identity in accordance =
with RFC 4474, although not feasible in most deployed SIP environments. =
Although RTC-Web is not specifying signalling, clearly it needs to say =
something about the problem, and can't just indicate to the user that =
communication is secure just because DTLS-SRTP is used.

Correct.

>=20
> Secondly, if interworking with legacy SIP, where DTLS-SRTP (or =
whatever we select) is terminated at a gateway, there is a real problem =
ensuring that the user knows the call is not necessarily secure =
end-to-end. Perhaps this argues towards option B, since if DTLS-SRTP is =
not used from browser to gateway, at least the browser should know to =
indicate a lesser level of security to the user.

If you are sitting in an Internet cafe using someone else's unencrypted =
wifi and a telephony service provider you trust would you rather 1) use =
DTLS-SRTP between you and the telephony provider and plain RTP past =
their gateway inside their internal network, or 2) use plain RTP on the =
wifi network and all the way to the telephony provider?

I agree that we can't display "this connection is secured everywhere" to =
the user... but look, HTTPS is the same. When I see a lock icon on my =
screen and click to see that yes, I'm really talking to who it says in =
the address bar, I still can't tell if they're terminating my HTTPS =
session in a front-end box and then using HTTP beyond it, or if the =
HTTPS really goes all the way to the server. And when I use HTTPS to =
talk to Gmail, there's no way to tell if an email I send will be sent =
over a TLS channel or if it'll just be plain SMTP from Gmail to my =
recipient.

>=20
> The whole issue of the browser knowing what it can tell the user =
concerning the security posture of a communication is a really difficult =
problem.

It isn't a difficult problem if what the user is told is confined to =
what is provably true. (See my previous Gmail example.)

Matthew Kaufman


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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: "Elwell, John" <john.elwell@siemens-enterprise.com>
Date: Thu, 28 Jul 2011 10:48:07 -0400
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On Jul 28, 2011, at 8:10 AM, Elwell, John wrote:

> [JRE] The problem with requiring continued RTP receipt (or even ANY RTP r=
eceipt) is sessions where RTP is one-way, e.g., when mic muted, camera off.=
 With many asymmetric possibilities coming out of the CLUE work, this could=
 be a problem, although it perhaps goes away if everything is multiplexed o=
nto the same port. It depends whether we can expect all legacy devices to b=
ehave in a way that would make this work.

Yes, but they don't turn off RTCP (assuming they do RTCP).  I'm not suggest=
ing that it's a 1:1 mechanism of receipt/send, nor that it even be very dyn=
amic/real-time.  But sure there will be some cases that it won't work.  But=
 that's no worse than the current state of things, since it won't work to b=
egin with for such devices if ICE is mandated.  We can't achieve 100% inter=
op regardless - I'm just trying for as much as possible.

With regards to CLUE, I don't consider their devices "legacy".  They'll be =
upgrading to handle CLUE's final spec, so they can be "redeemed".

-hadriel


From HKaplan@acmepacket.com  Thu Jul 28 08:13:56 2011
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Matthew Kaufman <matthew.kaufman@skype.net>
Date: Thu, 28 Jul 2011 11:13:53 -0400
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On Jul 28, 2011, at 8:25 AM, Matthew Kaufman wrote:

>> Secondly, if interworking with legacy SIP, where DTLS-SRTP (or whatever =
we select) is terminated at a gateway, there is a real problem ensuring tha=
t the user knows the call is not necessarily secure end-to-end. Perhaps thi=
s argues towards option B, since if DTLS-SRTP is not used from browser to g=
ateway, at least the browser should know to indicate a lesser level of secu=
rity to the user.
>=20
> If you are sitting in an Internet cafe using someone else's unencrypted w=
ifi and a telephony service provider you trust would you rather 1) use DTLS=
-SRTP between you and the telephony provider and plain RTP past their gatew=
ay inside their internal network, or 2) use plain RTP on the wifi network a=
nd all the way to the telephony provider?

I would be perfectly happy with using sdes-based SRTP.  But if the call wou=
ld otherwise fail altogether, I'd like the option to make the call no matte=
r what (ie, even if it ends up being cleartext).

(And since you're sitting in an Internet cafe in hearing distance of everyo=
ne, any sense of privacy is gone anyway. ;)


> I agree that we can't display "this connection is secured everywhere" to =
the user... but look, HTTPS is the same. When I see a lock icon on my scree=
n and click to see that yes, I'm really talking to who it says in the addre=
ss bar, I still can't tell if they're terminating my HTTPS session in a fro=
nt-end box and then using HTTP beyond it, or if the HTTPS really goes all t=
he way to the server.

But the browser ties the identity of the name I typed in the address bar to=
 the cert, so that if I typed "https://mybank.com", I know it's secure to m=
ybank.com.  Of course it could be cleartext after whomever has a cert for m=
ybank.com, and may even end up triggering protocols that exit mybank.com an=
d go to evil.com, but I implicitly trust mybank.com to be careful with bank=
 info.  That's not the same for DTLS-SRTP; until and unless the human being=
 verifies the keys/SAS with the far end, nothing is known.  The DTLS-SRTP m=
ay only reach the NAT-router in the Internet Cafe, for example, be terminat=
ed by it and a separate DTLS-SRTP between it and the bank.  And if I'm talk=
ing to an IVR system (as one frequently does with banks), there is no human=
 to verify the keys/SAS with.

-hadriel


From john.elwell@siemens-enterprise.com  Thu Jul 28 08:14:46 2011
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: Hadriel Kaplan <HKaplan@acmepacket.com>
Date: Thu, 28 Jul 2011 17:14:42 +0200
Thread-Topic: Strawman for how to prevent voice-hammer without ICE
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=20

> -----Original Message-----
> From: Hadriel Kaplan [mailto:HKaplan@acmepacket.com]=20
> Sent: 28 July 2011 15:48
> To: Elwell, John
> Cc: rtcweb@ietf.org
> Subject: Re: Strawman for how to prevent voice-hammer without ICE
>=20
>=20
> On Jul 28, 2011, at 8:10 AM, Elwell, John wrote:
>=20
> > [JRE] The problem with requiring continued RTP receipt (or=20
> even ANY RTP receipt) is sessions where RTP is one-way, e.g.,=20
> when mic muted, camera off. With many asymmetric=20
> possibilities coming out of the CLUE work, this could be a=20
> problem, although it perhaps goes away if everything is=20
> multiplexed onto the same port. It depends whether we can=20
> expect all legacy devices to behave in a way that would make=20
> this work.
>=20
> Yes, but they don't turn off RTCP (assuming they do RTCP). =20
> I'm not suggesting that it's a 1:1 mechanism of receipt/send,=20
> nor that it even be very dynamic/real-time.  But sure there=20
> will be some cases that it won't work.  But that's no worse=20
> than the current state of things, since it won't work to=20
> begin with for such devices if ICE is mandated.  We can't=20
> achieve 100% interop regardless - I'm just trying for as much=20
> as possible.
[JRE] This assumes RTP-RTCP multiplexing, which not many current devices su=
pport. There are so many things that RTC-Web is proposing to use that are n=
ot widely supported on existing devices that some sort of gateway looks ine=
vitable. Eliminating one particular instance of incompatibility might reduc=
e the amount of adaptation the gateway needs to perform, but it looks incre=
asingly unlikely that all adaptation can be eliminated. So is there any rea=
l value in trying to eliminate one aspect if others can't be eliminated? Pr=
obably yes, but not if it introduces other compromises, e.g., security. I t=
hink it is worth exploring this further.

John

>=20
> With regards to CLUE, I don't consider their devices=20
> "legacy".  They'll be upgrading to handle CLUE's final spec,=20
> so they can be "redeemed".
>=20
> -hadriel
>=20
> =

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Thread-Topic: Strawman for how to prevent voice-hammer without ICE
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On Jul 28, 2011, at 11:14 AM, Elwell, John wrote:

> [JRE] This assumes RTP-RTCP multiplexing, which not many current devices =
support.

No, I didn't mean to assume multiplexing.  Sorry if it was confusing.  I me=
ant if you're worried about using RTP reception to allow transmission becau=
se in some cases RTP isn't sent from the far end for periods of time, then =
one could use the reception of RTCP on the odd port to allow the sending of=
 RTP on the even. =20

I was already assuming multiplexing would not likely be used with legacy vo=
ip devices, since none of them do it today.

-hadriel



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To: Matthew Kaufman <matthew.kaufman@skype.net>
Date: Thu, 28 Jul 2011 11:58:15 -0400
Thread-Topic: [rtcweb] Strawman for how to prevent voice-hammer without ICE
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On Jul 28, 2011, at 7:38 AM, Matthew Kaufman wrote:

>=20
>> But I think there's a way to prevent the hammer attack without using ICE=
, and without requiring legacy VoIP devices to change whatsoever.
>>=20
>> One way would be to use the receipt of RTP as an indicator the far-end e=
xpects to receive it from you.  So have the browser generate RTP/RTCP packe=
ts, at a relatively slow rate, for a short duration (e.g., use the same rat=
e/retransmit timers of STUN connectivity checks in ICE).  If the browser re=
ceives RTP/RTCP packets, then the far-end expected to receive them as well =
and the browser can do normal RTP from then on.
>=20
> This is not as safe. There are devices/software that do things upon recei=
pt of RTP packets (like play them out through loudspeakers, or initiate rin=
ging of the softphone), and more important RTP packets are not carefully de=
signed to avoid imitating other types of traffic such as SNMP, whereas STUN=
 packets contain a relatively large header with a magic number that makes t=
hem much less likely to be misinterpreted by other protocols.

No it's not as "safe", but it may be safe enough. =20
With regards to being misinterpreted by others, since the web server javasc=
ript cannot define the actual payload of the RTP (correct?), and things lik=
e sequence numbers would also be randomly chosen by the browser, so it's no=
t like the javascript can purposefully craft an RTP packet to immitate othe=
r protocols.  For example it would be an astronmoical coincidence for any g=
iven RTP packet to be parsable as SNMP BER for a given ASN.1 dictionary.
(Also as an aside, if I recall correctly SNMP is actually distinguishable f=
rom RTP by the first byte)


> Additionally the STUN connectivity check used for ICE contains short-term=
 credentials and a transaction ID that is unknown to the signaling layer (o=
r Javascript) that ensure that you are in fact conducting a pairwise negoti=
ation with the far end that you are thinking of. With RTP/RTCP packets alon=
e you can create attacks both from the browser (by sending RTP/RTCP and usi=
ng something else to spoof enough replies that the test passes) or on the b=
rowser (by sending it RTP from somewhere else).

If we used RTCP as part of the check, the reports contain a last received s=
equence number which would not be known to the javascript.  (it's a hack to=
 be sure, but if we want to handle non-RTCWEB devices, a hack's better than=
 nothing)


> And of course ICE, or something like it, is required anyway for doing NAT=
 traversal.

Not when communicating with legacy voip devices and the PSTN.


> I don't believe this is the case. There's so many cases where one side se=
nds A+V but the other side sends audio only, and sending HD-video-rate traf=
fic to an unsuspecting endpoint is unacceptable.

I should probably write it up in a draft to be much more clear, but to the =
above point that's not an issue.  I'm not suggesting the browser be allowed=
 to do this RTP-based verification for any and all use-cases.  I'm only sug=
gesting it to handle the large population of legacy VoIP devices, including=
 to the PSTN.  Video without audio, or any future data-type streams, would =
not apply and we don't need to handle them this way and can restrict the br=
owser from allowing it.

-hadriel


From jdrosen@jdrosen.net  Thu Jul 28 09:58:07 2011
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Subject: Re: [rtcweb] Strawman for how to prevent voice-hammer without ICE
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Let me suggest a variation on this..

The rtcweb client can send RTP packet, voice-only, for a brief period of 
time (say, 2x the RTCP interval). It waits to receive an RTCP packet. 
The RTCP packet should have an RR which reflects back the SSRC sent by 
the client, if it does, the rtcweb client continues. If not, it stops 
sending.

The purpose of the RTCP SSRC check is to emulate what the STUN 
transaction ID provides; that there is something on the media path which 
is expecting this RTP packet. Not as good as STUN which also has the 
short term credential, but its something.

Thanks,
Jonathan R.

On 7/28/11 3:52 AM, Hadriel Kaplan wrote:
> Howdy,
> With regard to mandating ICE, such that an RTCWEB browser cannot send RTP without doing ICE successfully first... is that restriction purely to prevent the voice-hammer attacks?  If so, then it's unfortunate because obviously it would seriously reduce interop with non-RTCWEB VoIP devices.  But I think there's a way to prevent the hammer attack without using ICE, and without requiring legacy VoIP devices to change whatsoever.
>
> One way would be to use the receipt of RTP as an indicator the far-end expects to receive it from you.  So have the browser generate RTP/RTCP packets, at a relatively slow rate, for a short duration (e.g., use the same rate/retransmit timers of STUN connectivity checks in ICE).  If the browser receives RTP/RTCP packets, then the far-end expected to receive them as well and the browser can do normal RTP from then on.
>
> If this was a hammer attack, this doesn't generate any more load on the target than ICE, since ICE would have sent X number of STUN packets for Y time as well, and I'm suggesting the X and Y be the same values for the initial RTP packets during the "check" phase.
>
> The major weakness of this approach is that a malicious web-server trying to get your browser to do the voice hammer could send RTP to your browser, since it knows the address/ports of both sides, codec payload types, etc. (i.e., it can spoof being the attack target to make your browser think it's ok to do normal RTP)  But we could probably play games with RTCP SR/RR or even just require continued RTP receipt to send RTP, in order to mitigate this weakness or make it of low value to exploit.
>
> Does anyone else care about interop-ing with legacy non-RTCWEB voip devices?  I checked draft-ietf-rtcweb-use-cases-and-requirements-01 and I don't see it, so I'm not sure.
>
> -hadriel
>
> _______________________________________________
> 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

From HKaplan@acmepacket.com  Thu Jul 28 10:07:40 2011
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To: Jonathan Rosenberg <jdrosen@jdrosen.net>
Date: Thu, 28 Jul 2011 13:07:36 -0400
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On Jul 28, 2011, at 12:58 PM, Jonathan Rosenberg wrote:

> Let me suggest a variation on this..
>=20
> The rtcweb client can send RTP packet, voice-only, for a brief period of=
=20
> time (say, 2x the RTCP interval). It waits to receive an RTCP packet.=20
> The RTCP packet should have an RR which reflects back the SSRC sent by=20
> the client, if it does, the rtcweb client continues. If not, it stops=20
> sending.
>=20
> The purpose of the RTCP SSRC check is to emulate what the STUN=20
> transaction ID provides; that there is something on the media path which=
=20
> is expecting this RTP packet. Not as good as STUN which also has the=20
> short term credential, but its something.
>=20

I had considered using the SSRC before, but I think you're gonna have to le=
t the javascript know the SSRC to handle some SIP/SDP usages (since there's=
 an SSRC attribute for SDP, I assume someone needs it).  No?

-hadriel


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On Jul 28, 2011, at 11:13 AM, Hadriel Kaplan wrote:

>=20
> On Jul 28, 2011, at 8:25 AM, Matthew Kaufman wrote:
>=20
>>> Secondly, if interworking with legacy SIP, where DTLS-SRTP (or =
whatever we select) is terminated at a gateway, there is a real problem =
ensuring that the user knows the call is not necessarily secure =
end-to-end. Perhaps this argues towards option B, since if DTLS-SRTP is =
not used from browser to gateway, at least the browser should know to =
indicate a lesser level of security to the user.
>>=20
>> If you are sitting in an Internet cafe using someone else's =
unencrypted wifi and a telephony service provider you trust would you =
rather 1) use DTLS-SRTP between you and the telephony provider and plain =
RTP past their gateway inside their internal network, or 2) use plain =
RTP on the wifi network and all the way to the telephony provider?
>=20
> I would be perfectly happy with using sdes-based SRTP.  But if the =
call would otherwise fail altogether, I'd like the option to make the =
call no matter what (ie, even if it ends up being cleartext).

Why would the call "otherwise fail altogether"?

>=20
> (And since you're sitting in an Internet cafe in hearing distance of =
everyone, any sense of privacy is gone anyway. ;)
>=20

Have you ever called into a conference call from a public place, with =
your microphone muted?

>=20
>> I agree that we can't display "this connection is secured everywhere" =
to the user... but look, HTTPS is the same. When I see a lock icon on my =
screen and click to see that yes, I'm really talking to who it says in =
the address bar, I still can't tell if they're terminating my HTTPS =
session in a front-end box and then using HTTP beyond it, or if the =
HTTPS really goes all the way to the server.
>=20
> But the browser ties the identity of the name I typed in the address =
bar to the cert, so that if I typed "https://mybank.com", I know it's =
secure to mybank.com.  Of course it could be cleartext after whomever =
has a cert for mybank.com, and may even end up triggering protocols that =
exit mybank.com and go to evil.com, but I implicitly trust mybank.com to =
be careful with bank info. =20

Ok, so you trust your "banking service".

> That's not the same for DTLS-SRTP; until and unless the human being =
verifies the keys/SAS with the far end, nothing is known.

If you trust your "calling service" (and are connected to it with HTTPS) =
then you can use that trusted service to exchange the fingerprints and =
verify the keys of the far end to prevent MITM.

(You can also add a browser extension that uses a service that you trust =
but that isn't your calling service, and when you're calling other =
people with the same extension you can check the fingerprints in an =
automated way using that.)

If you *don't* trust your "calling service", then that's just like not =
trusting your banking service when you type in passwords.

>  The DTLS-SRTP may only reach the NAT-router in the Internet Cafe, for =
example, be terminated by it and a separate DTLS-SRTP between it and the =
bank.  And if I'm talking to an IVR system (as one frequently does with =
banks), there is no human to verify the keys/SAS with.

Correct. But there is a signaling channel, and if you trust it you can =
assure that there is no router doing MITM at the cafe. Perhaps you'll =
even be calling your bank's IVR using *their* calling service...

Matthew Kaufman


From harald@alvestrand.no  Thu Jul 28 10:22:11 2011
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Question that I could probably answer if I understood the DH key 
exchange thing:

Is it possible for anyone with packet-replacement access to the media 
path to perform a MITM attack against DH?

If so: Is it possible to deliver some token by the "high path" (where 
the SDES keys would go) that ensures that the DH key exchange is with 
someone possessing that token?

That would limit MITM attacks to attackers who had access to both the 
"high path" and the media path.

                                Harald


From ekr@rtfm.com  Thu Jul 28 10:28:17 2011
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That's effectively what DTLS-SRTP does.

-Ekr


On Thu, Jul 28, 2011 at 1:22 PM, Harald Alvestrand <harald@alvestrand.no> w=
rote:
> Question that I could probably answer if I understood the DH key exchange
> thing:
>
> Is it possible for anyone with packet-replacement access to the media pat=
h
> to perform a MITM attack against DH?
>
> If so: Is it possible to deliver some token by the "high path" (where the
> SDES keys would go) that ensures that the DH key exchange is with someone
> possessing that token?
>
> That would limit MITM attacks to attackers who had access to both the "hi=
gh
> path" and the media path.
>
> =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 Harald
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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On Jul 28, 2011, at 11:58 AM, Hadriel Kaplan wrote:

>=20
> On Jul 28, 2011, at 7:38 AM, Matthew Kaufman wrote:
>=20
>>=20
>> This is not as safe. There are devices/software that do things upon =
receipt of RTP packets (like play them out through loudspeakers, or =
initiate ringing of the softphone), and more important RTP packets are =
not carefully designed to avoid imitating other types of traffic such as =
SNMP, whereas STUN packets contain a relatively large header with a =
magic number that makes them much less likely to be misinterpreted by =
other protocols.
>=20
> No it's not as "safe", but it may be safe enough. =20

I disagree. You need both 1) a large number of bits of transaction ID =
(on the order of the 128 bits provided by STUN), and 2) a shared secret =
passed out of band between the two ends (for example, the short term =
credentials used by STUN connectivity checks).

Failing to provide both of the above opens possibilities for attacks =
that I believe will not be acceptable to browser vendors due to the =
relative danger of exposing this capability "behind the firewall."

> With regards to being misinterpreted by others, since the web server =
javascript cannot define the actual payload of the RTP (correct?)

If datagrams are also allowed, a remarkable amount of the data being =
sent can be chosen from the Javascript. The datagram proposals assume =
that ICE is used to get consent ahead of time.

> , and things like sequence numbers would also be randomly chosen by =
the browser, so it's not like the javascript can purposefully craft an =
RTP packet to immitate other protocols.  For example it would be an =
astronmoical coincidence for any given RTP packet to be parsable as SNMP =
BER for a given ASN.1 dictionary.
> (Also as an aside, if I recall correctly SNMP is actually =
distinguishable from RTP by the first byte)
>=20
>=20
>> Additionally the STUN connectivity check used for ICE contains =
short-term credentials and a transaction ID that is unknown to the =
signaling layer (or Javascript) that ensure that you are in fact =
conducting a pairwise negotiation with the far end that you are thinking =
of. With RTP/RTCP packets alone you can create attacks both from the =
browser (by sending RTP/RTCP and using something else to spoof enough =
replies that the test passes) or on the browser (by sending it RTP from =
somewhere else).
>=20
> If we used RTCP as part of the check, the reports contain a last =
received sequence number which would not be known to the javascript.  =
(it's a hack to be sure, but if we want to handle non-RTCWEB devices, a =
hack's better than nothing)

I don't believe this is enough bits, and it lacks the shared secret =
mechanism.

>=20
>=20
>> And of course ICE, or something like it, is required anyway for doing =
NAT traversal.
>=20
> Not when communicating with legacy voip devices and the PSTN.

Correct, but my point is that the ICE implementation must exist.

>=20
>=20
>> I don't believe this is the case. There's so many cases where one =
side sends A+V but the other side sends audio only, and sending =
HD-video-rate traffic to an unsuspecting endpoint is unacceptable.
>=20
> I should probably write it up in a draft to be much more clear, but to =
the above point that's not an issue.  I'm not suggesting the browser be =
allowed to do this RTP-based verification for any and all use-cases.  =
I'm only suggesting it to handle the large population of legacy VoIP =
devices, including to the PSTN.  Video without audio, or any future =
data-type streams, would not apply and we don't need to handle them this =
way and can restrict the browser from allowing it.

Again, I don't think it provides adequate protection to the other =
devices on the network behind the firewall that the browser is on. But =
do please write your proposal up in more detail.

Matthew Kaufman


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Subject: Re: [rtcweb] Strawman for how to prevent voice-hammer without ICE
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On Jul 28, 2011, at 12:58 PM, Jonathan Rosenberg wrote:

> Let me suggest a variation on this..
>=20
> The rtcweb client can send RTP packet, voice-only, for a brief period =
of time (say, 2x the RTCP interval). It waits to receive an RTCP packet. =
The RTCP packet should have an RR which reflects back the SSRC sent by =
the client, if it does, the rtcweb client continues. If not, it stops =
sending.
>=20
> The purpose of the RTCP SSRC check is to emulate what the STUN =
transaction ID provides; that there is something on the media path which =
is expecting this RTP packet. Not as good as STUN which also has the =
short term credential, but its something.

Lacks both the short term credential and is not enough bits to protect =
against attackers who can generate a flurry of forged RTCP packets. See =
my previous email.

Matthew Kaufman


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On Jul 28, 2011, at 1:22 PM, Harald Alvestrand wrote:

> Question that I could probably answer if I understood the DH key =
exchange thing:
>=20
> Is it possible for anyone with packet-replacement access to the media =
path to perform a MITM attack against DH?

Yes.

>=20
> If so: Is it possible to deliver some token by the "high path" (where =
the SDES keys would go) that ensures that the DH key exchange is with =
someone possessing that token?

You can do that, or you can check the fingerprints via the high path to =
authenticate each end and prove that there's no MITM on the media path.

>=20
> That would limit MITM attacks to attackers who had access to both the =
"high path" and the media path.
>=20

Correct. And the high path can be protected with HTTPS and the =
associated PKI.

Matthew Kaufman


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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.

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Hi,
In today's session, someone characterized VP8 as "royalty free".  On this
subject, some of you may find the linked blog post (and the article linked
therein) interesting.  It was posted today, during our session.  To
summarize, DOJ probe or not, MPEG-LA continues its efforts to form an
(presumably royalty-bearing) patent pool for VP8, they had initial pool
member meeting sometime in June, and talk about 12 potential member
companies.
http://fosspatents.blogspot.com/2011/07/googles-webm-vp8-allegedly-infringes
.html
Regards,
Stephan




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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; "><div style=3D"color: rgb(0, 0, =
0); font-family: Calibri, sans-serif; font-size: 14px; ">Hi,</div><div style=
=3D"color: rgb(0, 0, 0); font-family: Calibri, sans-serif; font-size: 14px; ">=
In today's session, someone characterized VP8 as "royalty free". &nbsp;On th=
is subject, some of you may find the linked blog post (and the article linke=
d therein) interesting. &nbsp;It was posted today, during our session. &nbsp=
;To summarize, DOJ probe or not, MPEG-LA continues its efforts to form an (p=
resumably royalty-bearing) patent pool for VP8, they had initial pool member=
 meeting sometime in June, and talk about 12 potential member companies.</di=
v><div><font class=3D"Apple-style-span" face=3D"Calibri,sans-serif">http://fossp=
atents.blogspot.com/2011/07/googles-webm-vp8-allegedly-infringes.html</font>=
</div><div style=3D"color: rgb(0, 0, 0); font-family: Calibri, sans-serif; fon=
t-size: 14px; ">Regards,</div><div style=3D"color: rgb(0, 0, 0); font-family: =
Calibri, sans-serif; font-size: 14px; ">Stephan</div><div style=3D"color: rgb(=
0, 0, 0); font-family: Calibri, sans-serif; font-size: 14px; "><br></div></b=
ody></html>

--B_3394713240_1517471--



From harald@alvestrand.no  Thu Jul 28 13:49:49 2011
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Subject: Re: [rtcweb] Strawman for how to prevent voice-hammer without ICE
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On 07/28/11 03:52, Hadriel Kaplan wrote:
> Howdy,
> With regard to mandating ICE, such that an RTCWEB browser cannot send RTP without doing ICE successfully first... is that restriction purely to prevent the voice-hammer attacks?  If so, then it's unfortunate because obviously it would seriously reduce interop with non-RTCWEB VoIP devices.  But I think there's a way to prevent the hammer attack without using ICE, and without requiring legacy VoIP devices to change whatsoever.
>
> One way would be to use the receipt of RTP as an indicator the far-end expects to receive it from you.  So have the browser generate RTP/RTCP packets, at a relatively slow rate, for a short duration (e.g., use the same rate/retransmit timers of STUN connectivity checks in ICE).  If the browser receives RTP/RTCP packets, then the far-end expected to receive them as well and the browser can do normal RTP from then on.
>
> If this was a hammer attack, this doesn't generate any more load on the target than ICE, since ICE would have sent X number of STUN packets for Y time as well, and I'm suggesting the X and Y be the same values for the initial RTP packets during the "check" phase.
>
> The major weakness of this approach is that a malicious web-server trying to get your browser to do the voice hammer could send RTP to your browser, since it knows the address/ports of both sides, codec payload types, etc. (i.e., it can spoof being the attack target to make your browser think it's ok to do normal RTP)  But we could probably play games with RTCP SR/RR or even just require continued RTP receipt to send RTP, in order to mitigate this weakness or make it of low value to exploit.
I think this approach is not paranoid enough.

The attacker will negotiate a channel claiming that you can reach him on 
10.0.0.2 (your server that he wants to voice-hammer), and then send you 
the five or so RTP packets you expect with a fake source address of 
10.0.0.2.

Then you, having seen exactly the packets that "authorize" sending 
traffic to 10.0.0.2, will be performing the voice-hammer attack against 
the server that the attacker otherwise couldn't reach.

You would have to send RTP packets yourself (which will correctly be 
dropped on the floor by 10.0.0.2) until the time at which you can start 
wondering about there being no RTCP packets from 10.0.0.2 in order to 
have a 2-way handshake - and that only works if the RTCP RR contains 
enough stuff the attacker can't predict that he can't just generate the 
RTCP too.

(This doesn't work with ICE, because the ICE handshake involves the 
recipient replying to your packet with some parameters that can only be 
found in the request, not in the negotiation).
> Does anyone else care about interop-ing with legacy non-RTCWEB voip devices?  I checked draft-ietf-rtcweb-use-cases-and-requirements-01 and I don't see it, so I'm not sure.
>
> -hadriel
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


From randell1@jesup.org  Thu Jul 28 14:56:10 2011
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On 7/28/2011 1:18 PM, Matthew Kaufman wrote:
>   we can't display "this connection is secured everywhere" to the user... but look, HTTPS is the same. When I see a lock icon on my screen and click to see that yes, I'm really talking to who it says in the address bar, I still can't tell if they're terminating my HTTPS session in a front-end box and then using HTTP beyond it, or if the HTTPS really goes all the way to the server.
>> But the browser ties the identity of the name I typed in the address bar to the cert, so that if I typed "https://mybank.com", I know it's secure to mybank.com.  Of course it could be cleartext after whomever has a cert for mybank.com, and may even end up triggering protocols that exit mybank.com and go to evil.com, but I implicitly trust mybank.com to be careful with bank info.
> Ok, so you trust your "banking service".

And more to the point, you trust the chain of certificates incorporated 
into the browser to verify it's actually the entity that owns mybank.com 
(and EV certs come into play here too).  Overall, that's a "good" chain 
of trust, though no such chain is perfect.

>> That's not the same for DTLS-SRTP; until and unless the human being verifies the keys/SAS with the far end, nothing is known.
> If you trust your "calling service" (and are connected to it with HTTPS) then you can use that trusted service to exchange the fingerprints and verify the keys of the far end to prevent MITM.

Right, this is trusting that not only they are who they say they are, 
but also trusting that they haven't been compromised.   And one may not 
have the same level of trust in that that you have (have to have) in 
your bank.

> (You can also add a browser extension that uses a service that you trust but that isn't your calling service, and when you're calling other people with the same extension you can check the fingerprints in an automated way using that.)

Definitely possible, and adds another largely-independent layer of trust.

> If you *don't* trust your "calling service", then that's just like not trusting your banking service when you type in passwords.

I may trust my calling service to connect me (to someone), but I may not 
trust them to not MITM me themselves (read CALEA or other national 
security services), and I may not trust them not to be compromised, 
since they have a far smaller monetary stake in avoiding it compared to 
a bank.

Another idea related to identity: as mentioned, the cert trust chain in 
a browser verifies that you're talking to mybank.com or myprovider.com.  
This can be leveraged to provide authentication by cert that you're 
talking to someone who's part of mybank.com, even if you went through 
myprovider.com to get there and myprovider.com might be compromised.  It 
doesn't directly help verify you're talking to Joe Your Neighbor, but 
other services can help with that (as mentioned - extensions, etc).

Effectively, for a fairly large subset of the net, we have a deployed 
PKI of a form.  It's not perfect (DNSSEC anyone; recent todo with bogus 
certs because a CA snafu) but for most uses that it covers it's pretty 
good.  Getting it to cover identifying the destination as a dependent of 
the primary cert may be an issue for many companies, however.

In any case, there may be a way to leverage this existing trust chain, 
both to authenticate who's calling you and who you're calling, for a 
subset of possible destinations.

-- 
Randell Jesup
randell-ietf@jesup.org


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On 7/28/2011 5:54 PM, Randell Jesup wrote:
>> (You can also add a browser extension that uses a service that you 
>> trust but that isn't your calling service, and when you're calling 
>> other people with the same extension you can check the fingerprints 
>> in an automated way using that.)
>
> Definitely possible, and adds another largely-independent layer of trust.
>

Replying to myself with something the Mozilla team was thinking about 
(we may come
back with more on this later, after talking to the people here about 
it):  an example of an
independent layer of identity trust would be the just-announced 
BrowserID initiative:

https://browserid.org/

The plan is to move this into the browser itself.

Also see 
http://identity.mozilla.com/post/7616727542/introducing-browserid-a-better-way-to-sign-in

Perhaps the browserid data could be used to facilitate authentication of 
SAS data, for example.

-- 
Randell Jesup
randell-ietf@jesup.org


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From: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
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Hi John,

I am interested in this use case and would be interested in seeing it
added.

Cheers,
Charles

> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
Behalf Of Elwell, John
> Sent: Tuesday, July 26, 2011 4:14 PM
> To: rtcweb@ietf.org
> Subject: [rtcweb] Possible new use case
>=20
> (Not sure whether this should go to the W3C list,  but trying the IETF
list to start with)
>=20
> I raised this during the WEBRTC session on Saturday.
>=20
> Does anyone have any interest in adding a use case concerning
recording of real-time media at a remote
> recorder? I noticed something on local recording in the proposed APIs,
but remote recording might have
> other impacts, e.g.,
> - taking a stream from a capture device (mic, camera) and sending it
not only to the remote browser
> but also to a remote recorder;
> - taking a stream from the remote browser and sending it to a remote
recorder (as well as rendering
> locally);
> - possibly mixing the two streams (in case of audio) and sending as a
single mixed stream to the
> recording device.
>=20
> John
>=20
>=20
> John Elwell
> Tel: +44 1908 817801 (office and mobile)
> Email: john.elwell@siemens-enterprise.com
> http://www.siemens-enterprise.com/uk/
>=20
> Siemens Enterprise Communications Limited.
> Registered office: Brickhill Street, Willen Lake, Milton Keynes, MK15
0DJ.
> Registered No: 5903714, England.
>=20
> Siemens Enterprise Communications Limited is a Trademark Licensee of
Siemens AG.
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Matthew Kaufman <matthew.kaufman@skype.net>
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On Jul 28, 2011, at 1:32 PM, Matthew Kaufman wrote:

>=20
> On Jul 28, 2011, at 1:22 PM, Harald Alvestrand wrote:
>=20
>> Question that I could probably answer if I understood the DH key exchang=
e thing:
>>=20
>> Is it possible for anyone with packet-replacement access to the media pa=
th to perform a MITM attack against DH?
>=20
> Yes.
>=20
>>=20
>> If so: Is it possible to deliver some token by the "high path" (where th=
e SDES keys would go) that ensures that the DH key exchange is with someone=
 possessing that token?
>=20
> You can do that, or you can check the fingerprints via the high path to a=
uthenticate each end and prove that there's no MITM on the media path.

That's assuming the high path itself is secure end-to-end.  Since this grou=
p has already agreed to allow HTTP rather than mandate HTTPS, and since it =
can go from your web service to SIP and beyond, such a guarantee is hard to=
 enforce.

-hadriel



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Date: Thu, 28 Jul 2011 18:44:40 -0400
Thread-Topic: [rtcweb] Retransmit: Summary of Alternatives for media keying
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Subject: Re: [rtcweb] Retransmit: Summary of Alternatives for media keying
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On Jul 28, 2011, at 1:18 PM, Matthew Kaufman wrote:

>=20
> On Jul 28, 2011, at 11:13 AM, Hadriel Kaplan wrote:
>=20
>> I would be perfectly happy with using sdes-based SRTP.  But if the call =
would otherwise fail altogether, I'd like the option to make the call no ma=
tter what (ie, even if it ends up being cleartext).
>=20
> Why would the call "otherwise fail altogether"?

Because I'm calling someone who has a legacy VoIP device or is on the PSTN,=
 and they're not going to support DTLS-SRTP.
Of course we could require the RTCWEB service to deploy "gateways" in order=
 to terminate DTLS-SRTP and do SDES-based SRTP or cleartext RTP to non-RTCW=
EB, but that's expensive and complex.  Not to mention its mis-leading - the=
 media is not secure end-to-end, and again the lock-icon model won't work.


> Have you ever called into a conference call from a public place, with you=
r microphone muted?

Nah, I talk too much.  ;)

-hadriel
p.s. Of course I'm speaking as an individual - my employer would probably b=
e ecstatic with requiring such gateways.=20


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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Harald Alvestrand <harald@alvestrand.no>
Date: Thu, 28 Jul 2011 18:55:46 -0400
Thread-Topic: [rtcweb] Strawman for how to prevent voice-hammer without ICE
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Subject: Re: [rtcweb] Strawman for how to prevent voice-hammer without ICE
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On Jul 28, 2011, at 4:49 PM, Harald Alvestrand wrote:

> I think this approach is not paranoid enough.
>=20
> The attacker will negotiate a channel claiming that you can reach him on=
=20
> 10.0.0.2 (your server that he wants to voice-hammer), and then send you=20
> the five or so RTP packets you expect with a fake source address of=20
> 10.0.0.2.
>=20
> Then you, having seen exactly the packets that "authorize" sending=20
> traffic to 10.0.0.2, will be performing the voice-hammer attack against=20
> the server that the attacker otherwise couldn't reach.

Yes, that was the weakness of the model as I described in my original email=
: that the malicious web-server can spoof the RTP from the device being att=
acked.
And I proposed that instead of it being a single "authorization" phase at t=
he beginning of the call, it could even be continuous/periodic.  Of course =
the web-server could continuously send spoofed RTP to the browser, but at t=
hat point it might as well do the attack directly spoofing the browser. (in=
 other words, the malicious website isn't gaining a very useful botnet of a=
ttackers, since I assume that's the concern)


> (This doesn't work with ICE, because the ICE handshake involves the=20
> recipient replying to your packet with some parameters that can only be=20
> found in the request, not in the negotiation).

Yes I know ICE solves this, but since ICE doesn't exist for legacy VoIP and=
 PSTN, a non-existent solution isn't very useful.

-hadriel


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Date: Thu, 28 Jul 2011 23:01:38 +0000
From: "Roy Yue(PeiYu)" <yuepeiyu@huawei.com>
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From stewe@stewe.org  Thu Jul 28 16:08:37 2011
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Date: Thu, 28 Jul 2011 19:08:24 -0400
From: Stephan Wenger <stewe@stewe.org>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
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Subject: Re: [rtcweb] VP8 royalty free?
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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.

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I was told that the blog has disappeared.
The cached version is still available:
http://webcache.googleusercontent.com/search?q=cache:IEfA8rZjUWsJ:fosspatent
s.blogspot.com/+fosspatents&cd=1&hl=en&ct=clnk&source=www.google.com
There's also the cited article:
http://www.streamingmedia.com/Articles/News/Featured-News/WebM-Patent-Fight-
Ahead-for-Google-76781.aspx
Stephan

From:  Stephan Wenger <stewe@stewe.org>
Date:  Thu, 28 Jul 2011 15:53:54 -0400
To:  "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject:  VP8 royalty free?

Hi,
In today's session, someone characterized VP8 as "royalty free".  On this
subject, some of you may find the linked blog post (and the article linked
therein) interesting.  It was posted today, during our session.  To
summarize, DOJ probe or not, MPEG-LA continues its efforts to form an
(presumably royalty-bearing) patent pool for VP8, they had initial pool
member meeting sometime in June, and talk about 12 potential member
companies.
http://fosspatents.blogspot.com/2011/07/googles-webm-vp8-allegedly-infringes
.html
Regards,
Stephan




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spatents.blogspot.com/+fosspatents&amp;cd=3D1&amp;hl=3Den&amp;ct=3Dclnk&amp;source=
=3Dwww.google.com">http://webcache.googleusercontent.com/search?q=3Dcache:IEfA8r=
ZjUWsJ:fosspatents.blogspot.com/+fosspatents&amp;cd=3D1&amp;hl=3Den&amp;ct=3Dclnk&=
amp;source=3Dwww.google.com</a></div><div>There's also the cited article:&nbsp=
;<a href=3D"http://www.streamingmedia.com/Articles/News/Featured-News/WebM-Pat=
ent-Fight-Ahead-for-Google-76781.aspx">http://www.streamingmedia.com/Article=
s/News/Featured-News/WebM-Patent-Fight-Ahead-for-Google-76781.aspx</a></div>=
<div>Stephan</div><div><br></div><span id=3D"OLK_SRC_BODY_SECTION"><div style=3D=
"font-family:Calibri; font-size:11pt; text-align:left; color:black; BORDER-B=
OTTOM: medium none; BORDER-LEFT: medium none; PADDING-BOTTOM: 0in; PADDING-L=
EFT: 0in; PADDING-RIGHT: 0in; BORDER-TOP: #b5c4df 1pt solid; BORDER-RIGHT: m=
edium none; PADDING-TOP: 3pt"><span style=3D"font-weight:bold">From: </span> S=
tephan Wenger &lt;<a href=3D"mailto:stewe@stewe.org">stewe@stewe.org</a>&gt;<b=
r><span style=3D"font-weight:bold">Date: </span> Thu, 28 Jul 2011 15:53:54 -04=
00<br><span style=3D"font-weight:bold">To: </span> "<a href=3D"mailto:rtcweb@iet=
f.org">rtcweb@ietf.org</a>" &lt;<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf=
.org</a>&gt;<br><span style=3D"font-weight:bold">Subject: </span> VP8 royalty =
free?<br></div><div><br></div><div><div style=3D"word-wrap: break-word; -webki=
t-nbsp-mode: space; -webkit-line-break: after-white-space; "><div style=3D"col=
or: rgb(0, 0, 0); font-family: Calibri, sans-serif; font-size: 14px; ">Hi,</=
div><div style=3D"color: rgb(0, 0, 0); font-family: Calibri, sans-serif; font-=
size: 14px; ">In today's session, someone characterized VP8 as "royalty free=
". &nbsp;On this subject, some of you may find the linked blog post (and the=
 article linked therein) interesting. &nbsp;It was posted today, during our =
session. &nbsp;To summarize, DOJ probe or not, MPEG-LA continues its efforts=
 to form an (presumably royalty-bearing) patent pool for VP8, they had initi=
al pool member meeting sometime in June, and talk about 12 potential member =
companies.</div><div><font class=3D"Apple-style-span" face=3D"Calibri,sans-serif=
"><a href=3D"http://fosspatents.blogspot.com/2011/07/googles-webm-vp8-allegedl=
y-infringes.html">http://fosspatents.blogspot.com/2011/07/googles-webm-vp8-a=
llegedly-infringes.html</a></font></div><div style=3D"color: rgb(0, 0, 0); fon=
t-family: Calibri, sans-serif; font-size: 14px; ">Regards,</div><div style=3D"=
color: rgb(0, 0, 0); font-family: Calibri, sans-serif; font-size: 14px; ">St=
ephan</div><div style=3D"color: rgb(0, 0, 0); font-family: Calibri, sans-serif=
; font-size: 14px; "><br></div></div></div></span></body></html>

--B_3394724910_2191743--



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From: =?ISO-8859-1?Q?=22Martin_J=2E_D=FCrst=22?= <duerst@it.aoyama.ac.jp>
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I get:

"Blog has been removed

Sorry, the blog at fosspatents.blogspot.com has been removed. This 
address is not available for new blogs."

(and same for http://fosspatents.blogspot.com/). Do you have a better 
address?

Regards,   Martin.

On 2011/07/29 4:53, Stephan Wenger wrote:
> Hi,
> In today's session, someone characterized VP8 as "royalty free".  On this
> subject, some of you may find the linked blog post (and the article linked
> therein) interesting.  It was posted today, during our session.  To
> summarize, DOJ probe or not, MPEG-LA continues its efforts to form an
> (presumably royalty-bearing) patent pool for VP8, they had initial pool
> member meeting sometime in June, and talk about 12 potential member
> companies.
> http://fosspatents.blogspot.com/2011/07/googles-webm-vp8-allegedly-infringes
> .html
> Regards,
> Stephan
>
>
>
>
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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I recommend taking blog postings and other press from this particular
person (Florian Mueller) with  (more than a pinch) of salt. 

His batting average on both knowledge of the law and prior art has been
very poor and he repeatedly has been portraying himself as an expert at
such matters.  (If you watch Groklaw, you'll know what I mean).
                    - Jim


On 07/28/2011 08:42 PM, "Martin J. Drst" wrote:
> I get:
>
> "Blog has been removed
>
> Sorry, the blog at fosspatents.blogspot.com has been removed. This
> address is not available for new blogs."
>
> (and same for http://fosspatents.blogspot.com/). Do you have a better
> address?
>
> Regards,   Martin.
>
> On 2011/07/29 4:53, Stephan Wenger wrote:
>> Hi,
>> In today's session, someone characterized VP8 as "royalty free".  On
>> this
>> subject, some of you may find the linked blog post (and the article
>> linked
>> therein) interesting.  It was posted today, during our session.  To
>> summarize, DOJ probe or not, MPEG-LA continues its efforts to form an
>> (presumably royalty-bearing) patent pool for VP8, they had initial pool
>> member meeting sometime in June, and talk about 12 potential member
>> companies.
>> http://fosspatents.blogspot.com/2011/07/googles-webm-vp8-allegedly-infringes
>>
>> .html
>> Regards,
>> Stephan
>>
>>
>>
>>
>>
>>
>>
>> _______________________________________________
>> 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  Thu Jul 28 20:12:46 2011
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From: Stephan Wenger <stewe@stewe.org>
To: Jim Gettys <jg@freedesktop.org>, "Martin J. =?ISO-8859-1?B?RPxyc3Q=?=" <duerst@it.aoyama.ac.jp>
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I will refrain from expressing an opinion of Mr. Mueller's views (as I
would when it comes to Groklaw), but please note that the blog post did
contain a link to another document from a different source that contains
quotes of Larry Horn, MPEG-LA's CEO.
I posted that link in a follow up to this list.
Stephan


On 7.28.2011 22:40 , "Jim Gettys" <jg@freedesktop.org> wrote:

>I recommend taking blog postings and other press from this particular
>person (Florian Mueller) with  (more than a pinch) of salt.
>
>His batting average on both knowledge of the law and prior art has been
>very poor and he repeatedly has been portraying himself as an expert at
>such matters.  (If you watch Groklaw, you'll know what I mean).
>                    - Jim
>
>
>On 07/28/2011 08:42 PM, "Martin J. D=FCrst" wrote:
>> I get:
>>
>> "Blog has been removed
>>
>> Sorry, the blog at fosspatents.blogspot.com has been removed. This
>> address is not available for new blogs."
>>
>> (and same for http://fosspatents.blogspot.com/). Do you have a better
>> address?
>>
>> Regards,   Martin.
>>
>> On 2011/07/29 4:53, Stephan Wenger wrote:
>>> Hi,
>>> In today's session, someone characterized VP8 as "royalty free".  On
>>> this
>>> subject, some of you may find the linked blog post (and the article
>>> linked
>>> therein) interesting.  It was posted today, during our session.  To
>>> summarize, DOJ probe or not, MPEG-LA continues its efforts to form an
>>> (presumably royalty-bearing) patent pool for VP8, they had initial pool
>>> member meeting sometime in June, and talk about 12 potential member
>>> companies.
>>>=20
>>>http://fosspatents.blogspot.com/2011/07/googles-webm-vp8-allegedly-infri
>>>nges
>>>
>>> .html
>>> Regards,
>>> Stephan
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> 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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On Jul 28, 2011, at 5:54 PM, Randell Jesup wrote:

> On 7/28/2011 1:18 PM, Matthew Kaufman wrote:
>>  we can't display "this connection is secured everywhere" to the =
user... but look, HTTPS is the same. When I see a lock icon on my screen =
and click to see that yes, I'm really talking to who it says in the =
address bar, I still can't tell if they're terminating my HTTPS session =
in a front-end box and then using HTTP beyond it, or if the HTTPS really =
goes all the way to the server.
>>> But the browser ties the identity of the name I typed in the address =
bar to the cert, so that if I typed "https://mybank.com", I know it's =
secure to mybank.com.  Of course it could be cleartext after whomever =
has a cert for mybank.com, and may even end up triggering protocols that =
exit mybank.com and go to evil.com, but I implicitly trust mybank.com to =
be careful with bank info.
>> Ok, so you trust your "banking service".
>=20
> And more to the point, you trust the chain of certificates =
incorporated into the browser to verify it's actually the entity that =
owns mybank.com (and EV certs come into play here too).  Overall, that's =
a "good" chain of trust, though no such chain is perfect.

It appears to be "good enough" for people to do banking using their web =
browser, which is pretty amazing if you think about it.

>=20
>>> That's not the same for DTLS-SRTP; until and unless the human being =
verifies the keys/SAS with the far end, nothing is known.
>> If you trust your "calling service" (and are connected to it with =
HTTPS) then you can use that trusted service to exchange the =
fingerprints and verify the keys of the far end to prevent MITM.
>=20
> Right, this is trusting that not only they are who they say they are, =
but also trusting that they haven't been compromised.   And one may not =
have the same level of trust in that that you have (have to have) in =
your bank.

I'm not sure how this is any different from any other type of calling =
scenario.

You always need to trust the person at the far end to not reveal the =
call, and trust the equipment they (and you) are using.

>> (You can also add a browser extension that uses a service that you =
trust but that isn't your calling service, and when you're calling other =
people with the same extension you can check the fingerprints in an =
automated way using that.)
>=20
> Definitely possible, and adds another largely-independent layer of =
trust.

Yes, but independent layers of trust are a feature, not a bug.

And what are you arguing here? My claim is that DTLS-SRTP is often =
*more* secure than SDES-style keying, and certainly no worse.

This is one of the things that backs up my claim. I still haven't heard =
why SDES or plain RTP would be superior from a security point of view.

>=20
>> If you *don't* trust your "calling service", then that's just like =
not trusting your banking service when you type in passwords.
>=20
> I may trust my calling service to connect me (to someone), but I may =
not trust them to not MITM me themselves (read CALEA or other national =
security services), and I may not trust them not to be compromised, =
since they have a far smaller monetary stake in avoiding it compared to =
a bank.

Sure, but again all of these things are EASIER if they use SDES. They =
can MITM you themselves by letting anyone get a copy of the traffic from =
anywhere and sending that person the keys, even after the fact. (Both =
passive attacks like this and getting the key later are prevented with =
DTLS-SRTP and DH key agreement).

>=20
> Another idea related to identity: as mentioned, the cert trust chain =
in a browser verifies that you're talking to mybank.com or =
myprovider.com.  This can be leveraged to provide authentication by cert =
that you're talking to someone who's part of mybank.com, even if you =
went through myprovider.com to get there and myprovider.com might be =
compromised.  It doesn't directly help verify you're talking to Joe Your =
Neighbor, but other services can help with that (as mentioned - =
extensions, etc).

Yes, all good ideas, but built upon DTLS-SRTP, and not SDES or plain =
RTP.

>=20
> Effectively, for a fairly large subset of the net, we have a deployed =
PKI of a form.  It's not perfect (DNSSEC anyone; recent todo with bogus =
certs because a CA snafu) but for most uses that it covers it's pretty =
good.  Getting it to cover identifying the destination as a dependent of =
the primary cert may be an issue for many companies, however.

I doubt it, but perhaps.

>=20
> In any case, there may be a way to leverage this existing trust chain, =
both to authenticate who's calling you and who you're calling, for a =
subset of possible destinations.

Almost certainly.

Matthew Kaufman


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From: Matthew Kaufman <matthew.kaufman@skype.net>
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On Jul 28, 2011, at 6:33 PM, Hadriel Kaplan wrote:

>=20
> On Jul 28, 2011, at 1:32 PM, Matthew Kaufman wrote:
>>=20
>> You can do that, or you can check the fingerprints via the high path =
to authenticate each end and prove that there's no MITM on the media =
path.
>=20
> That's assuming the high path itself is secure end-to-end.  Since this =
group has already agreed to allow HTTP rather than mandate HTTPS, and =
since it can go from your web service to SIP and beyond, such a =
guarantee is hard to enforce.


I don't believe we had complete consensus to allow HTTP rather than =
HTTPS... and of course you are free to not use calling services that =
aren't using HTTPS, just as you're free to not use web-based email =
providers or web-based social networking sites that don't require the =
use of HTTPS.

The guarantee of what happens beyond the service is difficult. The =
guarantee of what happens within the service goes as far as you can =
trust the service.

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 Jul 28, 2011, at 6:33 PM, Hadriel Kaplan =
wrote:</div><br class=3D"Apple-interchange-newline"><blockquote =
type=3D"cite"><div><br>On Jul 28, 2011, at 1:32 PM, Matthew Kaufman =
wrote:<br><blockquote type=3D"cite"><font class=3D"Apple-style-span" =
color=3D"#144FAE"><br></font></blockquote><blockquote type=3D"cite">You =
can do that, or you can check the fingerprints via the high path to =
authenticate each end and prove that there's no MITM on the media =
path.<br></blockquote><br>That's assuming the high path itself is secure =
end-to-end. &nbsp;Since this group has already agreed to allow HTTP =
rather than mandate HTTPS, and since it can go from your web service to =
SIP and beyond, such a guarantee is hard to =
enforce.<br></div></blockquote></div><br><div><br></div><div>I don't =
believe we had complete consensus to allow HTTP rather than HTTPS... and =
of course you are free to not use calling services that aren't using =
HTTPS, just as you're free to not use web-based email providers or =
web-based social networking sites that don't require the use of =
HTTPS.</div><div><br></div><div>The guarantee of what happens beyond the =
service is difficult. The guarantee of what happens within the service =
goes as far as you can trust the =
service.</div><div><br></div><div>Matthew =
Kaufman</div><div><br></div></body></html>=

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On Jul 28, 2011, at 6:44 PM, Hadriel Kaplan wrote:

>=20
> On Jul 28, 2011, at 1:18 PM, Matthew Kaufman wrote:
>=20
>>=20
>> On Jul 28, 2011, at 11:13 AM, Hadriel Kaplan wrote:
>>=20
>>> I would be perfectly happy with using sdes-based SRTP.  But if the =
call would otherwise fail altogether, I'd like the option to make the =
call no matter what (ie, even if it ends up being cleartext).
>>=20
>> Why would the call "otherwise fail altogether"?
>=20
> Because I'm calling someone who has a legacy VoIP device or is on the =
PSTN, and they're not going to support DTLS-SRTP.

I'm confused... do you want maximum security (because DTLS-SRTP is =
probably the best path to that of the choices) or maximum interoperation =
with legacy (because plain RTP is probably the easiest way to get that)

> Of course we could require the RTCWEB service to deploy "gateways" in =
order to terminate DTLS-SRTP and do SDES-based SRTP or cleartext RTP to =
non-RTCWEB, but that's expensive and complex.

Expensive? Maybe. Complex? Not really.

>  Not to mention its mis-leading - the media is not secure end-to-end, =
and again the lock-icon model won't work.

The lock icon model works just as well as the lock icon works on Gmail =
to indicate that your interaction with Gmail is protected but the email =
you send might be going via plain SMTP.

It might even be better than this, if we do a good job of sending =
requirements to W3C.

>=20
>=20
>> Have you ever called into a conference call from a public place, with =
your microphone muted?
>=20
> Nah, I talk too much.  ;)

Even so, you'd probably rather not be recorded by a 3rd party using the =
microphone in your headset if you can avoid it.

Matthew Kaufman


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Dear all,=0A=0Athe full recording (synchronized video, audio, slides and =
jabber room) =0Aof yesterday's RTCWEB session is available.=0A=0AYou can =
watch it by either clicking the proper link on the remote participation p=
age (http://www.ietf.org/meeting/81/remote-participation.html#Meetecho), =
or by directly accessing the following URL:=0Ahttp://www.meetecho.com/iet=
f81/recordings=0A=0AFor the chair(s): please feel free to put the link to=
 the recording in the minutes, if you think this might be useful.=0A=0AIn=
 case of problems with the playout, just drop an e-mail to =0Aietf-suppor=
t@meetecho.com.=0A=0ACheers,=0Athe Meetecho Team


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--_b3df2268-24ac-4fe7-b703-d9d5ea885170_
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[BA] Agree that it's worth exploring further=2C if only to document what th=
e security requirements are=2C and what is required to meet them.  The prob=
lem with the gateway argument is that it can be extended too far -- it is o=
ne thing to require a gateway for signaling=2C and quite another to require=
 transcoding of media=2C for example.  As long as we have legacy interop us=
e cases=2C and aren't asserting that RTCWEB represents a "singularity"=2C t=
hen we need to be able to handle them.=20


> [JRE] This assumes RTP-RTCP multiplexing=2C which not many current device=
s support. There are so many things that RTC-Web is proposing to use that a=
re not widely supported on existing devices that some sort of gateway looks=
 inevitable. Eliminating one particular instance of incompatibility might r=
educe the amount of adaptation the gateway needs to perform=2C but it looks=
 increasingly unlikely that all adaptation can be eliminated. So is there a=
ny real value in trying to eliminate one aspect if others can't be eliminat=
ed? Probably yes=2C but not if it introduces other compromises=2C e.g.=2C s=
ecurity. I think it is worth exploring this further.

 		 	   		  =

--_b3df2268-24ac-4fe7-b703-d9d5ea885170_
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<html>
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</head>
<body class=3D'hmmessage'><div dir=3D'ltr'>
[BA] Agree that it's worth exploring further=2C if only to document what th=
e security requirements are=2C and what is required to meet them.&nbsp=3B T=
he problem with the gateway argument is that it can be extended too far -- =
it is one thing to require a gateway for signaling=2C and quite another to =
require transcoding of media=2C for example.&nbsp=3B As long as we have leg=
acy interop use cases=2C and aren't asserting that RTCWEB represents a "sin=
gularity"=2C then we need to be able to handle them. <br><div><br><br>&gt=
=3B [JRE] This assumes RTP-RTCP multiplexing=2C which not many current devi=
ces support. There are so many things that RTC-Web is proposing to use that=
 are not widely supported on existing devices that some sort of gateway loo=
ks inevitable. Eliminating one particular instance of incompatibility might=
 reduce the amount of adaptation the gateway needs to perform=2C but it loo=
ks increasingly unlikely that all adaptation can be eliminated. So is there=
 any real value in trying to eliminate one aspect if others can't be elimin=
ated? Probably yes=2C but not if it introduces other compromises=2C e.g.=2C=
 security. I think it is worth exploring this further.<br><br></div> 		 	  =
 		  </div></body>
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--_b3df2268-24ac-4fe7-b703-d9d5ea885170_--

From randell1@jesup.org  Fri Jul 29 06:01:10 2011
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On 7/29/2011 1:12 AM, Matthew Kaufman wrote:

>>> (You can also add a browser extension that uses a service that you trust but that isn't your calling service, and when you're calling other people with the same extension you can check the fingerprints in an automated way using that.)
>> Definitely possible, and adds another largely-independent layer of trust.
> Yes, but independent layers of trust are a feature, not a bug.
>
> And what are you arguing here? My claim is that DTLS-SRTP is often *more* secure than SDES-style keying, and certainly no worse.

No disagreement at all.   I'm discussing that DTLS-SRTP is vulnerable 
(to a degree) to MITM
attacks, which is well-known, if you don't have a known-secure signaling 
channel.  I'm not making the
argument that Hadriel is.

>>> If you *don't* trust your "calling service", then that's just like not trusting your banking service when you type in passwords.
>> I may trust my calling service to connect me (to someone), but I may not trust them to not MITM me themselves (read CALEA or other national security services), and I may not trust them not to be compromised, since they have a far smaller monetary stake in avoiding it compared to a bank.
> Sure, but again all of these things are EASIER if they use SDES. They can MITM you themselves by letting anyone get a copy of the traffic from anywhere and sending that person the keys, even after the fact. (Both passive attacks like this and getting the key later are prevented with DTLS-SRTP and DH key agreement).

Absolutely.

> Another idea related to identity: as mentioned, the cert trust chain in a browser verifies that you're talking to mybank.com or myprovider.com.  This can be leveraged to provide authentication by cert that you're talking to someone who's part of mybank.com, even if you went through myprovider.com to get there and myprovider.com might be compromised.  It doesn't directly help verify you're talking to Joe Your Neighbor, but other services can help with that (as mentioned - extensions, etc).
> Yes, all good ideas, but built upon DTLS-SRTP, and not SDES or plain RTP.

Yes, that's the idea.

>> Effectively, for a fairly large subset of the net, we have a deployed PKI of a form.  It's not perfect (DNSSEC anyone; recent todo with bogus certs because a CA snafu) but for most uses that it covers it's pretty good.  Getting it to cover identifying the destination as a dependent of the primary cert may be an issue for many companies, however.
> I doubt it, but perhaps.

My concern was getting companies to deploy subsidiary certs to the 
browsers of call center agents, etc
(especially as these functions are often out-sourced).

>> In any case, there may be a way to leverage this existing trust chain, both to authenticate who's calling you and who you're calling, for a subset of possible destinations.
> Almost certainly.

Good!

-- 
Randell Jesup
randell-ietf@jesup.org


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On Jul 29, 2011, at 8:59 AM, Randell Jesup wrote:

> No disagreement at all.   I'm discussing that DTLS-SRTP is vulnerable=20
> (to a degree) to MITM
> attacks, which is well-known, if you don't have a known-secure signaling=
=20
> channel.  I'm not making the
> argument that Hadriel is.
>=20

But that *is* the argument I was making. (or at least trying to :) =20
I certainly wasn't claiming SDES nor RTP are as secure as DTLS-SRTP could b=
e.

What I said to start this whole thing was: the two alternatives should not =
be described as choosing between secure and insecure.  BOTH alternatives re=
quire the user to verify something for the call to be secure.  BOTH alterna=
tives have the potential to be very secure.  That is all.

BTW, I am assuming of course that even if we choose the alternative of DTLS=
+SDES+RTP, that DTLS would always be preferred, and if the peer cannot do i=
t then SDES, and if the peer can't do that then RTP. (assuming the human ha=
s set whatever browser knobs are necessary to enable/disable this stuff)
So between two RTCWEB browsers it would always be DTLS-SRTP.

-hadriel


From matthew.kaufman@skype.net  Fri Jul 29 07:30:40 2011
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On Jul 29, 2011, at 10:17 AM, Hadriel Kaplan wrote:

>=20
> On Jul 29, 2011, at 8:59 AM, Randell Jesup wrote:
>=20
>> No disagreement at all.   I'm discussing that DTLS-SRTP is vulnerable=20=

>> (to a degree) to MITM
>> attacks, which is well-known, if you don't have a known-secure =
signaling=20
>> channel.  I'm not making the
>> argument that Hadriel is.
>>=20
>=20
> But that *is* the argument I was making. (or at least trying to :) =20
> I certainly wasn't claiming SDES nor RTP are as secure as DTLS-SRTP =
could be.

We agree here.

>=20
> What I said to start this whole thing was: the two alternatives should =
not be described as choosing between secure and insecure.  BOTH =
alternatives require the user to verify something for the call to be =
secure.  BOTH alternatives have the potential to be very secure.  That =
is all.

I suppose that depends on what you mean by "the potential to be very =
secure"... certainly plain RTP *might* be secure, if it never leaves =
your desktop.

DTLS-SRTP + a security inspection UI gives the user a way to verify =
something *independently of the service provider* to see if the call is =
secure. SDES does not. The only way to be sure the service provider =
isn't colluding with someone to tap your call when you use SDES is to =
agree with the other party to mangle the key you receive from the =
service provider. (The "pessimist" approach to trust in this =
environment.)

>=20
> BTW, I am assuming of course that even if we choose the alternative of =
DTLS+SDES+RTP, that DTLS would always be preferred, and if the peer =
cannot do it then SDES, and if the peer can't do that then RTP. =
(assuming the human has set whatever browser knobs are necessary to =
enable/disable this stuff)

Agree. Though the human needs to be able to see a lot more about what is =
going on if we allow all three.

> So between two RTCWEB browsers it would always be DTLS-SRTP.

Agree.

Matthew Kaufman


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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Matthew Kaufman <matthew.kaufman@skype.net>
Date: Fri, 29 Jul 2011 10:52:53 -0400
Thread-Topic: [rtcweb] Retransmit: Summary of Alternatives for media keying
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>, Randell Jesup <randell1@jesup.org>
Subject: Re: [rtcweb] Retransmit: Summary of Alternatives for media keying
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On Jul 29, 2011, at 10:30 AM, Matthew Kaufman wrote:

>> What I said to start this whole thing was: the two alternatives should n=
ot be described as choosing between secure and insecure.  BOTH alternatives=
 require the user to verify something for the call to be secure.  BOTH alte=
rnatives have the potential to be very secure.  That is all.
>=20
> I suppose that depends on what you mean by "the potential to be very secu=
re"... certainly plain RTP *might* be secure, if it never leaves your deskt=
op.

Yeah I should have clarified.  I meant since the alternative of DTLS+SDES+R=
TP includes DTLS-SRTP, and the human can verify it, and I assume DTLS-SRTP =
would always be preferred... then it has the potential to be very secure. (=
namely if DTLS-SRTP is used and the human verifies it)

In other words, both Alternative A and B have the ability to use DTLS-SRTP =
in the same way.  The only difference is Alternative A (DTLS only) would fa=
il if trying to talk to someone who can't do DTLS-SRTP.  Alternative B woul=
d still succeed in letting people talk.

-hadriel


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Subject: Re: [rtcweb] Retransmit: Summary of Alternatives for media keying
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On 7/29/2011 10:30 AM, Matthew Kaufman wrote:

> What I said to start this whole thing was: the two alternatives should not be described as choosing between secure and insecure.  BOTH alternatives require the user to verify something for the call to be secure.  BOTH alternatives have the potential to be very secure.  That is all.
> I suppose that depends on what you mean by "the potential to be very secure"... certainly plain RTP *might* be secure, if it never leaves your desktop.
>
> DTLS-SRTP + a security inspection UI gives the user a way to verify something *independently of the service provider* to see if the call is secure. SDES does not. The only way to be sure the service provider isn't colluding with someone to tap your call when you use SDES is to agree with the other party to mangle the key you receive from the service provider. (The "pessimist" approach to trust in this environment.)
This is where there's an opening for browser extensions, potentially (mangling
the agreed-to-by-protocol keys, such as by using a pre-shared secret) - though that
would require some level of external interaction with the keying from JS/etc code.
That may be worth exposing to the W3/JS layer, perhaps.  (Should the JS layer have access
to the keys?  Be able to modify them (per above)?  Security implications?  What if it
wants to implement an "answering-machine" function via recording packets for later
replay?  Or will it have  access to decoded packets,  or will it only have access
to decoded media and have to re-encode it for "answering-machine" use?)

If the user and the person they're talking to has the option of using an extension
(or the app) to modify the keys at both ends, that allows for perfect(*) security via
pre-shared secrets of whatever form.

* Nothing is perfect


>> BTW, I am assuming of course that even if we choose the alternative of DTLS+SDES+RTP, that DTLS would always be preferred, and if the peer cannot do it then SDES, and if the peer can't do that then RTP. (assuming the human has set whatever browser knobs are necessary to enable/disable this stuff)
> Agree. Though the human needs to be able to see a lot more about what is going on if we allow all three.

Mostly that the user be allowed to see it, and that if one does put up 
any sort of Lock icon it would
only be for the "most" secure.  (Though this is a security-UI issue, not 
a technical one.)  You could
even not put up anything about the security level unless the protocol 
detected an attack or unless
the user's 3rd-party  verification tool agreed it was secure by some 
independent channel.

>> So between two RTCWEB browsers it would always be DTLS-SRTP.
> Agree.
Ditto

-- 
Randell Jesup
randell-ietf@jesup.org


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Subject: [rtcweb] Possible New Cases - Emergency Services Call & Text
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This was mentioned is the initial RTCWeb session at IETF 81 but just to make=
 sure it has been captured and understood.

(1) Please can a separate use case be added for an Emergency Services teleph=
ony call (eg 999 in UK, 112 in EU, 911 in NA, etc).=20

(2) Please can a further separate use case be added for an Emergency Service=
s real time text (eg 18000 in UK, via TextDirect operator to ES operator for=
 hearing impaired persons). FYI today this is a low speed in-band voice band=
 data call on the text leg. Presumably the replacement would be some kind of=
 Presence/Messaging replacement with assured delivery or similar.

Please note these are treated differently today in current networks and I wo=
uld suggest we aim to preserve the capabilities - subject to discussion in t=
he community, in the right WG - eg ECRIT. These are, using legacy terminolog=
y ...

A). Prioritisation in processing of Emergency Services calls above Ordinary C=
alls, particularly when server in overload.

B). Marking of Emergency Services calls to a specific A-/Calling Party Categ=
ory to allow them to be identified downstream.

C). Release control modified to B-/Called Party Control with changed RTCWeb b=
rowser client behaviour along with indication as to "Off Hook" and "On Hook"=
.

Paul=

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Scratching my head why you did not even mention the SOS URL,
http://www.ietf.org/rfc/rfc5031.txt

> Please note these are treated differently today in current networks and I
> would suggest we aim to preserve the capabilities - subject to discussion in
> the community, in the right WG - eg ECRIT. These are, using legacy terminology
> ...

Subject to discussion in yet another WG?

Henry


On 7/30/11 6:03 AM, "Paul Beaumont" <paulbeaumont.ietf@gmail.com> wrote:

> This was mentioned is the initial RTCWeb session at IETF 81 but just to make
> sure it has been captured and understood.
> 
> (1) Please can a separate use case be added for an Emergency Services
> telephony call (eg 999 in UK, 112 in EU, 911 in NA, etc).
> 
> (2) Please can a further separate use case be added for an Emergency Services
> real time text (eg 18000 in UK, via TextDirect operator to ES operator for
> hearing impaired persons). FYI today this is a low speed in-band voice band
> data call on the text leg. Presumably the replacement would be some kind of
> Presence/Messaging replacement with assured delivery or similar.
> 
> Please note these are treated differently today in current networks and I
> would suggest we aim to preserve the capabilities - subject to discussion in
> the community, in the right WG - eg ECRIT. These are, using legacy terminology
> ...
> 
> A). Prioritisation in processing of Emergency Services calls above Ordinary
> Calls, particularly when server in overload.
> 
> B). Marking of Emergency Services calls to a specific A-/Calling Party
> Category to allow them to be identified downstream.
> 
> C). Release control modified to B-/Called Party Control with changed RTCWeb
> browser client behaviour along with indication as to "Off Hook" and "On Hook".
> 
> Paul
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb



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Date: Sun, 31 Jul 2011 08:31:40 -0400
From: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] Possible New Cases - Emergency Services Call & Text
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On 07/30/11 07:03, Paul Beaumont wrote:
> This was mentioned is the initial RTCWeb session at IETF 81 but just to make sure it has been captured and understood.
>
> (1) Please can a separate use case be added for an Emergency Services telephony call (eg 999 in UK, 112 in EU, 911 in NA, etc).
I am skeptical to adding this use case. The most important issues and 
complications that ES brings to the table are with call setup, which I 
believe is outside of what RTCWEB should be focusing on. As long as 
there is nothing special about the media plane issues here, I would 
prefer not to take it on.
> (2) Please can a further separate use case be added for an Emergency Services real time text (eg 18000 in UK, via TextDirect operator to ES operator for hearing impaired persons). FYI today this is a low speed in-band voice band data call on the text leg. Presumably the replacement would be some kind of Presence/Messaging replacement with assured delivery or similar.
>
> Please note these are treated differently today in current networks and I would suggest we aim to preserve the capabilities - subject to discussion in the community, in the right WG - eg ECRIT. These are, using legacy terminology ...
>
> A). Prioritisation in processing of Emergency Services calls above Ordinary Calls, particularly when server in overload.
Are you talking about prioritization in the control plane (out of scope) 
in the media plane, or both? Note that the media plane doesn't 
necessarily have any servers in it.
> B). Marking of Emergency Services calls to a specific A-/Calling Party Category to allow them to be identified downstream.
Same question here.
> C). Release control modified to B-/Called Party Control with changed RTCWeb browser client behaviour along with indication as to "Off Hook" and "On Hook".
Does the "hook" map to anything we currently have in our APIs or concepts?
> Paul
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


From randell-ietf@jesup.org  Sun Jul 31 14:23:44 2011
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Subject: Re: [rtcweb] Possible New Cases - Emergency Services Call & Text
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On 7/31/2011 8:31 AM, Harald Alvestrand wrote:
> On 07/30/11 07:03, Paul Beaumont wrote:
>> This was mentioned is the initial RTCWeb session at IETF 81 but just 
>> to make sure it has been captured and understood.
>>
>> (1) Please can a separate use case be added for an Emergency Services 
>> telephony call (eg 999 in UK, 112 in EU, 911 in NA, etc).
> I am skeptical to adding this use case. The most important issues and 
> complications that ES brings to the table are with call setup, which I 
> believe is outside of what RTCWEB should be focusing on. As long as 
> there is nothing special about the media plane issues here, I would 
> prefer not to take it on.

The main media-plane issues that we should focus on (as opposed to the 
services built with it)
are ones that involve making capabilities available to the application, 
such as the aforementioned
ability to turn off VAD or to force a change from a "voice" codec to a 
more "general" codec (and
maybe to allow in this case automatic activation of the camera, but 
that's slightly arguable AND
with the one exception of VRS-911 there's no current support for cameras 
in 911 centers.  Most
of the rest of the requirements fall on the application/service.

-- 
Randell Jesup
randell-ietf@jesup.org

