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Subject: [rtcweb] Protocol Action: 'Web Real-Time Communication (WebRTC): Media Transport and Use of RTP' to Proposed Standard (draft-ietf-rtcweb-rtp-usage-25.txt)
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The IESG has approved the following document:
- 'Web Real-Time Communication (WebRTC): Media Transport and Use of RTP'
  (draft-ietf-rtcweb-rtp-usage-25.txt) as Proposed Standard

This document is the product of the Real-Time Communication in
WEB-browsers Working Group.

The IESG contact persons are Ben Campbell, Barry Leiba and Alissa Cooper.

A URL of this Internet Draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-rtp-usage/





Technical Summary

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

Working Group Summary

The document passed initial WG last call at version 14, with an 
outstanding issue about FEC remaining; that has since been split
into a new document (draft-ietf-rtcweb-fec).  Between that draft and 
this there were also additional comments on video orientation and 
media stream identification which resulted in new text.  I do not believe
there were areas where consensus is particularly rough at this point.

Document Quality

There are multiple implementations of the WebRTC framework
and they generally interoperate.  The review of the draft by the
working group was adequate, if prolonged.  It should be noted 
that some less-used aspects of RTP processing are well-understood 
by only a small number of participants, but the related matters 
appear to be resolved.

Personnel

Ted Hardie is the Document Shepherd.  Alissa Cooper is the Responsible
Area Director.  


RFC Editor Note

draft-ietf-avtcore-rtp-topologies-update should be changed from a 
normative reference to an informative reference.


From nobody Thu Jul  2 10:55:58 2015
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The IESG has received a request from the Real-Time Communication in
WEB-browsers WG (rtcweb) to consider the following document:
- 'STUN Usage for Consent Freshness'
  <draft-ietf-rtcweb-stun-consent-freshness-15.txt> as Proposed Standard

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action. Please send substantive comments to the
ietf@ietf.org mailing lists by 2015-07-16. Exceptionally, comments may be
sent to iesg@ietf.org instead. In either case, please retain the
beginning of the Subject line to allow automated sorting.

This document was previously last called on May 1 and has been updated as a result of comments submitted during that last call.

Abstract


   To prevent WebRTC applications, such as browsers, from launching
   attacks by sending media to unwilling victims, periodic consent to
   send needs to be obtained from remote endpoints.

   This document describes a consent mechanism using a new Session
   Traversal Utilities for NAT (STUN) usage.




The file can be obtained via
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-stun-consent-freshness/

IESG discussion can be tracked via
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-stun-consent-freshness/ballot/


No IPR declarations have been submitted directly on this I-D.



From nobody Sun Jul  5 17:09:30 2015
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the Real-Time Communication in WEB-browsers Working Group of the IETF.

        Title           : Javascript Session Establishment Protocol
        Authors         : Justin Uberti
                          Cullen Jennings
                          Eric Rescorla
	Filename        : draft-ietf-rtcweb-jsep-11.txt
	Pages           : 75
	Date            : 2015-07-05

Abstract:
   This document describes the mechanisms for allowing a Javascript
   application to control the signaling plane of a multimedia session
   via the interface specified in the W3C RTCPeerConnection API, and
   discusses how this relates to existing signaling protocols.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-jsep/

There's also a htmlized version available at:
https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-11

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-jsep-11


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

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


From nobody Mon Jul  6 01:35:24 2015
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Subject: Re: [rtcweb] LS groups (Re: Cullen Notes from RTCWeb interim on June 18, 2015)
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Den 29. juni 2015 19:24, skrev Cullen Jennings:
> 
> I think that LS makes more sense. Doing both LS and MSID was a comprise the WG came to long ago. We need for a say to synchronized audio and video and the IETF has already defined a widely deployed way of doing that so I don't see a gain in ignoring it and inventing something else. 

As I've said before, I've got no qualms about adding LS for
interoperability, for the cases where it makes sense.

I do have strong qualms about changing the definition of LS so that
stuff only interoperates some of the time (the "make it declarative"
solution).

I have also heard some people mumble about discarding the MSID mechanism
and only using LS; this is a proposal that I do not think will satisfy
the group's requirements.

And with regard to whether the WG came to the conclusion that we should
do both LS and MSID "long ago": I was present long ago, and I don't
recall the group coming to such a conclusion.

I've found a proposal for such a solution
(draft-jennings-rtcweb-plan-01, February 25 2013, section 7 bullet 7),
but that proposal seems to ignore the one-way nature of MediaStreams
(which is what led to the current isuse), and I've found no evidence of
a WG consensus on that part of that specification.

> 
> 
>> On Jun 22, 2015, at 5:19 AM, Harald Alvestrand <harald@alvestrand.no> wrote:
>>
>> On 06/18/2015 08:00 PM, Cullen Jennings (fluffy) wrote:
>>> * make sure to add note well slide to slide deck
>>>
>>> LS Behavior
>>> - Magnus points out that if JSEP overrides 5888, we will have issue with legacy things
>>> - Martin points needs more thought to make declarative groups
>>> - Martin raised point that we might be able to add some extra signaling that indicates that we can support this declarative LS usage
>>>
>>> Conclusion: Magnus will dig into more analysis of options on this and send to list. No decision on Option 1,2
>>>
>> Just FWIW:
>>
>> Personally, I don't see the added value in adding LS groups.
>>
>> We already have a well defined, declarative (=one-way) means of declaring synchronization in the form of the MSID values, which bind MediaStreamTracks into MediaStreams, which you're supposed to try syncing if you can (it's not a hard requirement in the spec that you should succeed).
>>
>> If we want to be extra helpful for non-WebRTC implementations, and end up in the case where we have one audio and one video track from one stream in one direction using the same media descriptions as an audio and video track in the other direction (which will be the normal case when your application is trying to emulate a single-channel VC unit), I have no particular problems with adding LS group markings to those media descriptions - but I don't want to spend resources changing the semantics of LS groups (and thereby imperiling the exact interworking we're trying to make easier) when we already have something that fulfils all the requirements for a pure WebRTC scenario.
>>
>> My $0.02.
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
> 


From nobody Mon Jul  6 13:08:53 2015
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From: Ted Hardie <ted.ietf@gmail.com>
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--001a11c24888dab3a9051a3a7868
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https://www.ietf.org/proceedings/93/agenda/agenda-93-rtcweb

corrections or suggestions to the list, please.

Ted, Cullen, and Sean

--001a11c24888dab3a9051a3a7868
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<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:tahoma,s=
ans-serif;font-size:small"><a href=3D"https://www.ietf.org/proceedings/93/a=
genda/agenda-93-rtcweb">https://www.ietf.org/proceedings/93/agenda/agenda-9=
3-rtcweb</a><br><br></div><div class=3D"gmail_default" style=3D"font-family=
:tahoma,sans-serif;font-size:small">corrections or suggestions to the list,=
 please.<br><br></div><div class=3D"gmail_default" style=3D"font-family:tah=
oma,sans-serif;font-size:small">Ted, Cullen, and Sean<br></div></div>

--001a11c24888dab3a9051a3a7868--


From nobody Mon Jul  6 14:20:28 2015
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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           : Transports for WebRTC
        Author          : Harald Alvestrand
	Filename        : draft-ietf-rtcweb-transports-09.txt
	Pages           : 15
	Date            : 2015-07-06

Abstract:
   This document describes the data transport protocols used by WebRTC,
   including the protocols used for interaction with intermediate boxes
   such as firewalls, relays and NAT boxes.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-transports/

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From nobody Mon Jul  6 15:01:13 2015
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Den 06. juli 2015 22:08, skrev Ted Hardie:
> https://www.ietf.org/proceedings/93/agenda/agenda-93-rtcweb
> 
> corrections or suggestions to the list, please.
> 
> Ted, Cullen, and Sean
> 
> 
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> 

Note - I have just submitted an update to -transports.

I don't think the change needs agenda time (it was just the DTLS
paragraph we discussed long ago), but if anyone has any other issues
with it, I'd appreciate it if they were filed at
https://github.com/rtcweb-wg/rtcweb-transport/issues so that I don't
lose track of them - and if the chairs think we need agenda time for
them, I'll be there.

Harald


From nobody Mon Jul  6 15:15:04 2015
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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           : WebRTC Gateways
        Authors         : Harald Alvestrand
                          Uwe Rauschenbach
	Filename        : draft-ietf-rtcweb-gateways-01.txt
	Pages           : 7
	Date            : 2015-07-06

Abstract:
   This document describes interoperability considerations for a class
   of WebRTC-compatible endpoints called "WebRTC gateways", which
   interconnect between WebRTC endpoints and devices that are not WebRTC
   endpoints.



The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-gateways/

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From nobody Mon Jul  6 15:19:12 2015
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Uwe and I have prepared a new -gateways draft (version -01).

I don't think there's much controversial in it, and there's only one
issue that I think needs WG time at this time:

Should it be Informational or Standards-track?

An informational document defines guidance for how gateways should
behave, but nobody needs to pay attention if they don't want to.

A standards-track document (which no other RTCWEB document should be
dependent on) defines requirements for how things that call themselves
"WebRTC gateways" should behave, but nobody needs to call their gateways
that, and anyway, there's no protocol police, so the difference isn't
all that much in practice.

More important: What do people want it to be?

Harald


From nobody Mon Jul  6 15:35:26 2015
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--047d7bb03f9c68fdaa051a3c8552
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On Mon, Jul 6, 2015 at 3:19 PM, Harald Alvestrand <harald@alvestrand.no>
wrote:

> Uwe and I have prepared a new -gateways draft (version -01).
>
> I don't think there's much controversial in it, and there's only one
> issue that I think needs WG time at this time:
>
> Should it be Informational or Standards-track?
>
> An informational document defines guidance for how gateways should
> behave, but nobody needs to pay attention if they don't want to.
>
> A standards-track document (which no other RTCWEB document should be
> dependent on) defines requirements for how things that call themselves
> "WebRTC gateways" should behave, but nobody needs to call their gateways
> that, and anyway, there's no protocol police, so the difference isn't
> all that much in practice.
>
> More important: What do people want it to be?
>
>
=E2=80=8BI think we adopted it in part because 3GPP had a dependency on it.=
  Does
it need to be standards track for them to reference it?

Ted=E2=80=8B




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

--047d7bb03f9c68fdaa051a3c8552
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<div dir=3D"ltr"><div class=3D"gmail_extra">On Mon, Jul 6, 2015 at 3:19 PM,=
 Harald Alvestrand <span dir=3D"ltr">&lt;<a href=3D"mailto:harald@alvestran=
d.no" target=3D"_blank">harald@alvestrand.no</a>&gt;</span> wrote:<br><div =
class=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"margin:0 0=
 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Uwe and I have prepare=
d a new -gateways draft (version -01).<br>
<br>
I don&#39;t think there&#39;s much controversial in it, and there&#39;s onl=
y one<br>
issue that I think needs WG time at this time:<br>
<br>
Should it be Informational or Standards-track?<br>
<br>
An informational document defines guidance for how gateways should<br>
behave, but nobody needs to pay attention if they don&#39;t want to.<br>
<br>
A standards-track document (which no other RTCWEB document should be<br>
dependent on) defines requirements for how things that call themselves<br>
&quot;WebRTC gateways&quot; should behave, but nobody needs to call their g=
ateways<br>
that, and anyway, there&#39;s no protocol police, so the difference isn&#39=
;t<br>
all that much in practice.<br>
<br>
More important: What do people want it to be?<br>
<br></blockquote><div><br><div class=3D"gmail_default" style=3D"font-family=
:tahoma,sans-serif;font-size:small">=E2=80=8BI think we adopted it in part =
because 3GPP had a dependency on it.=C2=A0 Does it need to be standards tra=
ck for them to reference it?<br><br></div><div class=3D"gmail_default" styl=
e=3D"font-family:tahoma,sans-serif;font-size:small">Ted=E2=80=8B</div><br><=
br>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex=
;border-left:1px #ccc solid;padding-left:1ex">
Harald<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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div><br></div></div>

--047d7bb03f9c68fdaa051a3c8552--


From nobody Mon Jul  6 15:48:53 2015
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From: Alan Johnston <alan.b.johnston@gmail.com>
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FYI - a minor update of the draft with the implementation status.  Anyone
interested in working on an open source ZRTP in JavaScript library, contact
me off list or in Prague.

- Alan -

---------- Forwarded message ----------
From: <internet-drafts@ietf.org>
Date: Mon, Jul 6, 2015 at 1:44 PM
Subject: New Version Notification for draft-johnston-rtcweb-zrtp-02.txt
To: Alan Johnston <alan.b.johnston@gmail.com>, Travis Cross <
tc@traviscross.com>, John Yoakum <yoakum@avaya.com>, Philip Zimmermann <
prz@mit.edu>, Jon Callas <jon@callas.org>



A new version of I-D, draft-johnston-rtcweb-zrtp-02.txt
has been successfully submitted by Alan Johnston and posted to the
IETF repository.

Name:           draft-johnston-rtcweb-zrtp
Revision:       02
Title:          Using ZRTP to Secure WebRTC
Document date:  2015-07-06
Group:          Individual Submission
Pages:          10
URL:
https://www.ietf.org/internet-drafts/draft-johnston-rtcweb-zrtp-02.txt
Status:         https://datatracker.ietf.org/doc/draft-johnston-rtcweb-zrtp/
Htmlized:       https://tools.ietf.org/html/draft-johnston-rtcweb-zrtp-02
Diff:
https://www.ietf.org/rfcdiff?url2=draft-johnston-rtcweb-zrtp-02

Abstract:
   WebRTC, Web Real-Time Communications, is a set of protocols and APIs
   used to enable web developers to add real-time communications into
   their web pages and applications with a few lines of JavaScript.
   WebRTC media flows are encrypted and authenticated by SRTP, the
   Secure Real-time Transport Protocol while the key agreement is
   provided by DTLS-SRTP, Datagram Transport Layer Security for Secure
   Real-time Transport Protocol.  However, without some third party
   identity service or certificate authority, WebRTC media flows have no
   protection against a man-in-the-middle (MitM) attack.  ZRTP, Media
   Path Key Agreement for Unicast Secure RTP, RFC 6189, does provide
   protection against MitM attackers using key continuity augmented with
   a Short Authentication String (SAS).  This specification describes
   how ZRTP can be used over the WebRTC data channel to provide MitM
   protection for WebRTC media flows keyed using DTLS-SRTP.  This
   provides users protection against MitM attackers without requiring
   browsers to support ZRTP or users to download a plugin or extension
   to implement ZRTP.




Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

The IETF Secretariat

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<div dir=3D"ltr">FYI - a minor update of the draft with the implementation =
status.=C2=A0 Anyone interested in working on an open source ZRTP in JavaSc=
ript library, contact me off list or in Prague.<div><br></div><div>- Alan -=
</div><div><br><div class=3D"gmail_quote">---------- Forwarded message ----=
------<br>From: <b class=3D"gmail_sendername"></b> <span dir=3D"ltr">&lt;<a=
 href=3D"mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a>&gt;<=
/span><br>Date: Mon, Jul 6, 2015 at 1:44 PM<br>Subject: New Version Notific=
ation for draft-johnston-rtcweb-zrtp-02.txt<br>To: Alan Johnston &lt;<a hre=
f=3D"mailto:alan.b.johnston@gmail.com">alan.b.johnston@gmail.com</a>&gt;, T=
ravis Cross &lt;<a href=3D"mailto:tc@traviscross.com">tc@traviscross.com</a=
>&gt;, John Yoakum &lt;<a href=3D"mailto:yoakum@avaya.com">yoakum@avaya.com=
</a>&gt;, Philip Zimmermann &lt;<a href=3D"mailto:prz@mit.edu">prz@mit.edu<=
/a>&gt;, Jon Callas &lt;<a href=3D"mailto:jon@callas.org">jon@callas.org</a=
>&gt;<br><br><br><br>
A new version of I-D, draft-johnston-rtcweb-zrtp-02.txt<br>
has been successfully submitted by Alan Johnston and posted to the<br>
IETF repository.<br>
<br>
Name:=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0draft-johnston-rtcweb-zrtp<br=
>
Revision:=C2=A0 =C2=A0 =C2=A0 =C2=A002<br>
Title:=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 Using ZRTP to Secure WebRTC<br>
Document date:=C2=A0 2015-07-06<br>
Group:=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 Individual Submission<br>
Pages:=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 10<br>
URL:=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 <a href=3D"https://www.ietf.o=
rg/internet-drafts/draft-johnston-rtcweb-zrtp-02.txt" rel=3D"noreferrer" ta=
rget=3D"_blank">https://www.ietf.org/internet-drafts/draft-johnston-rtcweb-=
zrtp-02.txt</a><br>
Status:=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0<a href=3D"https://datatracker.iet=
f.org/doc/draft-johnston-rtcweb-zrtp/" rel=3D"noreferrer" target=3D"_blank"=
>https://datatracker.ietf.org/doc/draft-johnston-rtcweb-zrtp/</a><br>
Htmlized:=C2=A0 =C2=A0 =C2=A0 =C2=A0<a href=3D"https://tools.ietf.org/html/=
draft-johnston-rtcweb-zrtp-02" rel=3D"noreferrer" target=3D"_blank">https:/=
/tools.ietf.org/html/draft-johnston-rtcweb-zrtp-02</a><br>
Diff:=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0<a href=3D"https://www.ietf.o=
rg/rfcdiff?url2=3Ddraft-johnston-rtcweb-zrtp-02" rel=3D"noreferrer" target=
=3D"_blank">https://www.ietf.org/rfcdiff?url2=3Ddraft-johnston-rtcweb-zrtp-=
02</a><br>
<br>
Abstract:<br>
=C2=A0 =C2=A0WebRTC, Web Real-Time Communications, is a set of protocols an=
d APIs<br>
=C2=A0 =C2=A0used to enable web developers to add real-time communications =
into<br>
=C2=A0 =C2=A0their web pages and applications with a few lines of JavaScrip=
t.<br>
=C2=A0 =C2=A0WebRTC media flows are encrypted and authenticated by SRTP, th=
e<br>
=C2=A0 =C2=A0Secure Real-time Transport Protocol while the key agreement is=
<br>
=C2=A0 =C2=A0provided by DTLS-SRTP, Datagram Transport Layer Security for S=
ecure<br>
=C2=A0 =C2=A0Real-time Transport Protocol.=C2=A0 However, without some thir=
d party<br>
=C2=A0 =C2=A0identity service or certificate authority, WebRTC media flows =
have no<br>
=C2=A0 =C2=A0protection against a man-in-the-middle (MitM) attack.=C2=A0 ZR=
TP, Media<br>
=C2=A0 =C2=A0Path Key Agreement for Unicast Secure RTP, RFC 6189, does prov=
ide<br>
=C2=A0 =C2=A0protection against MitM attackers using key continuity augment=
ed with<br>
=C2=A0 =C2=A0a Short Authentication String (SAS).=C2=A0 This specification =
describes<br>
=C2=A0 =C2=A0how ZRTP can be used over the WebRTC data channel to provide M=
itM<br>
=C2=A0 =C2=A0protection for WebRTC media flows keyed using DTLS-SRTP.=C2=A0=
 This<br>
=C2=A0 =C2=A0provides users protection against MitM attackers without requi=
ring<br>
=C2=A0 =C2=A0browsers to support ZRTP or users to download a plugin or exte=
nsion<br>
=C2=A0 =C2=A0to implement ZRTP.<br>
<br>
<br>
<br>
<br>
Please note that it may take a couple of minutes from the time of submissio=
n<br>
until the htmlized version and diff are available at <a href=3D"http://tool=
s.ietf.org" rel=3D"noreferrer" target=3D"_blank">tools.ietf.org</a>.<br>
<br>
The IETF Secretariat<br>
<br>
</div><br></div></div>

--047d7bacc110557ceb051a3cb5a3--


From nobody Tue Jul  7 04:33:31 2015
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From: Barry Dingle <btdingle@gmail.com>
Date: Tue, 7 Jul 2015 21:33:06 +1000
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-jsep-11.txt
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In JSEP 5.1.1 Implementation Requirements, the second requirement states:

   R-2   [RFC5764 <https://tools.ietf.org/html/rfc5764>] MUST be
supported for signaling the UDP/TLS/RTP/SAVPF
         [RFC5764 <https://tools.ietf.org/html/rfc5764>] and TCP/DTLS/RTP/SAVPF
         [I-D.nandakumar-mmusic-proto-iana-registration
<https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-11#ref-I-D.nandakumar-mmusic-proto-iana-registration>]
RTP profiles.


The second [RFC5764] appears to an editorial error.


However, I am not sure that the remaining text makes sense as RFC2764
appears to relate to UDP stacks only and appears to explicitly exclude
TCP stacks (see its Section 8.)


Do we need two requirements - one for UDP stacks and another for TCP stacks?


/Barry Dingle


On Mon, Jul 6, 2015 at 10:09 AM, <internet-drafts@ietf.org> wrote:

>
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
>  This draft is a work item of the Real-Time Communication in WEB-browsers
> Working Group of the IETF.
>
>         Title           : Javascript Session Establishment Protocol
>         Authors         : Justin Uberti
>                           Cullen Jennings
>                           Eric Rescorla
>         Filename        : draft-ietf-rtcweb-jsep-11.txt
>         Pages           : 75
>         Date            : 2015-07-05
>
> Abstract:
>    This document describes the mechanisms for allowing a Javascript
>    application to control the signaling plane of a multimedia session
>    via the interface specified in the W3C RTCPeerConnection API, and
>    discusses how this relates to existing signaling protocols.
>
>
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-jsep/
>
> There's also a htmlized version available at:
> https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-11
>
> A diff from the previous version is available at:
> https://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-jsep-11
>
>
> Please note that it may take a couple of minutes from the time of
> submission
> until the htmlized version and diff are available at tools.ietf.org.
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<div dir=3D"ltr">In JSEP 5.1.1 Implementation Requirements, the second requ=
irement states:=C2=A0<div><br></div><div><pre class=3D"" style=3D"font-size=
:13.3333330154419px;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)">   R=
-2   [<a href=3D"https://tools.ietf.org/html/rfc5764" title=3D"&quot;Datagr=
am Transport Layer Security (DTLS) Extension to Establish Keys for the Secu=
re Real-time Transport Protocol (SRTP)&quot;">RFC5764</a>] MUST be supporte=
d for signaling the UDP/TLS/RTP/SAVPF
         [<a href=3D"https://tools.ietf.org/html/rfc5764" title=3D"&quot;Da=
tagram Transport Layer Security (DTLS) Extension to Establish Keys for the =
Secure Real-time Transport Protocol (SRTP)&quot;">RFC5764</a>] and TCP/DTLS=
/RTP/SAVPF
         [<a href=3D"https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-11#=
ref-I-D.nandakumar-mmusic-proto-iana-registration">I-D.nandakumar-mmusic-pr=
oto-iana-registration</a>] RTP profiles. </pre><pre class=3D"" style=3D"fon=
t-size:13.3333330154419px;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)=
"><br></pre><pre class=3D"" style=3D"font-size:13.3333330154419px;margin-to=
p:0px;margin-bottom:0px;color:rgb(0,0,0)">The second [RFC5764] appears to a=
n editorial error. </pre><pre class=3D"" style=3D"font-size:13.333333015441=
9px;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><br></pre><pre class=
=3D"" style=3D"font-size:13.3333330154419px;margin-top:0px;margin-bottom:0p=
x;color:rgb(0,0,0)">However, I am not sure that the remaining text makes se=
nse as <span style=3D"font-size:13.3333330154419px;font-family:arial,sans-s=
erif">RFC2764 appears to relate to UDP stacks only and appears to explicitl=
y exclude TCP stacks (see its Section 8.)</span></pre><pre class=3D"" style=
=3D"font-size:13.3333330154419px;margin-top:0px;margin-bottom:0px;color:rgb=
(0,0,0)"><span style=3D"font-size:13.3333330154419px;font-family:arial,sans=
-serif"><br></span></pre><pre class=3D"" style=3D"font-size:13.333333015441=
9px;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><span style=3D"font-=
size:13.3333330154419px;font-family:arial,sans-serif">Do we need two requir=
ements - one for UDP stacks and another for TCP stacks? </span></pre></div>=
<div class=3D"gmail_extra"><div><div class=3D"gmail_signature"><div dir=3D"=
ltr"><div><div dir=3D"ltr"><br>/Barry=C2=A0Dingle<br>=C2=A0=C2=A0<br></div>=
</div></div></div></div>
<br><div class=3D"gmail_quote">On Mon, Jul 6, 2015 at 10:09 AM,  <span dir=
=3D"ltr">&lt;<a href=3D"mailto:internet-drafts@ietf.org" target=3D"_blank">=
internet-drafts@ietf.org</a>&gt;</span> wrote:<br><blockquote class=3D"gmai=
l_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left=
:1ex"><br>
A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.<br>
=C2=A0This draft is a work item of the Real-Time Communication in WEB-brows=
ers Working Group of the IETF.<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Title=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0:=
 Javascript Session Establishment Protocol<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Authors=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0: Just=
in Uberti<br>
=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 Cullen Jennings<br>
=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 Eric Rescorla<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Filename=C2=A0 =C2=A0 =C2=A0 =C2=A0 : draft-iet=
f-rtcweb-jsep-11.txt<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Pages=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0:=
 75<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Date=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 :=
 2015-07-05<br>
<br>
Abstract:<br>
=C2=A0 =C2=A0This document describes the mechanisms for allowing a Javascri=
pt<br>
=C2=A0 =C2=A0application to control the signaling plane of a multimedia ses=
sion<br>
=C2=A0 =C2=A0via the interface specified in the W3C RTCPeerConnection API, =
and<br>
=C2=A0 =C2=A0discusses how this relates to existing signaling protocols.<br=
>
<br>
<br>
The IETF datatracker status page for this draft is:<br>
<a href=3D"https://datatracker.ietf.org/doc/draft-ietf-rtcweb-jsep/" rel=3D=
"noreferrer" target=3D"_blank">https://datatracker.ietf.org/doc/draft-ietf-=
rtcweb-jsep/</a><br>
<br>
There&#39;s also a htmlized version available at:<br>
<a href=3D"https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-11" rel=3D"no=
referrer" target=3D"_blank">https://tools.ietf.org/html/draft-ietf-rtcweb-j=
sep-11</a><br>
<br>
A diff from the previous version is available at:<br>
<a href=3D"https://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-jsep-11" r=
el=3D"noreferrer" target=3D"_blank">https://www.ietf.org/rfcdiff?url2=3Ddra=
ft-ietf-rtcweb-jsep-11</a><br>
<br>
<br>
Please note that it may take a couple of minutes from the time of submissio=
n<br>
until the htmlized version and diff are available at <a href=3D"http://tool=
s.ietf.org" rel=3D"noreferrer" target=3D"_blank">tools.ietf.org</a>.<br>
<br>
Internet-Drafts are also available by anonymous FTP at:<br>
<a href=3D"ftp://ftp.ietf.org/internet-drafts/" rel=3D"noreferrer" target=
=3D"_blank">ftp://ftp.ietf.org/internet-drafts/</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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div><br></div></div>

--089e013c6b92f60811051a476365--


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Subject: [rtcweb] Agenda time for draft-jennings-behave-rtcweb-firewall-00 ?
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I'd like to get a brief amount of agenda time to discuss changes we =
would need to some of the WebRTC WG drafts to make the ideas proposed in =
draft-jennings-behave-rtcweb-firewall- work and make people aware of =
this work.=20

Would it be possible to get 10 minutes on the agenda ?=20

Thanks, Cullen



From nobody Tue Jul  7 09:17:12 2015
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On Tue, Jul 7, 2015 at 8:50 AM, Cullen Jennings <fluffy@iii.ca> wrote:

>
> I'd like to get a brief amount of agenda time to discuss changes we would
> need to some of the WebRTC WG drafts to make the ideas proposed in
> draft-jennings-behave-rtcweb-firewall- work and make people aware of this
> work.
>
> Would it be possible to get 10 minutes on the agenda ?
>
>
=E2=80=8BThere is time on day 2, at this point, so it shouldn't be a proble=
m.

Ted=E2=80=8B




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

--001a11c25b68a88202051a4b5a24
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<div dir=3D"ltr"><div class=3D"gmail_extra">On Tue, Jul 7, 2015 at 8:50 AM,=
 Cullen Jennings <span dir=3D"ltr">&lt;<a href=3D"mailto:fluffy@iii.ca" tar=
get=3D"_blank">fluffy@iii.ca</a>&gt;</span> wrote:<br><div class=3D"gmail_q=
uote"><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-l=
eft:1px #ccc solid;padding-left:1ex"><br>
I&#39;d like to get a brief amount of agenda time to discuss changes we wou=
ld need to some of the WebRTC WG drafts to make the ideas proposed in draft=
-jennings-behave-rtcweb-firewall- work and make people aware of this work.<=
br>
<br>
Would it be possible to get 10 minutes on the agenda ?<br>
<br></blockquote><div><br><div class=3D"gmail_default" style=3D"font-family=
:tahoma,sans-serif;font-size:small">=E2=80=8BThere is time on day 2, at thi=
s point, so it shouldn&#39;t be a problem.<br><br></div><div class=3D"gmail=
_default" style=3D"font-family:tahoma,sans-serif;font-size:small">Ted=E2=80=
=8B</div><br><br>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"mar=
gin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Thanks, Cullen<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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div><br></div></div>

--001a11c25b68a88202051a4b5a24--


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From: Ted Hardie <ted.ietf@gmail.com>
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Subject: [rtcweb] PERC pointer
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--047d7bb03f9cf457ff051a5008cb
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With the approval of the PERC working group, there is now IETF activity on
privacy enhanced RTP conferencing--including a draft on WebRTC:

https://www.ietf.org/internet-drafts/draft-westerlund-perc-webrtc-use-case-00.txt


They've requested agenda time for it in the PERC WG and discussion should
probably be on that list.

regards,

Ted
/no hat

--047d7bb03f9cf457ff051a5008cb
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<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:tahoma,s=
ans-serif;font-size:small">With the approval of the PERC working group, the=
re is now IETF activity on privacy enhanced RTP conferencing--including a d=
raft on WebRTC:<br><br><pre class=3D""><a href=3D"https://www.ietf.org/inte=
rnet-drafts/draft-westerlund-perc-webrtc-use-case-00.txt">https://www.ietf.=
org/internet-drafts/draft-westerlund-perc-webrtc-use-case-00.txt</a></pre><=
br></div><div class=3D"gmail_default" style=3D"font-family:tahoma,sans-seri=
f;font-size:small">They&#39;ve requested agenda time for it in the PERC WG =
and discussion should probably be on that list.<br><br></div><div class=3D"=
gmail_default" style=3D"font-family:tahoma,sans-serif;font-size:small">rega=
rds,<br><br></div><div class=3D"gmail_default" style=3D"font-family:tahoma,=
sans-serif;font-size:small">Ted<br></div><div class=3D"gmail_default" style=
=3D"font-family:tahoma,sans-serif;font-size:small">/no hat<br></div></div>

--047d7bb03f9cf457ff051a5008cb--


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From: Eric Rescorla <ekr@rtfm.com>
Date: Wed, 8 Jul 2015 07:39:35 -0700
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Subject: Re: [rtcweb] Agenda time for draft-jennings-behave-rtcweb-firewall-00 ?
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I have taken a brief look at this draft and I don't believe that the
proposed pinhole/inspection
algorithm provides the desired assurances of verifying consent:

1. If you open up an incoming/outgoing pinhole on outgoing packet, prior to
receiving an incoming STUN packet, then if the internal implementation
simply sends a new outgoing STUN packet every 5 seconds, then they
never need to rely on external consent.

2. If you just require *a* incoming STUN packet to be received in order to
extend the lifetime to 30 seconds, then an off-path attacker can forge
a STUN response that will cause the pinhole to be open for 30 seconds.
(Hence the need for the STUN transaction ID).

It seems to me that you should be instead requiring a complete STUN
transaction before allowing any non-STUN outgoing traffic. It would probably
be safe-ish to require it before allowing any incoming traffic as well,
though
that leaves the chance that you get a small amount of clipping with packet
drops or reordering.


WRT to the origin attribute, I have mixed feelings. I agree that your SNI
argument
is strong, but there is also a move to figure out how to encrypt SNI, so
every time
we extend that argument it makes that harder. OTOH, STUN origin will be
sent a lot less often than SNI, I imagine.


I don't understand Section 6. Is this guidance to firewall vendors or to
browser
implementors. If the former, the firewall has no way of knowing what
"requests"
are if the connection use HTTPS (and if is uses HTTP, you have DPI-type
methods for
determining whether something is voice or video. If this is guidance to
browser
vendors, I consider it extremely unlikely that we would implement it.


Point #4 in your security considerations section seems like an entirely lost
cause, unless you assume that attackers are not really trying.

-Ekr










On Tue, Jul 7, 2015 at 8:50 AM, Cullen Jennings <fluffy@iii.ca> wrote:

>
> I'd like to get a brief amount of agenda time to discuss changes we would
> need to some of the WebRTC WG drafts to make the ideas proposed in
> draft-jennings-behave-rtcweb-firewall- work and make people aware of this
> work.
>
> Would it be possible to get 10 minutes on the agenda ?
>
> Thanks, Cullen
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--089e011771a9f3e0bc051a5e1d40
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">I have taken a brief look at this draft and I don&#39;t be=
lieve that the proposed pinhole/inspection<div>algorithm provides the desir=
ed assurances of verifying consent:</div><div><br></div><div>1. If you open=
 up an incoming/outgoing pinhole on outgoing packet, prior to</div><div>rec=
eiving an incoming STUN packet, then if the internal implementation</div><d=
iv>simply sends a new outgoing STUN packet every 5 seconds, then they</div>=
<div>never need to rely on external consent.</div><div><br></div><div>2. If=
 you just require *a* incoming STUN packet to be received in order to</div>=
<div>extend the lifetime to 30 seconds, then an off-path attacker can forge=
</div><div>a STUN response that will cause the pinhole to be open for 30 se=
conds.</div><div>(Hence the need for the STUN transaction ID).</div><div><b=
r></div><div>It seems to me that you should be instead requiring a complete=
 STUN</div><div>transaction before allowing any non-STUN outgoing traffic. =
It would probably</div><div>be safe-ish to require it before allowing any i=
ncoming traffic as well, though</div><div>that leaves the chance that you g=
et a small amount of clipping with packet</div><div>drops or reordering.</d=
iv><div><br></div><div><br></div><div>WRT to the origin attribute, I have m=
ixed feelings. I agree that your SNI argument</div><div>is strong, but ther=
e is also a move to figure out how to encrypt SNI, so every time</div><div>=
we extend that argument it makes that harder. OTOH, STUN origin will be</di=
v><div>sent a lot less often than SNI, I imagine.</div><div><br></div><div>=
<br></div><div>I don&#39;t understand Section 6. Is this guidance to firewa=
ll vendors or to browser</div><div>implementors. If the former, the firewal=
l has no way of knowing what &quot;requests&quot;</div><div>are if the conn=
ection use HTTPS (and if is uses HTTP, you have DPI-type methods for</div><=
div>determining whether something is voice or video. If this is guidance to=
 browser</div><div>vendors, I consider it extremely unlikely that we would =
implement it.</div><div><br></div><div><br></div><div>Point #4 in your secu=
rity considerations section seems like an entirely lost</div><div>cause, un=
less you assume that attackers are not really trying.</div><div><br></div><=
div>-Ekr</div><div><br></div><div><br></div><div><br></div><div><br></div><=
div><br></div><div><br></div><div><br></div><div><br></div><div><br></div><=
/div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Tue, Jul =
7, 2015 at 8:50 AM, Cullen Jennings <span dir=3D"ltr">&lt;<a href=3D"mailto=
:fluffy@iii.ca" target=3D"_blank">fluffy@iii.ca</a>&gt;</span> wrote:<br><b=
lockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px =
#ccc solid;padding-left:1ex"><br>
I&#39;d like to get a brief amount of agenda time to discuss changes we wou=
ld need to some of the WebRTC WG drafts to make the ideas proposed in draft=
-jennings-behave-rtcweb-firewall- work and make people aware of this work.<=
br>
<br>
Would it be possible to get 10 minutes on the agenda ?<br>
<br>
Thanks, Cullen<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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div><br></div>

--089e011771a9f3e0bc051a5e1d40--


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Date: Wed, 08 Jul 2015 20:34:15 -0600
From: Peter Saint-Andre - &yet <peter@andyet.net>
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Subject: [rtcweb] two questions about draft-ietf-rtcweb-return
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First, this is a great document - very clear and easy to read. Kudos to 
the authors!

I have two questions.

1. Section 4.2 states:

    A virtual interface can produce
    "host", "server-reflexive", and "relay" candidates, but may be
    restricted to only some type of candidate (e.g.  UDP-only).

How does that restriction work? Is it effected by the nature of the 
firewall (e.g., Section 6.1 mentions the possibility of a firewall that 
blocks all UDP traffic), by the configuration of the TURN proxy, by 
local policy in the WebRTC user agent, or by some other method?

2. Section 5.1 states:

    When a proxy is configured, by Auto-Discovery or a proprietary means,
    the browser MUST NOT report a "relay" candidate representing the
    proxy.  [...]

    For a virtual interface representing a TURN proxy, this means that
    the browser MUST report the public-facing IP address and port
    acquired through TURN as a "host" candidate....

Have the authors considered how that rule might interact with the known 
(although poorly documented) improvement in call setup time that results 
from communicating only relayed candidates in the initial offer?

Thanks!

Peter

-- 
Peter Saint-Andre
https://andyet.com/


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--089e01183ebcfb5758051a71f878
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]On Wed, Jul 8, 2015 at 10:34 PM, Peter Saint-Andre - &yet <peter@andyet.net
> wrote:

> First, this is a great document - very clear and easy to read. Kudos to
> the authors!
>

Thanks!


> I have two questions.
>
> 1. Section 4.2 states:
>
>    A virtual interface can produce
>    "host", "server-reflexive", and "relay" candidates, but may be
>    restricted to only some type of candidate (e.g.  UDP-only).
>
> How does that restriction work? Is it effected by the nature of the
> firewall (e.g., Section 6.1 mentions the possibility of a firewall that
> blocks all UDP traffic), by the configuration of the TURN proxy, by local
> policy in the WebRTC user agent, or by some other method?
>

I would say "by the configuration of the TURN proxy".  This statement was
intended to be just a helpful reinforcement of the obvious: if a virtual
interface represents paths through a TURN proxy, and the proxy is UDP-only
(like most TURN implementations), then no TCP candidates will be generated
for that interface.

As always, language improvements welcome.


>
> 2. Section 5.1 states:
>
>    When a proxy is configured, by Auto-Discovery or a proprietary means,
>    the browser MUST NOT report a "relay" candidate representing the
>    proxy.  [...]
>
>    For a virtual interface representing a TURN proxy, this means that
>    the browser MUST report the public-facing IP address and port
>    acquired through TURN as a "host" candidate....
>
> Have the authors considered how that rule might interact with the known
> (although poorly documented) improvement in call setup time that results
> from communicating only relayed candidates in the initial offer?
>

My impression is that this improvement is only true when there are no
publicly routable "host" candidates.  If the proxy has a publicly routable
external interface (as seems likely to be the common case), then I expect
there should be no performance issue.

This does suggest a discussion of how best to compute priority of these
candidates, but I view that kind of performance tuning as a secondary
concern.


>
> Thanks!
>
> Peter
>
> --
> Peter Saint-Andre
> https://andyet.com/
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--089e01183ebcfb5758051a71f878
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">]On =
Wed, Jul 8, 2015 at 10:34 PM, Peter Saint-Andre - &amp;yet <span dir=3D"ltr=
">&lt;<a href=3D"mailto:peter@andyet.net" target=3D"_blank">peter@andyet.ne=
t</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margi=
n:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">First, this is a =
great document - very clear and easy to read. Kudos to the authors!<br></bl=
ockquote><div><br></div><div>Thanks!</div><div>=C2=A0</div><blockquote clas=
s=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;pad=
ding-left:1ex">
I have two questions.<br>
<br>
1. Section 4.2 states:<br>
<br>
=C2=A0 =C2=A0A virtual interface can produce<br>
=C2=A0 =C2=A0&quot;host&quot;, &quot;server-reflexive&quot;, and &quot;rela=
y&quot; candidates, but may be<br>
=C2=A0 =C2=A0restricted to only some type of candidate (e.g.=C2=A0 UDP-only=
).<br>
<br>
How does that restriction work? Is it effected by the nature of the firewal=
l (e.g., Section 6.1 mentions the possibility of a firewall that blocks all=
 UDP traffic), by the configuration of the TURN proxy, by local policy in t=
he WebRTC user agent, or by some other method?<br></blockquote><div><br></d=
iv><div>I would say &quot;by the configuration of the TURN proxy&quot;.=C2=
=A0 This statement was intended to be just a helpful reinforcement of the o=
bvious: if a virtual interface represents paths through a TURN proxy, and t=
he proxy is UDP-only (like most TURN implementations), then no TCP candidat=
es will be generated for that interface.</div><div><br></div><div>As always=
, language improvements welcome.</div><div>=C2=A0</div><blockquote class=3D=
"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding=
-left:1ex">
<br>
2. Section 5.1 states:<br>
<br>
=C2=A0 =C2=A0When a proxy is configured, by Auto-Discovery or a proprietary=
 means,<br>
=C2=A0 =C2=A0the browser MUST NOT report a &quot;relay&quot; candidate repr=
esenting the<br>
=C2=A0 =C2=A0proxy.=C2=A0 [...]<br>
<br>
=C2=A0 =C2=A0For a virtual interface representing a TURN proxy, this means =
that<br>
=C2=A0 =C2=A0the browser MUST report the public-facing IP address and port<=
br>
=C2=A0 =C2=A0acquired through TURN as a &quot;host&quot; candidate....<br>
<br>
Have the authors considered how that rule might interact with the known (al=
though poorly documented) improvement in call setup time that results from =
communicating only relayed candidates in the initial offer?<br></blockquote=
><div><br></div><div>My impression is that this improvement is only true wh=
en there are no publicly routable &quot;host&quot; candidates.=C2=A0 If the=
 proxy has a publicly routable external interface (as seems likely to be th=
e common case), then I expect there should be no performance issue.</div><d=
iv><br></div><div>This does suggest a discussion of how best to compute pri=
ority of these candidates, but I view that kind of performance tuning as a =
secondary concern.</div><div>=C2=A0<br></div><blockquote class=3D"gmail_quo=
te" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"=
>
<br>
Thanks!<span class=3D"HOEnZb"><font color=3D"#888888"><br>
<br>
Peter<br>
<br>
-- <br>
Peter Saint-Andre<br>
<a href=3D"https://andyet.com/" rel=3D"noreferrer" target=3D"_blank">https:=
//andyet.com/</a><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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</font></span></blockquote></div><br></div></div>

--089e01183ebcfb5758051a71f878--


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From: Simon Perreault <sperreault@jive.com>
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Subject: Re: [rtcweb] two questions about draft-ietf-rtcweb-return
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Le 2015-07-09 10:21, Benjamin Schwartz a écrit :
>     How does that restriction work? Is it effected by the nature of the
>     firewall (e.g., Section 6.1 mentions the possibility of a firewall
>     that blocks all UDP traffic), by the configuration of the TURN
>     proxy, by local policy in the WebRTC user agent, or by some other
>     method?
> 
> 
> I would say "by the configuration of the TURN proxy".  This statement
> was intended to be just a helpful reinforcement of the obvious: if a
> virtual interface represents paths through a TURN proxy, and the proxy
> is UDP-only (like most TURN implementations), then no TCP candidates
> will be generated for that interface.
> 
> As always, language improvements welcome.

Ok then here's one: "TURN server" is the preferred term, instead of
"TURN proxy". :)

>     Have the authors considered how that rule might interact with the
>     known (although poorly documented) improvement in call setup time
>     that results from communicating only relayed candidates in the
>     initial offer?
> 
> 
> My impression is that this improvement is only true when there are no
> publicly routable "host" candidates.  If the proxy has a publicly
> routable external interface (as seems likely to be the common case),
> then I expect there should be no performance issue.

I don't think that this is even true. I bet that most of the time when
you have a publicly routable "host" candidate you will still be behind a
NAT or a stateful firewall, and inbound connections will still be
denied. The IP address range means very little.

The fact is that we have ICE prioritization completely wrong. See RFC
5245 section 4.1.2.2 and note that the guidelines listed there relate to
steady-state performance. That is worthless. Nomination is what is
concerned with steady-state performance. Prioritization should only be
concerned with how long it takes until I have a working candidate pair.
This means that candidates with the highest chance of working and the
lowest expected setup time should be tried first. That's probably TURN
candidates. As soon as you have something that works you can start using
it. Then you can take your time and evaluate the other candidate pairs
more-or-less in depth, perhaps do some actual measurements, then
nominate the candidate pair that has the best steady-state performance.

Simon


From nobody Sat Jul 11 09:43:02 2015
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Subject: [rtcweb] Please require user consent for data channels
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Howdy all, this is mostly a re-surfacing of the discussion about IP
address leaking back in April[1], which unfortunately I did not
discover until recently.

One of the items in the new proposal was "WebRTC already requires
permission to access getUserMedia. Why not use that permission to
control interface enumeration?" That item didn't really get discussed
much in the thread, but I think it's one of the most important issues.

Why? There is now a documented case where a third party on nytimes.com
is using a fake webRTC datachannel to silently gather user local (and
potentially "real" ISP) IP addresses.

https://twitter.com/incloud/status/619624021123010560

So I'd like to voice my support for adding a consent requirement that
would prevent this type of silent behavior. A user that is
purposefully visiting a site that has a legitimate reason for using a
webRTC data channel will have the necessary context to give consent,
just like they have to get consent on getUserMedia.

I'm really unclear on why it's so important to have the silent webRTC
data channel behavior. P2P data connections should be held to the same
consent requirements just like P2P video and voice connections. In
addition to the silent third party local IP tracking, a silent P2P
data channel opens up users to silently becoming nodes in a web-based
file sharing network. For example, webtorrent[2] can be silently
embedded on a website so that all of the users to that website start
seeding content they don't know they are seeding.

Possible fix:

1. Require user consent before calling the callback on createOffer().
2. If the user has already given consent via getUserMedia, the user
consent requirement is satisfied.
3. If there's no previous gUM consent, then ask the user for consent.

This implementation requires no updates to current webRTC demos, since
you have to wait for a callback on createOffer anyway. I'm open to
other ways, such as making createDataChannel asynchronous like gUM,
but that would require updates to existing apps that use
createDataChannel.

Thanks!
Daniel Roesler

[1]: https://www.ietf.org/mail-archive/web/rtcweb/current/msg14494.htm
[2]: https://github.com/feross/webtorrent


From nobody Sat Jul 11 21:26:33 2015
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On 7/11/2015 9:42 AM, Daniel Roesler wrote:
> Howdy all, this is mostly a re-surfacing of the discussion about IP
> address leaking back in April[1], which unfortunately I did not
> discover until recently.
>
> One of the items in the new proposal was "WebRTC already requires
> permission to access getUserMedia. Why not use that permission to
> control interface enumeration?" That item didn't really get discussed
> much in the thread, but I think it's one of the most important issues.
>
> Why? There is now a documented case where a third party on nytimes.com
> is using a fake webRTC datachannel to silently gather user local (and
> potentially "real" ISP) IP addresses.
> ...

On the IPv6 Internet, the IP address you use to reach the web site is 
almost certainly the same as your "local" IP address. There's no 
additional information exposed by allowing an application to discover 
that information directly via JavaScript.

The IPv4 Internet is essentially out of addresses and in the process of 
being retired. I don't believe there's any reason at this point to 
disable functionality in order to improve compatibility with this legacy 
network.

Matthew Kaufman

ps. You can also gather all these addresses for any browser with Flash 
Player installed by asking Flash to connect via RTMFP to a server, 
whereupon it will report the full enumeration of available IPv4 and IPv6 
addresses to that server.


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Date: Sun, 12 Jul 2015 10:49:31 +0200
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Subject: Re: [rtcweb] Please require user consent for data channels
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Hi,

While acknowledging that there might be a potential privacy issue, I 
don't think agree on your analysis on the issue that you have made nor 
your proposal.

The information available in the ICE gathering process and made 
available to the JS app is the following:

-Host candidates, mostly private IP addresses
-Server reflexive candidates, exposing the public IP address as shown by 
the TURN/STUN server
-Relay candidates, hosted by the TURN server

My understanding of the issue is that the leak happens when the 
application is able to retrieve information not already available to it.
That happens either when the user is using a VPN and have multiple 
public IP addresses or using an external PROXY so the http/ws connection 
have a different originator public IP address. AFAIK there should be no 
potential danger in leaking the private IP4 addresses (here you got 
mine, in case you are interested 127.0.0.1 and 192.168.64.2 ;). I have 
not much background on IPv6, so please complete any missing info and how 
would it affect here.

I don't think it is a good idea to tight the solution of requesting user 
permission, to not leak IP information, as theoretically a user could be 
interested in using webrtc (audio/video/datachannels) and still don't 
disclose their public IP address (i.e. forcing all the data to go via a 
TURN server).

One possible solution would be that the browser don't gather/announce 
host&reflexive candidates when an HTTP proxy is set, and only use turn 
relay candidates instead. The second solution (which I think it is 
already discussed somewhere else) would be to allow the user to 
configure a TURN server to override the app settings and force all 
connection to go through it. (Then build a chained-TURN infrastructure 
into TOR and you could have a killer combo ;)

Regarding the issue that the browser becomes a CDN peer, not sure if 
anyone has ever raised the issue before. From my point of view, there is 
not, but in any case, I think it is orthogonal to the leak issue.

Best regards
Sergio



On 11/07/2015 18:42, Daniel Roesler wrote:
> Howdy all, this is mostly a re-surfacing of the discussion about IP
> address leaking back in April[1], which unfortunately I did not
> discover until recently.
>
> One of the items in the new proposal was "WebRTC already requires
> permission to access getUserMedia. Why not use that permission to
> control interface enumeration?" That item didn't really get discussed
> much in the thread, but I think it's one of the most important issues.
>
> Why? There is now a documented case where a third party on nytimes.com
> is using a fake webRTC datachannel to silently gather user local (and
> potentially "real" ISP) IP addresses.
>
> https://twitter.com/incloud/status/619624021123010560
>
> So I'd like to voice my support for adding a consent requirement that
> would prevent this type of silent behavior. A user that is
> purposefully visiting a site that has a legitimate reason for using a
> webRTC data channel will have the necessary context to give consent,
> just like they have to get consent on getUserMedia.
>
> I'm really unclear on why it's so important to have the silent webRTC
> data channel behavior. P2P data connections should be held to the same
> consent requirements just like P2P video and voice connections. In
> addition to the silent third party local IP tracking, a silent P2P
> data channel opens up users to silently becoming nodes in a web-based
> file sharing network. For example, webtorrent[2] can be silently
> embedded on a website so that all of the users to that website start
> seeding content they don't know they are seeding.
>
> Possible fix:
>
> 1. Require user consent before calling the callback on createOffer().
> 2. If the user has already given consent via getUserMedia, the user
> consent requirement is satisfied.
> 3. If there's no previous gUM consent, then ask the user for consent.
>
> This implementation requires no updates to current webRTC demos, since
> you have to wait for a callback on createOffer anyway. I'm open to
> other ways, such as making createDataChannel asynchronous like gUM,
> but that would require updates to existing apps that use
> createDataChannel.
>
> Thanks!
> Daniel Roesler
>
> [1]: https://www.ietf.org/mail-archive/web/rtcweb/current/msg14494.htm
> [2]: https://github.com/feross/webtorrent
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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From: Daniel Roesler <diafygi@gmail.com>
Date: Sun, 12 Jul 2015 12:19:03 -0700
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Thanks very much for the responses! Going to try and address both inline below.

On Sat, Jul 11, 2015 at 9:26 PM, Matthew Kaufman <matthew@matthew.at> wrote:
>
> On the IPv6 Internet, the IP address you use to reach the web site is almost
> certainly the same as your "local" IP address. There's no additional
> information exposed by allowing an application to discover that information
> directly via JavaScript.
>
> The IPv4 Internet is essentially out of addresses and in the process of
> being retired. I don't believe there's any reason at this point to disable
> functionality in order to improve compatibility with this legacy network.
>

I disagree with this sentiment. IPv4 will be around for many more
years and many ISPs still do not assign their customers IPv6
addresses. If IPv6 is replacing IPv4 so readily, why not only offer
WebRTC through IPv6? It's a huge double standard to say that we won't
support privacy issues of one network while at the same time
supporting connectivity though it.

>
> ps. You can also gather all these addresses for any browser with Flash
> Player installed by asking Flash to connect via RTMFP to a server, whereupon
> it will report the full enumeration of available IPv4 and IPv6 addresses to
> that server.

Yes, however flash can easily be blocked using click-to-enable (i.e.
user consent) and will not be installed on the majority of modern
browsers (mobile devices). WebRTC will be. What I'm asking for here is
a similar flash-blocking behavior to what is possible in Chromium
(built-in setting) and Firefox (via FlashBlock addons), so that users
can choose to click-to-enable based on context (just like they need to
already in getUserMedia).

On Sun, Jul 12, 2015 at 1:49 AM, Sergio Garcia Murillo
<sergio.garcia.murillo@gmail.com> wrote:
> Hi,
>
> While acknowledging that there might be a potential privacy issue, I don't
> think agree on your analysis on the issue that you have made nor your
> proposal.

I'd like to strongly encourage everyone here to stop treating this
issue as "might be a potential privacy issue". There have been
numerous news articles[1][2][3][4], discussion by
VPNs/Tor[5][6][7][8], my github proof-of-concept has over 1,500 stars
and 195 forks[9], the Mozilla Firefox bug has over 100 comments and is
marked as sec-moderate[10], and we are now seeing it exploited in the
wild to detect VPN users[11]. This is not "might be a potential
privacy issue". This is an existing, significant privacy issue.

[1]: https://torrentfreak.com/huge-security-flaw-leaks-vpn-users-real-ip-addresses-150130/
[2]: http://lifehacker.com/how-to-see-if-your-vpn-is-leaking-your-ip-address-and-1685180082
[3]: https://threatpost.com/webrtc-found-leaking-local-ip-addresses/110803
[4]: http://news.softpedia.com/news/WebRTC-in-Firefox-and-Chrome-Reveals-IPs-Behind-VPN-471881.shtml
[5]: https://www.expressvpn.com/blog/explaining-webrtc-ip-leak-vulnerability-affecting-web-browsers/
[6]: https://trac.torproject.org/projects/tor/ticket/5578
[7]: https://ipleak.net/
[8]: https://www.privateinternetaccess.com/forum/discussion/8204/
[9]: https://github.com/diafygi/webrtc-ips
[10]: https://bugzilla.mozilla.org/show_bug.cgi?id=959893
[11]: https://twitter.com/incloud/status/619624021123010560

>
> The information available in the ICE gathering process and made available to
> the JS app is the following:
>
> -Host candidates, mostly private IP addresses
> -Server reflexive candidates, exposing the public IP address as shown by the
> TURN/STUN server
> -Relay candidates, hosted by the TURN server
>
> My understanding of the issue is that the leak happens when the application
> is able to retrieve information not already available to it.
> That happens either when the user is using a VPN and have multiple public IP
> addresses or using an external PROXY so the http/ws connection have a
> different originator public IP address. AFAIK there should be no potential
> danger in leaking the private IP4 addresses (here you got mine, in case you
> are interested 127.0.0.1 and 192.168.64.2 ;). I have not much background on
> IPv6, so please complete any missing info and how would it affect here.

If you read the above articles and discussion surrounding the leak.
The concern is not around local or private IPs, it's around public IPs
for VPN users. Personal VPN use is extremely widespread
internationally, mostly for censorship and content-blocking
circumvention. Chinese users use them to get around the great
firewall. Australian users use them to be able to watch Netflix. Tor
is often not a suitable option because the web service they are trying
to reach blocks Tor (e.g. bank website) or is too data heavy (e.g.
streaming video).

>From the popularity of the issue, it's clear that the vast majority of
users thought that using a VPN hid their real IP address from the
websites they were visiting. It has been a huge shock (including for
me) that default VPN configurations let the browser (and thus WebRC)
know the real IP. Is that an operating system or VPN software issue?
Either way, due to WebRTC this real IP access issue has now spread to
javascript, so that any website can access your real IP.

"AFAIK there should be no potential danger in leaking the private IP4 addresses"

Here's a real-life example of the danger this has. China started
injecting malicious javascript into people's Baidu searches when those
searches came from outside China[12]. This was used to DDOS Github,
but what if that javascript contained a webrtc data channel setup? All
of a sudden, China could obtain the real Chinese IPs of citizens who
were using VPNs. The result of this could definitely be
life-threatening. Again, just saying "use Tor" means those citizens
cannot use many modern internet services (especially streaming video).
Personal VPNs are a fantastic compromise that provide access at quite
good speeds while retaining a good-enough-for-Netflix level or
privacy. WebRTC's real IP leaking removes this middle ground
compromise.

[12]: http://arstechnica.com/security/2015/03/github-battles-largest-ddos-in-sites-history-targeted-at-anti-censorship-tools/

>
> I don't think it is a good idea to tight the solution of requesting user
> permission, to not leak IP information, as theoretically a user could be
> interested in using webrtc (audio/video/datachannels) and still don't
> disclose their public IP address (i.e. forcing all the data to go via a TURN
> server).
>
> One possible solution would be that the browser don't gather/announce
> host&reflexive candidates when an HTTP proxy is set, and only use turn relay
> candidates instead. The second solution (which I think it is already
> discussed somewhere else) would be to allow the user to configure a TURN
> server to override the app settings and force all connection to go through
> it. (Then build a chained-TURN infrastructure into TOR and you could have a
> killer combo ;)

I'm fully supportive of removing the real IPs from the onicecandidate
results. However, as we saw in April's discussion thread and now in
this thread, identifying that IP is a complex topic that likely
doesn't have an easy answer and might require significant browser or
operating system-level changes. So, I'm suggesting here that an easier
solution is to make data channels require user-consent (again, just
like what already happens with audio and video channels).

>
> Regarding the issue that the browser becomes a CDN peer, not sure if anyone
> has ever raised the issue before. From my point of view, there is not, but
> in any case, I think it is orthogonal to the leak issue.

It is orthogonal to the privacy issue, but this user-consent proposal
also helps address this issue, too.

Thanks again for the responses!
Daniel


From nobody Sun Jul 12 12:36:04 2015
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On Sun, 12 Jul 2015 12:19:03 -0700
Daniel Roesler <diafygi@gmail.com> wrote:

> Thanks very much for the responses! Going to try and address both inline below.
> 
> On Sat, Jul 11, 2015 at 9:26 PM, Matthew Kaufman <matthew@matthew.at> wrote:
> >
> > On the IPv6 Internet, the IP address you use to reach the web site is almost
> > certainly the same as your "local" IP address. There's no additional
> > information exposed by allowing an application to discover that information
> > directly via JavaScript.
> >
> > The IPv4 Internet is essentially out of addresses and in the process of
> > being retired. I don't believe there's any reason at this point to disable
> > functionality in order to improve compatibility with this legacy network.
> >
> 
> I disagree with this sentiment. IPv4 will be around for many more
> years and many ISPs still do not assign their customers IPv6
> addresses. If IPv6 is replacing IPv4 so readily, why not only offer
> WebRTC through IPv6? It's a huge double standard to say that we won't
> support privacy issues of one network while at the same time
> supporting connectivity though it.
> 
> >
> > ps. You can also gather all these addresses for any browser with Flash
> > Player installed by asking Flash to connect via RTMFP to a server, whereupon
> > it will report the full enumeration of available IPv4 and IPv6 addresses to
> > that server.
> 
> Yes, however flash can easily be blocked using click-to-enable (i.e.
> user consent) and will not be installed on the majority of modern
> browsers (mobile devices). WebRTC will be. What I'm asking for here is
> a similar flash-blocking behavior to what is possible in Chromium
> (built-in setting) and Firefox (via FlashBlock addons), so that users
> can choose to click-to-enable based on context (just like they need to
> already in getUserMedia).
> 
> On Sun, Jul 12, 2015 at 1:49 AM, Sergio Garcia Murillo
> <sergio.garcia.murillo@gmail.com> wrote:
> > Hi,
> >
> > While acknowledging that there might be a potential privacy issue, I don't
> > think agree on your analysis on the issue that you have made nor your
> > proposal.
> 
> I'd like to strongly encourage everyone here to stop treating this
> issue as "might be a potential privacy issue". There have been
> numerous news articles[1][2][3][4], discussion by
> VPNs/Tor[5][6][7][8], my github proof-of-concept has over 1,500 stars
> and 195 forks[9], the Mozilla Firefox bug has over 100 comments and is
> marked as sec-moderate[10], and we are now seeing it exploited in the
> wild to detect VPN users[11]. This is not "might be a potential
> privacy issue". This is an existing, significant privacy issue.
> 
> [1]: https://torrentfreak.com/huge-security-flaw-leaks-vpn-users-real-ip-addresses-150130/
> [2]: http://lifehacker.com/how-to-see-if-your-vpn-is-leaking-your-ip-address-and-1685180082
> [3]: https://threatpost.com/webrtc-found-leaking-local-ip-addresses/110803
> [4]: http://news.softpedia.com/news/WebRTC-in-Firefox-and-Chrome-Reveals-IPs-Behind-VPN-471881.shtml
> [5]: https://www.expressvpn.com/blog/explaining-webrtc-ip-leak-vulnerability-affecting-web-browsers/
> [6]: https://trac.torproject.org/projects/tor/ticket/5578
> [7]: https://ipleak.net/
> [8]: https://www.privateinternetaccess.com/forum/discussion/8204/
> [9]: https://github.com/diafygi/webrtc-ips
> [10]: https://bugzilla.mozilla.org/show_bug.cgi?id=959893
> [11]: https://twitter.com/incloud/status/619624021123010560
> 
> >
> > The information available in the ICE gathering process and made available to
> > the JS app is the following:
> >
> > -Host candidates, mostly private IP addresses
> > -Server reflexive candidates, exposing the public IP address as shown by the
> > TURN/STUN server
> > -Relay candidates, hosted by the TURN server
> >
> > My understanding of the issue is that the leak happens when the application
> > is able to retrieve information not already available to it.
> > That happens either when the user is using a VPN and have multiple public IP
> > addresses or using an external PROXY so the http/ws connection have a
> > different originator public IP address. AFAIK there should be no potential
> > danger in leaking the private IP4 addresses (here you got mine, in case you
> > are interested 127.0.0.1 and 192.168.64.2 ;). I have not much background on
> > IPv6, so please complete any missing info and how would it affect here.
> 
> If you read the above articles and discussion surrounding the leak.
> The concern is not around local or private IPs, it's around public IPs
> for VPN users. Personal VPN use is extremely widespread
> internationally, mostly for censorship and content-blocking
> circumvention. Chinese users use them to get around the great
> firewall. Australian users use them to be able to watch Netflix. Tor
> is often not a suitable option because the web service they are trying
> to reach blocks Tor (e.g. bank website) or is too data heavy (e.g.
> streaming video).
> 
> >From the popularity of the issue, it's clear that the vast majority of
> users thought that using a VPN hid their real IP address from the
> websites they were visiting. It has been a huge shock (including for
> me) that default VPN configurations let the browser (and thus WebRC)
> know the real IP. Is that an operating system or VPN software issue?
> Either way, due to WebRTC this real IP access issue has now spread to
> javascript, so that any website can access your real IP.
> 
> "AFAIK there should be no potential danger in leaking the private IP4 addresses"
> 
> Here's a real-life example of the danger this has. China started
> injecting malicious javascript into people's Baidu searches when those
> searches came from outside China[12]. This was used to DDOS Github,
> but what if that javascript contained a webrtc data channel setup? All
> of a sudden, China could obtain the real Chinese IPs of citizens who
> were using VPNs. The result of this could definitely be
> life-threatening. Again, just saying "use Tor" means those citizens
> cannot use many modern internet services (especially streaming video).
> Personal VPNs are a fantastic compromise that provide access at quite
> good speeds while retaining a good-enough-for-Netflix level or
> privacy. WebRTC's real IP leaking removes this middle ground
> compromise.
> 
> [12]: http://arstechnica.com/security/2015/03/github-battles-largest-ddos-in-sites-history-targeted-at-anti-censorship-tools/
> 
> >
> > I don't think it is a good idea to tight the solution of requesting user
> > permission, to not leak IP information, as theoretically a user could be
> > interested in using webrtc (audio/video/datachannels) and still don't
> > disclose their public IP address (i.e. forcing all the data to go via a TURN
> > server).
> >
> > One possible solution would be that the browser don't gather/announce
> > host&reflexive candidates when an HTTP proxy is set, and only use turn relay
> > candidates instead. The second solution (which I think it is already
> > discussed somewhere else) would be to allow the user to configure a TURN
> > server to override the app settings and force all connection to go through
> > it. (Then build a chained-TURN infrastructure into TOR and you could have a
> > killer combo ;)
> 
> I'm fully supportive of removing the real IPs from the onicecandidate
> results. However, as we saw in April's discussion thread and now in
> this thread, identifying that IP is a complex topic that likely
> doesn't have an easy answer and might require significant browser or
> operating system-level changes. So, I'm suggesting here that an easier
> solution is to make data channels require user-consent (again, just
> like what already happens with audio and video channels).
> 


I don't see how that would fix anything. Setting up a recvonly audio/video PeerConnection won't get you any user-consent request either.

Forcing TURN in those cases seems more like it.

Lorenzo


> >
> > Regarding the issue that the browser becomes a CDN peer, not sure if anyone
> > has ever raised the issue before. From my point of view, there is not, but
> > in any case, I think it is orthogonal to the leak issue.
> 
> It is orthogonal to the privacy issue, but this user-consent proposal
> also helps address this issue, too.
> 
> Thanks again for the responses!
> Daniel
> 
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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> On 12 Jul 2015, at 21:19, Daniel Roesler <diafygi@gmail.com> wrote:
>=20
> Thanks very much for the responses! Going to try and address both =
inline below.
>=20
> On Sat, Jul 11, 2015 at 9:26 PM, Matthew Kaufman <matthew@matthew.at> =
wrote:
>>=20
>> On the IPv6 Internet, the IP address you use to reach the web site is =
almost
>> certainly the same as your "local" IP address. There's no additional
>> information exposed by allowing an application to discover that =
information
>> directly via JavaScript.
>>=20
>> The IPv4 Internet is essentially out of addresses and in the process =
of
>> being retired. I don't believe there's any reason at this point to =
disable
>> functionality in order to improve compatibility with this legacy =
network.
>>=20
>=20
> I disagree with this sentiment. IPv4 will be around for many more
> years and many ISPs still do not assign their customers IPv6
> addresses. If IPv6 is replacing IPv4 so readily, why not only offer
> WebRTC through IPv6? It's a huge double standard to say that we won't
> support privacy issues of one network while at the same time
> supporting connectivity though it.
>=20
>>=20
>> ps. You can also gather all these addresses for any browser with =
Flash
>> Player installed by asking Flash to connect via RTMFP to a server, =
whereupon
>> it will report the full enumeration of available IPv4 and IPv6 =
addresses to
>> that server.
>=20
> Yes, however flash can easily be blocked using click-to-enable (i.e.
> user consent) and will not be installed on the majority of modern
> browsers (mobile devices). WebRTC will be. What I'm asking for here is
> a similar flash-blocking behavior to what is possible in Chromium
> (built-in setting) and Firefox (via FlashBlock addons), so that users
> can choose to click-to-enable based on context (just like they need to
> already in getUserMedia).
>=20
> On Sun, Jul 12, 2015 at 1:49 AM, Sergio Garcia Murillo
> <sergio.garcia.murillo@gmail.com> wrote:
>> Hi,
>>=20
>> While acknowledging that there might be a potential privacy issue, I =
don't
>> think agree on your analysis on the issue that you have made nor your
>> proposal.
>=20
> I'd like to strongly encourage everyone here to stop treating this
> issue as "might be a potential privacy issue". There have been
> numerous news articles[1][2][3][4], discussion by
> VPNs/Tor[5][6][7][8], my github proof-of-concept has over 1,500 stars
> and 195 forks[9], the Mozilla Firefox bug has over 100 comments and is
> marked as sec-moderate[10], and we are now seeing it exploited in the
> wild to detect VPN users[11]. This is not "might be a potential
> privacy issue". This is an existing, significant privacy issue.
>=20
> [1]: =
https://torrentfreak.com/huge-security-flaw-leaks-vpn-users-real-ip-addres=
ses-150130/
> [2]: =
http://lifehacker.com/how-to-see-if-your-vpn-is-leaking-your-ip-address-an=
d-1685180082
> [3]: =
https://threatpost.com/webrtc-found-leaking-local-ip-addresses/110803
> [4]: =
http://news.softpedia.com/news/WebRTC-in-Firefox-and-Chrome-Reveals-IPs-Be=
hind-VPN-471881.shtml
> [5]: =
https://www.expressvpn.com/blog/explaining-webrtc-ip-leak-vulnerability-af=
fecting-web-browsers/
> [6]: https://trac.torproject.org/projects/tor/ticket/5578
> [7]: https://ipleak.net/
> [8]: https://www.privateinternetaccess.com/forum/discussion/8204/
> [9]: https://github.com/diafygi/webrtc-ips
> [10]: https://bugzilla.mozilla.org/show_bug.cgi?id=3D959893
> [11]: https://twitter.com/incloud/status/619624021123010560
>=20
>>=20
>> The information available in the ICE gathering process and made =
available to
>> the JS app is the following:
>>=20
>> -Host candidates, mostly private IP addresses
>> -Server reflexive candidates, exposing the public IP address as shown =
by the
>> TURN/STUN server
>> -Relay candidates, hosted by the TURN server
>>=20
>> My understanding of the issue is that the leak happens when the =
application
>> is able to retrieve information not already available to it.
>> That happens either when the user is using a VPN and have multiple =
public IP
>> addresses or using an external PROXY so the http/ws connection have a
>> different originator public IP address. AFAIK there should be no =
potential
>> danger in leaking the private IP4 addresses (here you got mine, in =
case you
>> are interested 127.0.0.1 and 192.168.64.2 ;). I have not much =
background on
>> IPv6, so please complete any missing info and how would it affect =
here.
>=20
> If you read the above articles and discussion surrounding the leak.
> The concern is not around local or private IPs, it's around public IPs
> for VPN users. Personal VPN use is extremely widespread
> internationally, mostly for censorship and content-blocking
> circumvention. Chinese users use them to get around the great
> firewall. Australian users use them to be able to watch Netflix. Tor
> is often not a suitable option because the web service they are trying
> to reach blocks Tor (e.g. bank website) or is too data heavy (e.g.
> streaming video).
>=20
>> =46rom the popularity of the issue, it's clear that the vast majority =
of
> users thought that using a VPN hid their real IP address from the
> websites they were visiting. It has been a huge shock (including for
> me) that default VPN configurations let the browser (and thus WebRC)
> know the real IP. Is that an operating system or VPN software issue?
> Either way, due to WebRTC this real IP access issue has now spread to
> javascript, so that any website can access your real IP.
>=20
> "AFAIK there should be no potential danger in leaking the private IP4 =
addresses"
>=20
> Here's a real-life example of the danger this has. China started
> injecting malicious javascript into people's Baidu searches when those
> searches came from outside China[12]. This was used to DDOS Github,
> but what if that javascript contained a webrtc data channel setup? All
> of a sudden, China could obtain the real Chinese IPs of citizens who
> were using VPNs. The result of this could definitely be
> life-threatening. Again, just saying "use Tor" means those citizens
> cannot use many modern internet services (especially streaming video).
> Personal VPNs are a fantastic compromise that provide access at quite
> good speeds while retaining a good-enough-for-Netflix level or
> privacy. WebRTC's real IP leaking removes this middle ground
> compromise.
>=20
> [12]: =
http://arstechnica.com/security/2015/03/github-battles-largest-ddos-in-sit=
es-history-targeted-at-anti-censorship-tools/
>=20
>>=20
>> I don't think it is a good idea to tight the solution of requesting =
user
>> permission, to not leak IP information, as theoretically a user could =
be
>> interested in using webrtc (audio/video/datachannels) and still don't
>> disclose their public IP address (i.e. forcing all the data to go via =
a TURN
>> server).
>>=20
>> One possible solution would be that the browser don't gather/announce
>> host&reflexive candidates when an HTTP proxy is set, and only use =
turn relay
>> candidates instead. The second solution (which I think it is already
>> discussed somewhere else) would be to allow the user to configure a =
TURN
>> server to override the app settings and force all connection to go =
through
>> it. (Then build a chained-TURN infrastructure into TOR and you could =
have a
>> killer combo ;)
>=20
> I'm fully supportive of removing the real IPs from the onicecandidate
> results. However, as we saw in April's discussion thread and now in
> this thread, identifying that IP is a complex topic that likely
> doesn't have an easy answer and might require significant browser or
> operating system-level changes. So, I'm suggesting here that an easier
> solution is to make data channels require user-consent (again, just
> like what already happens with audio and video channels).
>=20
>>=20
>> Regarding the issue that the browser becomes a CDN peer, not sure if =
anyone
>> has ever raised the issue before. =46rom my point of view, there is =
not, but
>> in any case, I think it is orthogonal to the leak issue.
>=20
> It is orthogonal to the privacy issue, but this user-consent proposal
> also helps address this issue, too.
Don't you ask for user consent for a peer connection?

Best regards
Michael
>=20
> Thanks again for the responses!
> Daniel
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


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> On 12 Jul 2015, at 20:19, Daniel Roesler <diafygi@gmail.com> wrote:
>=20
> If you read the above articles and discussion surrounding the leak.
> The concern is not around local or private IPs, it's around public IPs
> for VPN users. Personal VPN use is extremely widespread
> internationally, mostly for censorship and content-blocking
> circumvention. Chinese users use them to get around the great
> firewall. Australian users use them to be able to watch Netflix. Tor
> is often not a suitable option because the web service they are trying
> to reach blocks Tor (e.g. bank website) or is too data heavy (e.g.
> streaming video).

The problem is that those VPNs are misconfigured.
If you want your VPN to ensure you don=E2=80=99t reveal the ISP provided =
public IP,
then you shouldn=E2=80=99t allow it to be the default route. You set the =
default=20
route to the VPN - and then add a specific route to let just the VPN =
packets flow=20
out with that IP.

If you do this correctly the STUN address webRTC gets is that of the far =
end of the=20
VPN tunnel - which your website already knew.

I don=E2=80=99t think that just because some VPN service vendors can=E2=80=
=99t update their=20
scripts correctly we should freeze web protocols.

This doesn=E2=80=99t help http proxy services of course, but that =
isn=E2=80=99t a VPN.

Tim.=

--Apple-Mail=_D05EBB11-E9B6-45B5-A84E-CA1C149CA6BD
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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dutf-8"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" =
class=3D""><br class=3D""><div><blockquote type=3D"cite" class=3D""><div =
class=3D"">On 12 Jul 2015, at 20:19, Daniel Roesler &lt;<a =
href=3D"mailto:diafygi@gmail.com" class=3D"">diafygi@gmail.com</a>&gt; =
wrote:</div><br class=3D"Apple-interchange-newline"><div class=3D""><span =
style=3D"font-family: Helvetica; font-size: 12px; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; =
line-height: normal; orphans: auto; text-align: start; text-indent: 0px; =
text-transform: none; white-space: normal; widows: auto; word-spacing: =
0px; -webkit-text-stroke-width: 0px; float: none; display: inline =
!important;" class=3D"">If you read the above articles and discussion =
surrounding the leak.</span><br style=3D"font-family: Helvetica; =
font-size: 12px; font-style: normal; font-variant: normal; font-weight: =
normal; letter-spacing: normal; line-height: normal; orphans: auto; =
text-align: start; text-indent: 0px; text-transform: none; white-space: =
normal; widows: auto; word-spacing: 0px; -webkit-text-stroke-width: =
0px;" class=3D""><span style=3D"font-family: Helvetica; font-size: 12px; =
font-style: normal; font-variant: normal; font-weight: normal; =
letter-spacing: normal; line-height: normal; orphans: auto; text-align: =
start; text-indent: 0px; text-transform: none; white-space: normal; =
widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px; float: =
none; display: inline !important;" class=3D"">The concern is not around =
local or private IPs, it's around public IPs</span><br =
style=3D"font-family: Helvetica; font-size: 12px; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; =
line-height: normal; orphans: auto; text-align: start; text-indent: 0px; =
text-transform: none; white-space: normal; widows: auto; word-spacing: =
0px; -webkit-text-stroke-width: 0px;" class=3D""><span =
style=3D"font-family: Helvetica; font-size: 12px; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; =
line-height: normal; orphans: auto; text-align: start; text-indent: 0px; =
text-transform: none; white-space: normal; widows: auto; word-spacing: =
0px; -webkit-text-stroke-width: 0px; float: none; display: inline =
!important;" class=3D"">for VPN users. Personal VPN use is extremely =
widespread</span><br style=3D"font-family: Helvetica; font-size: 12px; =
font-style: normal; font-variant: normal; font-weight: normal; =
letter-spacing: normal; line-height: normal; orphans: auto; text-align: =
start; text-indent: 0px; text-transform: none; white-space: normal; =
widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px;" =
class=3D""><span style=3D"font-family: Helvetica; font-size: 12px; =
font-style: normal; font-variant: normal; font-weight: normal; =
letter-spacing: normal; line-height: normal; orphans: auto; text-align: =
start; text-indent: 0px; text-transform: none; white-space: normal; =
widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px; float: =
none; display: inline !important;" class=3D"">internationally, mostly =
for censorship and content-blocking</span><br style=3D"font-family: =
Helvetica; font-size: 12px; font-style: normal; font-variant: normal; =
font-weight: normal; letter-spacing: normal; line-height: normal; =
orphans: auto; text-align: start; text-indent: 0px; text-transform: =
none; white-space: normal; widows: auto; word-spacing: 0px; =
-webkit-text-stroke-width: 0px;" class=3D""><span style=3D"font-family: =
Helvetica; font-size: 12px; font-style: normal; font-variant: normal; =
font-weight: normal; letter-spacing: normal; line-height: normal; =
orphans: auto; text-align: start; text-indent: 0px; text-transform: =
none; white-space: normal; widows: auto; word-spacing: 0px; =
-webkit-text-stroke-width: 0px; float: none; display: inline =
!important;" class=3D"">circumvention. Chinese users use them to get =
around the great</span><br style=3D"font-family: Helvetica; font-size: =
12px; font-style: normal; font-variant: normal; font-weight: normal; =
letter-spacing: normal; line-height: normal; orphans: auto; text-align: =
start; text-indent: 0px; text-transform: none; white-space: normal; =
widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px;" =
class=3D""><span style=3D"font-family: Helvetica; font-size: 12px; =
font-style: normal; font-variant: normal; font-weight: normal; =
letter-spacing: normal; line-height: normal; orphans: auto; text-align: =
start; text-indent: 0px; text-transform: none; white-space: normal; =
widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px; float: =
none; display: inline !important;" class=3D"">firewall. Australian users =
use them to be able to watch Netflix. Tor</span><br style=3D"font-family: =
Helvetica; font-size: 12px; font-style: normal; font-variant: normal; =
font-weight: normal; letter-spacing: normal; line-height: normal; =
orphans: auto; text-align: start; text-indent: 0px; text-transform: =
none; white-space: normal; widows: auto; word-spacing: 0px; =
-webkit-text-stroke-width: 0px;" class=3D""><span style=3D"font-family: =
Helvetica; font-size: 12px; font-style: normal; font-variant: normal; =
font-weight: normal; letter-spacing: normal; line-height: normal; =
orphans: auto; text-align: start; text-indent: 0px; text-transform: =
none; white-space: normal; widows: auto; word-spacing: 0px; =
-webkit-text-stroke-width: 0px; float: none; display: inline =
!important;" class=3D"">is often not a suitable option because the web =
service they are trying</span><br style=3D"font-family: Helvetica; =
font-size: 12px; font-style: normal; font-variant: normal; font-weight: =
normal; letter-spacing: normal; line-height: normal; orphans: auto; =
text-align: start; text-indent: 0px; text-transform: none; white-space: =
normal; widows: auto; word-spacing: 0px; -webkit-text-stroke-width: =
0px;" class=3D""><span style=3D"font-family: Helvetica; font-size: 12px; =
font-style: normal; font-variant: normal; font-weight: normal; =
letter-spacing: normal; line-height: normal; orphans: auto; text-align: =
start; text-indent: 0px; text-transform: none; white-space: normal; =
widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px; float: =
none; display: inline !important;" class=3D"">to reach blocks Tor (e.g. =
bank website) or is too data heavy (e.g.</span><br style=3D"font-family: =
Helvetica; font-size: 12px; font-style: normal; font-variant: normal; =
font-weight: normal; letter-spacing: normal; line-height: normal; =
orphans: auto; text-align: start; text-indent: 0px; text-transform: =
none; white-space: normal; widows: auto; word-spacing: 0px; =
-webkit-text-stroke-width: 0px;" class=3D""><span style=3D"font-family: =
Helvetica; font-size: 12px; font-style: normal; font-variant: normal; =
font-weight: normal; letter-spacing: normal; line-height: normal; =
orphans: auto; text-align: start; text-indent: 0px; text-transform: =
none; white-space: normal; widows: auto; word-spacing: 0px; =
-webkit-text-stroke-width: 0px; float: none; display: inline =
!important;" class=3D"">streaming =
video).</span></div></blockquote></div><br class=3D""><div class=3D"">The =
problem is that those VPNs are misconfigured.</div><div class=3D"">If =
you want your VPN to ensure you don=E2=80=99t reveal the ISP provided =
public IP,</div><div class=3D"">then you shouldn=E2=80=99t allow it to =
be the default route. You set the default&nbsp;</div><div class=3D"">route=
 to the VPN - and then add a specific route to let just the VPN packets =
flow&nbsp;</div><div class=3D"">out with that IP.</div><div class=3D""><br=
 class=3D""></div><div class=3D"">If you do this correctly the STUN =
address webRTC gets is that of the far end of the&nbsp;</div><div =
class=3D"">VPN tunnel - which your website already knew.</div><div =
class=3D""><br class=3D""></div><div class=3D"">I don=E2=80=99t think =
that just because some VPN service vendors can=E2=80=99t update =
their&nbsp;</div><div class=3D"">scripts correctly we should freeze web =
protocols.</div><div class=3D""><br class=3D""></div><div class=3D"">This =
doesn=E2=80=99t help http proxy services of course, but that isn=E2=80=99t=
 a VPN.</div><div class=3D""><br class=3D""></div><div =
class=3D"">Tim.</div></body></html>=

--Apple-Mail=_D05EBB11-E9B6-45B5-A84E-CA1C149CA6BD--


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From: Bernard Aboba <bernard.aboba@gmail.com>
Date: Sun, 12 Jul 2015 15:02:58 -0700
To: Daniel Roesler <diafygi@gmail.com>
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Subject: Re: [rtcweb] Please require user consent for data channels
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Consent is an API issue not a protocol and therefore is owned by W3C, not IE=
TF. Why not take it up there?



> On Jul 11, 2015, at 9:42 AM, Daniel Roesler <diafygi@gmail.com> wrote:
>=20
> Howdy all, this is mostly a re-surfacing of the discussion about IP
> address leaking back in April[1], which unfortunately I did not
> discover until recently.
>=20
> One of the items in the new proposal was "WebRTC already requires
> permission to access getUserMedia. Why not use that permission to
> control interface enumeration?" That item didn't really get discussed
> much in the thread, but I think it's one of the most important issues.
>=20
> Why? There is now a documented case where a third party on nytimes.com
> is using a fake webRTC datachannel to silently gather user local (and
> potentially "real" ISP) IP addresses.
>=20
> https://twitter.com/incloud/status/619624021123010560
>=20
> So I'd like to voice my support for adding a consent requirement that
> would prevent this type of silent behavior. A user that is
> purposefully visiting a site that has a legitimate reason for using a
> webRTC data channel will have the necessary context to give consent,
> just like they have to get consent on getUserMedia.
>=20
> I'm really unclear on why it's so important to have the silent webRTC
> data channel behavior. P2P data connections should be held to the same
> consent requirements just like P2P video and voice connections. In
> addition to the silent third party local IP tracking, a silent P2P
> data channel opens up users to silently becoming nodes in a web-based
> file sharing network. For example, webtorrent[2] can be silently
> embedded on a website so that all of the users to that website start
> seeding content they don't know they are seeding.
>=20
> Possible fix:
>=20
> 1. Require user consent before calling the callback on createOffer().
> 2. If the user has already given consent via getUserMedia, the user
> consent requirement is satisfied.
> 3. If there's no previous gUM consent, then ask the user for consent.
>=20
> This implementation requires no updates to current webRTC demos, since
> you have to wait for a callback on createOffer anyway. I'm open to
> other ways, such as making createDataChannel asynchronous like gUM,
> but that would require updates to existing apps that use
> createDataChannel.
>=20
> Thanks!
> Daniel Roesler
>=20
> [1]: https://www.ietf.org/mail-archive/web/rtcweb/current/msg14494.htm
> [2]: https://github.com/feross/webtorrent
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From nobody Mon Jul 13 08:16:42 2015
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On 7/12/2015 12:19 PM, Daniel Roesler wrote:
> Thanks very much for the responses! Going to try and address both inline below.
>
> On Sat, Jul 11, 2015 at 9:26 PM, Matthew Kaufman <matthew@matthew.at> wrote:
>> On the IPv6 Internet, the IP address you use to reach the web site is almost
>> certainly the same as your "local" IP address. There's no additional
>> information exposed by allowing an application to discover that information
>> directly via JavaScript.
>>
>> The IPv4 Internet is essentially out of addresses and in the process of
>> being retired. I don't believe there's any reason at this point to disable
>> functionality in order to improve compatibility with this legacy network.
>>
> I disagree with this sentiment. IPv4 will be around for many more
> years

Maybe.

> and many ISPs still do not assign their customers IPv6
> addresses.

Only ones that don't expect to stay in business.

>   If IPv6 is replacing IPv4 so readily, why not only offer
> WebRTC through IPv6?

I would support such a proposal.

> It's a huge double standard to say that we won't
> support privacy issues of one network while at the same time
> supporting connectivity though it.

IPv6 won't have that "privacy", so there's no reason for IPv4 users to 
continue to expect such.

Matthew Kaufman


From nobody Mon Jul 13 11:55:00 2015
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Michael Tuexen wrote:
> Don't you ask for user consent for a peer connection?

No.

Since there seems to be some misunderstanding on this thread: consent is 
required for getUserMedia (camera/microphone) access. This is true 
regardless of whether or not you send that data anywhere over the 
network. User consent is not required for a peer connection (since you 
cannot use it to send any data that you could not otherwise have just 
relayed through the server, e.g., via WebSockets).

Consent *is* required by the other side to continue receiving your data 
(to prevent DDoS attacks), but that is automatic and does not require 
user intervention (other than viewing a page/running a program designed 
to operate the other end of the PeerConnection).

These issues are all detailed in 
<https://tools.ietf.org/html/draft-ietf-rtcweb-security-arch>. I would 
encourage those participating in this thread to read that document. We 
could always benefit from additional review.


From nobody Mon Jul 13 15:14:48 2015
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Subject: Re: [rtcweb] Agenda time for draft-jennings-behave-rtcweb-firewall-00 ?
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I have read this draft and the latest draft-ietf-rtcweb-return-00 and =
draft-ietf-tram-turn-server-discovery-03 drafts from the point of view =
of finally getting a good and working solution to the old NAT/Firewall =
traversal issue with real-time communication, here specifically WebRTC =
using ICE.

=20

I find all these drafts to be in the right direction and should be =
worked trough to RFCs.

=20

Regarding the comment on Section 6 (=E2=80=9CThe IETF is designing =
systems to send interactive audio and video such that it looks like =
HTTPS and HTTP to the proxies and firewalls =E2=80=9C) and below:

> extremely unlikely that we would implement it.

=20

I hope this means that the IETF should go no further in the direction of =
=E2=80=9Cfooling real-time traffic=E2=80=9D through firewalls, but =
rather make RFCs that can allows good traversal of real-time traffic for =
users on networks that want such traffic, and that we realize:

=20

i) The network administrator wanting to block or make real-time traffic =
bad and useless can always do that (exemplified in this behave draft, =
but maybe the intent has to be clarified). And that taking further steps =
in the direction of =E2=80=9Cfooling real-time traffic=E2=80=9D through =
firewalls =E2=80=9Chas some risk of impacting normal HTTP traffic, so =
there is a desire to provide guidance around ways to do that in this =
draft.=E2=80=9D  (quote from this behave draft)

=20

ii) There may be legitimate reasons to e.g. allow surfing, but not voice =
on a network. We cannot/should not rule against that, even if we all are =
for net neutrality and against evil purposes for blocking e.g. voice and =
SIP (which still sometimes seem to be the case). The network admin and =
the firewall are in charge. If we want to use good real-time =
communication, we have to use networks that wants to provide it =
=E2=80=93 It should be a selling feature for using the network access. =
Our efforts could possibly be advice how evil blocking could be =
exposed/displayed so users can select other network accesses for =
real-time communication.

=20

I have lost track of where these repeatedly popping up suggestions to =
=E2=80=9Cfool real-time traffic=E2=80=9D through the most restrictive =
firewalls are, but the only real solution is to make these firewalls =
less restrictive or to parallel them, which is what the RETURN + TURN =
server DISCOVERY draft may achieve in a good way.

=20

I think there could be (maybe is) consensus around this, and as a =
comment to this behave draft=E2=80=9D:

- Yes, it guides how a firewall could be less restrictive, but still =
pretty secure, and allow STUN-based traversal of  WebRTC real-time =
traffic through the network default gateway.

=20

However, that is not sufficient, it should in its =E2=80=9Cguidance to =
firewall vendors=E2=80=9D (which I believe the draft is), also point out =
the firewall traversal clarified in the RETURN draft (From the =
introduction):

=E2=80=9CThis TURN server may be placed inside the network, with a

   firewall configuration allowing it to communicate with the public

   internet, or it may be operated by a third party outside the network,

   with a firewall configuration that allows hosts inside the network to

   communicate with it.  Use of the specified TURN server may be the

   only way for clients on the network to achieve a high quality WebRTC

   experience.  This scenario is required to be supported by the WebRTC

   requirements document [I-D.ietf-rtcweb-use-cases-and-requirements]

   Section 3.3.5.1.=E2=80=9D

=20

A firewall (designed to be secure AND to handle real-time communication =
without having to be paralleled by an SBC or a TURN server) could very =
well include such TURN server (or TURN Proxy or Border TURN server as it =
is called in the RETURN draft).

=20

Also, a network service provider offering such (sealed) TURN server (or =
TURN Proxy or Border TURN server), typically has an enterprise firewall =
in front of it: The firewall vendor should then be adviced to recommend =
his customers to open a pinhole from the LAN to that TURN server (not to =
use an own additional border TURN server requiring double RETURN or =
something like that).

=20

So, I suggest to add guidance about these things to this behave draft.

=20

=20

Having expressed this, I believe the sealed, configuration and =
authentication concept of the RETURN draft, and possible the DISCOVERY =
draft needs to be clarified/extended, which will follow in next email.

=20

/Karl

=20

PS: I am separately posting this and the next email to the TRAM WG also, =
since they were also address about this draft and is responsible for the =
DISCOVERY draft

=20

=20

Fr=C3=A5n: rtcweb [mailto:rtcweb-bounces@ietf.org] F=C3=B6r Eric =
Rescorla
Skickat: den 8 juli 2015 16:40
Till: Cullen Jennings
Kopia: rtcweb@ietf.org
=C3=84mne: Re: [rtcweb] Agenda time for =
draft-jennings-behave-rtcweb-firewall-00 ?

=20

I have taken a brief look at this draft and I don't believe that the =
proposed pinhole/inspection

algorithm provides the desired assurances of verifying consent:

=20

1. If you open up an incoming/outgoing pinhole on outgoing packet, prior =
to

receiving an incoming STUN packet, then if the internal implementation

simply sends a new outgoing STUN packet every 5 seconds, then they

never need to rely on external consent.

=20

2. If you just require *a* incoming STUN packet to be received in order =
to

extend the lifetime to 30 seconds, then an off-path attacker can forge

a STUN response that will cause the pinhole to be open for 30 seconds.

(Hence the need for the STUN transaction ID).

=20

It seems to me that you should be instead requiring a complete STUN

transaction before allowing any non-STUN outgoing traffic. It would =
probably

be safe-ish to require it before allowing any incoming traffic as well, =
though

that leaves the chance that you get a small amount of clipping with =
packet

drops or reordering.

=20

=20

WRT to the origin attribute, I have mixed feelings. I agree that your =
SNI argument

is strong, but there is also a move to figure out how to encrypt SNI, so =
every time

we extend that argument it makes that harder. OTOH, STUN origin will be

sent a lot less often than SNI, I imagine.

=20

=20

I don't understand Section 6. Is this guidance to firewall vendors or to =
browser

implementors. If the former, the firewall has no way of knowing what =
"requests"

are if the connection use HTTPS (and if is uses HTTP, you have DPI-type =
methods for

determining whether something is voice or video. If this is guidance to =
browser

vendors, I consider it extremely unlikely that we would implement it.

=20

=20

Point #4 in your security considerations section seems like an entirely =
lost

cause, unless you assume that attackers are not really trying.

=20

-Ekr

=20

=20

=20

On Tue, Jul 7, 2015 at 8:50 AM, Cullen Jennings <fluffy@iii.ca> wrote:


I'd like to get a brief amount of agenda time to discuss changes we =
would need to some of the WebRTC WG drafts to make the ideas proposed in =
draft-jennings-behave-rtcweb-firewall- work and make people aware of =
this work.

Would it be possible to get 10 minutes on the agenda ?

Thanks, Cullen


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

=20


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<o:idmap v:ext=3D"edit" data=3D"1" />
</o:shapelayout></xml><![endif]--></head><body lang=3DSV link=3Dblue =
vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>I =
have read this draft and the latest </span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>draft-ietf-rt=
cweb-return-00<span style=3D'color:blue'> and =
</span>draft-ietf-tram-turn-server-discovery-03<span =
style=3D'color:blue'> drafts from the point of view of finally getting a =
good and working solution to the old NAT/Firewall traversal issue with =
real-time communication, here specifically WebRTC using =
ICE.<o:p></o:p></span></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>I =
find all these drafts to be in the right direction and should be worked =
trough to RFCs.<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Re=
garding the comment on Section 6 (=E2=80=9C</span><span lang=3DEN-US =
style=3D'font-family:"Courier New"'>The IETF is designing systems to =
send interactive audio and video such that it looks like HTTPS and HTTP =
to the proxies and firewalls</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'> =
=E2=80=9C) and below:</span><span lang=3DEN-US =
style=3D'font-family:"Courier New"'><o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&g=
t; </span><span lang=3DEN-US>extremely unlikely that we would implement =
it.<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>I =
hope this means that the IETF should go no further in the direction of =
=E2=80=9Cfooling real-time traffic=E2=80=9D through firewalls, but =
rather make RFCs that can allows good traversal of real-time traffic for =
users on networks that want such traffic, and that we =
realize:<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>i)=
 The network administrator wanting to block or make real-time traffic =
bad and useless can always do that (exemplified in this behave draft, =
but maybe the intent has to be clarified). And that taking further steps =
in the direction of =E2=80=9Cfooling real-time traffic=E2=80=9D through =
firewalls =E2=80=9C</span><span lang=3DEN-US =
style=3D'font-family:"Courier New"'>has some risk of impacting normal =
HTTP traffic, so there is a desire to provide guidance around ways to do =
that in this draft.</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>=E2=
=80=9D =C2=A0(quote from this behave draft)<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>ii=
) There may be legitimate reasons to e.g. allow surfing, but not voice =
on a network. We cannot/should not rule against that, even if we all are =
for net neutrality and against evil purposes for blocking e.g. voice and =
SIP (which still sometimes seem to be the case). The network admin and =
the firewall are in charge. If we want to use good real-time =
communication, we have to use networks that wants to provide it =
=E2=80=93 It should be a selling feature for using the network access. =
Our efforts could possibly be advice how evil blocking could be =
exposed/displayed so users can select other network accesses for =
real-time communication.<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>I =
have lost track of where these repeatedly popping up suggestions to =
=E2=80=9Cfool real-time traffic=E2=80=9D through the most restrictive =
firewalls are, but the only real solution is to make these firewalls =
less restrictive or to parallel them, which is what the RETURN + TURN =
server DISCOVERY draft may achieve in a good =
way.<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>I =
think there could be (maybe is) consensus around this, and as a comment =
to this behave draft=E2=80=9D:<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>- =
Yes, it guides how a firewall could be less restrictive, but still =
pretty secure, and allow STUN-based traversal of =C2=A0WebRTC real-time =
traffic through the network default gateway.<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Ho=
wever, that is not sufficient, it should in its =E2=80=9C</span><span =
lang=3DEN-US>guidance to firewall vendors=E2=80=9D (</span><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>wh=
ich I believe the draft is), also point out the firewall traversal =
clarified in the RETURN draft (From the =
introduction):<o:p></o:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>=E2=
=80=9C</span><span lang=3DEN-US style=3D'font-family:"Courier New"'>This =
TURN server may be placed inside the network, with =
a<o:p></o:p></span></p><p class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-family:"Courier New"'>=C2=A0=C2=A0 firewall configuration =
allowing it to communicate with the public<o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US style=3D'font-family:"Courier =
New"'>=C2=A0=C2=A0 internet, or it may be operated by a third party =
outside the network,<o:p></o:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US style=3D'font-family:"Courier New"'>=C2=A0=C2=A0 with a =
firewall configuration that allows hosts inside the network =
to<o:p></o:p></span></p><p class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-family:"Courier New"'>=C2=A0=C2=A0 communicate with =
it.=C2=A0 Use of the specified TURN server may be =
the<o:p></o:p></span></p><p class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-family:"Courier New"'>=C2=A0=C2=A0 only way for clients on =
the network to achieve a high quality WebRTC<o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US style=3D'font-family:"Courier =
New"'>=C2=A0=C2=A0 experience.=C2=A0 This scenario is required to be =
supported by the WebRTC<o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US style=3D'font-family:"Courier =
New"'>=C2=A0=C2=A0 requirements document =
[I-D.ietf-rtcweb-use-cases-and-requirements]<o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US style=3D'font-family:"Courier =
New"'>=C2=A0=C2=A0 Section 3.3.5.1.=E2=80=9D<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>A =
firewall (designed to be secure AND to handle real-time communication =
without having to be paralleled by an SBC or a TURN server) could very =
well include such TURN server (or TURN Proxy or Border TURN server as it =
is called in the RETURN draft).<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Al=
so, a network service provider offering such (sealed) TURN server (or =
TURN Proxy or Border TURN server), typically has an enterprise firewall =
in front of it: The firewall vendor should then be adviced to recommend =
his customers to open a pinhole from the LAN to that TURN server (not to =
use an own additional border TURN server requiring double RETURN or =
something like that).<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>So=
, I suggest to add guidance about these things to this behave =
draft.<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Ha=
ving expressed this, I believe the sealed, configuration and =
authentication concept of the RETURN draft, and possible the DISCOVERY =
draft needs to be clarified/extended, which will follow in next =
email.<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>/K=
arl<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>PS=
: I am separately posting this and the next email to the TRAM WG also, =
since they were also address about this draft and is responsible for the =
DISCOVERY draft<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><div style=3D'border:none;border-top:solid =
#B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm'><p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Fr=C3=A5n:</=
span></b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> rtcweb =
[mailto:rtcweb-bounces@ietf.org] <b>F=C3=B6r </b>Eric =
Rescorla<br><b>Skickat:</b> den 8 juli 2015 16:40<br><b>Till:</b> Cullen =
Jennings<br><b>Kopia:</b> rtcweb@ietf.org<br><b>=C3=84mne:</b> Re: =
[rtcweb] Agenda time for draft-jennings-behave-rtcweb-firewall-00 =
?<o:p></o:p></span></p></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><div><p class=3DMsoNormal>I have =
taken a brief look at this draft and I don't believe that the proposed =
pinhole/inspection<o:p></o:p></p><div><p class=3DMsoNormal>algorithm =
provides the desired assurances of verifying =
consent:<o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal>1. If you open up an incoming/outgoing pinhole on =
outgoing packet, prior to<o:p></o:p></p></div><div><p =
class=3DMsoNormal>receiving an incoming STUN packet, then if the =
internal implementation<o:p></o:p></p></div><div><p =
class=3DMsoNormal>simply sends a new outgoing STUN packet every 5 =
seconds, then they<o:p></o:p></p></div><div><p class=3DMsoNormal>never =
need to rely on external consent.<o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal>2. If you just require *a* incoming STUN packet to be =
received in order to<o:p></o:p></p></div><div><p =
class=3DMsoNormal>extend the lifetime to 30 seconds, then an off-path =
attacker can forge<o:p></o:p></p></div><div><p class=3DMsoNormal>a STUN =
response that will cause the pinhole to be open for 30 =
seconds.<o:p></o:p></p></div><div><p class=3DMsoNormal>(Hence the need =
for the STUN transaction ID).<o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal>It seems to me that you should be instead requiring a =
complete STUN<o:p></o:p></p></div><div><p class=3DMsoNormal>transaction =
before allowing any non-STUN outgoing traffic. It would =
probably<o:p></o:p></p></div><div><p class=3DMsoNormal>be safe-ish to =
require it before allowing any incoming traffic as well, =
though<o:p></o:p></p></div><div><p class=3DMsoNormal>that leaves the =
chance that you get a small amount of clipping with =
packet<o:p></o:p></p></div><div><p class=3DMsoNormal>drops or =
reordering.<o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal>WRT to the origin attribute, I have mixed feelings. I =
agree that your SNI argument<o:p></o:p></p></div><div><p =
class=3DMsoNormal>is strong, but there is also a move to figure out how =
to encrypt SNI, so every time<o:p></o:p></p></div><div><p =
class=3DMsoNormal>we extend that argument it makes that harder. OTOH, =
STUN origin will be<o:p></o:p></p></div><div><p class=3DMsoNormal>sent a =
lot less often than SNI, I imagine.<o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p class=3DMsoNormal>I =
don't understand Section 6. Is this guidance to firewall vendors or to =
browser<o:p></o:p></p></div><div><p class=3DMsoNormal>implementors. If =
the former, the firewall has no way of knowing what =
&quot;requests&quot;<o:p></o:p></p></div><div><p class=3DMsoNormal>are =
if the connection use HTTPS (and if is uses HTTP, you have DPI-type =
methods for<o:p></o:p></p></div><div><p class=3DMsoNormal>determining =
whether something is voice or video. If this is guidance to =
browser<o:p></o:p></p></div><div><p class=3DMsoNormal>vendors, I =
consider it extremely unlikely that we would implement =
it.<o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal>Point #4 in your security considerations section seems =
like an entirely lost<o:p></o:p></p></div><div><p =
class=3DMsoNormal>cause, unless you assume that attackers are not really =
trying.<o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal>-Ekr<o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><div><p class=3DMsoNormal>On Tue, =
Jul 7, 2015 at 8:50 AM, Cullen Jennings &lt;<a =
href=3D"mailto:fluffy@iii.ca" target=3D"_blank">fluffy@iii.ca</a>&gt; =
wrote:<o:p></o:p></p><p class=3DMsoNormal><br>I'd like to get a brief =
amount of agenda time to discuss changes we would need to some of the =
WebRTC WG drafts to make the ideas proposed in =
draft-jennings-behave-rtcweb-firewall- work and make people aware of =
this work.<br><br>Would it be possible to get 10 minutes on the agenda =
?<br><br>Thanks, =
Cullen<br><br><br>_______________________________________________<br>rtcw=
eb mailing list<br><a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br><a =
href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" =
target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><o:p></=
o:p></p></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div></div></body></html>
------=_NextPart_000_0087_01D0BDCA.129E1DA0--


From nobody Mon Jul 13 15:19:28 2015
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From: "Karl Stahl" <karl.stahl@intertex.se>
To: "'Benjamin Schwartz'" <bemasc@webrtc.org>
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Subject: [rtcweb]   draft-ietf-rtcweb-return-00.txt; Sealed, Configuration and TURN discovery:
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We need this draft implemented in WebRTC browsers as soon as possible, =
so we
can get good WebRTC in difficult places (for firewall traversal and for
quality). Is there any implementation e.g. in Chrome that can be tried?

Now having stated that I am positive to this draft, I have some
concerns/questions/suggestions:

1) We need more levels than sealed/leaky

It is not acceptable that WebRTC traffic between parties on a typical =
LAN is
forced through a TURN server, just because of a sealed configuration
required for good WebRTC traffic over the WAN. This consequence is =
pointed
out under "5.11.  Reusing the same TURN server", but IS a problem even =
if it
"work as expected".

Such TURN resource consumption, waste of bandwidth and possible =
performance
degradation do not happen today, and can easily be avoided.

I realize this consequence of "sealed" is required to meet some of the
requirements under "3.2.  Independent Path Control", but we simply need =
an
"exo-sealed" level, in addition to the proposed strictly sealed and =
leaky,
allowing local endpoints to connect media directly, while external =
traffic
is forced to use the TURN server (rather than STUN through the default
gateway).=20

Such "exo-sealed" configuration should be the default configuration if a
network provided TURN server is made available for good (non-evil) =
purposes.


2) Mobile devices need to work on various networks, without individual
configuration=20

I am not clear about the thoughts behind the "configuration" of sealed =
level
and requirements for authentication of the turn server, but if there are
methods where an enterprise network (LAN) admin can provide credentials =
and
force a certain sealed level upon the WebRTC-browser when used at that =
LAN,
or if the user has the possibility to configure manually, I am OK with =
that
this can and should OVERRIDE any default configuration or other
configuration (on that particular network).

However, for mobile devices - Don't we all happily connect our browsers =
via
WiFi here and there? - there must be a way for a network provider with a
good intent to offer an exo-sealed TURN server that automatically should =
be
used, e.g. to achieve the purpose of offering better RTC quality than is
available through the default gateway WAN path on that network.=20

A network provided TURN-server can easily be made available ONLY to the
network users that it is serving, so there is no need for authentication =
and
credentials to protect against illegal usage of a valuable resource, =
like it
is with the application provided turn server. Network provided TURN =
servers
are simply deployed for the network users to be use without other
authentication than being granted an IP address and a default gateway to =
use
on that particular network in itself.

The same goes for trusting the TURN server with our real-time traffic - =
If
we would trust the same network provider with the same real-time traffic
through the default gateway - we should also use the offered TURN server
without other "authentication" than that it is available on our network:
Data through the default gateway, RTC through the TURN path is why the =
TURN
server is provided!

But do we know that it is not someone with the evil purpose of stealing =
our
(encrypted!) real time traffic that is offering the TURN server (instead =
of
the network provider), which leads us to my point 3). (If we really =
think
there is a risk of and danger with such attacks...)


3) Is it now good enough to simply use the auto discovered TURN server? =
Hope
so!

There has been some recent improvements to
draft-ietf-tram-turn-server-discovery-03 and hopefully that is =
sufficient
for assuming that the auto-discovered TURN server is offered by our =
network
provider (a service provider or the enterprise itself) that we trust =
anyway,
and the TURN server then simply shall be used and considered exo-sealed.
(The current RETURN draft says a non configured TURN server should be
leaky.)

Some concerns regarding the DISCOVERY draft and whether these things =
should
be specified in the RETURN document:

- Is the above the case with all discovery methods proposed? Or shall =
the
RETURN document make one method mandatory to implement?

- Can that method then be the Anycast method (which by far seems to be =
the
most convenient to deploy and to implement and should work for all =
network
types and in all devices operating systems). I think what is said under
"9.3.  Anycast" under "9.  Security Considerations" is sufficient for
trusting such discovered TURN server. What should then go into the =
RETURN
draft - or should the DISCOVERY draft be more specific?

I don't fully understand the DISCOVERY draft's: "... three discovery
mechanisms. The reason for providing multiple mechanisms is to maximize =
the
opportunity for discovery,"

- Can we really put the burden upon all WebRTC devices to implement all
methods and I doubt network providers will deploy more than one method. =
That
seems risking rather than maximizing discovery...


It is high time to get this done - can we?

/Karl









-----Ursprungligt meddelande-----
Fr=E5n: rtcweb [mailto:rtcweb-bounces@ietf.org] F=F6r =
internet-drafts@ietf.org
Skickat: den 14 maj 2015 00:29
Till: i-d-announce@ietf.org
Kopia: rtcweb@ietf.org
=C4mne: [rtcweb] I-D Action: draft-ietf-rtcweb-return-00.txt


A New Internet-Draft is available from the on-line Internet-Drafts
directories.
 This draft is a work item of the Real-Time Communication in =
WEB-browsers
Working Group of the IETF.

        Title           : Recursively Encapsulated TURN (RETURN) for
Connectivity and Privacy in WebRTC
        Authors         : Benjamin M. Schwartz
                          Justin Uberti
	Filename        : draft-ietf-rtcweb-return-00.txt
	Pages           : 18
	Date            : 2015-05-13

Abstract:
   In the context of WebRTC, the concept of a local TURN proxy has been
   suggested, but not reviewed in detail.  WebRTC applications are
   already using TURN to enhance connectivity and privacy.  This
   document explains how local TURN proxies and WebRTC applications can
   work together.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-return/

There's also a htmlized version available at:
https://tools.ietf.org/html/draft-ietf-rtcweb-return-00


Please note that it may take a couple of minutes from the time of =
submission
until the htmlized version and diff are available at tools.ietf.org.

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

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


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Subject: Re: [rtcweb] draft-ietf-rtcweb-return-00.txt; Sealed, Configuration and TURN discovery:
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--089e01183ebc65a076051acb9588
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On Mon, Jul 13, 2015 at 6:19 PM, Karl Stahl <karl.stahl@intertex.se> wrote:

> We need this draft implemented in WebRTC browsers as soon as possible, so
> we
> can get good WebRTC in difficult places (for firewall traversal and for
> quality). Is there any implementation e.g. in Chrome that can be tried?
>
> Now having stated that I am positive to this draft, I have some
> concerns/questions/suggestions:
>
> 1) We need more levels than sealed/leaky
>
> It is not acceptable that WebRTC traffic between parties on a typical LAN
> is
> forced through a TURN server, just because of a sealed configuration
> required for good WebRTC traffic over the WAN. This consequence is pointe=
d
> out under "5.11.  Reusing the same TURN server", but IS a problem even if
> it
> "work as expected".
>
> Such TURN resource consumption, waste of bandwidth and possible performan=
ce
> degradation do not happen today, and can easily be avoided.
>
> I realize this consequence of "sealed" is required to meet some of the
> requirements under "3.2.  Independent Path Control", but we simply need a=
n
> "exo-sealed" level, in addition to the proposed strictly sealed and leaky=
,
> allowing local endpoints to connect media directly, while external traffi=
c
> is forced to use the TURN server (rather than STUN through the default
> gateway).
>

How do you propose to tell what is "local"?  I don't know how to define
that procedurally.

I think my main concern here is that it requires us to "lie" to the remote
party, by sending candidates that we may then refuse to use even if they
work.  This seems very strange.

Such "exo-sealed" configuration should be the default configuration if a
> network provided TURN server is made available for good (non-evil)
> purposes.
>
>
> 2) Mobile devices need to work on various networks, without individual
> configuration
>
> I am not clear about the thoughts behind the "configuration" of sealed
> level
> and requirements for authentication of the turn server, but if there are
> methods where an enterprise network (LAN) admin can provide credentials a=
nd
> force a certain sealed level upon the WebRTC-browser when used at that LA=
N,
> or if the user has the possibility to configure manually, I am OK with th=
at
> this can and should OVERRIDE any default configuration or other
> configuration (on that particular network).
>
> However, for mobile devices - Don't we all happily connect our browsers v=
ia
> WiFi here and there? - there must be a way for a network provider with a
> good intent to offer an exo-sealed TURN server that automatically should =
be
> used, e.g. to achieve the purpose of offering better RTC quality than is
> available through the default gateway WAN path on that network.
>
> A network provided TURN-server can easily be made available ONLY to the
> network users that it is serving, so there is no need for authentication
> and
> credentials to protect against illegal usage of a valuable resource, like
> it
> is with the application provided turn server. Network provided TURN serve=
rs
> are simply deployed for the network users to be use without other
> authentication than being granted an IP address and a default gateway to
> use
> on that particular network in itself.
>
> The same goes for trusting the TURN server with our real-time traffic - I=
f
> we would trust the same network provider with the same real-time traffic
> through the default gateway - we should also use the offered TURN server
> without other "authentication" than that it is available on our network:
> Data through the default gateway, RTC through the TURN path is why the TU=
RN
> server is provided!
>
> But do we know that it is not someone with the evil purpose of stealing o=
ur
> (encrypted!) real time traffic that is offering the TURN server (instead =
of
> the network provider), which leads us to my point 3). (If we really think
> there is a risk of and danger with such attacks...)
>
>
> 3) Is it now good enough to simply use the auto discovered TURN server?
> Hope
> so!
>
> There has been some recent improvements to
> draft-ietf-tram-turn-server-discovery-03 and hopefully that is sufficient
> for assuming that the auto-discovered TURN server is offered by our netwo=
rk
> provider (a service provider or the enterprise itself) that we trust
> anyway,
> and the TURN server then simply shall be used and considered exo-sealed.
> (The current RETURN draft says a non configured TURN server should be
> leaky.)
>

The security considerations in that document suggest that not all discovery
mechanisms have equal additional exposure to local network interference.  I
would appreciate clearer guidance on that front.  It seems to me that DHCP
is the only listed mechanism that creates minimal additional exposure to
network interference.


> Some concerns regarding the DISCOVERY draft and whether these things shou=
ld
> be specified in the RETURN document:
>
> - Is the above the case with all discovery methods proposed? Or shall the
> RETURN document make one method mandatory to implement?
>
> - Can that method then be the Anycast method (which by far seems to be th=
e
> most convenient to deploy and to implement and should work for all networ=
k
> types and in all devices operating systems). I think what is said under
> "9.3.  Anycast" under "9.  Security Considerations" is sufficient for
> trusting such discovered TURN server. What should then go into the RETURN
> draft - or should the DISCOVERY draft be more specific?
>
> I don't fully understand the DISCOVERY draft's: "... three discovery
> mechanisms. The reason for providing multiple mechanisms is to maximize t=
he
> opportunity for discovery,"
>
> - Can we really put the burden upon all WebRTC devices to implement all
> methods and I doubt network providers will deploy more than one method.
> That
> seems risking rather than maximizing discovery...
>
>
> It is high time to get this done - can we?
>
> /Karl
>
>
>
>
>
>
>
>
>
> -----Ursprungligt meddelande-----
> Fr=C3=A5n: rtcweb [mailto:rtcweb-bounces@ietf.org] F=C3=B6r internet-draf=
ts@ietf.org
> Skickat: den 14 maj 2015 00:29
> Till: i-d-announce@ietf.org
> Kopia: rtcweb@ietf.org
> =C3=84mne: [rtcweb] I-D Action: draft-ietf-rtcweb-return-00.txt
>
>
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
>  This draft is a work item of the Real-Time Communication in WEB-browsers
> Working Group of the IETF.
>
>         Title           : Recursively Encapsulated TURN (RETURN) for
> Connectivity and Privacy in WebRTC
>         Authors         : Benjamin M. Schwartz
>                           Justin Uberti
>         Filename        : draft-ietf-rtcweb-return-00.txt
>         Pages           : 18
>         Date            : 2015-05-13
>
> Abstract:
>    In the context of WebRTC, the concept of a local TURN proxy has been
>    suggested, but not reviewed in detail.  WebRTC applications are
>    already using TURN to enhance connectivity and privacy.  This
>    document explains how local TURN proxies and WebRTC applications can
>    work together.
>
>
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-return/
>
> There's also a htmlized version available at:
> https://tools.ietf.org/html/draft-ietf-rtcweb-return-00
>
>
> Please note that it may take a couple of minutes from the time of
> submission
> until the htmlized version and diff are available at tools.ietf.org.
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--089e01183ebc65a076051acb9588
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On M=
on, Jul 13, 2015 at 6:19 PM, Karl Stahl <span dir=3D"ltr">&lt;<a href=3D"ma=
ilto:karl.stahl@intertex.se" target=3D"_blank">karl.stahl@intertex.se</a>&g=
t;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0=
 .8ex;border-left:1px #ccc solid;padding-left:1ex">We need this draft imple=
mented in WebRTC browsers as soon as possible, so we<br>
can get good WebRTC in difficult places (for firewall traversal and for<br>
quality). Is there any implementation e.g. in Chrome that can be tried?<br>
<br>
Now having stated that I am positive to this draft, I have some<br>
concerns/questions/suggestions:<br>
<br>
1) We need more levels than sealed/leaky<br>
<br>
It is not acceptable that WebRTC traffic between parties on a typical LAN i=
s<br>
forced through a TURN server, just because of a sealed configuration<br>
required for good WebRTC traffic over the WAN. This consequence is pointed<=
br>
out under &quot;5.11.=C2=A0 Reusing the same TURN server&quot;, but IS a pr=
oblem even if it<br>
&quot;work as expected&quot;.<br>
<br>
Such TURN resource consumption, waste of bandwidth and possible performance=
<br>
degradation do not happen today, and can easily be avoided.<br>
<br>
I realize this consequence of &quot;sealed&quot; is required to meet some o=
f the<br>
requirements under &quot;3.2.=C2=A0 Independent Path Control&quot;, but we =
simply need an<br>
&quot;exo-sealed&quot; level, in addition to the proposed strictly sealed a=
nd leaky,<br>
allowing local endpoints to connect media directly, while external traffic<=
br>
is forced to use the TURN server (rather than STUN through the default<br>
gateway).<br></blockquote><div><br></div><div>How do you propose to tell wh=
at is &quot;local&quot;?=C2=A0 I don&#39;t know how to define that procedur=
ally.</div><div><br></div><div>I think my main concern here is that it requ=
ires us to &quot;lie&quot; to the remote party, by sending candidates that =
we may then refuse to use even if they work.=C2=A0 This seems very strange.=
</div><div><br></div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 =
0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Such &quot;exo-sealed&quot; configuration should be the default configurati=
on if a<br>
network provided TURN server is made available for good (non-evil) purposes=
.<br>
<br>
<br>
2) Mobile devices need to work on various networks, without individual<br>
configuration<br>
<br>
I am not clear about the thoughts behind the &quot;configuration&quot; of s=
ealed level<br>
and requirements for authentication of the turn server, but if there are<br=
>
methods where an enterprise network (LAN) admin can provide credentials and=
<br>
force a certain sealed level upon the WebRTC-browser when used at that LAN,=
<br>
or if the user has the possibility to configure manually, I am OK with that=
<br>
this can and should OVERRIDE any default configuration or other<br>
configuration (on that particular network).<br>
<br>
However, for mobile devices - Don&#39;t we all happily connect our browsers=
 via<br>
WiFi here and there? - there must be a way for a network provider with a<br=
>
good intent to offer an exo-sealed TURN server that automatically should be=
<br>
used, e.g. to achieve the purpose of offering better RTC quality than is<br=
>
available through the default gateway WAN path on that network.<br>
<br>
A network provided TURN-server can easily be made available ONLY to the<br>
network users that it is serving, so there is no need for authentication an=
d<br>
credentials to protect against illegal usage of a valuable resource, like i=
t<br>
is with the application provided turn server. Network provided TURN servers=
<br>
are simply deployed for the network users to be use without other<br>
authentication than being granted an IP address and a default gateway to us=
e<br>
on that particular network in itself.<br>
<br>
The same goes for trusting the TURN server with our real-time traffic - If<=
br>
we would trust the same network provider with the same real-time traffic<br=
>
through the default gateway - we should also use the offered TURN server<br=
>
without other &quot;authentication&quot; than that it is available on our n=
etwork:<br>
Data through the default gateway, RTC through the TURN path is why the TURN=
<br>
server is provided!<br>
<br>
But do we know that it is not someone with the evil purpose of stealing our=
<br>
(encrypted!) real time traffic that is offering the TURN server (instead of=
<br>
the network provider), which leads us to my point 3). (If we really think<b=
r>
there is a risk of and danger with such attacks...)<br>
<br>
<br>
3) Is it now good enough to simply use the auto discovered TURN server? Hop=
e<br>
so!<br>
<br>
There has been some recent improvements to<br>
draft-ietf-tram-turn-server-discovery-03 and hopefully that is sufficient<b=
r>
for assuming that the auto-discovered TURN server is offered by our network=
<br>
provider (a service provider or the enterprise itself) that we trust anyway=
,<br>
and the TURN server then simply shall be used and considered exo-sealed.<br=
>
(The current RETURN draft says a non configured TURN server should be<br>
leaky.)<br></blockquote><div><br></div><div>The security considerations in =
that document suggest that not all discovery mechanisms have equal addition=
al exposure to local network interference.=C2=A0 I would appreciate clearer=
 guidance on that front.=C2=A0 It seems to me that DHCP is the only listed =
mechanism that creates minimal additional exposure to network interference.=
</div><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 =
0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Some concerns regarding the DISCOVERY draft and whether these things should=
<br>
be specified in the RETURN document:<br>
<br>
- Is the above the case with all discovery methods proposed? Or shall the<b=
r>
RETURN document make one method mandatory to implement?<br>
<br>
- Can that method then be the Anycast method (which by far seems to be the<=
br>
most convenient to deploy and to implement and should work for all network<=
br>
types and in all devices operating systems). I think what is said under<br>
&quot;9.3.=C2=A0 Anycast&quot; under &quot;9.=C2=A0 Security Considerations=
&quot; is sufficient for<br>
trusting such discovered TURN server. What should then go into the RETURN<b=
r>
draft - or should the DISCOVERY draft be more specific?<br>
<br>
I don&#39;t fully understand the DISCOVERY draft&#39;s: &quot;... three dis=
covery<br>
mechanisms. The reason for providing multiple mechanisms is to maximize the=
<br>
opportunity for discovery,&quot;<br>
<br>
- Can we really put the burden upon all WebRTC devices to implement all<br>
methods and I doubt network providers will deploy more than one method. Tha=
t<br>
seems risking rather than maximizing discovery...<br>
<br>
<br>
It is high time to get this done - can we?<br>
<br>
/Karl<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
-----Ursprungligt meddelande-----<br>
Fr=C3=A5n: rtcweb [mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org" target=
=3D"_blank">rtcweb-bounces@ietf.org</a>] F=C3=B6r <a href=3D"mailto:interne=
t-drafts@ietf.org" target=3D"_blank">internet-drafts@ietf.org</a><br>
Skickat: den 14 maj 2015 00:29<br>
Till: <a href=3D"mailto:i-d-announce@ietf.org" target=3D"_blank">i-d-announ=
ce@ietf.org</a><br>
Kopia: <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org=
</a><br>
=C3=84mne: [rtcweb] I-D Action: draft-ietf-rtcweb-return-00.txt<br>
<br>
<br>
A New Internet-Draft is available from the on-line Internet-Drafts<br>
directories.<br>
=C2=A0This draft is a work item of the Real-Time Communication in WEB-brows=
ers<br>
Working Group of the IETF.<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Title=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0:=
 Recursively Encapsulated TURN (RETURN) for<br>
Connectivity and Privacy in WebRTC<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Authors=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0: Benj=
amin M. Schwartz<br>
=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 Justin Uberti<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Filename=C2=A0 =C2=A0 =C2=A0 =C2=A0 : draft-iet=
f-rtcweb-return-00.txt<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Pages=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0:=
 18<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Date=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 :=
 2015-05-13<br>
<br>
Abstract:<br>
=C2=A0 =C2=A0In the context of WebRTC, the concept of a local TURN proxy ha=
s been<br>
=C2=A0 =C2=A0suggested, but not reviewed in detail.=C2=A0 WebRTC applicatio=
ns are<br>
=C2=A0 =C2=A0already using TURN to enhance connectivity and privacy.=C2=A0 =
This<br>
=C2=A0 =C2=A0document explains how local TURN proxies and WebRTC applicatio=
ns can<br>
=C2=A0 =C2=A0work together.<br>
<br>
<br>
The IETF datatracker status page for this draft is:<br>
<a href=3D"https://datatracker.ietf.org/doc/draft-ietf-rtcweb-return/" rel=
=3D"noreferrer" target=3D"_blank">https://datatracker.ietf.org/doc/draft-ie=
tf-rtcweb-return/</a><br>
<br>
There&#39;s also a htmlized version available at:<br>
<a href=3D"https://tools.ietf.org/html/draft-ietf-rtcweb-return-00" rel=3D"=
noreferrer" target=3D"_blank">https://tools.ietf.org/html/draft-ietf-rtcweb=
-return-00</a><br>
<br>
<br>
Please note that it may take a couple of minutes from the time of submissio=
n<br>
until the htmlized version and diff are available at <a href=3D"http://tool=
s.ietf.org" rel=3D"noreferrer" target=3D"_blank">tools.ietf.org</a>.<br>
<br>
Internet-Drafts are also available by anonymous FTP at:<br>
<a href=3D"ftp://ftp.ietf.org/internet-drafts/" rel=3D"noreferrer" target=
=3D"_blank">ftp://ftp.ietf.org/internet-drafts/</a><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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br>
</blockquote></div><br></div></div>

--089e01183ebc65a076051acb9588--


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From: ranjit@ranjitvoip.com
To: Ted Hardie <ted.ietf@gmail.com>
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Subject: Re: [rtcweb] Updated -gateways draft
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Hi
It would be good if it is a standards track as 3GPP depends on it and 
all implementations would confirm to it . having it as Informational is 
really not of much use.

- Ranjit

On 2015-07-06 5:35 pm, Ted Hardie wrote:
> On Mon, Jul 6, 2015 at 3:19 PM, Harald Alvestrand
> <harald@alvestrand.no> wrote:
> 
>> Uwe and I have prepared a new -gateways draft (version -01).
>> 
>> I don't think there's much controversial in it, and there's only
>> one
>> issue that I think needs WG time at this time:
>> 
>> Should it be Informational or Standards-track?
>> 
>> An informational document defines guidance for how gateways should
>> behave, but nobody needs to pay attention if they don't want to.
>> 
>> A standards-track document (which no other RTCWEB document should
>> be
>> dependent on) defines requirements for how things that call
>> themselves
>> "WebRTC gateways" should behave, but nobody needs to call their
>> gateways
>> that, and anyway, there's no protocol police, so the difference
>> isn't
>> all that much in practice.
>> 
>> More important: What do people want it to be?
> 
> â€‹I think we adopted it in part because 3GPP had a dependency on it.
> Does it need to be standards track for them to reference it?
> 
> Tedâ€‹
> 
>> Harald
>> 
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb [1]
> 
> 
> 
> Links:
> ------
> [1] https://www.ietf.org/mailman/listinfo/rtcweb
> 
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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Den 14. juli 2015 05:18, skrev ranjit@ranjitvoip.com:
> Hi
> It would be good if it is a standards track as 3GPP depends on it and
> all implementations would confirm to it . having it as Informational is
> really not of much use.

Could you share some more detail here?

- What 3GPP spec is depending on it?
- What does the 3GPP spec depend on being in the draft?
- What's the rules for referencing an IETF spec from a 3GPP spec, and
why does the category matter?

Harald


> 
> - Ranjit
> 
> On 2015-07-06 5:35 pm, Ted Hardie wrote:
>> On Mon, Jul 6, 2015 at 3:19 PM, Harald Alvestrand
>> <harald@alvestrand.no> wrote:
>>
>>> Uwe and I have prepared a new -gateways draft (version -01).
>>>
>>> I don't think there's much controversial in it, and there's only
>>> one
>>> issue that I think needs WG time at this time:
>>>
>>> Should it be Informational or Standards-track?
>>>
>>> An informational document defines guidance for how gateways should
>>> behave, but nobody needs to pay attention if they don't want to.
>>>
>>> A standards-track document (which no other RTCWEB document should
>>> be
>>> dependent on) defines requirements for how things that call
>>> themselves
>>> "WebRTC gateways" should behave, but nobody needs to call their
>>> gateways
>>> that, and anyway, there's no protocol police, so the difference
>>> isn't
>>> all that much in practice.
>>>
>>> More important: What do people want it to be?
>>
>> â€‹I think we adopted it in part because 3GPP had a dependency on it.
>> Does it need to be standards track for them to reference it?
>>
>> Tedâ€‹
>>
>>> Harald
>>>
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb [1]
>>
>>
>>
>> Links:
>> ------
>> [1] https://www.ietf.org/mailman/listinfo/rtcweb
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
> 


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Hi,

I have reviewed this latest version and find nothing that warrants a 
comment, except:

Can you please spell my name correctly in the acknowledgement section ;-).

Cheers

Magnus

The IESG skrev den 2015-07-02 19:55:
>
> The IESG has received a request from the Real-Time Communication in
> WEB-browsers WG (rtcweb) to consider the following document:
> - 'STUN Usage for Consent Freshness'
>    <draft-ietf-rtcweb-stun-consent-freshness-15.txt> as Proposed Standard
>
> The IESG plans to make a decision in the next few weeks, and solicits
> final comments on this action. Please send substantive comments to the
> ietf@ietf.org mailing lists by 2015-07-16. Exceptionally, comments may be
> sent to iesg@ietf.org instead. In either case, please retain the
> beginning of the Subject line to allow automated sorting.
>
> This document was previously last called on May 1 and has been updated as a result of comments submitted during that last call.
>
> Abstract
>
>
>     To prevent WebRTC applications, such as browsers, from launching
>     attacks by sending media to unwilling victims, periodic consent to
>     send needs to be obtained from remote endpoints.
>
>     This document describes a consent mechanism using a new Session
>     Traversal Utilities for NAT (STUN) usage.
>
>
>
>
> The file can be obtained via
> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-stun-consent-freshness/
>
> IESG discussion can be tracked via
> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-stun-consent-freshness/ballot/
>
>
> No IPR declarations have been submitted directly on this I-D.
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>


-- 

Magnus Westerlund

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Subject: Re: [rtcweb] JSEP: Imageattr and CVO
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Hi,

3GPP SA4 had a meeting last week. Current status is that the issue is 
recognized in 3GPP SA4 and the required detailed changes to 3GPP TS 
26.114 are being worked out. Their next meeting is 24 - 28 Aug 2015.

Cheers

Magnus

Magnus Westerlund skrev den 2015-06-22 11:34:
> Hi,
>
> I got an action point at the interim to check on what 3GPP and GSMA says
> about their usage of a=imageattr and CVO (Coordination of Video
> Orientation).
>
> 3GPP SA4 in TS 26.114 (Which defines CVO) there is discussion about the
> usage of a=imageattr when a endpoint is not CVO capable, but it appears
> to fail to be explicit about interpretation of imageattr when using CVO.
> As the text says that for a non CVO capable endpoint one have to express
> both orientations (x,y) and (y,x) in imageattr and perform the rotation
> prior to encoding.
>
> This appear to support the proposed interpretation for JSEP, i.e. that
> one expresses the supported max resolution without rotation using
> imageattr.
>
> Bo Burman intend to prepare an contribution to 3GPP SA4 in their
> upcoming meeting in two weeks time and see if SA4 can come to agreement
> on this interpretation and write it into their specifications. So in
> time for Prague we will hopefully be able to conclude on this.
>
> Cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> Färögatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>


-- 

Magnus Westerlund

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Services, Media and Network features, Ericsson Research EAB/TXM
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Ericsson AB                 | Phone  +46 10 7148287
Färögatan 6                 | Mobile +46 73 0949079
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To: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Hi Magnus,

Sorry for the typo.
I will note the same and correct it in the next revision.

Regards,
Ram

-----Original Message-----
From: Magnus Westerlund <magnus.westerlund@ericsson.com>
Date: Tuesday, 14 July 2015 10:00 pm
To: "rtcweb@ietf.org" <rtcweb@ietf.org>,
"draft-ietf-rtcweb-stun-consent-freshness@tools.ietf.org"
<draft-ietf-rtcweb-stun-consent-freshness@tools.ietf.org>
Subject: Re: [rtcweb] Last Call:
<draft-ietf-rtcweb-stun-consent-freshness-15.txt> (STUN Usage for Consent
Freshness) to Proposed Standard
Resent-From: <magnus.westerlund@ericsson.com>
Resent-To: <draft-ietf-rtcweb-stun-consent-freshness@ietf.org>
Resent-Date: Tuesday, 14 July 2015 10:00 pm

>Hi,
>
>I have reviewed this latest version and find nothing that warrants a
>comment, except:
>
>Can you please spell my name correctly in the acknowledgement section ;-).
>
>Cheers
>
>Magnus
>
>The IESG skrev den 2015-07-02 19:55:
>>
>> The IESG has received a request from the Real-Time Communication in
>> WEB-browsers WG (rtcweb) to consider the following document:
>> - 'STUN Usage for Consent Freshness'
>>    <draft-ietf-rtcweb-stun-consent-freshness-15.txt> as Proposed
>>Standard
>>
>> The IESG plans to make a decision in the next few weeks, and solicits
>> final comments on this action. Please send substantive comments to the
>> ietf@ietf.org mailing lists by 2015-07-16. Exceptionally, comments may
>>be
>> sent to iesg@ietf.org instead. In either case, please retain the
>> beginning of the Subject line to allow automated sorting.
>>
>> This document was previously last called on May 1 and has been updated
>>as a result of comments submitted during that last call.
>>
>> Abstract
>>
>>
>>     To prevent WebRTC applications, such as browsers, from launching
>>     attacks by sending media to unwilling victims, periodic consent to
>>     send needs to be obtained from remote endpoints.
>>
>>     This document describes a consent mechanism using a new Session
>>     Traversal Utilities for NAT (STUN) usage.
>>
>>
>>
>>
>> The file can be obtained via
>>=20
>>https://datatracker.ietf.org/doc/draft-ietf-rtcweb-stun-consent-freshness
>>/
>>
>> IESG discussion can be tracked via
>>=20
>>https://datatracker.ietf.org/doc/draft-ietf-rtcweb-stun-consent-freshness
>>/ballot/
>>
>>
>> No IPR declarations have been submitted directly on this I-D.
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>>
>
>
>--=20
>
>Magnus Westerlund
>
>----------------------------------------------------------------------
>Services, Media and Network features, Ericsson Research EAB/TXM
>----------------------------------------------------------------------
>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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Date: Tue, 14 Jul 2015 11:18:24 -0700
Message-ID: <CA+9kkMBD4NRsM6Kq3zmF1L2GDJzjBKBbgvJZfcf2i5jFm2PjgQ@mail.gmail.com>
From: Ted Hardie <ted.ietf@gmail.com>
To: "Orit Levin (LCA)" <oritl@microsoft.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
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Cc: "appsdir@ietf.org" <appsdir@ietf.org>, Eliot Lear <lear@cisco.com>
Subject: Re: [rtcweb] [appsdir] Fwd: Request for Apps Directorate review of rtcweb-security drafts
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(adding the WG)

Hi Orit,

Thanks for the review.

regards,

Ted Hardie

On Tue, Jul 14, 2015 at 11:07 AM, Orit Levin (LCA) <oritl@microsoft.com>
wrote:

> Disclaimer: I am familiar with RTC and WebRTC, but I haven't followed the
> RTCWeb work in the last few years. I read the two drafts for the first time.
> Therefore, my apologies for potentially raising points that have been
> discussed in the past.
>
> >From the APP (or ART) review perspective, my main comment is on the topic
> of "Web-Based Peer Authentication". While it is not clear from the text,
> which parts are already standardized concepts (e.g., by W3C) and which are
> introduced here for the first time, the described approach seems to be (1)
> useful to web applications beyond WebRTC, (2) under development and thus
> probably not stable, and (3) too detailed for "an architecture" document.
> As such, the concept of " Web-Based Peer Authentication" probably needs to
> be introduced and standardized under the Security Area and then adopted by
> WebRTC and other apps.
>
> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-security/
> This document contains an excellent collection of security threats to
> WebRTC and discusses many different ideas for their potential mitigation.
> That being said, it is not an easy read. Specifically, from high level to
> more specific:
> 1. The intended status of the document is Standards Track, while a typical
> status of a "security threat model" RFC is Informational. This discrepancy
> probably lies in the current scope of the document. See the next comment
> below.
> 2. The scope of the document is not clear. In addition to describing the
> security threats (and the requirements or user expectations related to
> their prevention), in many places the document lists potential solutions
> with their perceived limitations and/or applicability. Examples are
> sections: 4.2.1, 4.2.2, 4.2.3, 4.2.4, 4.3.2.1, 4.3.2.2, 4.3.2.3. To improve
> the document's readability and reduce overlaps,  the solutions should be
> introduced and explained in the companion "architecture" document.
> Solutions' applicability to different use cases needs to be provided in a
> non-bias way.
> 3. Apart from the brief Introduction with a few examples, there is no
> description of the WebRTC threat model putting all main types of threat in
> one place. For example, the "Communications Security" aspect is introduced
> for the first time in Section 4.3 only. The start of Section "4. Security
> for WebRTC Applications" could be a logical place for introducing the
> WebRTC threat model.
> 4. The "simple WebRTC system" presented in the Introduction in Figure 1 is
> inadequate to illustrate many of the scenarios and points discussed in the
> document. For example, see the discussion around media origin vs. web
> origin in Section 4.1.3.
> 5. Section "3. The Browser Threat Model" is an overview of the existing
> web security architecture and some current practices. As such, references
> to relevant standards (e.g., W3C) and common practices need to be added
> within the text to help the reader to understand the status of the
> information and to follow the sources.
> 6. The last paragraph in Section 4.1 conveys a key point in the whole
> WebRTC threat model. As such,  it needs to be moved to the start of the
> discussion where the threat model is introduced (see comments above).
> 7. Stylistically, the document would benefit from removing non-standards
> vocabulary, i.e., "to bug", "to bug me", "no real way", "it is important to
> require", "extremely aggressive intermediates",  and "for obvious reasons",
> etc.
> 8. Suggestions for renaming some of the sections to improve readability:
> "3. The Browser Threat Model" -> " 3. Overview of the Existing Web Threat
> Model"
> "4.1. Access to Local Devices" -> "4.1. Consent to Access Local Resources"
> "4.1.2. Calling Scenarios and User Expectations" -> "4.1.2. Scope of
> Consent in Various Calling Scenarios"
> "4.1.2.1. Dedicated Calling Services" -> "4.1.2.1. Long-term Consent"
> "4.1.2.2. Calling the site You're On" -> "4.1.2.2. Short-term Consent"
> "4.1.3. Origin-Based Security" -> "4.1.3. Web Attackers and Origin-based
> Security"
> "4.1.4. Security Properties of the Calling Page" -> "4.1.4. Network
> Attackers and  Security Properties of the Calling Page"
> "4.2. Communications Consent Verification" -> "4.2. Consent to Receive
> Traffic"
> "4.3.3. Malicious Peers" -> "4.3.3. Out of Scope"
>
> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-security-arch/
> This document is in a much better state in terms of its organization and
> readability. It introduces the security architecture using an example and
> then lists possible approaches to implement it. Some approaches are
> described on a high level by referencing other documents, while others are
> introduced for the first time in this document together with their
> implementation details. By just reading the document, it is difficult to
> judge whether these details are sufficient or stable enough for
> implementation. In general, I doubt that this is the best approach for an
> "architecture" document, which is supposed to serve as a long term stable
> reference.
> More specific comments:
> 1. Different terminology is used throughout the document to address same
> or close concepts. It would help a lot to explain the concepts upfront and
> then to use the terminology consistently. Currently it includes "client",
> "endpoint",  "peer", "user"; "implementation", "browser", "chrome", "API",
> "JS"; "application", "server", etc.
> 2. The discussion is phrased in terms of "authentication" and "trust". In
> think that "authorization" instead of "trust" would be a better technical
> term, for example, in Section 3.1.
> 3. In section 4. Overview, sentence "the tension between security (or
> performance) and privacy" is not clear, especially because it doesn't seem
> to correspond to Section 6.2. (BTW "tradeoff" would be a better word than
> "tension".)
> 4. Section "4.1. Initial Signaling", the text is inconsistent in terms of
> whether Alice and Bob share or use different identity providers.
> 5. Section "4.1. Initial Signaling", reference to PeerConnection needs to
> be added.
> 6. Section "4.2. Media Consent Verification" and Section "5.3
> Communications Consent": the name/terminology of this functionality needs
> to be harmonized. The ICE-based solution with its pros, cons as well as the
> alternatives (applicable to different use cases) need to be moved from the
> companion threat model document to this "architecture" one.
> 7. A suggestion: rename "5. Detailed Technical Description" to "5.
> Security Architecture Components".
> 8. In Section 5.2., "Implementations which support some form of direct
> user authentication..." -> "Implementations that support some form of
> direct user authentication..."
> 9. Section "5.6. Web-Based Peer Authentication" and section "5.6.2.
> Overview of operation": It is not clear, which parts are standardized
> concepts (e.g., by W3C) and which are introduced here for the first time.
> If exist, references to existing standards need to be added. The described
> architecture and the technical details seem to be (1) under development and
> thus probably not stable, (2) too detailed for "an architecture" document
> and (3) useful beyond WebRTC.  As such, the concept of the " Web-Based Peer
> Authentication" probably needs be introduced under the Security Area and
> then adopted by WebRTC.
> 10. The web-based peer authentication using IdP is not the only mechanism
> applicable to WebRTC. Moreover, the web-based peer authentication  is not
> the only approach applicable to WebRTC. Alternatives are listed in the
> companion document and need to be moved here to the "architecture" document
> with their pros and cons presented in an unbiased way.
> 11. Stylistically, the document would benefit from removing non-standards
> vocabulary, i.e., "some Web server", "can't really trust", "that much",
> "fairly conventional", "simply have", "a little anticlimactic", etc. ;-)
>
> Thanks,
> Orit.
>
> > -----Original Message-----
> > From: appsdir [mailto:appsdir-bounces@ietf.org] On Behalf Of Orit Levin
> > (LCA)
> > Sent: Thursday, June 25, 2015 1:17 PM
> > To: Eliot Lear; appsdir@ietf.org
> > Subject: Re: [appsdir] Fwd: Request for Apps Directorate review of
> rtcweb-
> > security drafts
> >
> > Perfect! I have a very long flight on the 7th... So right after then or
> earlier.
> > Orit.
> >
> > > -----Original Message-----
> > > From: Eliot Lear [mailto:lear@cisco.com]
> > > Sent: Thursday, June 25, 2015 12:55 PM
> > > To: Orit Levin (LCA); appsdir@ietf.org
> > > Subject: Re: [appsdir] Fwd: Request for Apps Directorate review of
> rtcweb-
> > > security drafts
> > >
> > > Thank you Orit!  Can you try to have them done in about two weeks?
> > >
> > > Eliot
> > >
> > > On 6/25/15 9:52 PM, Orit Levin (LCA) wrote:
> > > > I will be happy to take a look at them!
> > > > Orit.
> > > >
> > > >> -----Original Message-----
> > > >> From: appsdir [mailto:appsdir-bounces@ietf.org] On Behalf Of Eliot
> > Lear
> > > >> Sent: Thursday, June 25, 2015 11:44 AM
> > > >> To: appsdir@ietf.org
> > > >> Subject: [appsdir] Fwd: Request for Apps Directorate review of
> rtcweb-
> > > >> security drafts
> > > >>
> > > >> Dear Appsdir folk,
> > > >>
> > > >> Could I have a volunteer to review the following two drafts?
> > > >>
> > > >> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-security/
> > > >>
> > > >> and
> > > >>
> > > >> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-security-arch/
> > > >>
> > > >> Ted is exempt, having made the request ;-)
> > > >>
> > > >> Eliot
> > >
> >
> > _______________________________________________
> > appsdir mailing list
> > appsdir@ietf.org
> > https://www.ietf.org/mailman/listinfo/appsdir
>
> _______________________________________________
> appsdir mailing list
> appsdir@ietf.org
> https://www.ietf.org/mailman/listinfo/appsdir
>

--f46d0444038a2f94a8051ad9dd9c
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:tahoma,s=
ans-serif;font-size:small">(adding the WG)<br><br></div><div class=3D"gmail=
_default" style=3D"font-family:tahoma,sans-serif;font-size:small">Hi Orit,<=
br><br></div><div class=3D"gmail_default" style=3D"font-family:tahoma,sans-=
serif;font-size:small">Thanks for the review.<br><br></div><div class=3D"gm=
ail_default" style=3D"font-family:tahoma,sans-serif;font-size:small">regard=
s,<br><br></div><div class=3D"gmail_default" style=3D"font-family:tahoma,sa=
ns-serif;font-size:small">Ted Hardie<br></div><div class=3D"gmail_extra"><b=
r><div class=3D"gmail_quote">On Tue, Jul 14, 2015 at 11:07 AM, Orit Levin (=
LCA) <span dir=3D"ltr">&lt;<a href=3D"mailto:oritl@microsoft.com" target=3D=
"_blank">oritl@microsoft.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">Disclaimer: I am familiar with RTC and WebRTC, but I haven&#39;t =
followed the RTCWeb work in the last few years. I read the two drafts for t=
he first time.<br>
Therefore, my apologies for potentially raising points that have been discu=
ssed in the past.<br>
<br>
&gt;From the APP (or ART) review perspective, my main comment is on the top=
ic of &quot;Web-Based Peer Authentication&quot;. While it is not clear from=
 the text, which parts are already standardized concepts (e.g., by W3C) and=
 which are introduced here for the first time, the described approach seems=
 to be (1) useful to web applications beyond WebRTC, (2) under development =
and thus probably not stable, and (3) too detailed for &quot;an architectur=
e&quot; document.=C2=A0 As such, the concept of &quot; Web-Based Peer Authe=
ntication&quot; probably needs to be introduced and standardized under the =
Security Area and then adopted by WebRTC and other apps.<br>
<br>
<a href=3D"https://datatracker.ietf.org/doc/draft-ietf-rtcweb-security/" re=
l=3D"noreferrer" target=3D"_blank">https://datatracker.ietf.org/doc/draft-i=
etf-rtcweb-security/</a><br>
This document contains an excellent collection of security threats to WebRT=
C and discusses many different ideas for their potential mitigation. That b=
eing said, it is not an easy read. Specifically, from high level to more sp=
ecific:<br>
1. The intended status of the document is Standards Track, while a typical =
status of a &quot;security threat model&quot; RFC is Informational. This di=
screpancy probably lies in the current scope of the document. See the next =
comment below.<br>
2. The scope of the document is not clear. In addition to describing the se=
curity threats (and the requirements or user expectations related to their =
prevention), in many places the document lists potential solutions with the=
ir perceived limitations and/or applicability. Examples are sections: 4.2.1=
, 4.2.2, 4.2.3, 4.2.4, 4.3.2.1, 4.3.2.2, 4.3.2.3. To improve the document&#=
39;s readability and reduce overlaps,=C2=A0 the solutions should be introdu=
ced and explained in the companion &quot;architecture&quot; document. Solut=
ions&#39; applicability to different use cases needs to be provided in a no=
n-bias way.<br>
3. Apart from the brief Introduction with a few examples, there is no descr=
iption of the WebRTC threat model putting all main types of threat in one p=
lace. For example, the &quot;Communications Security&quot; aspect is introd=
uced for the first time in Section 4.3 only. The start of Section &quot;4. =
Security for WebRTC Applications&quot; could be a logical place for introdu=
cing the WebRTC threat model.<br>
4. The &quot;simple WebRTC system&quot; presented in the Introduction in Fi=
gure 1 is inadequate to illustrate many of the scenarios and points discuss=
ed in the document. For example, see the discussion around media origin vs.=
 web origin in Section 4.1.3.<br>
5. Section &quot;3. The Browser Threat Model&quot; is an overview of the ex=
isting web security architecture and some current practices. As such, refer=
ences to relevant standards (e.g., W3C) and common practices need to be add=
ed within the text to help the reader to understand the status of the infor=
mation and to follow the sources.<br>
6. The last paragraph in Section 4.1 conveys a key point in the whole WebRT=
C threat model. As such,=C2=A0 it needs to be moved to the start of the dis=
cussion where the threat model is introduced (see comments above).<br>
7. Stylistically, the document would benefit from removing non-standards vo=
cabulary, i.e., &quot;to bug&quot;, &quot;to bug me&quot;, &quot;no real wa=
y&quot;, &quot;it is important to require&quot;, &quot;extremely aggressive=
 intermediates&quot;,=C2=A0 and &quot;for obvious reasons&quot;, etc.<br>
8. Suggestions for renaming some of the sections to improve readability:<br=
>
&quot;3. The Browser Threat Model&quot; -&gt; &quot; 3. Overview of the Exi=
sting Web Threat Model&quot;<br>
&quot;4.1. Access to Local Devices&quot; -&gt; &quot;4.1. Consent to Access=
 Local Resources&quot;<br>
&quot;4.1.2. Calling Scenarios and User Expectations&quot; -&gt; &quot;4.1.=
2. Scope of Consent in Various Calling Scenarios&quot;<br>
&quot;4.1.2.1. Dedicated Calling Services&quot; -&gt; &quot;4.1.2.1. Long-t=
erm Consent&quot;<br>
&quot;4.1.2.2. Calling the site You&#39;re On&quot; -&gt; &quot;4.1.2.2. Sh=
ort-term Consent&quot;<br>
&quot;4.1.3. Origin-Based Security&quot; -&gt; &quot;4.1.3. Web Attackers a=
nd Origin-based Security&quot;<br>
&quot;4.1.4. Security Properties of the Calling Page&quot; -&gt; &quot;4.1.=
4. Network Attackers and=C2=A0 Security Properties of the Calling Page&quot=
;<br>
&quot;4.2. Communications Consent Verification&quot; -&gt; &quot;4.2. Conse=
nt to Receive Traffic&quot;<br>
&quot;4.3.3. Malicious Peers&quot; -&gt; &quot;4.3.3. Out of Scope&quot;<br=
>
<br>
<a href=3D"https://datatracker.ietf.org/doc/draft-ietf-rtcweb-security-arch=
/" rel=3D"noreferrer" target=3D"_blank">https://datatracker.ietf.org/doc/dr=
aft-ietf-rtcweb-security-arch/</a><br>
This document is in a much better state in terms of its organization and re=
adability. It introduces the security architecture using an example and the=
n lists possible approaches to implement it. Some approaches are described =
on a high level by referencing other documents, while others are introduced=
 for the first time in this document together with their implementation det=
ails. By just reading the document, it is difficult to judge whether these =
details are sufficient or stable enough for implementation. In general, I d=
oubt that this is the best approach for an &quot;architecture&quot; documen=
t, which is supposed to serve as a long term stable reference.<br>
More specific comments:<br>
1. Different terminology is used throughout the document to address same or=
 close concepts. It would help a lot to explain the concepts upfront and th=
en to use the terminology consistently. Currently it includes &quot;client&=
quot;, &quot;endpoint&quot;,=C2=A0 &quot;peer&quot;, &quot;user&quot;; &quo=
t;implementation&quot;, &quot;browser&quot;, &quot;chrome&quot;, &quot;API&=
quot;, &quot;JS&quot;; &quot;application&quot;, &quot;server&quot;, etc.<br=
>
2. The discussion is phrased in terms of &quot;authentication&quot; and &qu=
ot;trust&quot;. In think that &quot;authorization&quot; instead of &quot;tr=
ust&quot; would be a better technical term, for example, in Section 3.1.<br=
>
3. In section 4. Overview, sentence &quot;the tension between security (or =
performance) and privacy&quot; is not clear, especially because it doesn&#3=
9;t seem to correspond to Section 6.2. (BTW &quot;tradeoff&quot; would be a=
 better word than &quot;tension&quot;.)<br>
4. Section &quot;4.1. Initial Signaling&quot;, the text is inconsistent in =
terms of whether Alice and Bob share or use different identity providers.<b=
r>
5. Section &quot;4.1. Initial Signaling&quot;, reference to PeerConnection =
needs to be added.<br>
6. Section &quot;4.2. Media Consent Verification&quot; and Section &quot;5.=
3 Communications Consent&quot;: the name/terminology of this functionality =
needs to be harmonized. The ICE-based solution with its pros, cons as well =
as the alternatives (applicable to different use cases) need to be moved fr=
om the companion threat model document to this &quot;architecture&quot; one=
.<br>
7. A suggestion: rename &quot;5. Detailed Technical Description&quot; to &q=
uot;5. Security Architecture Components&quot;.<br>
8. In Section 5.2., &quot;Implementations which support some form of direct=
 user authentication...&quot; -&gt; &quot;Implementations that support some=
 form of direct user authentication...&quot;<br>
9. Section &quot;5.6. Web-Based Peer Authentication&quot; and section &quot=
;5.6.2. Overview of operation&quot;: It is not clear, which parts are stand=
ardized concepts (e.g., by W3C) and which are introduced here for the first=
 time. If exist, references to existing standards need to be added. The des=
cribed architecture and the technical details seem to be (1) under developm=
ent and thus probably not stable, (2) too detailed for &quot;an architectur=
e&quot; document and (3) useful beyond WebRTC.=C2=A0 As such, the concept o=
f the &quot; Web-Based Peer Authentication&quot; probably needs be introduc=
ed under the Security Area and then adopted by WebRTC.<br>
10. The web-based peer authentication using IdP is not the only mechanism a=
pplicable to WebRTC. Moreover, the web-based peer authentication=C2=A0 is n=
ot the only approach applicable to WebRTC. Alternatives are listed in the c=
ompanion document and need to be moved here to the &quot;architecture&quot;=
 document with their pros and cons presented in an unbiased way.<br>
11. Stylistically, the document would benefit from removing non-standards v=
ocabulary, i.e., &quot;some Web server&quot;, &quot;can&#39;t really trust&=
quot;, &quot;that much&quot;, &quot;fairly conventional&quot;, &quot;simply=
 have&quot;, &quot;a little anticlimactic&quot;, etc. ;-)<br>
<br>
Thanks,<br>
Orit.<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
&gt; -----Original Message-----<br>
&gt; From: appsdir [mailto:<a href=3D"mailto:appsdir-bounces@ietf.org">apps=
dir-bounces@ietf.org</a>] On Behalf Of Orit Levin<br>
&gt; (LCA)<br>
&gt; Sent: Thursday, June 25, 2015 1:17 PM<br>
&gt; To: Eliot Lear; <a href=3D"mailto:appsdir@ietf.org">appsdir@ietf.org</=
a><br>
&gt; Subject: Re: [appsdir] Fwd: Request for Apps Directorate review of rtc=
web-<br>
&gt; security drafts<br>
&gt;<br>
&gt; Perfect! I have a very long flight on the 7th... So right after then o=
r earlier.<br>
&gt; Orit.<br>
&gt;<br>
&gt; &gt; -----Original Message-----<br>
&gt; &gt; From: Eliot Lear [mailto:<a href=3D"mailto:lear@cisco.com">lear@c=
isco.com</a>]<br>
&gt; &gt; Sent: Thursday, June 25, 2015 12:55 PM<br>
&gt; &gt; To: Orit Levin (LCA); <a href=3D"mailto:appsdir@ietf.org">appsdir=
@ietf.org</a><br>
&gt; &gt; Subject: Re: [appsdir] Fwd: Request for Apps Directorate review o=
f rtcweb-<br>
&gt; &gt; security drafts<br>
&gt; &gt;<br>
&gt; &gt; Thank you Orit!=C2=A0 Can you try to have them done in about two =
weeks?<br>
&gt; &gt;<br>
&gt; &gt; Eliot<br>
&gt; &gt;<br>
&gt; &gt; On 6/25/15 9:52 PM, Orit Levin (LCA) wrote:<br>
&gt; &gt; &gt; I will be happy to take a look at them!<br>
&gt; &gt; &gt; Orit.<br>
&gt; &gt; &gt;<br>
&gt; &gt; &gt;&gt; -----Original Message-----<br>
&gt; &gt; &gt;&gt; From: appsdir [mailto:<a href=3D"mailto:appsdir-bounces@=
ietf.org">appsdir-bounces@ietf.org</a>] On Behalf Of Eliot<br>
&gt; Lear<br>
&gt; &gt; &gt;&gt; Sent: Thursday, June 25, 2015 11:44 AM<br>
&gt; &gt; &gt;&gt; To: <a href=3D"mailto:appsdir@ietf.org">appsdir@ietf.org=
</a><br>
&gt; &gt; &gt;&gt; Subject: [appsdir] Fwd: Request for Apps Directorate rev=
iew of rtcweb-<br>
&gt; &gt; &gt;&gt; security drafts<br>
&gt; &gt; &gt;&gt;<br>
&gt; &gt; &gt;&gt; Dear Appsdir folk,<br>
&gt; &gt; &gt;&gt;<br>
&gt; &gt; &gt;&gt; Could I have a volunteer to review the following two dra=
fts?<br>
&gt; &gt; &gt;&gt;<br>
&gt; &gt; &gt;&gt; <a href=3D"https://datatracker.ietf.org/doc/draft-ietf-r=
tcweb-security/" rel=3D"noreferrer" target=3D"_blank">https://datatracker.i=
etf.org/doc/draft-ietf-rtcweb-security/</a><br>
&gt; &gt; &gt;&gt;<br>
&gt; &gt; &gt;&gt; and<br>
&gt; &gt; &gt;&gt;<br>
&gt; &gt; &gt;&gt; <a href=3D"https://datatracker.ietf.org/doc/draft-ietf-r=
tcweb-security-arch/" rel=3D"noreferrer" target=3D"_blank">https://datatrac=
ker.ietf.org/doc/draft-ietf-rtcweb-security-arch/</a><br>
&gt; &gt; &gt;&gt;<br>
&gt; &gt; &gt;&gt; Ted is exempt, having made the request ;-)<br>
&gt; &gt; &gt;&gt;<br>
&gt; &gt; &gt;&gt; Eliot<br>
&gt; &gt;<br>
&gt;<br>
&gt; _______________________________________________<br>
&gt; appsdir mailing list<br>
&gt; <a href=3D"mailto:appsdir@ietf.org">appsdir@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/appsdir" rel=3D"noref=
errer" target=3D"_blank">https://www.ietf.org/mailman/listinfo/appsdir</a><=
br>
<br>
_______________________________________________<br>
appsdir mailing list<br>
<a href=3D"mailto:appsdir@ietf.org">appsdir@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/appsdir" rel=3D"noreferrer=
" target=3D"_blank">https://www.ietf.org/mailman/listinfo/appsdir</a><br>
</div></div></blockquote></div><br></div></div>

--f46d0444038a2f94a8051ad9dd9c--


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Harald Alvestrand <harald@alvestrand.no>, "ranjit@ranjitvoip.com" <ranjit@ranjitvoip.com>, Ted Hardie <ted.ietf@gmail.com>
Thread-Topic: [rtcweb] Updated -gateways draft
Thread-Index: AQHQuDnLo8SeEdvf+k2CJHOD/34edp3O5hsAgAtPjwCAADOdAIABxtog
Date: Wed, 15 Jul 2015 07:42:04 +0000
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To: Harald Alvestrand <harald@alvestrand.no>, "ranjit@ranjitvoip.com" <ranjit@ranjitvoip.com>, Ted Hardie <ted.ietf@gmail.com>
Thread-Topic: [rtcweb] Updated -gateways draft
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Date: Wed, 15 Jul 2015 08:02:25 +0000
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Subject: Re: [rtcweb] Updated -gateways draft
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From nobody Wed Jul 15 01:50:27 2015
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To: "Schwarz, Albrecht (Albrecht)" <albrecht.schwarz@alcatel-lucent.com>,  "ranjit@ranjitvoip.com" <ranjit@ranjitvoip.com>, Ted Hardie <ted.ietf@gmail.com>
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Subject: Re: [rtcweb] Updated -gateways draft
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Thank you very much for this information!

It would seem to me to make sense to have an informative reference from
the gateway draft to the H.248.WEBRTC draft, so that people can find a
real worked example of such a gateway.

The text-over-IP specification would seem to require additional work
outside the scope of RTCWEB - this has been discussed before, and the
decision was to declare it out of scope. I think this will be typical
for applications that connect to gateways - they will require more
things than just WebRTC for a complete specification; thus, having the
example to point to is a Good Thing.

Harald

Den 15. juli 2015 10:02, skrev Schwarz, Albrecht (Albrecht):
> Dear All,
> some background info:
> 
> the 3GPP architecture provides a WebRTC gateway function (since 3GPP Release 12).
> The 3GPP WebRTC gateway entity is based on the decomposed gateway model according ITU-T H.248, i.e., H.248 is used as (vertical) gateway control protocol.
> The 3GPP WebRTC gateway function is mapped on the existing 3GPP P-CSCF/IMS-ALG entity ("the controller part") and the IMS access gateway ("the media gateway part").
> Thus, the 3GPP WebRTC gateway represents actually an "IMS WebRTC gateway" (in Rel-12 and 13).
> 
> The 3GPP specs (from WebRTC gateway perspective) are
> a) 3GPP 23.334, which covers signalling requirements and procedures for the gateway control interface and
> b) 3GPP 29.334, which specifies the protocol solution, known as so called "H.248 profile specification" (in context of H.248).
> 
> The 3GPP WebRTC gateway work is complemented by ITU-T SG16, as the technology owner of the H.248 protocol, as responsible instance for the development of H.248 protocol extensions (called H.248 packages).
> 
> The associated ITU-T work item is "H.248.WEBRTC", - and there will be a first Recommendation for "H.248 WebRTC gateways" soon, under Rec number ITU-T H.248.94.
> The June draft may be found under:
>  http://wftp3.itu.int/av-arch/avc-site/2013-2016/1506_Che/TD-11.zip 
> The October meeting input for next IETU-T meeting is available (for ITU-T delegates) under:
>  https://www.itu.int/md/dologin_md.asp?lang=en&id=T13-SG16-151012-TD-WP1-0278!!MSW-E 
> 
> The scope of the ITU-T H.248 WebRTC gateway function is aligned to IETF RTCWEB requirements in order to provide full support from gateway side. Hence, the "ITU-T H.248 WebRTC gateway function" represents currently a superset in comparison to the "3GPP Rel-12/13 IMS H.248 WebRTC gateway function". All major capabilities are supported, but there' are also some differences ...
> 
> Harald, draft H.248.94 (H.248.WEBRTC) is already refering to draft-rtcweb-gateways, - because the H.248 WebRTC gateway actually just realizes such a WebRTC gateway entity as outlined by the rtcweb draft.
>  
> BTW: I'd like to take the chance to point out some gateway specific observations, which should be addressed as well by the gateway draft. See H.248.WEBRTC, Appendix L6 Living List â€“ Distributed Text-over-IP endpoints for WebRTC data 'text'
> 
> Coming back to the original question: whether it should be Informational or Standards-track, - I don't know. But it is apparent that both, the 3GPP and ITU-T work on WebRTC gateways needs to be tightly coupled and linked to the IETF gateway draft in my opinion.
> 
> Regards,
> Albrecht
> 
> 
> 
> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Harald Alvestrand
> Sent: Dienstag, 14. Juli 2015 08:24
> To: ranjit@ranjitvoip.com; Ted Hardie
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Updated -gateways draft
> 
> Den 14. juli 2015 05:18, skrev ranjit@ranjitvoip.com:
>> Hi
>> It would be good if it is a standards track as 3GPP depends on it and 
>> all implementations would confirm to it . having it as Informational 
>> is really not of much use.
> 
> Could you share some more detail here?
> 
> - What 3GPP spec is depending on it?
> - What does the 3GPP spec depend on being in the draft?
> - What's the rules for referencing an IETF spec from a 3GPP spec, and why does the category matter?
> 
> Harald
> 
> 
>>
>> - Ranjit
>>
>> On 2015-07-06 5:35 pm, Ted Hardie wrote:
>>> On Mon, Jul 6, 2015 at 3:19 PM, Harald Alvestrand 
>>> <harald@alvestrand.no> wrote:
>>>
>>>> Uwe and I have prepared a new -gateways draft (version -01).
>>>>
>>>> I don't think there's much controversial in it, and there's only one 
>>>> issue that I think needs WG time at this time:
>>>>
>>>> Should it be Informational or Standards-track?
>>>>
>>>> An informational document defines guidance for how gateways should 
>>>> behave, but nobody needs to pay attention if they don't want to.
>>>>
>>>> A standards-track document (which no other RTCWEB document should be 
>>>> dependent on) defines requirements for how things that call 
>>>> themselves "WebRTC gateways" should behave, but nobody needs to call 
>>>> their gateways that, and anyway, there's no protocol police, so the 
>>>> difference isn't all that much in practice.
>>>>
>>>> More important: What do people want it to be?
>>>
>>> â€‹I think we adopted it in part because 3GPP had a dependency on it.
>>> Does it need to be standards track for them to reference it?
>>>
>>> Tedâ€‹
>>>
>>>> Harald
>>>>
>>>> _______________________________________________
>>>> rtcweb mailing list
>>>> rtcweb@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/rtcweb [1]
>>>
>>>
>>>
>>> Links:
>>> ------
>>> [1] 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 nobody Wed Jul 15 05:38:06 2015
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Date: Wed, 15 Jul 2015 14:38:03 +0200
From: carlo von lynX <lynX@i.know.you.are.psyced.org>
To: rtcweb@ietf.org
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Subject: [rtcweb] Prompt for mic, camera *and* network access
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It's not such a big deal, even casual readers understand the
problem and suggest the simple solution:

Someone called Michael writes:
    "local ip doesnâ€™t seem that useful for marketers except as a user fingerprinting tool. They already have your public ip, this helps them differentiate between people behind nat. itâ€™s a bit icky but not such a big deal. This issue blows up again when someone starts using it maliciously, which Iâ€™m sure will happen soon enough. I donâ€™t get why exactly we donâ€™t just prompt for this the same way we do camera and mic, it wouldnâ€™t be a huge deal to work that into the spec. That being said, I donâ€™t think itâ€™s actually as big of a deal as it has been made either"

(from https://bloggeek.me/webrtc-new-york-times/ - yet another privacy
 issue with mainstream web browsers, this time thanks to webrtc)

The reason why this is a big deal is because they do NOT already have 
the public IP for some people and de-anonymizing Tor or VPN users can
have life-threatening consequences as described below.
Let me look at your official statements on the topic:

https://tools.ietf.org/html/draft-ietf-rtcweb-security-arch-11#page-15
says on the topic of "5.4.  IP Location Privacy" :

	"The API MUST provide a mechanism to allow the JS to
      suppress ICE negotiation (though perhaps to allow candidate
      gathering) until the user has decided to answer the call"

Eh? This is not solving the problem at all since the JS isn't in
the slightest trustworthy.

	"The API MUST provide a mechanism for the calling application 
      JS to indicate that only TURN candidates are to be used."

Dito.

	"The API MUST provide a mechanism for the calling
      application to reconfigure an existing call to add non-TURN
      candidates.  Taken together, this and the previous requirement
      allow ICE negotiation to start immediately on incoming call
      notification, thus reducing post-dial delay, but also to avoid
      disclosing the user's IP address until they have decided to
      answer."

Yet nothing of it actually solves the issue it is supposed to solve,
protecting IP Location Privacy, except when the involved websites
are interested in protecting users from each other - which is just
the least interesting threat model case.

Further below you wrote:

	"Note that the protections provided here assume a non-malicious
   calling service.  As the calling service always knows the users
   status and (absent the use of a technology like Tor) their IP
   address, they can violate the users privacy at will.  Users who wish
   privacy against the calling sites they are using must use separate
   privacy enhancing technologies such as Tor. Combined WebRTC/Tor
   implementations SHOULD arrange to route the media as well as the
   signaling through Tor. Currently this will produce very suboptimal
   performance."

Performance is irrelevant here. The problem is that WebRTC APIs are
available for abuse even if people are not in the slightest interested
in making actual use of them. The entire security assessment document
is bypassing the real threat, as if Tor by itself could solve all
problems and it were enough to just mention it.

So I'll repeat Michael's sort-of key question:

    "I donâ€™t get why exactly we donâ€™t just prompt for this the same way we do camera and mic,"

It could be a single transaction, asking the user to allow access to
all three things in order to allow for rtc apps to be initiated.
There isn't even a need to make an *additional* prompt.

On Tue, Apr 21, 2015 at 05:46:58PM +0200, carlo von lynX wrote:
> Envision the typical young dissident with his Windows PC using
> some VPN to access critical information. My impression is these
> folks are not aware of how much safer they could be by using
> TBB with Tor bridges, and for their purposes all the risks of
> getting fingerprinted aren't even relevant *as long as the
> authorities can't find a way to follow the fingerprinted browser
> to its physical position* (something the NSA probably can, but
> the NSA is outside of this threat model). So until everyone in 
> their social surroundings hasn't been taken away to prison, they 
> will continue as always - and if the web browser in the meantime
> has introduced a way for authorities to reveal their whereabouts,
> that could cost some lives.
> 
> Possibly all the other bugs in the JS "security" model are not 
> half as dangerous to these people than the ability for an authority
> to inject JS code that will reveal the physical IP number of a
> dissident.
> 
> Correct me if I got something wrong but to me it looks like we
> have a serious case of lives at risk, in particular since this
> weakness is now known to the general public.. those authorities
> may be slow and dissidents like to make fun of them in order to
> feel less helpless.. but these authorities might just pick any 
> time soon to start flying webrtc drones into living people.
>  
> I hope I got something wrong.

If nobody answered, I must assume I didn't get anything wrong
(otherwise somebody would have been proud to show me how wrong I am).

No, there exists no technology mailing list that doesn't have
a political aspect to it, because technology is political at all
times. But of course you can go on pretending people like me are
a big bunch of trolls.. amazing how many trolls there are out there.

Try facing the responsibility that lies upon you and you alone
because corporations are by definition non-ethical. You, the
technician, are the only ethical element in the equation. If you
don't speak up on this issue, you will be responsible. Time is
not on your side and the later you act, the more expensive it
will get. Acting irresponsible now will harm the public image
of WebRTC more and more. As Michael suggests it is only a question
of time until some wikileak tells a story of WebRTC being abused
for catching dissidents, or whatever else could possibly go wrong.

I don't care, I don't have any stakes in this business. I don't 
mind if more browsers follow the good example of Torbrowser to 
disable WebRTC entirely.


-- 
  E-mail is public! Talk to me in private using encryption:
         http://loupsycedyglgamf.onion/LynX/
          irc://loupsycedyglgamf.onion:67/lynX
         https://psyced.org:34443/LynX/


From nobody Wed Jul 15 06:49:01 2015
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Subject: Re: [rtcweb] Prompt for mic, camera *and* network access
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<Rant>
There are 4 ip addresses involved here -
Lets assume a user with a PC - wifi connected to an ADSL router and =
using a VPN

1) External VPN address : the IP address of the exit node from the VPN - =
this is already known to everyone.
2) Internal VPN address : the IP address of the VPN interface on the PC =
end of the VPN tunnel - not known - but not =E2=80=98secret'
3) ISP address : the address allocated by the ISP for the ADSL router - =
this is the one that TOR and other VPNs mask
4) local IP address: the address returned by the router - typically =
192.168.0.2 or some such - this is not known=20

With current VPN configs (and I assume TOR) webRTC reveals all of these.
The critical one is  ISP address (correct me if I=E2=80=99m wrong) - =
This can reveal location and identity via the ISP=E2=80=99s business =
database.

A small change to the VPN configuration will hide the ISP address - you =
just have to set the default route to 2) rather than 3)
then the STUN queries will return the external VPN address - which is =
what you revealed anyway by loading the page.

'Just ask the user to enable webRTC' is a sledgehammer to crack a nut.=20=


You are forgoing the huge advantages (including privacy) of P2P crypto =
in webRTC just because the existing VPN=20
suppliers can=E2=80=99t figure out a default route and update their =
scripts.

The cynical part of me wonders if people with skin in the VPN game =
don=E2=80=99t want an easy to use P2P crypto in the browser=E2=80=A6..
</Rant>

Tim.
P.S. actually I=E2=80=99m =E2=80=98surprisingly reasonable=E2=80=99 in =
person - so I=E2=80=99m happy to discuss off-line if that helps.
P.P.S I have a second rant about why you want P2P to travel over the =
local net where possible - but that can wait=E2=80=A6.


> On 15 Jul 2015, at 13:38, carlo von lynX =
<lynX@i.know.you.are.psyced.org> wrote:
>=20
> It's not such a big deal, even casual readers understand the
> problem and suggest the simple solution:
>=20
> Someone called Michael writes:
>    "local ip doesn=E2=80=99t seem that useful for marketers except as =
a user fingerprinting tool. They already have your public ip, this helps =
them differentiate between people behind nat. it=E2=80=99s a bit icky =
but not such a big deal. This issue blows up again when someone starts =
using it maliciously, which I=E2=80=99m sure will happen soon enough. I =
don=E2=80=99t get why exactly we don=E2=80=99t just prompt for this the =
same way we do camera and mic, it wouldn=E2=80=99t be a huge deal to =
work that into the spec. That being said, I don=E2=80=99t think it=E2=80=99=
s actually as big of a deal as it has been made either"
>=20
> (from https://bloggeek.me/webrtc-new-york-times/ - yet another privacy
> issue with mainstream web browsers, this time thanks to webrtc)
>=20
> The reason why this is a big deal is because they do NOT already have=20=

> the public IP for some people and de-anonymizing Tor or VPN users can
> have life-threatening consequences as described below.
> Let me look at your official statements on the topic:
>=20
> https://tools.ietf.org/html/draft-ietf-rtcweb-security-arch-11#page-15
> says on the topic of "5.4.  IP Location Privacy" :
>=20
> 	"The API MUST provide a mechanism to allow the JS to
>      suppress ICE negotiation (though perhaps to allow candidate
>      gathering) until the user has decided to answer the call"
>=20
> Eh? This is not solving the problem at all since the JS isn't in
> the slightest trustworthy.
>=20
> 	"The API MUST provide a mechanism for the calling application=20
>      JS to indicate that only TURN candidates are to be used."
>=20
> Dito.
>=20
> 	"The API MUST provide a mechanism for the calling
>      application to reconfigure an existing call to add non-TURN
>      candidates.  Taken together, this and the previous requirement
>      allow ICE negotiation to start immediately on incoming call
>      notification, thus reducing post-dial delay, but also to avoid
>      disclosing the user's IP address until they have decided to
>      answer."
>=20
> Yet nothing of it actually solves the issue it is supposed to solve,
> protecting IP Location Privacy, except when the involved websites
> are interested in protecting users from each other - which is just
> the least interesting threat model case.
>=20
> Further below you wrote:
>=20
> 	"Note that the protections provided here assume a non-malicious
>   calling service.  As the calling service always knows the users
>   status and (absent the use of a technology like Tor) their IP
>   address, they can violate the users privacy at will.  Users who wish
>   privacy against the calling sites they are using must use separate
>   privacy enhancing technologies such as Tor. Combined WebRTC/Tor
>   implementations SHOULD arrange to route the media as well as the
>   signaling through Tor. Currently this will produce very suboptimal
>   performance."
>=20
> Performance is irrelevant here. The problem is that WebRTC APIs are
> available for abuse even if people are not in the slightest interested
> in making actual use of them. The entire security assessment document
> is bypassing the real threat, as if Tor by itself could solve all
> problems and it were enough to just mention it.
>=20
> So I'll repeat Michael's sort-of key question:
>=20
>    "I don=E2=80=99t get why exactly we don=E2=80=99t just prompt for =
this the same way we do camera and mic,"
>=20
> It could be a single transaction, asking the user to allow access to
> all three things in order to allow for rtc apps to be initiated.
> There isn't even a need to make an *additional* prompt.
>=20
> On Tue, Apr 21, 2015 at 05:46:58PM +0200, carlo von lynX wrote:
>> Envision the typical young dissident with his Windows PC using
>> some VPN to access critical information. My impression is these
>> folks are not aware of how much safer they could be by using
>> TBB with Tor bridges, and for their purposes all the risks of
>> getting fingerprinted aren't even relevant *as long as the
>> authorities can't find a way to follow the fingerprinted browser
>> to its physical position* (something the NSA probably can, but
>> the NSA is outside of this threat model). So until everyone in=20
>> their social surroundings hasn't been taken away to prison, they=20
>> will continue as always - and if the web browser in the meantime
>> has introduced a way for authorities to reveal their whereabouts,
>> that could cost some lives.
>>=20
>> Possibly all the other bugs in the JS "security" model are not=20
>> half as dangerous to these people than the ability for an authority
>> to inject JS code that will reveal the physical IP number of a
>> dissident.
>>=20
>> Correct me if I got something wrong but to me it looks like we
>> have a serious case of lives at risk, in particular since this
>> weakness is now known to the general public.. those authorities
>> may be slow and dissidents like to make fun of them in order to
>> feel less helpless.. but these authorities might just pick any=20
>> time soon to start flying webrtc drones into living people.
>>=20
>> I hope I got something wrong.
>=20
> If nobody answered, I must assume I didn't get anything wrong
> (otherwise somebody would have been proud to show me how wrong I am).
>=20
> No, there exists no technology mailing list that doesn't have
> a political aspect to it, because technology is political at all
> times. But of course you can go on pretending people like me are
> a big bunch of trolls.. amazing how many trolls there are out there.
>=20
> Try facing the responsibility that lies upon you and you alone
> because corporations are by definition non-ethical. You, the
> technician, are the only ethical element in the equation. If you
> don't speak up on this issue, you will be responsible. Time is
> not on your side and the later you act, the more expensive it
> will get. Acting irresponsible now will harm the public image
> of WebRTC more and more. As Michael suggests it is only a question
> of time until some wikileak tells a story of WebRTC being abused
> for catching dissidents, or whatever else could possibly go wrong.
>=20
> I don't care, I don't have any stakes in this business. I don't=20
> mind if more browsers follow the good example of Torbrowser to=20
> disable WebRTC entirely.
>=20
>=20
> --=20
>  E-mail is public! Talk to me in private using encryption:
>         http://loupsycedyglgamf.onion/LynX/
>          irc://loupsycedyglgamf.onion:67/lynX
>         https://psyced.org:34443/LynX/
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From nobody Wed Jul 15 06:52:59 2015
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From: Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com>
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Date: Wed, 15 Jul 2015 15:52:59 +0200
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Subject: Re: [rtcweb] Prompt for mic, camera *and* network access
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+1

I would add another type for completeness:

5) external HTTP Proxy, which behaves like 1)

Best regards
Sergio


On 15/07/2015 15:48, tim panton wrote:
> <Rant>
> There are 4 ip addresses involved here -
> Lets assume a user with a PC - wifi connected to an ADSL router and using a VPN
>
> 1) External VPN address : the IP address of the exit node from the VPN - this is already known to everyone.
> 2) Internal VPN address : the IP address of the VPN interface on the PC end of the VPN tunnel - not known - but not â€˜secret'
> 3) ISP address : the address allocated by the ISP for the ADSL router - this is the one that TOR and other VPNs mask
> 4) local IP address: the address returned by the router - typically 192.168.0.2 or some such - this is not known
>
> With current VPN configs (and I assume TOR) webRTC reveals all of these.
> The critical one is  ISP address (correct me if Iâ€™m wrong) - This can reveal location and identity via the ISPâ€™s business database.
>
> A small change to the VPN configuration will hide the ISP address - you just have to set the default route to 2) rather than 3)
> then the STUN queries will return the external VPN address - which is what you revealed anyway by loading the page.
>
> 'Just ask the user to enable webRTC' is a sledgehammer to crack a nut.
>
> You are forgoing the huge advantages (including privacy) of P2P crypto in webRTC just because the existing VPN
> suppliers canâ€™t figure out a default route and update their scripts.
>
> The cynical part of me wonders if people with skin in the VPN game donâ€™t want an easy to use P2P crypto in the browserâ€¦..
> </Rant>
>
> Tim.
> P.S. actually Iâ€™m â€˜surprisingly reasonableâ€™ in person - so Iâ€™m happy to discuss off-line if that helps.
> P.P.S I have a second rant about why you want P2P to travel over the local net where possible - but that can waitâ€¦.
>
>
>> On 15 Jul 2015, at 13:38, carlo von lynX <lynX@i.know.you.are.psyced.org> wrote:
>>
>> It's not such a big deal, even casual readers understand the
>> problem and suggest the simple solution:
>>
>> Someone called Michael writes:
>>     "local ip doesnâ€™t seem that useful for marketers except as a user fingerprinting tool. They already have your public ip, this helps them differentiate between people behind nat. itâ€™s a bit icky but not such a big deal. This issue blows up again when someone starts using it maliciously, which Iâ€™m sure will happen soon enough. I donâ€™t get why exactly we donâ€™t just prompt for this the same way we do camera and mic, it wouldnâ€™t be a huge deal to work that into the spec. That being said, I donâ€™t think itâ€™s actually as big of a deal as it has been made either"
>>
>> (from https://bloggeek.me/webrtc-new-york-times/ - yet another privacy
>> issue with mainstream web browsers, this time thanks to webrtc)
>>
>> The reason why this is a big deal is because they do NOT already have
>> the public IP for some people and de-anonymizing Tor or VPN users can
>> have life-threatening consequences as described below.
>> Let me look at your official statements on the topic:
>>
>> https://tools.ietf.org/html/draft-ietf-rtcweb-security-arch-11#page-15
>> says on the topic of "5.4.  IP Location Privacy" :
>>
>> 	"The API MUST provide a mechanism to allow the JS to
>>       suppress ICE negotiation (though perhaps to allow candidate
>>       gathering) until the user has decided to answer the call"
>>
>> Eh? This is not solving the problem at all since the JS isn't in
>> the slightest trustworthy.
>>
>> 	"The API MUST provide a mechanism for the calling application
>>       JS to indicate that only TURN candidates are to be used."
>>
>> Dito.
>>
>> 	"The API MUST provide a mechanism for the calling
>>       application to reconfigure an existing call to add non-TURN
>>       candidates.  Taken together, this and the previous requirement
>>       allow ICE negotiation to start immediately on incoming call
>>       notification, thus reducing post-dial delay, but also to avoid
>>       disclosing the user's IP address until they have decided to
>>       answer."
>>
>> Yet nothing of it actually solves the issue it is supposed to solve,
>> protecting IP Location Privacy, except when the involved websites
>> are interested in protecting users from each other - which is just
>> the least interesting threat model case.
>>
>> Further below you wrote:
>>
>> 	"Note that the protections provided here assume a non-malicious
>>    calling service.  As the calling service always knows the users
>>    status and (absent the use of a technology like Tor) their IP
>>    address, they can violate the users privacy at will.  Users who wish
>>    privacy against the calling sites they are using must use separate
>>    privacy enhancing technologies such as Tor. Combined WebRTC/Tor
>>    implementations SHOULD arrange to route the media as well as the
>>    signaling through Tor. Currently this will produce very suboptimal
>>    performance."
>>
>> Performance is irrelevant here. The problem is that WebRTC APIs are
>> available for abuse even if people are not in the slightest interested
>> in making actual use of them. The entire security assessment document
>> is bypassing the real threat, as if Tor by itself could solve all
>> problems and it were enough to just mention it.
>>
>> So I'll repeat Michael's sort-of key question:
>>
>>     "I donâ€™t get why exactly we donâ€™t just prompt for this the same way we do camera and mic,"
>>
>> It could be a single transaction, asking the user to allow access to
>> all three things in order to allow for rtc apps to be initiated.
>> There isn't even a need to make an *additional* prompt.
>>
>> On Tue, Apr 21, 2015 at 05:46:58PM +0200, carlo von lynX wrote:
>>> Envision the typical young dissident with his Windows PC using
>>> some VPN to access critical information. My impression is these
>>> folks are not aware of how much safer they could be by using
>>> TBB with Tor bridges, and for their purposes all the risks of
>>> getting fingerprinted aren't even relevant *as long as the
>>> authorities can't find a way to follow the fingerprinted browser
>>> to its physical position* (something the NSA probably can, but
>>> the NSA is outside of this threat model). So until everyone in
>>> their social surroundings hasn't been taken away to prison, they
>>> will continue as always - and if the web browser in the meantime
>>> has introduced a way for authorities to reveal their whereabouts,
>>> that could cost some lives.
>>>
>>> Possibly all the other bugs in the JS "security" model are not
>>> half as dangerous to these people than the ability for an authority
>>> to inject JS code that will reveal the physical IP number of a
>>> dissident.
>>>
>>> Correct me if I got something wrong but to me it looks like we
>>> have a serious case of lives at risk, in particular since this
>>> weakness is now known to the general public.. those authorities
>>> may be slow and dissidents like to make fun of them in order to
>>> feel less helpless.. but these authorities might just pick any
>>> time soon to start flying webrtc drones into living people.
>>>
>>> I hope I got something wrong.
>> If nobody answered, I must assume I didn't get anything wrong
>> (otherwise somebody would have been proud to show me how wrong I am).
>>
>> No, there exists no technology mailing list that doesn't have
>> a political aspect to it, because technology is political at all
>> times. But of course you can go on pretending people like me are
>> a big bunch of trolls.. amazing how many trolls there are out there.
>>
>> Try facing the responsibility that lies upon you and you alone
>> because corporations are by definition non-ethical. You, the
>> technician, are the only ethical element in the equation. If you
>> don't speak up on this issue, you will be responsible. Time is
>> not on your side and the later you act, the more expensive it
>> will get. Acting irresponsible now will harm the public image
>> of WebRTC more and more. As Michael suggests it is only a question
>> of time until some wikileak tells a story of WebRTC being abused
>> for catching dissidents, or whatever else could possibly go wrong.
>>
>> I don't care, I don't have any stakes in this business. I don't
>> mind if more browsers follow the good example of Torbrowser to
>> disable WebRTC entirely.
>>
>>
>> -- 
>>   E-mail is public! Talk to me in private using encryption:
>>          http://loupsycedyglgamf.onion/LynX/
>>           irc://loupsycedyglgamf.onion:67/lynX
>>          https://psyced.org:34443/LynX/
>>
>> _______________________________________________
>> 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 15, 2015 5:38 AM, "carlo von lynX" <lynX@i.know.you.are.psyced.org>
wrote:
>
> As Michael suggests it is only a question
> of time until some wikileak tells a story of WebRTC being abused
> for catching dissidents, or whatever else could possibly go wrong.
>
> I don't care, I don't have any stakes in this business. I don't
> mind if more browsers follow the good example of Torbrowser to
> disable WebRTC entirely.
>

FYI, in the leaked Hacking Team emails, they discussed using the WebRTC
exploit.

https://wikileaks.org/hackingteam/emails/emailid/119217

--001a1149781c28c80a051aeaee29
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<p dir="ltr">On Jul 15, 2015 5:38 AM, &quot;carlo von lynX&quot; &lt;<a href="mailto:lynX@i.know.you.are.psyced.org">lynX@i.know.you.are.psyced.org</a>&gt; wrote:<br>
&gt;<br>
&gt; As Michael suggests it is only a question<br>
&gt; of time until some wikileak tells a story of WebRTC being abused<br>
&gt; for catching dissidents, or whatever else could possibly go wrong.<br>
&gt;<br>
&gt; I don&#39;t care, I don&#39;t have any stakes in this business. I don&#39;t<br>
&gt; mind if more browsers follow the good example of Torbrowser to<br>
&gt; disable WebRTC entirely.<br>
&gt;</p>
<p dir="ltr">FYI, in the leaked Hacking Team emails, they discussed using the WebRTC exploit.</p>
<p dir="ltr"><a href="https://wikileaks.org/hackingteam/emails/emailid/119217">https://wikileaks.org/hackingteam/emails/emailid/119217</a></p>

--001a1149781c28c80a051aeaee29--


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> On Jul 15, 2015, at 2:02 AM, Schwarz, Albrecht (Albrecht) =
<albrecht.schwarz@alcatel-lucent.com> wrote:
>=20
> Coming back to the original question: whether it should be =
Informational or Standards-track, - I don't know. But it is apparent =
that both, the 3GPP and ITU-T work on WebRTC gateways needs to be =
tightly coupled and linked to the IETF gateway draft in my opinion.

I'd be happy for Keith to correct me but my understanding is that 3GPP =
does not really differentiate much on the RFC track and can have =
normative references to informational RFC.=20



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Concerning the HT mail, of course they find a local IP of little
interest when you already p0wn the entire computer.

On Wed, Jul 15, 2015 at 02:48:52PM +0100, tim panton wrote:
> <Rant>

You are welcome to rant AFTER you have done your homework.
Instead you postponed it by several weeks which made it worse.

> There are 4 ip addresses involved here -
> Lets assume a user with a PC - wifi connected to an ADSL router and using a VPN
> 
> 1) External VPN address : the IP address of the exit node from the VPN - this is already known to everyone.
> 2) Internal VPN address : the IP address of the VPN interface on the PC end of the VPN tunnel - not known - but not â€˜secret'
> 3) ISP address : the address allocated by the ISP for the ADSL router - this is the one that TOR and other VPNs mask
> 4) local IP address: the address returned by the router - typically 192.168.0.2 or some such - this is not known 
> 
> With current VPN configs (and I assume TOR) webRTC reveals all of these.
> The critical one is  ISP address (correct me if Iâ€™m wrong) - This can reveal location and identity via the ISPâ€™s business database.

Point one, all of this discussion and explanation belongs into
draft-ietf-rtcweb-security-arch. That is your homework for today.

> A small change to the VPN configuration will hide the ISP address - you just have to set the default route to 2) rather than 3)
> then the STUN queries will return the external VPN address - which is what you revealed anyway by loading the page.

This does not sound like something your average Egyptian
dissident will be able and sufficiently early informed about
to go about.

I frequently see this pattern of saying privacy problem X
can be solved by manual intervention Y and it is not our
problem if the general public will be unable to do such a
manual intervention.

Yes it is your problem.

It is not okay to dismiss problems this way.

Also the "solution" you suggest AFAIU does not work for people
who use Tor in a non-NATted environment, for example on a
fixed system in a university library with a static IP address.
We know that browsers will allow fingerprinting even through
Tor but it is a different calibre of problem if a browser
actually allows the attacker to find out the REAL public IP
of a person.

> 'Just ask the user to enable webRTC' is a sledgehammer to crack a nut. 

No. It's like asking the user to click on the telephone icon
before making a phone call. It is normal. What is abnormal is
to have a web browser that is supposed to browse the web be
enabled to do crazy non-web things without asking anyone.

You guys are completely losing the perspective of normal human beings.

And it's nothing new, you know these documents yourself:

https://www.torproject.org/projects/torbrowser/design/#proxy-obedience

WebRTC not respecting SOCKS configuration is a huge no-go.

https://www.torproject.org/projects/torbrowser/design/#fingerprinting-linkability

And if local IP information is so super harmless, then why is it
an obvious reason for TBB devs to disable the entire thing?
I presume for the reasons I stated before.

So when you do your homework please include all the points that
I am making. Add your solid arguments to counter them.

> You are forgoing the huge advantages (including privacy) of P2P crypto in webRTC just because the existing VPN 
> suppliers canâ€™t figure out a default route and update their scripts.

I don't know about VPN suppliers, I never used any VPN really,
but it sounds like you have a moral obligation to venture out
and inform them about the security problem you are introducing
and to make sure they protect their users before anyone gets
harmed.

Also don't tell me any fairy tales about P2P crypto in web
browsers. There is no architectural way you can provide E2E
crypto that isn't in one way or another MITM'able by the
involved servers. You are at *best* adding to the choice of
opportunistic cryptography. Don't be so excessively proud
of that considering that REAL P2P with REAL E2E authentication
is out there and would have deserved better attention.

> The cynical part of me wonders if people with skin in the VPN game donâ€™t want an easy to use P2P crypto in the browserâ€¦..

No idea about that, but I am certain the number of people with skin
in the WebRTC game are in the majority on this mailing list - and
you wouldn't like to be ignored in any privacy debate a priori just
because you're in the WebRTC game, right? But if you go on this way,
that is likely to be the outcome. By the time you start thinking of
the consequences of your actions your credibility may already be
gone downhill.

> </Rant>

Thanks for exposing your feelings.

> P.S. actually Iâ€™m â€˜surprisingly reasonableâ€™ in person - so Iâ€™m happy to discuss off-line if that helps.

Me too, but this is very much on topic.

> P.P.S I have a second rant about why you want P2P to travel over the local net where possible - but that can waitâ€¦.

I already have P2P over the local net, so the local IP
is indeed relevant in my configuration.


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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: Cullen Jennings <fluffy@iii.ca>, "Schwarz, Albrecht (Albrecht)" <albrecht.schwarz@alcatel-lucent.com>
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In general, 3GPP can quite happily normatively reference both informational=
 and standards track documents. The key is of course that the referenced ma=
terial is suitable, i.e. that normatively referenced material is specified =
in a suitable fashion to be a normative requirements. In previous mails I h=
ave given at least one example of such requirements.

However Albrecht does introduce a different complication.

3GPP does also normatively reference ITU-T documents for the H.248 function=
ality, and if the ITU-T documents need then to normatively reference the IE=
TF document to support that, then if I recall correctly, ITU-T does require=
 that to be a standards track document.

What is clear is that we will have a mixture of normative material and info=
rmational material. It is important that that distinction is clear. If that=
 is so, what is the issue with having a standards track document?

Keith=20

> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of=20
> Cullen Jennings
> Sent: 15 July 2015 16:43
> To: Schwarz, Albrecht (Albrecht)
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Updated -gateways draft
>=20
>=20
> > On Jul 15, 2015, at 2:02 AM, Schwarz, Albrecht (Albrecht)=20
> <albrecht.schwarz@alcatel-lucent.com> wrote:
> >=20
> > Coming back to the original question: whether it should be=20
> Informational or Standards-track, - I don't know. But it is=20
> apparent that both, the 3GPP and ITU-T work on WebRTC=20
> gateways needs to be tightly coupled and linked to the IETF=20
> gateway draft in my opinion.
>=20
> I'd be happy for Keith to correct me but my understanding is=20
> that 3GPP does not really differentiate much on the RFC track=20
> and can have normative references to informational RFC.=20
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> =


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> On 15 Jul 2015, at 17:41, carlo von lynX =
<lynX@i.know.you.are.psyced.org <mailto:lynX@i.know.you.are.psyced.org>> =
wrote:
>=20
> Concerning the HT mail, of course they find a local IP of little
> interest when you already p0wn the entire computer.
>=20
> On Wed, Jul 15, 2015 at 02:48:52PM +0100, tim panton wrote:
>> <Rant>
>=20
> You are welcome to rant AFTER you have done your homework.
> Instead you postponed it by several weeks which made it worse.

Um, if you check back in the archives, I brought this issue up earlier =
this year.
I think it is important.

https://mailarchive.ietf.org/arch/msg/rtcweb/VnStGD7Emmvc1-QOPRtY2cFF79E =
<https://mailarchive.ietf.org/arch/msg/rtcweb/VnStGD7Emmvc1-QOPRtY2cFF79E>=


>=20
>> There are 4 ip addresses involved here -
>> Lets assume a user with a PC - wifi connected to an ADSL router and =
using a VPN
>>=20
>> 1) External VPN address : the IP address of the exit node from the =
VPN - this is already known to everyone.
>> 2) Internal VPN address : the IP address of the VPN interface on the =
PC end of the VPN tunnel - not known - but not =E2=80=98secret'
>> 3) ISP address : the address allocated by the ISP for the ADSL router =
- this is the one that TOR and other VPNs mask
>> 4) local IP address: the address returned by the router - typically =
192.168.0.2 or some such - this is not known=20
>>=20
>> With current VPN configs (and I assume TOR) webRTC reveals all of =
these.
>> The critical one is  ISP address (correct me if I=E2=80=99m wrong) - =
This can reveal location and identity via the ISP=E2=80=99s business =
database.
>=20
> Point one, all of this discussion and explanation belongs into
> draft-ietf-rtcweb-security-arch. That is your homework for today.

Thanks.

>=20
>> A small change to the VPN configuration will hide the ISP address - =
you just have to set the default route to 2) rather than 3)
>> then the STUN queries will return the external VPN address - which is =
what you revealed anyway by loading the page.
>=20
> This does not sound like something your average Egyptian
> dissident will be able and sufficiently early informed about
> to go about.

I have zero expectation that the Egyptian dissident should do this.=20
I do expect their software provider to keep up with developments.

>=20
> I frequently see this pattern of saying privacy problem X
> can be solved by manual intervention Y and it is not our
> problem if the general public will be unable to do such a
> manual intervention.
>=20
> Yes it is your problem.

That isn=E2=80=99t what I meant. I expect secure software providers to =
keep their
software abreast of new developments.=20

>=20
> It is not okay to dismiss problems this way.

I=E2=80=99m really sorry you think I=E2=80=99m dismissing this problem, =
I=E2=80=99m not.
I happen to have the (no doubt under informed) opinion that
this isn=E2=80=99t as simple as it has been made out to be.

>=20
> Also the "solution" you suggest AFAIU does not work for people
> who use Tor in a non-NATted environment, for example on a
> fixed system in a university library with a static IP address.

That is an interesting case. In that instance hiding the local IP would =
be=20
valuable. I thought however those users would normally run a TOR os off=20=

a usb stick in a VM. That would probably mask the local IP.


> We know that browsers will allow fingerprinting even through
> Tor but it is a different calibre of problem if a browser
> actually allows the attacker to find out the REAL public IP
> of a person.

I totally agree. My solution is imho a better fix, since it
also protects against many other apps that might get started from
a browser page (email, dns, ipv6 tunnels, flash, mp3 players etc).


>=20
>> 'Just ask the user to enable webRTC' is a sledgehammer to crack a =
nut.=20
>=20
> No. It's like asking the user to click on the telephone icon
> before making a phone call. It is normal. What is abnormal is
> to have a web browser that is supposed to browse the web be
> enabled to do crazy non-web things without asking anyone.

No, really not. I want the VPN folks and the Tor folks to update their
scripts after consultations about the best way forward.

>=20
> You guys are completely losing the perspective of normal human beings.
>=20
> And it's nothing new, you know these documents yourself:
>=20
> https://www.torproject.org/projects/torbrowser/design/#proxy-obedience =
<https://www.torproject.org/projects/torbrowser/design/#proxy-obedience>
>=20
> WebRTC not respecting SOCKS configuration is a huge no-go.

I=E2=80=99m SOCKs ignorant.

>=20
> =
https://www.torproject.org/projects/torbrowser/design/#fingerprinting-link=
ability =
<https://www.torproject.org/projects/torbrowser/design/#fingerprinting-lin=
kability>
>=20
> And if local IP information is so super harmless, then why is it
> an obvious reason for TBB devs to disable the entire thing?
> I presume for the reasons I stated before.

I have no idea, but I fear it is because no one has split these issues =
out.

>=20
> So when you do your homework please include all the points that
> I am making. Add your solid arguments to counter them.
>=20
>> You are forgoing the huge advantages (including privacy) of P2P =
crypto in webRTC just because the existing VPN=20
>> suppliers can=E2=80=99t figure out a default route and update their =
scripts.
>=20
> I don't know about VPN suppliers, I never used any VPN really,
> but it sounds like you have a moral obligation to venture out
> and inform them about the security problem you are introducing
> and to make sure they protect their users before anyone gets
> harmed.

Yep, I=E2=80=99ve been doing that, I clearly haven=E2=80=99t spoken to =
enough people
but there is only so much homework I can do.

>=20
> Also don't tell me any fairy tales about P2P crypto in web
> browsers. There is no architectural way you can provide E2E
> crypto that isn't in one way or another MITM'able by the
> involved servers. You are at *best* adding to the choice of
> opportunistic cryptography. Don't be so excessively proud
> of that considering that REAL P2P with REAL E2E authentication
> is out there and would have deserved better attention.

I think putting PFS strong crypto in the hands of 2bn people and=20
1m web developers is something to be slightly proud of.

I=E2=80=99m of the view that there are _useful_ instances of P2P crypto
in-browser - we will clearly have to disagree about that.


>=20
>> The cynical part of me wonders if people with skin in the VPN game =
don=E2=80=99t want an easy to use P2P crypto in the browser=E2=80=A6..
>=20
> No idea about that, but I am certain the number of people with skin
> in the WebRTC game are in the majority on this mailing list - and
> you wouldn't like to be ignored in any privacy debate a priori just
> because you're in the WebRTC game, right? But if you go on this way,
> that is likely to be the outcome. By the time you start thinking of
> the consequences of your actions your credibility may already be
> gone downhill.

I=E2=80=99m not ignoring anyone or their views, just challenging the =
conclusions they
have drawn by providing a different viewpoint.

T.



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charset=3Dutf-8"><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dutf-8"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" =
class=3D""><br class=3D""><div class=3D""><blockquote type=3D"cite" =
class=3D""><div class=3D"">On 15 Jul 2015, at 17:41, carlo von lynX =
&lt;<a href=3D"mailto:lynX@i.know.you.are.psyced.org" =
class=3D"">lynX@i.know.you.are.psyced.org</a>&gt; wrote:</div><br =
class=3D"Apple-interchange-newline"><div class=3D"">Concerning the HT =
mail, of course they find a local IP of little<br class=3D"">interest =
when you already p0wn the entire computer.<br class=3D""><br class=3D"">On=
 Wed, Jul 15, 2015 at 02:48:52PM +0100, tim panton wrote:<br =
class=3D""><blockquote type=3D"cite" class=3D"">&lt;Rant&gt;<br =
class=3D""></blockquote><br class=3D"">You are welcome to rant AFTER you =
have done your homework.<br class=3D"">Instead you postponed it by =
several weeks which made it worse.<br class=3D""></div></blockquote><div =
class=3D""><br class=3D""></div><div class=3D"">Um, if you check back in =
the archives, I brought this issue up earlier this year.</div><div =
class=3D"">I think it is important.</div><div class=3D""><br =
class=3D""></div><a =
href=3D"https://mailarchive.ietf.org/arch/msg/rtcweb/VnStGD7Emmvc1-QOPRtY2=
cFF79E" =
class=3D"">https://mailarchive.ietf.org/arch/msg/rtcweb/VnStGD7Emmvc1-QOPR=
tY2cFF79E</a></div><div class=3D""><br class=3D""><blockquote =
type=3D"cite" class=3D""><div class=3D""><br class=3D""><blockquote =
type=3D"cite" class=3D"">There are 4 ip addresses involved here -<br =
class=3D"">Lets assume a user with a PC - wifi connected to an ADSL =
router and using a VPN<br class=3D""><br class=3D"">1) External VPN =
address : the IP address of the exit node from the VPN - this is already =
known to everyone.<br class=3D"">2) Internal VPN address : the IP =
address of the VPN interface on the PC end of the VPN tunnel - not known =
- but not =E2=80=98secret'<br class=3D"">3) ISP address : the address =
allocated by the ISP for the ADSL router - this is the one that TOR and =
other VPNs mask<br class=3D"">4) local IP address: the address returned =
by the router - typically 192.168.0.2 or some such - this is not known =
<br class=3D""><br class=3D"">With current VPN configs (and I assume =
TOR) webRTC reveals all of these.<br class=3D"">The critical one is =
&nbsp;ISP address (correct me if I=E2=80=99m wrong) - This can reveal =
location and identity via the ISP=E2=80=99s business database.<br =
class=3D""></blockquote><br class=3D"">Point one, all of this discussion =
and explanation belongs into<br =
class=3D"">draft-ietf-rtcweb-security-arch. That is your homework for =
today.<br class=3D""></div></blockquote><div class=3D""><br =
class=3D""></div><div class=3D"">Thanks.</div><br class=3D""><blockquote =
type=3D"cite" class=3D""><div class=3D""><br class=3D""><blockquote =
type=3D"cite" class=3D"">A small change to the VPN configuration will =
hide the ISP address - you just have to set the default route to 2) =
rather than 3)<br class=3D"">then the STUN queries will return the =
external VPN address - which is what you revealed anyway by loading the =
page.<br class=3D""></blockquote><br class=3D"">This does not sound like =
something your average Egyptian<br class=3D"">dissident will be able and =
sufficiently early informed about<br class=3D"">to go about.<br =
class=3D""></div></blockquote><div class=3D""><br class=3D""></div><div =
class=3D"">I have zero expectation that the Egyptian dissident should do =
this.&nbsp;</div><div class=3D"">I do expect their software provider to =
keep up with developments.</div><br class=3D""><blockquote type=3D"cite" =
class=3D""><div class=3D""><br class=3D"">I frequently see this pattern =
of saying privacy problem X<br class=3D"">can be solved by manual =
intervention Y and it is not our<br class=3D"">problem if the general =
public will be unable to do such a<br class=3D"">manual intervention.<br =
class=3D""><br class=3D"">Yes it is your problem.<br =
class=3D""></div></blockquote><div class=3D""><br class=3D""></div><div =
class=3D"">That isn=E2=80=99t what I meant. I expect secure software =
providers to keep their</div><div class=3D"">software abreast of new =
developments.&nbsp;</div><br class=3D""><blockquote type=3D"cite" =
class=3D""><div class=3D""><br class=3D"">It is not okay to dismiss =
problems this way.<br class=3D""></div></blockquote><div class=3D""><br =
class=3D""></div>I=E2=80=99m really sorry you think I=E2=80=99m =
dismissing this problem, I=E2=80=99m not.</div><div class=3D"">I happen =
to have the (no doubt under informed) opinion that</div><div =
class=3D"">this isn=E2=80=99t as simple as it has been made out to =
be.</div><div class=3D""><br class=3D""><blockquote type=3D"cite" =
class=3D""><div class=3D""><br class=3D"">Also the "solution" you =
suggest AFAIU does not work for people<br class=3D"">who use Tor in a =
non-NATted environment, for example on a<br class=3D"">fixed system in a =
university library with a static IP address.<br =
class=3D""></div></blockquote><div class=3D""><br class=3D""></div><div =
class=3D"">That is an interesting case. In that instance hiding the =
local IP would be&nbsp;</div><div class=3D"">valuable. I thought however =
those users would normally run a TOR os off&nbsp;</div><div class=3D"">a =
usb stick in a VM. That would probably mask the local IP.</div><div =
class=3D""><br class=3D""></div><br class=3D""><blockquote type=3D"cite" =
class=3D""><div class=3D"">We know that browsers will allow =
fingerprinting even through<br class=3D"">Tor but it is a different =
calibre of problem if a browser<br class=3D"">actually allows the =
attacker to find out the REAL public IP<br class=3D"">of a person.<br =
class=3D""></div></blockquote><div class=3D""><br class=3D""></div><div =
class=3D"">I totally agree. My solution is imho a better fix, since =
it</div><div class=3D"">also protects against many other apps that might =
get started from</div><div class=3D"">a browser page (email, dns, ipv6 =
tunnels, flash, mp3 players etc).</div><div class=3D""><br =
class=3D""></div><br class=3D""><blockquote type=3D"cite" class=3D""><div =
class=3D""><br class=3D""><blockquote type=3D"cite" class=3D"">'Just ask =
the user to enable webRTC' is a sledgehammer to crack a nut. <br =
class=3D""></blockquote><br class=3D"">No. It's like asking the user to =
click on the telephone icon<br class=3D"">before making a phone call. It =
is normal. What is abnormal is<br class=3D"">to have a web browser that =
is supposed to browse the web be<br class=3D"">enabled to do crazy =
non-web things without asking anyone.<br =
class=3D""></div></blockquote><div class=3D""><br class=3D""></div><div =
class=3D"">No, really not. I want the VPN folks and the Tor folks to =
update their</div><div class=3D"">scripts after consultations about the =
best way forward.</div><br class=3D""><blockquote type=3D"cite" =
class=3D""><div class=3D""><br class=3D"">You guys are completely losing =
the perspective of normal human beings.</div></blockquote><blockquote =
type=3D"cite" class=3D""><div class=3D""><br class=3D"">And it's nothing =
new, you know these documents yourself:<br class=3D""><br class=3D""><a =
href=3D"https://www.torproject.org/projects/torbrowser/design/#proxy-obedi=
ence" =
class=3D"">https://www.torproject.org/projects/torbrowser/design/#proxy-ob=
edience</a><br class=3D""><br class=3D"">WebRTC not respecting SOCKS =
configuration is a huge no-go.<br class=3D""></div></blockquote><div =
class=3D""><br class=3D""></div><div class=3D"">I=E2=80=99m SOCKs =
ignorant.</div><br class=3D""><blockquote type=3D"cite" class=3D""><div =
class=3D""><br class=3D""><a =
href=3D"https://www.torproject.org/projects/torbrowser/design/#fingerprint=
ing-linkability" =
class=3D"">https://www.torproject.org/projects/torbrowser/design/#fingerpr=
inting-linkability</a><br class=3D""><br class=3D"">And if local IP =
information is so super harmless, then why is it<br class=3D"">an =
obvious reason for TBB devs to disable the entire thing?<br class=3D"">I =
presume for the reasons I stated before.<br =
class=3D""></div></blockquote><div class=3D""><br class=3D""></div><div =
class=3D"">I have no idea, but I fear it is because no one has split =
these issues out.</div><br class=3D""><blockquote type=3D"cite" =
class=3D""><div class=3D""><br class=3D"">So when you do your homework =
please include all the points that<br class=3D"">I am making. Add your =
solid arguments to counter them.<br class=3D""><br class=3D""><blockquote =
type=3D"cite" class=3D"">You are forgoing the huge advantages (including =
privacy) of P2P crypto in webRTC just because the existing VPN <br =
class=3D"">suppliers can=E2=80=99t figure out a default route and update =
their scripts.<br class=3D""></blockquote><br class=3D"">I don't know =
about VPN suppliers, I never used any VPN really,<br class=3D"">but it =
sounds like you have a moral obligation to venture out<br class=3D"">and =
inform them about the security problem you are introducing<br =
class=3D"">and to make sure they protect their users before anyone =
gets<br class=3D"">harmed.<br class=3D""></div></blockquote><div =
class=3D""><br class=3D""></div><div class=3D"">Yep, I=E2=80=99ve been =
doing that, I clearly haven=E2=80=99t spoken to enough people</div><div =
class=3D"">but there is only so much homework I can do.</div><br =
class=3D""><blockquote type=3D"cite" class=3D""><div class=3D""><br =
class=3D"">Also don't tell me any fairy tales about P2P crypto in web<br =
class=3D"">browsers. There is no architectural way you can provide =
E2E<br class=3D"">crypto that isn't in one way or another MITM'able by =
the<br class=3D"">involved servers. You are at *best* adding to the =
choice of<br class=3D"">opportunistic cryptography. Don't be so =
excessively proud<br class=3D"">of that considering that REAL P2P with =
REAL E2E authentication<br class=3D"">is out there and would have =
deserved better attention.<br class=3D""></div></blockquote><div =
class=3D""><br class=3D""></div><div class=3D"">I think putting PFS =
strong crypto in the hands of 2bn people and&nbsp;</div><div class=3D"">1m=
 web developers is something to be slightly proud of.</div><div =
class=3D""><br class=3D""></div><div class=3D"">I=E2=80=99m of the view =
that there are _useful_ instances of P2P crypto</div><div =
class=3D"">in-browser - we will clearly have to disagree about =
that.</div><div class=3D""><br class=3D""></div><br class=3D""><blockquote=
 type=3D"cite" class=3D""><div class=3D""><br class=3D""><blockquote =
type=3D"cite" class=3D"">The cynical part of me wonders if people with =
skin in the VPN game don=E2=80=99t want an easy to use P2P crypto in the =
browser=E2=80=A6..<br class=3D""></blockquote><br class=3D"">No idea =
about that, but I am certain the number of people with skin<br =
class=3D"">in the WebRTC game are in the majority on this mailing list - =
and<br class=3D"">you wouldn't like to be ignored in any privacy debate =
a priori just<br class=3D"">because you're in the WebRTC game, right? =
But if you go on this way,<br class=3D"">that is likely to be the =
outcome. By the time you start thinking of<br class=3D"">the =
consequences of your actions your credibility may already be<br =
class=3D"">gone downhill.<br class=3D""></div></blockquote><div =
class=3D""><br class=3D""></div><div class=3D"">I=E2=80=99m not ignoring =
anyone or their views, just challenging the conclusions they</div><div =
class=3D"">have drawn by providing a different viewpoint.</div><div =
class=3D""><br class=3D""></div><div class=3D"">T.</div><div =
class=3D""><br class=3D""></div></div><br class=3D""></body></html>=

--Apple-Mail=_C287A5FA-58DD-4346-8110-DCB60E98EAB0--


From nobody Wed Jul 15 13:47:28 2015
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Date: Wed, 15 Jul 2015 22:47:19 +0200
From: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] Prompt for mic, camera *and* network access
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On 07/15/2015 06:41 PM, carlo von lynX wrote:
> I don't know about VPN suppliers, I never used any VPN really,
> but it sounds like you have a moral obligation to venture out
> and inform them about the security problem you are introducing
> and to make sure they protect their users before anyone gets
> harmed.
Sorry, I lost it.

You're lecturing us (many of whom have used VPNs regularly for the last
20 years, some of whom have implemented them) about what we should tell
VPN suppliers, and you haven't even *used* a VPN?

May I suggest that the strong opinions being delivered here might be
better served with slightly more respect for the fact that some of us
actually have some experience worth considering?

Just *explaining* this problem is complicated. And most of the
"headline" statements made about this problem are either completely
wrong or very incomplete.
Being clear about what needs protection, and from what threat, is just
the first step on the way to finding a consensus solution.

And I don't think we're even there yet.

(I recently learned of a couple of reasons why people think the local IP
address (the 192.168 ones) are valuable to learn for some services - in
that particular case, many would consider that particular purpose to
have value to society. But it shows that once we reveal information,
that information can be used in many different ways - some less obvious
than others.)

Harald


--=20
Surveillance is pervasive. Go Dark.



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I could clarify the situation for Draft ITU-T H.248.94 (ex H.248.WebRTC):

=3D> the reference to draft-rtcweb-gateways is INFORMATIVE only, and there =
isn't any real need for a normative reference because the scope of this ITU=
-T Recommendation is on "H.248 profile GUIDELINES"

("That's the reason why draft-rtcweb-gateways is listed in the Bibliography=
 section.")

-Albrecht


-----Original Message-----
From: DRAGE, Keith (Keith)=20
Sent: Mittwoch, 15. Juli 2015 21:13
To: Cullen Jennings; Schwarz, Albrecht (Albrecht)
Cc: rtcweb@ietf.org
Subject: RE: [rtcweb] Updated -gateways draft

In general, 3GPP can quite happily normatively reference both informational=
 and standards track documents. The key is of course that the referenced ma=
terial is suitable, i.e. that normatively referenced material is specified =
in a suitable fashion to be a normative requirements. In previous mails I h=
ave given at least one example of such requirements.

However Albrecht does introduce a different complication.

3GPP does also normatively reference ITU-T documents for the H.248 function=
ality, and if the ITU-T documents need then to normatively reference the IE=
TF document to support that, then if I recall correctly, ITU-T does require=
 that to be a standards track document.

What is clear is that we will have a mixture of normative material and info=
rmational material. It is important that that distinction is clear. If that=
 is so, what is the issue with having a standards track document?

Keith=20

> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Cullen=20
> Jennings
> Sent: 15 July 2015 16:43
> To: Schwarz, Albrecht (Albrecht)
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Updated -gateways draft
>=20
>=20
> > On Jul 15, 2015, at 2:02 AM, Schwarz, Albrecht (Albrecht)
> <albrecht.schwarz@alcatel-lucent.com> wrote:
> >=20
> > Coming back to the original question: whether it should be
> Informational or Standards-track, - I don't know. But it is apparent=20
> that both, the 3GPP and ITU-T work on WebRTC gateways needs to be=20
> tightly coupled and linked to the IETF gateway draft in my opinion.
>=20
> I'd be happy for Keith to correct me but my understanding is that 3GPP=20
> does not really differentiate much on the RFC track and can have=20
> normative references to informational RFC.
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


From nobody Thu Jul 16 01:42:28 2015
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Harald, sorry if after getting ignored once and getting a rant
on my second inquiry I start getting grumpy.. still let's stay
on topic and figure this issue out.. I still hope I and everyone
in blogland is wrong on this and exposing local IP is actually
not problematic.. but the webrtc documentations should elaborate
on this scenario and evaluate carefully in-depth on anything
that could possibly go wrong, if they intend to recover credibility
on this terrain rather than strengthening the general feeling that
browser vendors are bulldozering over humans and citizen anytime
the business model suggests so.

On Wed, Jul 15, 2015 at 08:27:31PM +0100, tim panton wrote:
> Um, if you check back in the archives, I brought this issue up earlier this year.
> I think it is important.
> 
> https://mailarchive.ietf.org/arch/msg/rtcweb/VnStGD7Emmvc1-QOPRtY2cFF79E <https://mailarchive.ietf.org/arch/msg/rtcweb/VnStGD7Emmvc1-QOPRtY2cFF79E>

Thank you. I see you brought up the same issues. Would have
appreciated receiving such a pointer back in April when I first
asked. Unfortunately I cannot read the follow-up discussion as
somebody at IETF had the great idea of making the archives
depend on Javascript execution. I don't grant execution of foreign
code on my machines to any unencrypted or untrustworthy sources.
Is there any reply I should read? What's its hash code?

> >> A small change to the VPN configuration will hide the ISP address - you just have to set the default route to 2) rather than 3)
> >> then the STUN queries will return the external VPN address - which is what you revealed anyway by loading the page.
> > 
> > This does not sound like something your average Egyptian
> > dissident will be able and sufficiently early informed about
> > to go about.
> 
> I have zero expectation that the Egyptian dissident should do this. 
> I do expect their software provider to keep up with developments.

Yes, I dare to speak out of my 25 years of experience on the
Internet even if I didn't have to use much VPN ever. I can
imagine plenty of constellations where the VPN is NOT supposed
to take over all communications on the system but rather offer
a side route into a company's servers or suchlike. Therefore
I worry if the software provider can always come up with a
configuration that still does what the customers expect and
at the same time deals with the new backdoor introduced by WebRTC.
Of course anyone with better understanding can help clear up
these worries.

> That isnâ€™t what I meant. I expect secure software providers to keep their
> software abreast of new developments. 

Uh, not sure if that is an established pattern in the industry.
Usually somebody does whatever fits into their business model
best, then there is a public outcry and possibly some bloodspill,
then all the players look at how they can make a good impression
by reducing the collateral damages.. but if that doesn't work out
cheap and easy.. tant pis. Let's continue with business.

> I happen to have the (no doubt under informed) opinion that
> this isnâ€™t as simple as it has been made out to be.

Well, in case of doubt it is important to pick the conservative
option until the issue is properly analyzed rather than hoping
for the general public to forget about the problem.

> > Also the "solution" you suggest AFAIU does not work for people
> > who use Tor in a non-NATted environment, for example on a
> > fixed system in a university library with a static IP address.
> 
> That is an interesting case. In that instance hiding the local IP would be 
> valuable. I thought however those users would normally run a TOR os off 
> a usb stick in a VM. That would probably mask the local IP.

Now that this is more clear, have another look at my original mail
that describes how such a dissident would use any Windows OS system
standing around. I doubt anything advanced like TAILS sticks are
out there as merely possessing one could be life-threatening.

We are talking about people that act out of a need to act with
very little understanding of technology. They would likely
only start wondering if what they did was safe after they got
caught. By introducing new vulnerabilities we may be tipping
some very shaky balances to the wrong side. It is quite likely
that a whole lot of people unconsciously rely on the principle
that a web browser configured to use a proxy will actually do so.

> > We know that browsers will allow fingerprinting even through
> > Tor but it is a different calibre of problem if a browser
> > actually allows the attacker to find out the REAL public IP
> > of a person.
> 
> I totally agree. My solution is imho a better fix, since it
> also protects against many other apps that might get started from
> a browser page (email, dns, ipv6 tunnels, flash, mp3 players etc).

If you can convince Microsoft to update all the OS's in such countries
in a way that VPN can no longer be configured to function in parallel
to direct Internet access, that could be a viable solution to this
problem. Introduces a new problem of not being backward compatible
to other use cases of VPNs, correct?

Again, expecting the dissidents to be magically internetworked and
inform each other quickly enough on safer practices that only a few
like us know about and may actually not be safer since they imply
singling people out of the masses of Windows users... acting differently
than the rest of the population can be a death sentence.. well then your 
proposed solution to the problem may not work out for real humans
with real problems.

> >> 'Just ask the user to enable webRTC' is a sledgehammer to crack a nut. 
> > 
> > No. It's like asking the user to click on the telephone icon
> > before making a phone call. It is normal. What is abnormal is
> > to have a web browser that is supposed to browse the web be
> > enabled to do crazy non-web things without asking anyone.
> 
> No, really not. I want the VPN folks and the Tor folks to update their
> scripts after consultations about the best way forward.

Breaking the proxy principle is neither a Tor nor a VPN problem.
It's just browser vendors breaking a promise given to users that
configured proxies would be respected.

> Iâ€™m SOCKs ignorant.

So we all have our knowledge deficiencies...
good to be honest about them.

> > And if local IP information is so super harmless, then why is it
> > an obvious reason for TBB devs to disable the entire thing?
> > I presume for the reasons I stated before.
> 
> I have no idea, but I fear it is because no one has split these issues out.

So let's dissect some more until we know what we are doing.

> > I don't know about VPN suppliers, I never used any VPN really,
> > but it sounds like you have a moral obligation to venture out
> > and inform them about the security problem you are introducing
> > and to make sure they protect their users before anyone gets
> > harmed.
> 
> Yep, Iâ€™ve been doing that, I clearly havenâ€™t spoken to enough people
> but there is only so much homework I can do.

True.

> > Also don't tell me any fairy tales about P2P crypto in web
> > browsers. There is no architectural way you can provide E2E
> > crypto that isn't in one way or another MITM'able by the
> > involved servers. You are at *best* adding to the choice of
> > opportunistic cryptography. Don't be so excessively proud
> > of that considering that REAL P2P with REAL E2E authentication
> > is out there and would have deserved better attention.
> 
> I think putting PFS strong crypto in the hands of 2bn people and 
> 1m web developers is something to be slightly proud of.

Perspective accepted.

> Iâ€™m of the view that there are _useful_ instances of P2P crypto
> in-browser - we will clearly have to disagree about that.

I think we could improve the user's ability to ensure end-to-end
authentication beyond server-based negotiation. Something like
rendering persistent public keys into QR codes and having people
exchange those QR codes at cocktail parties and hackathons.

Servers would still be able to provide the signaling code, but users
would be granted a guarantee that the E2E connection is authentic.

> Iâ€™m not ignoring anyone or their views, just challenging the conclusions they
> have drawn by providing a different viewpoint.

Moi aussi.


-- 
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From nobody Thu Jul 16 06:43:12 2015
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From: Victor Pascual Avila <victor.pascual.avila@gmail.com>
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--001a1136cbd85ee8cf051afe40a1
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On Sat, Jul 11, 2015 at 6:42 PM, Daniel Roesler <diafygi@gmail.com> wrote:

> There is now a documented case where a third party on nytimes.com
> is using a fake webRTC datachannel to silently gather user local (and
> potentially "real" ISP) IP addresses.
>
> https://twitter.com/incloud/status/619624021123010560
>

FYI - The company behind the NY Times IP tracking code turned it off. See
our blog: https://webrtchacks.com/dear-ny-times/#comment-17504

Cheers,
-Victor

--001a1136cbd85ee8cf051afe40a1
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On S=
at, Jul 11, 2015 at 6:42 PM, Daniel Roesler <span dir=3D"ltr">&lt;<a href=
=3D"mailto:diafygi@gmail.com" target=3D"_blank">diafygi@gmail.com</a>&gt;</=
span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0=
px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-le=
ft-style:solid;padding-left:1ex"><div id=3D":2tl" class=3D"" style=3D"overf=
low:hidden">There is now a documented case where a third party on <a href=
=3D"http://nytimes.com" rel=3D"noreferrer" target=3D"_blank"><span class=3D=
"">nytimes</span>.com</a><br>
is using a fake <span class=3D"">webRTC</span> datachannel to silently gath=
er user local (and<br>
potentially &quot;real&quot; ISP) IP addresses.<br>
<br>
<a href=3D"https://twitter.com/incloud/status/619624021123010560" rel=3D"no=
referrer" target=3D"_blank">https://twitter.com/incloud/status/619624021123=
010560</a></div></blockquote></div><br><span style=3D"color:rgb(20,24,35);f=
ont-family:helvetica,arial,sans-serif;font-size:12px;line-height:16.0799999=
237061px;background-color:rgb(246,247,248)">FYI - The company behind the NY=
 Times IP tracking code turned it off. See our blog:=C2=A0</span><font colo=
r=3D"#141823" face=3D"helvetica, arial, sans-serif"><span style=3D"font-siz=
e:12px;line-height:16.0799999237061px"><a href=3D"https://webrtchacks.com/d=
ear-ny-times/#comment-17504">https://webrtchacks.com/dear-ny-times/#comment=
-17504</a></span></font></div><div class=3D"gmail_extra"><font color=3D"#14=
1823" face=3D"helvetica, arial, sans-serif"><span style=3D"font-size:12px;l=
ine-height:16.0799999237061px"><br></span></font></div><div class=3D"gmail_=
extra"><font color=3D"#141823" face=3D"helvetica, arial, sans-serif"><span =
style=3D"font-size:12px;line-height:16.0799999237061px">Cheers,</span></fon=
t></div><div class=3D"gmail_extra"><font color=3D"#141823" face=3D"helvetic=
a, arial, sans-serif"><span style=3D"font-size:12px;line-height:16.07999992=
37061px">-Victor</span></font><br>
</div></div>

--001a1136cbd85ee8cf051afe40a1--


From nobody Thu Jul 16 12:51:41 2015
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Good points.I will update the draft but in the meantime there is a copy =
at=20
http://fluffy.github.io/iceTrav/

The summary of the algorithm is roughly:

=E2=80=A2The firewall looks for any outgoing STUN requests to STUN port =
(3478). When it finds one, it stores the 3 tuple of the source address =
port and protocol=3DUDP and for the next 30 seconds checks any packets =
from this 3 tuple to see if they are ICE connectivity checks.=20

=E2=80=A2When firewall sees an ICE connectivity check from the inside =
host (which is a STUN Binding Request containing the ice username), it =
stores that ice username value, and forwards the packet.  Thereafter, =
STUN Binding Requests and Binding Responses with matching 4-tuple =
(inside IP address, protocol=3DUDP, inside port, and matching ice =
username) are allowed in any direction (inside->outside and =
outside->inside).  For each 5-tuple, if a STUN Binding Request has a =
Success STUN Binding Response (with same transaction ID), in either =
direction, that 5-tuple is promoted to allow non-STUN traffic (DTLS, =
SRTP, and anything else). =20

=E2=80=A2Both promoted and non-promoted pinholes are closed after 30 =
seconds of not seeing a STUN Binding Request and Response transaction, =
in either direction.

=E2=80=A2(Note to match the ice username external requests will need the =
halves on either side of =E2=80=9C:=E2=80=9D swapped)

(and thanks to  Dan Wing who came up with much of the above proposal )=20=


more inline=20

> On Jul 8, 2015, at 8:39 AM, Eric Rescorla <ekr@rtfm.com> wrote:
>=20
> I have taken a brief look at this draft and I don't believe that the =
proposed pinhole/inspection
> algorithm provides the desired assurances of verifying consent:
>=20
> 1. If you open up an incoming/outgoing pinhole on outgoing packet, =
prior to
> receiving an incoming STUN packet, then if the internal implementation
> simply sends a new outgoing STUN packet every 5 seconds, then they
> never need to rely on external consent.
>=20
> 2. If you just require *a* incoming STUN packet to be received in =
order to
> extend the lifetime to 30 seconds, then an off-path attacker can forge
> a STUN response that will cause the pinhole to be open for 30 seconds.
> (Hence the need for the STUN transaction ID).

Yah, don't want outside stuff to extended it. Intent was only inside =
stuff should extend.=20

>=20
> It seems to me that you should be instead requiring a complete STUN
> transaction before allowing any non-STUN outgoing traffic. It would =
probably
> be safe-ish to require it before allowing any incoming traffic as =
well, though
> that leaves the chance that you get a small amount of clipping with =
packet
> drops or reordering.

Agree - have changed to that.=20

>=20
>=20
> WRT to the origin attribute, I have mixed feelings. I agree that your =
SNI argument
> is strong, but there is also a move to figure out how to encrypt SNI, =
so every time
> we extend that argument it makes that harder. OTOH, STUN origin will =
be
> sent a lot less often than SNI, I imagine.

Sounds like this is going to get discussed in Tram. Unfortunately I =
think we have to choose between least evil on this topic.=20

>=20
>=20
> I don't understand Section 6. Is this guidance to firewall vendors or =
to browser
> implementors. If the former, the firewall has no way of knowing what =
"requests"
> are if the connection use HTTPS (and if is uses HTTP, you have =
DPI-type methods for
> determining whether something is voice or video. If this is guidance =
to browser
> vendors, I consider it extremely unlikely that we would implement it.

advice for firewall vendors for more or less all TLS connections that =
might be web traffic. It's horrific of course but since they are =
implementing things along these lines we might as well be specific on =
the algorithm so we understand the breakage. Or go deal with the root =
causes that theses sort of things (basically trying to hide WebRTC =
traffic inside HTTP) . I imagine this algorithm is not the best, it was =
just a rough place holder. Its needs a bit of tuning to interact with =
dash in a reasonable way.=20


>=20
>=20
> Point #4 in your security considerations section seems like an =
entirely lost
> cause, unless you assume that attackers are not really trying.
>=20
> -Ekr
>=20
>=20
>=20


From nobody Thu Jul 16 14:13:59 2015
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On 7/16/15 1:42 AM, carlo von lynX wrote:
> I don't grant execution of foreign code on my machines to any 
> unencrypted or untrustworthy sources.

If you can get everyone to do that, we won't even need RTCWEB

Matthew Kaufman


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From: "Parthasarathi R" <partha@parthasarathi.co.in>
To: "'Schwarz, Albrecht \(Albrecht\)'" <albrecht.schwarz@alcatel-lucent.com>,  "'DRAGE, Keith \(Keith\)'" <keith.drage@alcatel-lucent.com>, "'Cullen Jennings'" <fluffy@iii.ca>
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Hi all,

The main intent of this draft is to define WebRTC gateway entity in IETF
itself as there are lot of discussions wherein it is not clear which WebRTC
entity under discussion. This draft has to be in standards track as the
draft relax some of the normative statements of WebRTC endpoint namely

1) ICE-lite
2) Non-bundle SDP support only
3) No need of TURN support
4) No need to support data channel.

These information are clarified in Sec 2 of the -gateways draft. Of course,
these relaxation has some impact in other WebRTC endpoints as well. The
example is to support non-bundle SDP in the incoming offer to the WebRTC
endpoint.

Of course, we may see more in the relaxation list as the other RTCWeb drafts
are yet to be matured as RFC. As Keith mentioned, some of the normative text
is required in this draft. IMO, the standards track looks more appropriate
for this draft.

Thanks
Partha

> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Schwarz,
> Albrecht (Albrecht)
> Sent: Thursday, July 16, 2015 12:20 PM
> To: DRAGE, Keith (Keith); Cullen Jennings
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Updated -gateways draft
> 
> I could clarify the situation for Draft ITU-T H.248.94 (ex
> H.248.WebRTC):
> 
> => the reference to draft-rtcweb-gateways is INFORMATIVE only, and
> there isn't any real need for a normative reference because the scope
> of this ITU-T Recommendation is on "H.248 profile GUIDELINES"
> 
> ("That's the reason why draft-rtcweb-gateways is listed in the
> Bibliography section.")
> 
> -Albrecht
> 
> 
> -----Original Message-----
> From: DRAGE, Keith (Keith)
> Sent: Mittwoch, 15. Juli 2015 21:13
> To: Cullen Jennings; Schwarz, Albrecht (Albrecht)
> Cc: rtcweb@ietf.org
> Subject: RE: [rtcweb] Updated -gateways draft
> 
> In general, 3GPP can quite happily normatively reference both
> informational and standards track documents. The key is of course that
> the referenced material is suitable, i.e. that normatively referenced
> material is specified in a suitable fashion to be a normative
> requirements. In previous mails I have given at least one example of
> such requirements.
> 
> However Albrecht does introduce a different complication.
> 
> 3GPP does also normatively reference ITU-T documents for the H.248
> functionality, and if the ITU-T documents need then to normatively
> reference the IETF document to support that, then if I recall
> correctly, ITU-T does require that to be a standards track document.
> 
> What is clear is that we will have a mixture of normative material and
> informational material. It is important that that distinction is clear.
> If that is so, what is the issue with having a standards track
> document?
> 
> Keith
> 
> > -----Original Message-----
> > From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Cullen
> > Jennings
> > Sent: 15 July 2015 16:43
> > To: Schwarz, Albrecht (Albrecht)
> > Cc: rtcweb@ietf.org
> > Subject: Re: [rtcweb] Updated -gateways draft
> >
> >
> > > On Jul 15, 2015, at 2:02 AM, Schwarz, Albrecht (Albrecht)
> > <albrecht.schwarz@alcatel-lucent.com> wrote:
> > >
> > > Coming back to the original question: whether it should be
> > Informational or Standards-track, - I don't know. But it is apparent
> > that both, the 3GPP and ITU-T work on WebRTC gateways needs to be
> > tightly coupled and linked to the IETF gateway draft in my opinion.
> >
> > I'd be happy for Keith to correct me but my understanding is that
> 3GPP
> > does not really differentiate much on the RFC track and can have
> > normative references to informational RFC.
> >
> >
> > _______________________________________________
> > 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 nobody Thu Jul 16 21:06:48 2015
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On 7/16/2015 4:42 AM, carlo von lynX wrote:
>
>> I=E2=80=99m of the view that there are _useful_ instances of P2P crypt=
o
>> in-browser - we will clearly have to disagree about that.
> I think we could improve the user's ability to ensure end-to-end
> authentication beyond server-based negotiation. Something like
> rendering persistent public keys into QR codes and having people
> exchange those QR codes at cocktail parties and hackathons.
>
> Servers would still be able to provide the signaling code, but users
> would be granted a guarantee that the E2E connection is authentic.

Have you read the 'identity' support in the webrtc/rtcweb specs?  Or=20
played with early implementations at=20
https://mozilla.github.io/webrtc-landing/pc_test.html?

This is something we've cared about, since the beginning of this work. =20
We're also working out speccing calls where the JS application can't get=20
access to the datastreams (via WebAudio/canvas/etc).  There's a lot of=20
work to do to spec this all out, and more to be done to implement it -=20
but these things are part of the goals of WebRTC.  Witness also that we=20
decided to not only not require SDES (which puts keys directly in the=20
hands of the servers), but to actively require not supporting SDES. =20
We've also moved to requires PFS encryption=20
(https://hacks.mozilla.org/2015/02/webrtc-requires-perfect-forward-secrec=
y-pfs-starting-in-firefox-38/)

I'll also note that WebRTC+DataChannels can be used to build=20
non-centralized encrypted communication services using (for example)=20
distributed-hash addressing and a web of datachannel connections, such=20
that there's no single easily-blocked locus for communications (like=20
blocking twitter.com).  There's a *bunch* of work to build a good,=20
secure service around this - but now it's feasible to do with just some=20
JS shared on a thumb drive (or any number of ways online - done right,=20
perhaps even in high-density QR-ish codes with compressed code to=20
bootstrap into the network.  The webtorrent clients are the first step=20
towards this.

--=20
Randell Jesup -- rjesup a t mozilla d o t com
Please please please don't email randell-ietf@jesup.org!  Way too much sp=
am


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Subject: [rtcweb] Dreaming of serverless P2P over the web
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Sorry for summoning up a change of topic.
The main thread on the safety of browser users affected by the
webrtc apis against their will is still open and unresolved.
Here we just talk about the young dreams of a Javascript generation:

On Fri, Jul 17, 2015 at 12:06:15AM -0400, Randell Jesup wrote:
> Have you read the 'identity' support in the webrtc/rtcweb specs?  Or
> played with early implementations at
> https://mozilla.github.io/webrtc-landing/pc_test.html?

I know that the JS community is very adamant exploring fertile
new grounds doing crazy things... I was to "data terra nemo"
just recently.. I understand there is plenty of good intentions.
I presume you have heard of where they likely lead to.

> I'll also note that WebRTC+DataChannels can be used to build
> non-centralized encrypted communication services using (for example)
> distributed-hash addressing and a web of datachannel connections,

Yes, these kind of good intentions.. the juvenile enthusiasm that
thinks something as complicated as DHTs can be quickly thrown into
a browser with "some JS on a thumb drive."

My knowledge indicates the security issues with cheap off-the-mill DHT
tech are impressive. It took gnunet a dozen of years to develop a
sybil-attack resistant DHT routing system. Go ahead and show
me how you will ship THAT in a bunch of .js files. Or do you think
the world will be done any good if you throw even more insecure
web apps at it?

>From my understanding of technology JS dreamers are building castles 
on houses of cards and distracting entire communities of young minds
from contributing to technologies that could actually work to
repair the damage the Internet has afflicted democracy.

But maybe you have some papers proving how a web-based distributed
hash table can be made sybil-attack resistant and secure from
eavesdropping. To me that is only achievable with such large
amounts of low-level code and packet routing that it totally 
blows the architectural options of a web browser.

So what's your take on this.. will you admit your plan is
unlikely to work out or come up with wild arguments just so
you can continue down the path of good intentions?


-- 
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Date: Fri, 17 Jul 2015 10:51:53 -0700
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On Sun, Jul 12, 2015 at 1:15 PM, tim panton <tim@phonefromhere.com> wrote:
>
> The problem is that those VPNs are misconfigured.
>

It appears that both built-in L2TP and PPTP VPN options for OSX cannot
be configured to hide the real IP address from WebRTC, even if you
select the option to route all traffic through the VPN[1]. You need to
install an OpenVPN client like Tunnelblick in order to hide your real
IP from WebRTC. As soon as I get some time on a Windows box, I'll
investigate the default behavior there.

[1]: https://i.imgur.com/Hyq1zLz.png

Since this is the behavior for built-in VPN options, and even if you
select the "send all traffic over VPN" setting, I disagree with the
sentiment that it's the user's fault or that the VPN has been
misconfigured. WebRTC should assume default behavior for operating
system VPN behavior and apply additional security steps to protect the
user when the operating system falls short. Blaming the operating
system is no excuse when you know the problem exists and you have the
means to protect the user.


On Mon, Jul 13, 2015 at 11:54 AM, Timothy B. Terriberry
<tterriberry@mozilla.com> wrote:
>
> These issues are all detailed in
> <https://tools.ietf.org/html/draft-ietf-rtcweb-security-arch>. I would
> encourage those participating in this thread to read that document. We could
> always benefit from additional review.

-------<snip from Section 5.4>-------
   API Requirement:  The API MUST provide a mechanism to allow the JS to
      suppress ICE negotiation (though perhaps to allow candidate
      gathering) until the user has decided to answer the call [note:
      determining when the call has been answered is a question for the
      JS.]  This enables a user to prevent a peer from learning their IP
      address if they elect not to answer a call and also from learning
      whether the user is online.
-------</snip from Section 5.4>-------

I'd like to request that "suppress ICE negotiation (though perhaps to
allow candidate gathering)" be changed to "suppress ICE negotiation
and candidate gathering". That would be enough to require user consent
before local and VPN IP addresses are gathered.

Daniel


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--90e6ba181b8246c571051b15f02b
Content-Type: text/plain; charset=UTF-8

On Fri, Jul 17, 2015 at 10:51 AM, Daniel Roesler <diafygi@gmail.com> wrote:

> On Sun, Jul 12, 2015 at 1:15 PM, tim panton <tim@phonefromhere.com> wrote:
> >
> > The problem is that those VPNs are misconfigured.
> >
>
> It appears that both built-in L2TP and PPTP VPN options for OSX cannot
> be configured to hide the real IP address from WebRTC, even if you
> select the option to route all traffic through the VPN[1]. You need to
> install an OpenVPN client like Tunnelblick in order to hide your real
> IP from WebRTC. As soon as I get some time on a Windows box, I'll
> investigate the default behavior there.
>
> [1]: https://i.imgur.com/Hyq1zLz.png
>
> Since this is the behavior for built-in VPN options, and even if you
> select the "send all traffic over VPN" setting, I disagree with the
> sentiment that it's the user's fault or that the VPN has been
> misconfigured. WebRTC should assume default behavior for operating
> system VPN behavior and apply additional security steps to protect the
> user when the operating system falls short. Blaming the operating
> system is no excuse when you know the problem exists and you have the
> means to protect the user.


Rather than trying to decide who's responsibility this is, I suggest it
would be
more productive to ask whether there's some way for the browser to determine
whether the VPN has been configured in this way. If there is, then I at
least
would be willing to investigate whether it's possible for the browser to
behave differently in these cases.

For starters, I would ask what netstat -rn shows in these cases.



> On Mon, Jul 13, 2015 at 11:54 AM, Timothy B. Terriberry
> <tterriberry@mozilla.com> wrote:
> >
> > These issues are all detailed in
> > <https://tools.ietf.org/html/draft-ietf-rtcweb-security-arch>. I would
> > encourage those participating in this thread to read that document. We
> could
> > always benefit from additional review.
>
> -------<snip from Section 5.4>-------
>    API Requirement:  The API MUST provide a mechanism to allow the JS to
>       suppress ICE negotiation (though perhaps to allow candidate
>       gathering) until the user has decided to answer the call [note:
>       determining when the call has been answered is a question for the
>       JS.]  This enables a user to prevent a peer from learning their IP
>       address if they elect not to answer a call and also from learning
>       whether the user is online.
> -------</snip from Section 5.4>-------
>
> I'd like to request that "suppress ICE negotiation (though perhaps to
> allow candidate gathering)" be changed to "suppress ICE negotiation
> and candidate gathering". That would be enough to require user consent
> before local and VPN IP addresses are gathered.


This wouldn't address your issue, because this is under *API* control.
the purpose of this section is not to protect the user from the site but
rather to allow the site to provide privacy features for the user.


-Ekr

--90e6ba181b8246c571051b15f02b
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Fri, Jul 17, 2015 at 10:51 AM, Daniel Roesler <span dir=3D"ltr">&lt;=
<a href=3D"mailto:diafygi@gmail.com" target=3D"_blank">diafygi@gmail.com</a=
>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 =
0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><span class=3D"">On S=
un, Jul 12, 2015 at 1:15 PM, tim panton &lt;<a href=3D"mailto:tim@phonefrom=
here.com">tim@phonefromhere.com</a>&gt; wrote:<br>
&gt;<br>
&gt; The problem is that those VPNs are misconfigured.<br>
&gt;<br>
<br>
</span>It appears that both built-in L2TP and PPTP VPN options for OSX cann=
ot<br>
be configured to hide the real IP address from WebRTC, even if you<br>
select the option to route all traffic through the VPN[1]. You need to<br>
install an OpenVPN client like Tunnelblick in order to hide your real<br>
IP from WebRTC. As soon as I get some time on a Windows box, I&#39;ll<br>
investigate the default behavior there.<br>
<br>
[1]: <a href=3D"https://i.imgur.com/Hyq1zLz.png" rel=3D"noreferrer" target=
=3D"_blank">https://i.imgur.com/Hyq1zLz.png</a><br>
<br>
Since this is the behavior for built-in VPN options, and even if you<br>
select the &quot;send all traffic over VPN&quot; setting, I disagree with t=
he<br>
sentiment that it&#39;s the user&#39;s fault or that the VPN has been<br>
misconfigured. WebRTC should assume default behavior for operating<br>
system VPN behavior and apply additional security steps to protect the<br>
user when the operating system falls short. Blaming the operating<br>
system is no excuse when you know the problem exists and you have the<br>
means to protect the user.</blockquote><div><br></div><div>Rather than tryi=
ng to decide who&#39;s responsibility this is, I suggest it would be</div><=
div>more productive to ask whether there&#39;s some way for the browser to =
determine</div><div>whether the VPN has been configured in this way. If the=
re is, then I at least</div><div>would be willing to investigate whether it=
&#39;s possible for the browser to</div><div>behave differently in these ca=
ses.</div><div><br></div><div>For starters, I would ask what netstat -rn sh=
ows in these cases.</div><div><br></div><div>=C2=A0</div><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex"><span class=3D"">
On Mon, Jul 13, 2015 at 11:54 AM, Timothy B. Terriberry<br>
&lt;<a href=3D"mailto:tterriberry@mozilla.com">tterriberry@mozilla.com</a>&=
gt; wrote:<br>
&gt;<br>
&gt; These issues are all detailed in<br>
&gt; &lt;<a href=3D"https://tools.ietf.org/html/draft-ietf-rtcweb-security-=
arch" rel=3D"noreferrer" target=3D"_blank">https://tools.ietf.org/html/draf=
t-ietf-rtcweb-security-arch</a>&gt;. I would<br>
&gt; encourage those participating in this thread to read that document. We=
 could<br>
&gt; always benefit from additional review.<br>
<br>
</span>-------&lt;snip from Section 5.4&gt;-------<br>
=C2=A0 =C2=A0API Requirement:=C2=A0 The API MUST provide a mechanism to all=
ow the JS to<br>
=C2=A0 =C2=A0 =C2=A0 suppress ICE negotiation (though perhaps to allow cand=
idate<br>
=C2=A0 =C2=A0 =C2=A0 gathering) until the user has decided to answer the ca=
ll [note:<br>
=C2=A0 =C2=A0 =C2=A0 determining when the call has been answered is a quest=
ion for the<br>
=C2=A0 =C2=A0 =C2=A0 JS.]=C2=A0 This enables a user to prevent a peer from =
learning their IP<br>
=C2=A0 =C2=A0 =C2=A0 address if they elect not to answer a call and also fr=
om learning<br>
=C2=A0 =C2=A0 =C2=A0 whether the user is online.<br>
-------&lt;/snip from Section 5.4&gt;-------<br>
<br>
I&#39;d like to request that &quot;suppress ICE negotiation (though perhaps=
 to<br>
allow candidate gathering)&quot; be changed to &quot;suppress ICE negotiati=
on<br>
and candidate gathering&quot;. That would be enough to require user consent=
<br>
before local and VPN IP addresses are gathered.</blockquote><div><br></div>=
<div>This wouldn&#39;t address your issue, because this is under *API* cont=
rol.</div><div>the purpose of this section is not to protect the user from =
the site but</div><div>rather to allow the site to provide privacy features=
 for the user.</div><div><br></div><div><br></div><div>-Ekr</div><div><br><=
/div><div><br></div></div><br></div></div>

--90e6ba181b8246c571051b15f02b--


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From: Justin Uberti <juberti@google.com>
Date: Fri, 17 Jul 2015 11:04:45 -0700
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Subject: Re: [rtcweb] Please require user consent for data channels
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--047d7bd767780b33d3051b1607b6
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>
>
> I'm really unclear on why it's so important to have the silent webRTC
> data channel behavior. P2P data connections should be held to the same
> consent requirements just like P2P video and voice connections. In
> addition to the silent third party local IP tracking, a silent P2P
> data channel opens up users to silently becoming nodes in a web-based
> file sharing network. For example, webtorrent[2] can be silently
> embedded on a website so that all of the users to that website start
> seeding content they don't know they are seeding.
>
>
>
This is already possible using non-WebRTC technologies, e.g. web sockets.
As such, I don't think that permission should be needed to create a data
channel, or receive (but not send) media.

That said, I agree that WebRTC should not allow drive-by harvesting of IP
addresses, and we intend to make changes to Chrome to prevent this.

--047d7bd767780b33d3051b1607b6
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote"><blo=
ckquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #c=
cc solid;padding-left:1ex"><br>
I&#39;m really unclear on why it&#39;s so important to have the silent webR=
TC<br>
data channel behavior. P2P data connections should be held to the same<br>
consent requirements just like P2P video and voice connections. In<br>
addition to the silent third party local IP tracking, a silent P2P<br>
data channel opens up users to silently becoming nodes in a web-based<br>
file sharing network. For example, webtorrent[2] can be silently<br>
embedded on a website so that all of the users to that website start<br>
seeding content they don&#39;t know they are seeding.<br>
<br><br></blockquote><div><br></div><div>This is already possible using non=
-WebRTC technologies, e.g. web sockets. As such, I don&#39;t think that per=
mission should be needed to create a data channel, or receive (but not send=
) media.</div><div><br></div><div>That said, I agree that WebRTC should not=
 allow drive-by harvesting of IP addresses, and we intend to make changes t=
o Chrome to prevent this.</div></div><br></div></div>

--047d7bd767780b33d3051b1607b6--


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On Fri, Jul 17, 2015 at 11:04 AM, Justin Uberti <juberti@google.com> wrote:
>
> This is already possible using non-WebRTC technologies, e.g. web sockets. As
> such, I don't think that permission should be needed to create a data
> channel, or receive (but not send) media.
>

The big difference is that Web Sockets starts off with an HTTP
request, so it can be filtered by various plugins (PrivacyBadger,
uBlock, Ghostery, etc.). WebRTC is invisible to those services since
it does not have any HTTP requests at all. In fact, is WebRTC the only
browser standard (i.e. not flash) that can fire off a DNS request
that's not (at least at first) for an HTTP address[1]? Lacking an
initial HTTP request makes it impossible for plugins to selectively
filter these requests, so they have to fall back to an all-or-nothing
config setting approach (which hurts WebRTC adoption in
general)[2][3].

> That said, I agree that WebRTC should not allow drive-by harvesting of IP
> addresses, and we intend to make changes to Chrome to prevent this.

Great to hear! Thanks! Is there a bug that is tracking this?


[1]: https://www.w3.org/wiki/Privacy/IPAddresses#Using_wildcard_DNS_entries_as_persistent_identifiers
[2]: https://github.com/EFForg/privacybadgerfirefox/issues/394
[3]: https://github.com/gorhill/uBlock/releases/tag/0.9.9.3

Daniel


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--047d7bd76778c8c164051b16775b
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The title is somewhat inaccurate, but
https://code.google.com/p/chromium/issues/detail?id=457492 is the current
bug for this.
See also https://code.google.com/p/chromium/issues/detail?id=333752, from
which the above bug was spun out.

On Fri, Jul 17, 2015 at 11:22 AM, Daniel Roesler <diafygi@gmail.com> wrote:

> On Fri, Jul 17, 2015 at 11:04 AM, Justin Uberti <juberti@google.com>
> wrote:
> >
> > This is already possible using non-WebRTC technologies, e.g. web
> sockets. As
> > such, I don't think that permission should be needed to create a data
> > channel, or receive (but not send) media.
> >
>
> The big difference is that Web Sockets starts off with an HTTP
> request, so it can be filtered by various plugins (PrivacyBadger,
> uBlock, Ghostery, etc.). WebRTC is invisible to those services since
> it does not have any HTTP requests at all. In fact, is WebRTC the only
> browser standard (i.e. not flash) that can fire off a DNS request
> that's not (at least at first) for an HTTP address[1]? Lacking an
> initial HTTP request makes it impossible for plugins to selectively
> filter these requests, so they have to fall back to an all-or-nothing
> config setting approach (which hurts WebRTC adoption in
> general)[2][3].
>
> > That said, I agree that WebRTC should not allow drive-by harvesting of IP
> > addresses, and we intend to make changes to Chrome to prevent this.
>
> Great to hear! Thanks! Is there a bug that is tracking this?
>
>
> [1]:
> https://www.w3.org/wiki/Privacy/IPAddresses#Using_wildcard_DNS_entries_as_persistent_identifiers
> [2]: https://github.com/EFForg/privacybadgerfirefox/issues/394
> [3]: https://github.com/gorhill/uBlock/releases/tag/0.9.9.3
>
> Daniel
>

--047d7bd76778c8c164051b16775b
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<div dir=3D"ltr">The title is somewhat inaccurate, but=C2=A0<a href=3D"http=
s://code.google.com/p/chromium/issues/detail?id=3D457492" target=3D"_blank"=
>https://code.google.com/p/chromium/issues/detail?id=3D457492</a> is the cu=
rrent bug for this.=C2=A0<div>See also=C2=A0<a href=3D"https://code.google.=
com/p/chromium/issues/detail?id=3D333752">https://code.google.com/p/chromiu=
m/issues/detail?id=3D333752</a>, from which the above bug was spun out.</di=
v></div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Fri, J=
ul 17, 2015 at 11:22 AM, Daniel Roesler <span dir=3D"ltr">&lt;<a href=3D"ma=
ilto:diafygi@gmail.com" target=3D"_blank">diafygi@gmail.com</a>&gt;</span> =
wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bord=
er-left:1px #ccc solid;padding-left:1ex"><span class=3D"">On Fri, Jul 17, 2=
015 at 11:04 AM, Justin Uberti &lt;<a href=3D"mailto:juberti@google.com">ju=
berti@google.com</a>&gt; wrote:<br>
&gt;<br>
&gt; This is already possible using non-WebRTC technologies, e.g. web socke=
ts. As<br>
&gt; such, I don&#39;t think that permission should be needed to create a d=
ata<br>
&gt; channel, or receive (but not send) media.<br>
&gt;<br>
<br>
</span>The big difference is that Web Sockets starts off with an HTTP<br>
request, so it can be filtered by various plugins (PrivacyBadger,<br>
uBlock, Ghostery, etc.). WebRTC is invisible to those services since<br>
it does not have any HTTP requests at all. In fact, is WebRTC the only<br>
browser standard (i.e. not flash) that can fire off a DNS request<br>
that&#39;s not (at least at first) for an HTTP address[1]? Lacking an<br>
initial HTTP request makes it impossible for plugins to selectively<br>
filter these requests, so they have to fall back to an all-or-nothing<br>
config setting approach (which hurts WebRTC adoption in<br>
general)[2][3].<br>
<span class=3D""><br>
&gt; That said, I agree that WebRTC should not allow drive-by harvesting of=
 IP<br>
&gt; addresses, and we intend to make changes to Chrome to prevent this.<br=
>
<br>
</span>Great to hear! Thanks! Is there a bug that is tracking this?<br>
<br>
<br>
[1]: <a href=3D"https://www.w3.org/wiki/Privacy/IPAddresses#Using_wildcard_=
DNS_entries_as_persistent_identifiers" rel=3D"noreferrer" target=3D"_blank"=
>https://www.w3.org/wiki/Privacy/IPAddresses#Using_wildcard_DNS_entries_as_=
persistent_identifiers</a><br>
[2]: <a href=3D"https://github.com/EFForg/privacybadgerfirefox/issues/394" =
rel=3D"noreferrer" target=3D"_blank">https://github.com/EFForg/privacybadge=
rfirefox/issues/394</a><br>
[3]: <a href=3D"https://github.com/gorhill/uBlock/releases/tag/0.9.9.3" rel=
=3D"noreferrer" target=3D"_blank">https://github.com/gorhill/uBlock/release=
s/tag/0.9.9.3</a><br>
<span class=3D"HOEnZb"><font color=3D"#888888"><br>
Daniel<br>
</font></span></blockquote></div><br></div>

--047d7bd76778c8c164051b16775b--


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From: Benjamin Schwartz <bemasc@webrtc.org>
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Subject: Re: [rtcweb] draft-jennings-behave-rtcweb-firewall-00
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--089e01183ebc5cc8d5051b1684ee
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Regarding the use of ORIGIN, the current text says

> Does adding the ORIGIN reduce user privacy? ... when the original TLS
connection for the HTTP was made, the Server Name Indication (SNI) in the
TLS of the HTTPS connection also revealed facebook.com, largely for the
same reasons - so that the firewall would be able to see which applications
are using the network.

I expect you're aware that this may not be true in TLS 1.3.

On Thu, Jul 16, 2015 at 12:51 PM, Cullen Jennings <fluffy@iii.ca> wrote:

> Good points.I will update the draft but in the meantime there is a copy a=
t
> http://fluffy.github.io/iceTrav/
>
> The summary of the algorithm is roughly:
>
> =E2=80=A2The firewall looks for any outgoing STUN requests to STUN port (=
3478).
> When it finds one, it stores the 3 tuple of the source address port and
> protocol=3DUDP and for the next 30 seconds checks any packets from this 3
> tuple to see if they are ICE connectivity checks.
>
> =E2=80=A2When firewall sees an ICE connectivity check from the inside hos=
t (which
> is a STUN Binding Request containing the ice username), it stores that ic=
e
> username value, and forwards the packet.  Thereafter, STUN Binding Reques=
ts
> and Binding Responses with matching 4-tuple (inside IP address,
> protocol=3DUDP, inside port, and matching ice username) are allowed in an=
y
> direction (inside->outside and outside->inside).  For each 5-tuple, if a
> STUN Binding Request has a Success STUN Binding Response (with same
> transaction ID), in either direction, that 5-tuple is promoted to allow
> non-STUN traffic (DTLS, SRTP, and anything else).
>
> =E2=80=A2Both promoted and non-promoted pinholes are closed after 30 seco=
nds of
> not seeing a STUN Binding Request and Response transaction, in either
> direction.
>
> =E2=80=A2(Note to match the ice username external requests will need the =
halves on
> either side of =E2=80=9C:=E2=80=9D swapped)
>
> (and thanks to  Dan Wing who came up with much of the above proposal )
>
> more inline
>
> > On Jul 8, 2015, at 8:39 AM, Eric Rescorla <ekr@rtfm.com> wrote:
> >
> > I have taken a brief look at this draft and I don't believe that the
> proposed pinhole/inspection
> > algorithm provides the desired assurances of verifying consent:
> >
> > 1. If you open up an incoming/outgoing pinhole on outgoing packet, prio=
r
> to
> > receiving an incoming STUN packet, then if the internal implementation
> > simply sends a new outgoing STUN packet every 5 seconds, then they
> > never need to rely on external consent.
> >
> > 2. If you just require *a* incoming STUN packet to be received in order
> to
> > extend the lifetime to 30 seconds, then an off-path attacker can forge
> > a STUN response that will cause the pinhole to be open for 30 seconds.
> > (Hence the need for the STUN transaction ID).
>
> Yah, don't want outside stuff to extended it. Intent was only inside stuf=
f
> should extend.
>
> >
> > It seems to me that you should be instead requiring a complete STUN
> > transaction before allowing any non-STUN outgoing traffic. It would
> probably
> > be safe-ish to require it before allowing any incoming traffic as well,
> though
> > that leaves the chance that you get a small amount of clipping with
> packet
> > drops or reordering.
>
> Agree - have changed to that.
>
> >
> >
> > WRT to the origin attribute, I have mixed feelings. I agree that your
> SNI argument
> > is strong, but there is also a move to figure out how to encrypt SNI, s=
o
> every time
> > we extend that argument it makes that harder. OTOH, STUN origin will be
> > sent a lot less often than SNI, I imagine.
>
> Sounds like this is going to get discussed in Tram. Unfortunately I think
> we have to choose between least evil on this topic.
>
> >
> >
> > I don't understand Section 6. Is this guidance to firewall vendors or t=
o
> browser
> > implementors. If the former, the firewall has no way of knowing what
> "requests"
> > are if the connection use HTTPS (and if is uses HTTP, you have DPI-type
> methods for
> > determining whether something is voice or video. If this is guidance to
> browser
> > vendors, I consider it extremely unlikely that we would implement it.
>
> advice for firewall vendors for more or less all TLS connections that
> might be web traffic. It's horrific of course but since they are
> implementing things along these lines we might as well be specific on the
> algorithm so we understand the breakage. Or go deal with the root causes
> that theses sort of things (basically trying to hide WebRTC traffic insid=
e
> HTTP) . I imagine this algorithm is not the best, it was just a rough pla=
ce
> holder. Its needs a bit of tuning to interact with dash in a reasonable w=
ay.
>
>
> >
> >
> > Point #4 in your security considerations section seems like an entirely
> lost
> > cause, unless you assume that attackers are not really trying.
> >
> > -Ekr
> >
> >
> >
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--089e01183ebc5cc8d5051b1684ee
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div>Regarding the use of ORIGIN, the current text says</d=
iv><div><br></div>&gt; Does adding the ORIGIN reduce user privacy?=C2=A0...=
 when the original TLS connection for the HTTP was made, the Server Name In=
dication (SNI) in the TLS of the HTTPS connection also revealed <a href=3D"=
http://facebook.com">facebook.com</a>, largely for the same reasons - so th=
at the firewall would be able to see which applications are using the netwo=
rk.<div><br></div><div>I expect you&#39;re aware that this may not be true =
in TLS 1.3.<br><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On=
 Thu, Jul 16, 2015 at 12:51 PM, Cullen Jennings <span dir=3D"ltr">&lt;<a hr=
ef=3D"mailto:fluffy@iii.ca" target=3D"_blank">fluffy@iii.ca</a>&gt;</span> =
wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bord=
er-left:1px #ccc solid;padding-left:1ex">Good points.I will update the draf=
t but in the meantime there is a copy at<br>
<a href=3D"http://fluffy.github.io/iceTrav/" rel=3D"noreferrer" target=3D"_=
blank">http://fluffy.github.io/iceTrav/</a><br>
<br>
The summary of the algorithm is roughly:<br>
<br>
=E2=80=A2The firewall looks for any outgoing STUN requests to STUN port (34=
78). When it finds one, it stores the 3 tuple of the source address port an=
d protocol=3DUDP and for the next 30 seconds checks any packets from this 3=
 tuple to see if they are ICE connectivity checks.<br>
<br>
=E2=80=A2When firewall sees an ICE connectivity check from the inside host =
(which is a STUN Binding Request containing the ice username), it stores th=
at ice username value, and forwards the packet.=C2=A0 Thereafter, STUN Bind=
ing Requests and Binding Responses with matching 4-tuple (inside IP address=
, protocol=3DUDP, inside port, and matching ice username) are allowed in an=
y direction (inside-&gt;outside and outside-&gt;inside).=C2=A0 For each 5-t=
uple, if a STUN Binding Request has a Success STUN Binding Response (with s=
ame transaction ID), in either direction, that 5-tuple is promoted to allow=
 non-STUN traffic (DTLS, SRTP, and anything else).<br>
<br>
=E2=80=A2Both promoted and non-promoted pinholes are closed after 30 second=
s of not seeing a STUN Binding Request and Response transaction, in either =
direction.<br>
<br>
=E2=80=A2(Note to match the ice username external requests will need the ha=
lves on either side of =E2=80=9C:=E2=80=9D swapped)<br>
<br>
(and thanks to=C2=A0 Dan Wing who came up with much of the above proposal )=
<br>
<br>
more inline<br>
<br>
&gt; On Jul 8, 2015, at 8:39 AM, Eric Rescorla &lt;<a href=3D"mailto:ekr@rt=
fm.com">ekr@rtfm.com</a>&gt; wrote:<br>
&gt;<br>
&gt; I have taken a brief look at this draft and I don&#39;t believe that t=
he proposed pinhole/inspection<br>
&gt; algorithm provides the desired assurances of verifying consent:<br>
&gt;<br>
&gt; 1. If you open up an incoming/outgoing pinhole on outgoing packet, pri=
or to<br>
&gt; receiving an incoming STUN packet, then if the internal implementation=
<br>
&gt; simply sends a new outgoing STUN packet every 5 seconds, then they<br>
&gt; never need to rely on external consent.<br>
&gt;<br>
&gt; 2. If you just require *a* incoming STUN packet to be received in orde=
r to<br>
&gt; extend the lifetime to 30 seconds, then an off-path attacker can forge=
<br>
&gt; a STUN response that will cause the pinhole to be open for 30 seconds.=
<br>
&gt; (Hence the need for the STUN transaction ID).<br>
<br>
Yah, don&#39;t want outside stuff to extended it. Intent was only inside st=
uff should extend.<br>
<br>
&gt;<br>
&gt; It seems to me that you should be instead requiring a complete STUN<br=
>
&gt; transaction before allowing any non-STUN outgoing traffic. It would pr=
obably<br>
&gt; be safe-ish to require it before allowing any incoming traffic as well=
, though<br>
&gt; that leaves the chance that you get a small amount of clipping with pa=
cket<br>
&gt; drops or reordering.<br>
<br>
Agree - have changed to that.<br>
<br>
&gt;<br>
&gt;<br>
&gt; WRT to the origin attribute, I have mixed feelings. I agree that your =
SNI argument<br>
&gt; is strong, but there is also a move to figure out how to encrypt SNI, =
so every time<br>
&gt; we extend that argument it makes that harder. OTOH, STUN origin will b=
e<br>
&gt; sent a lot less often than SNI, I imagine.<br>
<br>
Sounds like this is going to get discussed in Tram. Unfortunately I think w=
e have to choose between least evil on this topic.<br>
<br>
&gt;<br>
&gt;<br>
&gt; I don&#39;t understand Section 6. Is this guidance to firewall vendors=
 or to browser<br>
&gt; implementors. If the former, the firewall has no way of knowing what &=
quot;requests&quot;<br>
&gt; are if the connection use HTTPS (and if is uses HTTP, you have DPI-typ=
e methods for<br>
&gt; determining whether something is voice or video. If this is guidance t=
o browser<br>
&gt; vendors, I consider it extremely unlikely that we would implement it.<=
br>
<br>
advice for firewall vendors for more or less all TLS connections that might=
 be web traffic. It&#39;s horrific of course but since they are implementin=
g things along these lines we might as well be specific on the algorithm so=
 we understand the breakage. Or go deal with the root causes that theses so=
rt of things (basically trying to hide WebRTC traffic inside HTTP) . I imag=
ine this algorithm is not the best, it was just a rough place holder. Its n=
eeds a bit of tuning to interact with dash in a reasonable way.<br>
<br>
<br>
&gt;<br>
&gt;<br>
&gt; Point #4 in your security considerations section seems like an entirel=
y lost<br>
&gt; cause, unless you assume that attackers are not really trying.<br>
&gt;<br>
&gt; -Ekr<br>
&gt;<br>
&gt;<br>
&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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div><br></div></div></div>

--089e01183ebc5cc8d5051b1684ee--


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Subject: Re: [rtcweb] Please require user consent for data channels
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If I have understood the looong thread correctly, the issue is that 
Chrome overrides the OS default route and sends the STUN requests via 
all available interfaces, therefore leaking the IP addresses on the process.

AFAIK, that behavior is a Chrome implementation decision (wrong in my 
opinion, but understandable) and it is not specified on WebRTC, so while 
acknowledging the importance of the issue, this is not something that we 
should discuss in this mailing list, except if we decide that we should 
decide to explicitly forbid this behavior and add it to the security draft.

I will send a reply to discuss-webrtc to continue the discussion from a 
Chrome point of view.

Best regards
Sergio


On 17/07/2015 20:36, Justin Uberti wrote:
> The title is somewhat inaccurate, but 
> https://code.google.com/p/chromium/issues/detail?id=457492 is the 
> current bug for this.
> See also https://code.google.com/p/chromium/issues/detail?id=333752, 
> from which the above bug was spun out.
>
> On Fri, Jul 17, 2015 at 11:22 AM, Daniel Roesler <diafygi@gmail.com 
> <mailto:diafygi@gmail.com>> wrote:
>
>     On Fri, Jul 17, 2015 at 11:04 AM, Justin Uberti
>     <juberti@google.com <mailto:juberti@google.com>> wrote:
>     >
>     > This is already possible using non-WebRTC technologies, e.g. web
>     sockets. As
>     > such, I don't think that permission should be needed to create a
>     data
>     > channel, or receive (but not send) media.
>     >
>
>     The big difference is that Web Sockets starts off with an HTTP
>     request, so it can be filtered by various plugins (PrivacyBadger,
>     uBlock, Ghostery, etc.). WebRTC is invisible to those services since
>     it does not have any HTTP requests at all. In fact, is WebRTC the only
>     browser standard (i.e. not flash) that can fire off a DNS request
>     that's not (at least at first) for an HTTP address[1]? Lacking an
>     initial HTTP request makes it impossible for plugins to selectively
>     filter these requests, so they have to fall back to an all-or-nothing
>     config setting approach (which hurts WebRTC adoption in
>     general)[2][3].
>
>     > That said, I agree that WebRTC should not allow drive-by
>     harvesting of IP
>     > addresses, and we intend to make changes to Chrome to prevent this.
>
>     Great to hear! Thanks! Is there a bug that is tracking this?
>
>
>     [1]:
>     https://www.w3.org/wiki/Privacy/IPAddresses#Using_wildcard_DNS_entries_as_persistent_identifiers
>     [2]: https://github.com/EFForg/privacybadgerfirefox/issues/394
>     [3]: https://github.com/gorhill/uBlock/releases/tag/0.9.9.3
>
>     Daniel
>
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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    <div class="moz-cite-prefix">If I have understood the looong thread
      correctly, the issue is that Chrome overrides the OS default route
      and sends the STUN requests via all available interfaces,
      therefore leaking the IP addresses on the process.<br>
      <br>
      AFAIK, that behavior is a Chrome implementation decision (wrong in
      my opinion, but understandable) and it is not specified on WebRTC,
      so while acknowledging the importance of the issue, this is not
      something that we should discuss in this mailing list, except if
      we decide that we should decide to explicitly forbid this behavior
      and add it to the security draft.<br>
      <br>
      I will send a reply to discuss-webrtc to continue the discussion
      from a Chrome point of view. <br>
      <br>
      Best regards<br>
      Sergio<br>
      <br>
      <br>
      On 17/07/2015 20:36, Justin Uberti wrote:<br>
    </div>
    <blockquote
cite="mid:CAOJ7v-24VCW6kkn7LOLkqZzhYEU0r=nmd_F7Zns1rnyqKN6xAg@mail.gmail.com"
      type="cite">
      <div dir="ltr">The title is somewhat inaccurate, but <a
          moz-do-not-send="true"
          href="https://code.google.com/p/chromium/issues/detail?id=457492"
          target="_blank"><a class="moz-txt-link-freetext" href="https://code.google.com/p/chromium/issues/detail?id=457492">https://code.google.com/p/chromium/issues/detail?id=457492</a></a>
        is the current bug for this. 
        <div>See also <a moz-do-not-send="true"
            href="https://code.google.com/p/chromium/issues/detail?id=333752">https://code.google.com/p/chromium/issues/detail?id=333752</a>,
          from which the above bug was spun out.</div>
      </div>
      <div class="gmail_extra"><br>
        <div class="gmail_quote">On Fri, Jul 17, 2015 at 11:22 AM,
          Daniel Roesler <span dir="ltr">&lt;<a moz-do-not-send="true"
              href="mailto:diafygi@gmail.com" target="_blank">diafygi@gmail.com</a>&gt;</span>
          wrote:<br>
          <blockquote class="gmail_quote" style="margin:0 0 0
            .8ex;border-left:1px #ccc solid;padding-left:1ex"><span
              class="">On Fri, Jul 17, 2015 at 11:04 AM, Justin Uberti
              &lt;<a moz-do-not-send="true"
                href="mailto:juberti@google.com">juberti@google.com</a>&gt;
              wrote:<br>
              &gt;<br>
              &gt; This is already possible using non-WebRTC
              technologies, e.g. web sockets. As<br>
              &gt; such, I don't think that permission should be needed
              to create a data<br>
              &gt; channel, or receive (but not send) media.<br>
              &gt;<br>
              <br>
            </span>The big difference is that Web Sockets starts off
            with an HTTP<br>
            request, so it can be filtered by various plugins
            (PrivacyBadger,<br>
            uBlock, Ghostery, etc.). WebRTC is invisible to those
            services since<br>
            it does not have any HTTP requests at all. In fact, is
            WebRTC the only<br>
            browser standard (i.e. not flash) that can fire off a DNS
            request<br>
            that's not (at least at first) for an HTTP address[1]?
            Lacking an<br>
            initial HTTP request makes it impossible for plugins to
            selectively<br>
            filter these requests, so they have to fall back to an
            all-or-nothing<br>
            config setting approach (which hurts WebRTC adoption in<br>
            general)[2][3].<br>
            <span class=""><br>
              &gt; That said, I agree that WebRTC should not allow
              drive-by harvesting of IP<br>
              &gt; addresses, and we intend to make changes to Chrome to
              prevent this.<br>
              <br>
            </span>Great to hear! Thanks! Is there a bug that is
            tracking this?<br>
            <br>
            <br>
            [1]: <a moz-do-not-send="true"
href="https://www.w3.org/wiki/Privacy/IPAddresses#Using_wildcard_DNS_entries_as_persistent_identifiers"
              rel="noreferrer" target="_blank">https://www.w3.org/wiki/Privacy/IPAddresses#Using_wildcard_DNS_entries_as_persistent_identifiers</a><br>
            [2]: <a moz-do-not-send="true"
              href="https://github.com/EFForg/privacybadgerfirefox/issues/394"
              rel="noreferrer" target="_blank">https://github.com/EFForg/privacybadgerfirefox/issues/394</a><br>
            [3]: <a moz-do-not-send="true"
              href="https://github.com/gorhill/uBlock/releases/tag/0.9.9.3"
              rel="noreferrer" target="_blank">https://github.com/gorhill/uBlock/releases/tag/0.9.9.3</a><br>
            <span class="HOEnZb"><font color="#888888"><br>
                Daniel<br>
              </font></span></blockquote>
        </div>
        <br>
      </div>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_______________________________________________
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: Daniel Roesler <diafygi@gmail.com>
Date: Fri, 17 Jul 2015 12:20:00 -0700
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On Fri, Jul 17, 2015 at 12:11 PM, Sergio Garcia Murillo
<sergio.garcia.murillo@gmail.com> wrote:
> If I have understood the looong thread correctly, the issue is that Chrome
> overrides the OS default route and sends the STUN requests via all available
> interfaces, therefore leaking the IP addresses on the process.

Firefox has the same behavior in OSX. So it might also be OSX not
really sending all the traffic through the VPN (like the setting
says).

Would love someone more competent than me to investigate in both OSX
and Windows. I'd be willing to fund VPN subscriptions for them to use
with their research (Private Internet Access supports L2TP, PPTP,
OpenVPN, SOCKS, and they have their own client, but other VPNs are
reported as having their own clients leak the IP, so I'm willing to
fund those, too).

Daniel


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--047d7bd76778116050051b180001
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What you describe is the proper operation of RFC5245, so this isn't a
Chrome implementation decision (Firefox behaves the same way, and I assume
Edge will as well).

However, we have been thinking about whether refinements to the behaviors
specified in RFC5245 are warranted in the web context.

On Fri, Jul 17, 2015 at 12:11 PM, Sergio Garcia Murillo <
sergio.garcia.murillo@gmail.com> wrote:

>  If I have understood the looong thread correctly, the issue is that
> Chrome overrides the OS default route and sends the STUN requests via all
> available interfaces, therefore leaking the IP addresses on the process.
>
> AFAIK, that behavior is a Chrome implementation decision (wrong in my
> opinion, but understandable) and it is not specified on WebRTC, so while
> acknowledging the importance of the issue, this is not something that we
> should discuss in this mailing list, except if we decide that we should
> decide to explicitly forbid this behavior and add it to the security draft.
>
> I will send a reply to discuss-webrtc to continue the discussion from a
> Chrome point of view.
>
> Best regards
> Sergio
>
>
> On 17/07/2015 20:36, Justin Uberti wrote:
>
> The title is somewhat inaccurate, but
> <https://code.google.com/p/chromium/issues/detail?id=457492>
> https://code.google.com/p/chromium/issues/detail?id=457492 is the current
> bug for this.
> See also https://code.google.com/p/chromium/issues/detail?id=333752, from
> which the above bug was spun out.
>
> On Fri, Jul 17, 2015 at 11:22 AM, Daniel Roesler <diafygi@gmail.com>
> wrote:
>
>> On Fri, Jul 17, 2015 at 11:04 AM, Justin Uberti <juberti@google.com>
>> wrote:
>> >
>> > This is already possible using non-WebRTC technologies, e.g. web
>> sockets. As
>> > such, I don't think that permission should be needed to create a data
>> > channel, or receive (but not send) media.
>> >
>>
>> The big difference is that Web Sockets starts off with an HTTP
>> request, so it can be filtered by various plugins (PrivacyBadger,
>> uBlock, Ghostery, etc.). WebRTC is invisible to those services since
>> it does not have any HTTP requests at all. In fact, is WebRTC the only
>> browser standard (i.e. not flash) that can fire off a DNS request
>> that's not (at least at first) for an HTTP address[1]? Lacking an
>> initial HTTP request makes it impossible for plugins to selectively
>> filter these requests, so they have to fall back to an all-or-nothing
>> config setting approach (which hurts WebRTC adoption in
>> general)[2][3].
>>
>> > That said, I agree that WebRTC should not allow drive-by harvesting of
>> IP
>> > addresses, and we intend to make changes to Chrome to prevent this.
>>
>> Great to hear! Thanks! Is there a bug that is tracking this?
>>
>>
>> [1]:
>> https://www.w3.org/wiki/Privacy/IPAddresses#Using_wildcard_DNS_entries_as_persistent_identifiers
>> [2]: https://github.com/EFForg/privacybadgerfirefox/issues/394
>> [3]: https://github.com/gorhill/uBlock/releases/tag/0.9.9.3
>>
>> Daniel
>>
>
>
>
> _______________________________________________
> rtcweb mailing listrtcweb@ietf.orghttps://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--047d7bd76778116050051b180001
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">What you describe is the proper operation of RFC5245, so t=
his isn&#39;t a Chrome implementation decision (Firefox behaves the same wa=
y, and I assume Edge will as well).<div><br></div><div>However, we have bee=
n thinking about whether refinements to the behaviors specified in RFC5245 =
are warranted in the web context.<br><div class=3D"gmail_extra"><br><div cl=
ass=3D"gmail_quote">On Fri, Jul 17, 2015 at 12:11 PM, Sergio Garcia Murillo=
 <span dir=3D"ltr">&lt;<a href=3D"mailto:sergio.garcia.murillo@gmail.com" t=
arget=3D"_blank">sergio.garcia.murillo@gmail.com</a>&gt;</span> wrote:<br><=
blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px=
 #ccc solid;padding-left:1ex">
 =20
   =20
 =20
  <div bgcolor=3D"#FFFFFF" text=3D"#000000">
    <div>If I have understood the looong thread
      correctly, the issue is that Chrome overrides the OS default route
      and sends the STUN requests via all available interfaces,
      therefore leaking the IP addresses on the process.<br>
      <br>
      AFAIK, that behavior is a Chrome implementation decision (wrong in
      my opinion, but understandable) and it is not specified on WebRTC,
      so while acknowledging the importance of the issue, this is not
      something that we should discuss in this mailing list, except if
      we decide that we should decide to explicitly forbid this behavior
      and add it to the security draft.<br>
      <br>
      I will send a reply to discuss-webrtc to continue the discussion
      from a Chrome point of view. <br>
      <br>
      Best regards<br>
      Sergio<br>
      <br>
      <br>
      On 17/07/2015 20:36, Justin Uberti wrote:<br>
    </div>
    <blockquote type=3D"cite">
      <div dir=3D"ltr">The title is somewhat inaccurate, but=C2=A0<a href=
=3D"https://code.google.com/p/chromium/issues/detail?id=3D457492" target=3D=
"_blank"></a><a href=3D"https://code.google.com/p/chromium/issues/detail?id=
=3D457492" target=3D"_blank">https://code.google.com/p/chromium/issues/deta=
il?id=3D457492</a>
        is the current bug for this.=C2=A0
        <span class=3D""><div>See also=C2=A0<a href=3D"https://code.google.=
com/p/chromium/issues/detail?id=3D333752" target=3D"_blank">https://code.go=
ogle.com/p/chromium/issues/detail?id=3D333752</a>,
          from which the above bug was spun out.</div>
      </span></div><span class=3D"">
      <div class=3D"gmail_extra"><br>
        <div class=3D"gmail_quote">On Fri, Jul 17, 2015 at 11:22 AM,
          Daniel Roesler <span dir=3D"ltr">&lt;<a href=3D"mailto:diafygi@gm=
ail.com" target=3D"_blank">diafygi@gmail.com</a>&gt;</span>
          wrote:<br>
          <blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bord=
er-left:1px #ccc solid;padding-left:1ex"><span>On Fri, Jul 17, 2015 at 11:0=
4 AM, Justin Uberti
              &lt;<a href=3D"mailto:juberti@google.com" target=3D"_blank">j=
uberti@google.com</a>&gt;
              wrote:<br>
              &gt;<br>
              &gt; This is already possible using non-WebRTC
              technologies, e.g. web sockets. As<br>
              &gt; such, I don&#39;t think that permission should be needed
              to create a data<br>
              &gt; channel, or receive (but not send) media.<br>
              &gt;<br>
              <br>
            </span>The big difference is that Web Sockets starts off
            with an HTTP<br>
            request, so it can be filtered by various plugins
            (PrivacyBadger,<br>
            uBlock, Ghostery, etc.). WebRTC is invisible to those
            services since<br>
            it does not have any HTTP requests at all. In fact, is
            WebRTC the only<br>
            browser standard (i.e. not flash) that can fire off a DNS
            request<br>
            that&#39;s not (at least at first) for an HTTP address[1]?
            Lacking an<br>
            initial HTTP request makes it impossible for plugins to
            selectively<br>
            filter these requests, so they have to fall back to an
            all-or-nothing<br>
            config setting approach (which hurts WebRTC adoption in<br>
            general)[2][3].<br>
            <span><br>
              &gt; That said, I agree that WebRTC should not allow
              drive-by harvesting of IP<br>
              &gt; addresses, and we intend to make changes to Chrome to
              prevent this.<br>
              <br>
            </span>Great to hear! Thanks! Is there a bug that is
            tracking this?<br>
            <br>
            <br>
            [1]: <a href=3D"https://www.w3.org/wiki/Privacy/IPAddresses#Usi=
ng_wildcard_DNS_entries_as_persistent_identifiers" rel=3D"noreferrer" targe=
t=3D"_blank">https://www.w3.org/wiki/Privacy/IPAddresses#Using_wildcard_DNS=
_entries_as_persistent_identifiers</a><br>
            [2]: <a href=3D"https://github.com/EFForg/privacybadgerfirefox/=
issues/394" rel=3D"noreferrer" target=3D"_blank">https://github.com/EFForg/=
privacybadgerfirefox/issues/394</a><br>
            [3]: <a href=3D"https://github.com/gorhill/uBlock/releases/tag/=
0.9.9.3" rel=3D"noreferrer" target=3D"_blank">https://github.com/gorhill/uB=
lock/releases/tag/0.9.9.3</a><br>
            <span><font color=3D"#888888"><br>
                Daniel<br>
              </font></span></blockquote>
        </div>
        <br>
      </div>
      <br>
      <fieldset></fieldset>
      <br>
      </span><span class=3D""><pre>________________________________________=
_______
rtcweb mailing list
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
    </span></blockquote>
    <br>
  </div>

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

--047d7bd76778116050051b180001--


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Subject: Re: [rtcweb] Please require user consent for data channels
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On a related point. I'm not sure it's very compelling but there is
a passing reference in [1] to using WebRTC to discover local IP
addresses as a way of assisting in an attack on rc4 - presumably
the NAT'd source address offers some more known plaintext to an
attacker that can presumably see the WPA encrypted packets and hence
helps speed up the attack. See [1] p12 for the mention, though it
lacks details.

I guess the point is that folks will find unexpected ways to
(ab)use such knowledge once it's available. The right reaction of
course in this case is to not use rc4, but equally it perhaps
provides another example of the unexpected consequences of
exposing addresses.

S.

[1] http://www.rc4nomore.com/vanhoef-usenix2015.pdf


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Hi Justin,

Thanks for pointing that out, can you point me were in RFC5245 is that 
behavior described? I fail to find it.

The only references I found about multihoming are about HOST candidates, 
performing connectivity checks and priorization of candidates, but I 
cannot find a reference on sending STUN request to the TURN/STUN server 
via all the available interfaces in order to retrieve the server 
reflexive candidates that is where the leak is originating. Please 
correct me again if I am wrong.

Anyway, this raises another attack vector, as the IP may be leaked not 
directly via the ICE reflexive candidates, but via the connectivity 
checks. That is, the "malicious" server may just not get the IP via the 
server reflexive candidates, but doing an SDP O/A with remote ice 
candidates of a fake ICE server, that correlates the STUN connectivity 
checks with the web visit (via ice username for example).  They don't 
even need a STUN/TURN server deployed.

Best regards
Sergio


On 17/07/2015 22:25, Justin Uberti wrote:
> What you describe is the proper operation of RFC5245, so this isn't a 
> Chrome implementation decision (Firefox behaves the same way, and I 
> assume Edge will as well).
>
> However, we have been thinking about whether refinements to the 
> behaviors specified in RFC5245 are warranted in the web context.
>
> On Fri, Jul 17, 2015 at 12:11 PM, Sergio Garcia Murillo 
> <sergio.garcia.murillo@gmail.com 
> <mailto:sergio.garcia.murillo@gmail.com>> wrote:
>
>     If I have understood the looong thread correctly, the issue is
>     that Chrome overrides the OS default route and sends the STUN
>     requests via all available interfaces, therefore leaking the IP
>     addresses on the process.
>
>     AFAIK, that behavior is a Chrome implementation decision (wrong in
>     my opinion, but understandable) and it is not specified on WebRTC,
>     so while acknowledging the importance of the issue, this is not
>     something that we should discuss in this mailing list, except if
>     we decide that we should decide to explicitly forbid this behavior
>     and add it to the security draft.
>
>     I will send a reply to discuss-webrtc to continue the discussion
>     from a Chrome point of view.
>
>     Best regards
>     Sergio
>
>
>     On 17/07/2015 20:36, Justin Uberti wrote:
>>     The title is somewhat inaccurate, but
>>     https://code.google.com/p/chromium/issues/detail?id=457492 is the
>>     current bug for this.
>>     See also
>>     https://code.google.com/p/chromium/issues/detail?id=333752, from
>>     which the above bug was spun out.
>>
>>     On Fri, Jul 17, 2015 at 11:22 AM, Daniel Roesler
>>     <diafygi@gmail.com <mailto:diafygi@gmail.com>> wrote:
>>
>>         On Fri, Jul 17, 2015 at 11:04 AM, Justin Uberti
>>         <juberti@google.com <mailto:juberti@google.com>> wrote:
>>         >
>>         > This is already possible using non-WebRTC technologies,
>>         e.g. web sockets. As
>>         > such, I don't think that permission should be needed to
>>         create a data
>>         > channel, or receive (but not send) media.
>>         >
>>
>>         The big difference is that Web Sockets starts off with an HTTP
>>         request, so it can be filtered by various plugins (PrivacyBadger,
>>         uBlock, Ghostery, etc.). WebRTC is invisible to those
>>         services since
>>         it does not have any HTTP requests at all. In fact, is WebRTC
>>         the only
>>         browser standard (i.e. not flash) that can fire off a DNS request
>>         that's not (at least at first) for an HTTP address[1]? Lacking an
>>         initial HTTP request makes it impossible for plugins to
>>         selectively
>>         filter these requests, so they have to fall back to an
>>         all-or-nothing
>>         config setting approach (which hurts WebRTC adoption in
>>         general)[2][3].
>>
>>         > That said, I agree that WebRTC should not allow drive-by
>>         harvesting of IP
>>         > addresses, and we intend to make changes to Chrome to
>>         prevent this.
>>
>>         Great to hear! Thanks! Is there a bug that is tracking this?
>>
>>
>>         [1]:
>>         https://www.w3.org/wiki/Privacy/IPAddresses#Using_wildcard_DNS_entries_as_persistent_identifiers
>>         [2]: https://github.com/EFForg/privacybadgerfirefox/issues/394
>>         [3]: https://github.com/gorhill/uBlock/releases/tag/0.9.9.3
>>
>>         Daniel
>>
>>
>>
>>
>>     _______________________________________________
>>     rtcweb mailing list
>>     rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>>     https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>     _______________________________________________
>     rtcweb mailing list
>     rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>     https://www.ietf.org/mailman/listinfo/rtcweb
>
>


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    <div class="moz-cite-prefix">Hi Justin,<br>
      <br>
      Thanks for pointing that out, can you point me were in RFC5245 is
      that behavior described? I fail to find it. <br>
      <br>
      The only references I found about multihoming are about HOST
      candidates, performing connectivity checks and priorization of
      candidates, but I cannot find a reference on sending STUN request
      to the TURN/STUN server via all the available interfaces in order
      to retrieve the server reflexive candidates that is where the leak
      is originating. Please correct me again if I am wrong.<br>
      <br>
      Anyway, this raises another attack vector, as the IP may be leaked
      not directly via the ICE reflexive candidates, but via the
      connectivity checks. That is, the "malicious" server may just not
      get the IP via the server reflexive candidates, but doing an SDP
      O/A with remote ice candidates of a fake ICE server, that
      correlates the STUN connectivity checks with the web visit (via
      ice username for example).Â  They don't even need a STUN/TURN
      server deployed.<br>
      <br>
      Best regards<br>
      Sergio<br>
      <br>
      <br>
      On 17/07/2015 22:25, Justin Uberti wrote:<br>
    </div>
    <blockquote
cite="mid:CAOJ7v-3t9BQabR2e4EHs4G0Sec4sU9DFC2aiSXXYrat+an+RYg@mail.gmail.com"
      type="cite">
      <div dir="ltr">What you describe is the proper operation of
        RFC5245, so this isn't a Chrome implementation decision (Firefox
        behaves the same way, and I assume Edge will as well).
        <div><br>
        </div>
        <div>However, we have been thinking about whether refinements to
          the behaviors specified in RFC5245 are warranted in the web
          context.<br>
          <div class="gmail_extra"><br>
            <div class="gmail_quote">On Fri, Jul 17, 2015 at 12:11 PM,
              Sergio Garcia Murillo <span dir="ltr">&lt;<a
                  moz-do-not-send="true"
                  href="mailto:sergio.garcia.murillo@gmail.com"
                  target="_blank"><a class="moz-txt-link-abbreviated" href="mailto:sergio.garcia.murillo@gmail.com">sergio.garcia.murillo@gmail.com</a></a>&gt;</span>
              wrote:<br>
              <blockquote class="gmail_quote" style="margin:0 0 0
                .8ex;border-left:1px #ccc solid;padding-left:1ex">
                <div bgcolor="#FFFFFF" text="#000000">
                  <div>If I have understood the looong thread correctly,
                    the issue is that Chrome overrides the OS default
                    route and sends the STUN requests via all available
                    interfaces, therefore leaking the IP addresses on
                    the process.<br>
                    <br>
                    AFAIK, that behavior is a Chrome implementation
                    decision (wrong in my opinion, but understandable)
                    and it is not specified on WebRTC, so while
                    acknowledging the importance of the issue, this is
                    not something that we should discuss in this mailing
                    list, except if we decide that we should decide to
                    explicitly forbid this behavior and add it to the
                    security draft.<br>
                    <br>
                    I will send a reply to discuss-webrtc to continue
                    the discussion from a Chrome point of view. <br>
                    <br>
                    Best regards<br>
                    Sergio<br>
                    <br>
                    <br>
                    On 17/07/2015 20:36, Justin Uberti wrote:<br>
                  </div>
                  <blockquote type="cite">
                    <div dir="ltr">The title is somewhat inaccurate,
                      butÂ <a moz-do-not-send="true"
                        href="https://code.google.com/p/chromium/issues/detail?id=457492"
                        target="_blank">https://code.google.com/p/chromium/issues/detail?id=457492</a>
                      is the current bug for this.Â  <span class="">
                        <div>See alsoÂ <a moz-do-not-send="true"
                            href="https://code.google.com/p/chromium/issues/detail?id=333752"
                            target="_blank">https://code.google.com/p/chromium/issues/detail?id=333752</a>,
                          from which the above bug was spun out.</div>
                      </span></div>
                    <span class="">
                      <div class="gmail_extra"><br>
                        <div class="gmail_quote">On Fri, Jul 17, 2015 at
                          11:22 AM, Daniel Roesler <span dir="ltr">&lt;<a
                              moz-do-not-send="true"
                              href="mailto:diafygi@gmail.com"
                              target="_blank"><a class="moz-txt-link-abbreviated" href="mailto:diafygi@gmail.com">diafygi@gmail.com</a></a>&gt;</span>
                          wrote:<br>
                          <blockquote class="gmail_quote"
                            style="margin:0 0 0 .8ex;border-left:1px
                            #ccc solid;padding-left:1ex"><span>On Fri,
                              Jul 17, 2015 at 11:04 AM, Justin Uberti
                              &lt;<a moz-do-not-send="true"
                                href="mailto:juberti@google.com"
                                target="_blank">juberti@google.com</a>&gt;

                              wrote:<br>
                              &gt;<br>
                              &gt; This is already possible using
                              non-WebRTC technologies, e.g. web sockets.
                              As<br>
                              &gt; such, I don't think that permission
                              should be needed to create a data<br>
                              &gt; channel, or receive (but not send)
                              media.<br>
                              &gt;<br>
                              <br>
                            </span>The big difference is that Web
                            Sockets starts off with an HTTP<br>
                            request, so it can be filtered by various
                            plugins (PrivacyBadger,<br>
                            uBlock, Ghostery, etc.). WebRTC is invisible
                            to those services since<br>
                            it does not have any HTTP requests at all.
                            In fact, is WebRTC the only<br>
                            browser standard (i.e. not flash) that can
                            fire off a DNS request<br>
                            that's not (at least at first) for an HTTP
                            address[1]? Lacking an<br>
                            initial HTTP request makes it impossible for
                            plugins to selectively<br>
                            filter these requests, so they have to fall
                            back to an all-or-nothing<br>
                            config setting approach (which hurts WebRTC
                            adoption in<br>
                            general)[2][3].<br>
                            <span><br>
                              &gt; That said, I agree that WebRTC should
                              not allow drive-by harvesting of IP<br>
                              &gt; addresses, and we intend to make
                              changes to Chrome to prevent this.<br>
                              <br>
                            </span>Great to hear! Thanks! Is there a bug
                            that is tracking this?<br>
                            <br>
                            <br>
                            [1]: <a moz-do-not-send="true"
href="https://www.w3.org/wiki/Privacy/IPAddresses#Using_wildcard_DNS_entries_as_persistent_identifiers"
                              rel="noreferrer" target="_blank">https://www.w3.org/wiki/Privacy/IPAddresses#Using_wildcard_DNS_entries_as_persistent_identifiers</a><br>
                            [2]: <a moz-do-not-send="true"
                              href="https://github.com/EFForg/privacybadgerfirefox/issues/394"
                              rel="noreferrer" target="_blank">https://github.com/EFForg/privacybadgerfirefox/issues/394</a><br>
                            [3]: <a moz-do-not-send="true"
                              href="https://github.com/gorhill/uBlock/releases/tag/0.9.9.3"
                              rel="noreferrer" target="_blank">https://github.com/gorhill/uBlock/releases/tag/0.9.9.3</a><br>
                            <span><font color="#888888"><br>
                                Daniel<br>
                              </font></span></blockquote>
                        </div>
                        <br>
                      </div>
                      <br>
                      <fieldset></fieldset>
                      <br>
                    </span><span class="">
                      <pre>_______________________________________________
rtcweb mailing list
<a moz-do-not-send="true" href="mailto:rtcweb@ietf.org" target="_blank">rtcweb@ietf.org</a>
<a moz-do-not-send="true" href="https://www.ietf.org/mailman/listinfo/rtcweb" target="_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
                    </span></blockquote>
                  <br>
                </div>
                <br>
                _______________________________________________<br>
                rtcweb mailing list<br>
                <a moz-do-not-send="true" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
                <a moz-do-not-send="true"
                  href="https://www.ietf.org/mailman/listinfo/rtcweb"
                  rel="noreferrer" target="_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
                <br>
              </blockquote>
            </div>
            <br>
          </div>
        </div>
      </div>
    </blockquote>
    <br>
  </body>
</html>

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Date: Fri, 17 Jul 2015 14:41:34 -0700
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--001a113ec94e6f5133051b190e9c
Content-Type: text/plain; charset=UTF-8

On Fri, Jul 17, 2015 at 2:03 PM, Sergio Garcia Murillo <
sergio.garcia.murillo@gmail.com> wrote:

>  Hi Justin,
>
> Thanks for pointing that out, can you point me were in RFC5245 is that
> behavior described? I fail to find it.
>
> The only references I found about multihoming are about HOST candidates,
> performing connectivity checks and priorization of candidates, but I cannot
> find a reference on sending STUN request to the TURN/STUN server via all
> the available interfaces in order to retrieve the server reflexive
> candidates that is where the leak is originating. Please correct me again
> if I am wrong.
>

Section 2.1, paragraphs 2 and 3 talks about getting all interfaces, and
then getting STUN and TURN candidates from said interfaces.

>
> Anyway, this raises another attack vector, as the IP may be leaked not
> directly via the ICE reflexive candidates, but via the connectivity checks.
> That is, the "malicious" server may just not get the IP via the server
> reflexive candidates, but doing an SDP O/A with remote ice candidates of a
> fake ICE server, that correlates the STUN connectivity checks with the web
> visit (via ice username for example).  They don't even need a STUN/TURN
> server deployed.
>
> Best regards
> Sergio
>
>
> On 17/07/2015 22:25, Justin Uberti wrote:
>
> What you describe is the proper operation of RFC5245, so this isn't a
> Chrome implementation decision (Firefox behaves the same way, and I assume
> Edge will as well).
>
>  However, we have been thinking about whether refinements to the
> behaviors specified in RFC5245 are warranted in the web context.
>
> On Fri, Jul 17, 2015 at 12:11 PM, Sergio Garcia Murillo <
> <sergio.garcia.murillo@gmail.com>sergio.garcia.murillo@gmail.com> wrote:
>
>>  If I have understood the looong thread correctly, the issue is that
>> Chrome overrides the OS default route and sends the STUN requests via all
>> available interfaces, therefore leaking the IP addresses on the process.
>>
>> AFAIK, that behavior is a Chrome implementation decision (wrong in my
>> opinion, but understandable) and it is not specified on WebRTC, so while
>> acknowledging the importance of the issue, this is not something that we
>> should discuss in this mailing list, except if we decide that we should
>> decide to explicitly forbid this behavior and add it to the security draft.
>>
>> I will send a reply to discuss-webrtc to continue the discussion from a
>> Chrome point of view.
>>
>> Best regards
>> Sergio
>>
>>
>> On 17/07/2015 20:36, Justin Uberti wrote:
>>
>> The title is somewhat inaccurate, but
>> https://code.google.com/p/chromium/issues/detail?id=457492 is the
>> current bug for this.
>> See also https://code.google.com/p/chromium/issues/detail?id=333752,
>> from which the above bug was spun out.
>>
>> On Fri, Jul 17, 2015 at 11:22 AM, Daniel Roesler < <diafygi@gmail.com>
>> diafygi@gmail.com> wrote:
>>
>>> On Fri, Jul 17, 2015 at 11:04 AM, Justin Uberti <juberti@google.com>
>>> wrote:
>>> >
>>> > This is already possible using non-WebRTC technologies, e.g. web
>>> sockets. As
>>> > such, I don't think that permission should be needed to create a data
>>> > channel, or receive (but not send) media.
>>> >
>>>
>>> The big difference is that Web Sockets starts off with an HTTP
>>> request, so it can be filtered by various plugins (PrivacyBadger,
>>> uBlock, Ghostery, etc.). WebRTC is invisible to those services since
>>> it does not have any HTTP requests at all. In fact, is WebRTC the only
>>> browser standard (i.e. not flash) that can fire off a DNS request
>>> that's not (at least at first) for an HTTP address[1]? Lacking an
>>> initial HTTP request makes it impossible for plugins to selectively
>>> filter these requests, so they have to fall back to an all-or-nothing
>>> config setting approach (which hurts WebRTC adoption in
>>> general)[2][3].
>>>
>>> > That said, I agree that WebRTC should not allow drive-by harvesting of
>>> IP
>>> > addresses, and we intend to make changes to Chrome to prevent this.
>>>
>>> Great to hear! Thanks! Is there a bug that is tracking this?
>>>
>>>
>>> [1]:
>>> https://www.w3.org/wiki/Privacy/IPAddresses#Using_wildcard_DNS_entries_as_persistent_identifiers
>>> [2]: https://github.com/EFForg/privacybadgerfirefox/issues/394
>>> [3]: https://github.com/gorhill/uBlock/releases/tag/0.9.9.3
>>>
>>> Daniel
>>>
>>
>>
>>
>>  _______________________________________________
>> rtcweb mailing listrtcweb@ietf.orghttps://www.ietf.org/mailman/listinfo/rtcweb
>>
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>>
>
>

--001a113ec94e6f5133051b190e9c
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Fri, Jul 17, 2015 at 2:03 PM, Sergio Garcia Murillo <span dir=3D"ltr=
">&lt;<a href=3D"mailto:sergio.garcia.murillo@gmail.com" target=3D"_blank">=
sergio.garcia.murillo@gmail.com</a>&gt;</span> wrote:<br><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex">
 =20
   =20
 =20
  <div bgcolor=3D"#FFFFFF" text=3D"#000000">
    <div>Hi Justin,<br>
      <br>
      Thanks for pointing that out, can you point me were in RFC5245 is
      that behavior described? I fail to find it. <br>
      <br>
      The only references I found about multihoming are about HOST
      candidates, performing connectivity checks and priorization of
      candidates, but I cannot find a reference on sending STUN request
      to the TURN/STUN server via all the available interfaces in order
      to retrieve the server reflexive candidates that is where the leak
      is originating. Please correct me again if I am wrong.<br></div></div=
></blockquote><div><br></div><div>Section 2.1, paragraphs 2 and 3 talks abo=
ut getting all interfaces, and then getting STUN and TURN candidates from s=
aid interfaces.</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 =
0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div bgcolor=3D"#FFFFFF=
" text=3D"#000000"><div>
      <br>
      Anyway, this raises another attack vector, as the IP may be leaked
      not directly via the ICE reflexive candidates, but via the
      connectivity checks. That is, the &quot;malicious&quot; server may ju=
st not
      get the IP via the server reflexive candidates, but doing an SDP
      O/A with remote ice candidates of a fake ICE server, that
      correlates the STUN connectivity checks with the web visit (via
      ice username for example).=C2=A0 They don&#39;t even need a STUN/TURN
      server deployed.<br>
      <br>
      Best regards<br>
      Sergio<span class=3D""><br>
      <br>
      <br>
      On 17/07/2015 22:25, Justin Uberti wrote:<br>
    </span></div>
    <blockquote type=3D"cite">
      <div dir=3D"ltr"><span class=3D"">What you describe is the proper ope=
ration of
        RFC5245, so this isn&#39;t a Chrome implementation decision (Firefo=
x
        behaves the same way, and I assume Edge will as well).
        <div><br>
        </div>
        </span><div><span class=3D"">However, we have been thinking about w=
hether refinements to
          the behaviors specified in RFC5245 are warranted in the web
          context.<br>
          </span><div class=3D"gmail_extra"><br>
            <div class=3D"gmail_quote"><span class=3D"">On Fri, Jul 17, 201=
5 at 12:11 PM,
              Sergio Garcia Murillo <span dir=3D"ltr">&lt;<a href=3D"mailto=
:sergio.garcia.murillo@gmail.com" target=3D"_blank"></a><a href=3D"mailto:s=
ergio.garcia.murillo@gmail.com" target=3D"_blank">sergio.garcia.murillo@gma=
il.com</a>&gt;</span>
              wrote:<br>
              </span><blockquote class=3D"gmail_quote" style=3D"margin:0 0 =
0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
                <div bgcolor=3D"#FFFFFF" text=3D"#000000"><span class=3D"">
                  <div>If I have understood the looong thread correctly,
                    the issue is that Chrome overrides the OS default
                    route and sends the STUN requests via all available
                    interfaces, therefore leaking the IP addresses on
                    the process.<br>
                    <br>
                    AFAIK, that behavior is a Chrome implementation
                    decision (wrong in my opinion, but understandable)
                    and it is not specified on WebRTC, so while
                    acknowledging the importance of the issue, this is
                    not something that we should discuss in this mailing
                    list, except if we decide that we should decide to
                    explicitly forbid this behavior and add it to the
                    security draft.<br>
                    <br>
                    I will send a reply to discuss-webrtc to continue
                    the discussion from a Chrome point of view. <br>
                    <br>
                    Best regards<br>
                    Sergio<br>
                    <br>
                    <br>
                    On 17/07/2015 20:36, Justin Uberti wrote:<br>
                  </div>
                  </span><blockquote type=3D"cite">
                    <div dir=3D"ltr">The title is somewhat inaccurate,
                      but=C2=A0<a href=3D"https://code.google.com/p/chromiu=
m/issues/detail?id=3D457492" target=3D"_blank">https://code.google.com/p/ch=
romium/issues/detail?id=3D457492</a>
                      is the current bug for this.=C2=A0 <span class=3D""><=
span>
                        <div>See also=C2=A0<a href=3D"https://code.google.c=
om/p/chromium/issues/detail?id=3D333752" target=3D"_blank">https://code.goo=
gle.com/p/chromium/issues/detail?id=3D333752</a>,
                          from which the above bug was spun out.</div>
                      </span></span></div><div><div class=3D"h5">
                    <span>
                      <div class=3D"gmail_extra"><br>
                        <div class=3D"gmail_quote">On Fri, Jul 17, 2015 at
                          11:22 AM, Daniel Roesler <span dir=3D"ltr">&lt;<a=
 href=3D"mailto:diafygi@gmail.com" target=3D"_blank"></a><a href=3D"mailto:=
diafygi@gmail.com" target=3D"_blank">diafygi@gmail.com</a>&gt;</span>
                          wrote:<br>
                          <blockquote class=3D"gmail_quote" style=3D"margin=
:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><span>On Fri,
                              Jul 17, 2015 at 11:04 AM, Justin Uberti
                              &lt;<a href=3D"mailto:juberti@google.com" tar=
get=3D"_blank">juberti@google.com</a>&gt;

                              wrote:<br>
                              &gt;<br>
                              &gt; This is already possible using
                              non-WebRTC technologies, e.g. web sockets.
                              As<br>
                              &gt; such, I don&#39;t think that permission
                              should be needed to create a data<br>
                              &gt; channel, or receive (but not send)
                              media.<br>
                              &gt;<br>
                              <br>
                            </span>The big difference is that Web
                            Sockets starts off with an HTTP<br>
                            request, so it can be filtered by various
                            plugins (PrivacyBadger,<br>
                            uBlock, Ghostery, etc.). WebRTC is invisible
                            to those services since<br>
                            it does not have any HTTP requests at all.
                            In fact, is WebRTC the only<br>
                            browser standard (i.e. not flash) that can
                            fire off a DNS request<br>
                            that&#39;s not (at least at first) for an HTTP
                            address[1]? Lacking an<br>
                            initial HTTP request makes it impossible for
                            plugins to selectively<br>
                            filter these requests, so they have to fall
                            back to an all-or-nothing<br>
                            config setting approach (which hurts WebRTC
                            adoption in<br>
                            general)[2][3].<br>
                            <span><br>
                              &gt; That said, I agree that WebRTC should
                              not allow drive-by harvesting of IP<br>
                              &gt; addresses, and we intend to make
                              changes to Chrome to prevent this.<br>
                              <br>
                            </span>Great to hear! Thanks! Is there a bug
                            that is tracking this?<br>
                            <br>
                            <br>
                            [1]: <a href=3D"https://www.w3.org/wiki/Privacy=
/IPAddresses#Using_wildcard_DNS_entries_as_persistent_identifiers" rel=3D"n=
oreferrer" target=3D"_blank">https://www.w3.org/wiki/Privacy/IPAddresses#Us=
ing_wildcard_DNS_entries_as_persistent_identifiers</a><br>
                            [2]: <a href=3D"https://github.com/EFForg/priva=
cybadgerfirefox/issues/394" rel=3D"noreferrer" target=3D"_blank">https://gi=
thub.com/EFForg/privacybadgerfirefox/issues/394</a><br>
                            [3]: <a href=3D"https://github.com/gorhill/uBlo=
ck/releases/tag/0.9.9.3" rel=3D"noreferrer" target=3D"_blank">https://githu=
b.com/gorhill/uBlock/releases/tag/0.9.9.3</a><br>
                            <span><font color=3D"#888888"><br>
                                Daniel<br>
                              </font></span></blockquote>
                        </div>
                        <br>
                      </div>
                      <br>
                      <fieldset></fieldset>
                      <br>
                    </span><span>
                      <pre>_______________________________________________
rtcweb mailing list
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
                    </span></div></div></blockquote>
                  <br>
                </div><div><div class=3D"h5">
                <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" re=
l=3D"noreferrer" target=3D"_blank">https://www.ietf.org/mailman/listinfo/rt=
cweb</a><br>
                <br>
              </div></div></blockquote>
            </div>
            <br>
          </div>
        </div>
      </div>
    </blockquote>
    <br>
  </div>

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

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From: Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com>
Message-ID: <55A9860D.8030903@gmail.com>
Date: Sat, 18 Jul 2015 00:47:41 +0200
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Subject: Re: [rtcweb] Please require user consent for data channels
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On 17/07/2015 23:41, Justin Uberti wrote:
>
>
> On Fri, Jul 17, 2015 at 2:03 PM, Sergio Garcia Murillo 
> <sergio.garcia.murillo@gmail.com 
> <mailto:sergio.garcia.murillo@gmail.com>> wrote:
>
>     Hi Justin,
>
>     Thanks for pointing that out, can you point me were in RFC5245 is
>     that behavior described? I fail to find it.
>
>     The only references I found about multihoming are about HOST
>     candidates, performing connectivity checks and priorization of
>     candidates, but I cannot find a reference on sending STUN request
>     to the TURN/STUN server via all the available interfaces in order
>     to retrieve the server reflexive candidates that is where the leak
>     is originating. Please correct me again if I am wrong.
>
>
> Section 2.1, paragraphs 2 and 3 talks about getting all interfaces, 
> and then getting STUN and TURN candidates from said interfaces.
>

That was the ones I was looking at, but my understanding of them is 
different:

    If an agent is multihomed, it obtains a candidate from each IP
    address.  Depending on the location of the PEER (the other agent in
    the session) on the IP network relative to the agent, the agent may
    be reachable by the peer through one or more of those IP addresses.
    Consider, for example, an agent that has a local IP address on a
    private net 10 network (I1), and a second connected to the public
    Internet (I2).  A candidate from I1 will be directly reachable when
    communicating with a peer on the same private net 10 network, while a
    candidate from I2 will be directly reachable when communicating with
    a peer on the public Internet.  Rather than trying to guess which IP
    address will work prior to sending an offer, the offering agent
    includes both candidates in its offer.

Here, what I understand is that they are talking about Host ICE candidates, an then, yes, one per interface, no issue here (If my understanding of the leak problem is correct).

    Next, the agent uses STUN or TURN to obtain additional candidates.
    These come in two flavors: translated addresses on the public side of
    a NAT (SERVER REFLEXIVE CANDIDATES) and addresses on TURN servers
    (RELAYED CANDIDATES).  When TURN servers are utilized, both types of
    candidates are obtained from the TURN server.  If only STUN servers
    are utilized, only server reflexive candidates are obtained from
    them.  The relationship of these candidates to the host candidate is
    shown in Figure 2.  In this figure, both types of candidates are
    discovered using TURN.  In the figure, the notation X:x means IP
    address X and UDP port x.

In this one is talking about getting server reflexive and relay candidates, but in no way I see that the stun server should be contacted by all the interfaces or overriding default routing from OS.

What am I missing?

Best regards
Sergio


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    <div class="moz-cite-prefix">On 17/07/2015 23:41, Justin Uberti
      wrote:<br>
    </div>
    <blockquote
cite="mid:CAOJ7v-1ui7349NzK6NZNRHPbnHWZajctk4cDgMKqRZSv47EYdA@mail.gmail.com"
      type="cite">
      <div dir="ltr"><br>
        <div class="gmail_extra"><br>
          <div class="gmail_quote">On Fri, Jul 17, 2015 at 2:03 PM,
            Sergio Garcia Murillo <span dir="ltr">&lt;<a
                moz-do-not-send="true"
                href="mailto:sergio.garcia.murillo@gmail.com"
                target="_blank"><a class="moz-txt-link-abbreviated" href="mailto:sergio.garcia.murillo@gmail.com">sergio.garcia.murillo@gmail.com</a></a>&gt;</span>
            wrote:<br>
            <blockquote class="gmail_quote" style="margin:0 0 0
              .8ex;border-left:1px #ccc solid;padding-left:1ex">
              <div bgcolor="#FFFFFF" text="#000000">
                <div>Hi Justin,<br>
                  <br>
                  Thanks for pointing that out, can you point me were in
                  RFC5245 is that behavior described? I fail to find it.
                  <br>
                  <br>
                  The only references I found about multihoming are
                  about HOST candidates, performing connectivity checks
                  and priorization of candidates, but I cannot find a
                  reference on sending STUN request to the TURN/STUN
                  server via all the available interfaces in order to
                  retrieve the server reflexive candidates that is where
                  the leak is originating. Please correct me again if I
                  am wrong.<br>
                </div>
              </div>
            </blockquote>
            <div><br>
            </div>
            <div>Section 2.1, paragraphs 2 and 3 talks about getting all
              interfaces, and then getting STUN and TURN candidates from
              said interfaces.</div>
            <br>
          </div>
        </div>
      </div>
    </blockquote>
    <br>
    That was the ones I was looking at, but my understanding of them is
    different:<br>
    <br>
    <pre class="newpage" style="font-size: 13.3333330154419px; margin-top: 0px; margin-bottom: 0px; page-break-before: always; color: rgb(0, 0, 0); font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: auto; text-align: start; text-indent: 0px; text-transform: none; widows: 1; word-spacing: 0px; -webkit-text-stroke-width: 0px;">   If an agent is multihomed, it obtains a candidate from each IP
   address.  Depending on the location of the PEER (the other agent in
   the session) on the IP network relative to the agent, the agent may
   be reachable by the peer through one or more of those IP addresses.
   Consider, for example, an agent that has a local IP address on a
   private net 10 network (I1), and a second connected to the public
   Internet (I2).  A candidate from I1 will be directly reachable when
   communicating with a peer on the same private net 10 network, while a
   candidate from I2 will be directly reachable when communicating with
   a peer on the public Internet.  Rather than trying to guess which IP
   address will work prior to sending an offer, the offering agent
   includes both candidates in its offer.

Here, what I understand is that they are talking about Host ICE candidates, an then, yes, one per interface, no issue here (If my understanding of the leak problem is correct).

   Next, the agent uses STUN or TURN to obtain additional candidates.
   These come in two flavors: translated addresses on the public side of
   a NAT (SERVER REFLEXIVE CANDIDATES) and addresses on TURN servers
   (RELAYED CANDIDATES).  When TURN servers are utilized, both types of
   candidates are obtained from the TURN server.  If only STUN servers
   are utilized, only server reflexive candidates are obtained from
   them.  The relationship of these candidates to the host candidate is
   shown in Figure 2.  In this figure, both types of candidates are
   discovered using TURN.  In the figure, the notation X:x means IP
   address X and UDP port x.

In this one is talking about getting server reflexive and relay candidates, but in no way I see that the stun server should be contacted by all the interfaces or overriding default routing from OS.

What am I missing?

Best regards
Sergio
</pre>
  </body>
</html>

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From: Justin Uberti <juberti@google.com>
Date: Fri, 17 Jul 2015 16:09:56 -0700
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On Fri, Jul 17, 2015 at 3:47 PM, Sergio Garcia Murillo <
sergio.garcia.murillo@gmail.com> wrote:

>  On 17/07/2015 23:41, Justin Uberti wrote:
>
>
>
> On Fri, Jul 17, 2015 at 2:03 PM, Sergio Garcia Murillo <
> <sergio.garcia.murillo@gmail.com>sergio.garcia.murillo@gmail.com> wrote:
>
>>  Hi Justin,
>>
>> Thanks for pointing that out, can you point me were in RFC5245 is that
>> behavior described? I fail to find it.
>>
>> The only references I found about multihoming are about HOST candidates,
>> performing connectivity checks and priorization of candidates, but I cannot
>> find a reference on sending STUN request to the TURN/STUN server via all
>> the available interfaces in order to retrieve the server reflexive
>> candidates that is where the leak is originating. Please correct me again
>> if I am wrong.
>>
>
>  Section 2.1, paragraphs 2 and 3 talks about getting all interfaces, and
> then getting STUN and TURN candidates from said interfaces.
>
>
> That was the ones I was looking at, but my understanding of them is
> different:
>
>    If an agent is multihomed, it obtains a candidate from each IP
>    address.  Depending on the location of the PEER (the other agent in
>    the session) on the IP network relative to the agent, the agent may
>    be reachable by the peer through one or more of those IP addresses.
>    Consider, for example, an agent that has a local IP address on a
>    private net 10 network (I1), and a second connected to the public
>    Internet (I2).  A candidate from I1 will be directly reachable when
>    communicating with a peer on the same private net 10 network, while a
>    candidate from I2 will be directly reachable when communicating with
>    a peer on the public Internet.  Rather than trying to guess which IP
>    address will work prior to sending an offer, the offering agent
>    includes both candidates in its offer.
>
> Here, what I understand is that they are talking about Host ICE candidates, an then, yes, one per interface, no issue here (If my understanding of the leak problem is correct).
>
>
Agreed.

>    Next, the agent uses STUN or TURN to obtain additional candidates.
>    These come in two flavors: translated addresses on the public side of
>    a NAT (SERVER REFLEXIVE CANDIDATES) and addresses on TURN servers
>    (RELAYED CANDIDATES).  When TURN servers are utilized, both types of
>    candidates are obtained from the TURN server.  If only STUN servers
>    are utilized, only server reflexive candidates are obtained from
>    them.  The relationship of these candidates to the host candidate is
>    shown in Figure 2.  In this figure, both types of candidates are
>    discovered using TURN.  In the figure, the notation X:x means IP
>    address X and UDP port x.
>
> In this one is talking about getting server reflexive and relay candidates, but in no way I see that the stun server should be contacted by all the interfaces or overriding default routing from OS.
>
>
It is pretty clear from the diagram on page 10 that for each host candidate
X:x, the agent will gather a srflx candidate X1':x1' and relay candidate
Y:y. This is also spelled out in the text underneath the diagram:

   When the agent sends the TURN Allocate request from IP address and
   port X:x, the NAT (assuming there is one) will create a binding
   X1':x1', mapping this server reflexive candidate to the host
   candidate X:x.

Ergo, one tries to gather srflx and relay candidates for each host
candidate X:x.

--047d7b10c87181c0df051b1a4a2c
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Fri, Jul 17, 2015 at 3:47 PM, Sergio Garcia Murillo <span dir=3D"ltr=
">&lt;<a href=3D"mailto:sergio.garcia.murillo@gmail.com" target=3D"_blank">=
sergio.garcia.murillo@gmail.com</a>&gt;</span> wrote:<br><blockquote class=
=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;bo=
rder-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
 =20
   =20
 =20
  <div bgcolor=3D"#FFFFFF" text=3D"#000000">
    <div>On 17/07/2015 23:41, Justin Uberti
      wrote:<br>
    </div><span class=3D"">
    <blockquote type=3D"cite">
      <div dir=3D"ltr"><br>
        <div class=3D"gmail_extra"><br>
          <div class=3D"gmail_quote">On Fri, Jul 17, 2015 at 2:03 PM,
            Sergio Garcia Murillo <span dir=3D"ltr">&lt;<a href=3D"mailto:s=
ergio.garcia.murillo@gmail.com" target=3D"_blank"></a><a href=3D"mailto:ser=
gio.garcia.murillo@gmail.com" target=3D"_blank">sergio.garcia.murillo@gmail=
.com</a>&gt;</span>
            wrote:<br>
            <blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0=
.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-s=
tyle:solid;padding-left:1ex">
              <div bgcolor=3D"#FFFFFF" text=3D"#000000">
                <div>Hi Justin,<br>
                  <br>
                  Thanks for pointing that out, can you point me were in
                  RFC5245 is that behavior described? I fail to find it.
                  <br>
                  <br>
                  The only references I found about multihoming are
                  about HOST candidates, performing connectivity checks
                  and priorization of candidates, but I cannot find a
                  reference on sending STUN request to the TURN/STUN
                  server via all the available interfaces in order to
                  retrieve the server reflexive candidates that is where
                  the leak is originating. Please correct me again if I
                  am wrong.<br>
                </div>
              </div>
            </blockquote>
            <div><br>
            </div>
            <div>Section 2.1, paragraphs 2 and 3 talks about getting all
              interfaces, and then getting STUN and TURN candidates from
              said interfaces.</div>
            <br>
          </div>
        </div>
      </div>
    </blockquote>
    <br></span>
    That was the ones I was looking at, but my understanding of them is
    different:<br>
    <br>
    <pre style=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px;colo=
r:rgb(0,0,0);font-style:normal;font-variant:normal;font-weight:normal;lette=
r-spacing:normal;line-height:normal;text-align:start;text-indent:0px;text-t=
ransform:none;word-spacing:0px">   If an agent is multihomed, it obtains a =
candidate from each IP
   address.  Depending on the location of the PEER (the other agent in
   the session) on the IP network relative to the agent, the agent may
   be reachable by the peer through one or more of those IP addresses.
   Consider, for example, an agent that has a local IP address on a
   private net 10 network (I1), and a second connected to the public
   Internet (I2).  A candidate from I1 will be directly reachable when
   communicating with a peer on the same private net 10 network, while a
   candidate from I2 will be directly reachable when communicating with
   a peer on the public Internet.  Rather than trying to guess which IP
   address will work prior to sending an offer, the offering agent
   includes both candidates in its offer.

Here, what I understand is that they are talking about Host ICE candidates,=
 an then, yes, one per interface, no issue here (If my understanding of the=
 leak problem is correct).</pre></div></blockquote><div><br></div><div>Agre=
ed.</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8e=
x;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-styl=
e:solid;padding-left:1ex"><div bgcolor=3D"#FFFFFF" text=3D"#000000"><pre st=
yle=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0=
);font-style:normal;font-variant:normal;font-weight:normal;letter-spacing:n=
ormal;line-height:normal;text-align:start;text-indent:0px;text-transform:no=
ne;word-spacing:0px">   Next, the agent uses STUN or TURN to obtain additio=
nal candidates.
   These come in two flavors: translated addresses on the public side of
   a NAT (SERVER REFLEXIVE CANDIDATES) and addresses on TURN servers
   (RELAYED CANDIDATES).  When TURN servers are utilized, both types of
   candidates are obtained from the TURN server.  If only STUN servers
   are utilized, only server reflexive candidates are obtained from
   them.  The relationship of these candidates to the host candidate is
   shown in Figure 2.  In this figure, both types of candidates are
   discovered using TURN.  In the figure, the notation X:x means IP
   address X and UDP port x.

In this one is talking about getting server reflexive and relay candidates,=
 but in no way I see that the stun server should be contacted by all the in=
terfaces or overriding default routing from OS.</pre></div></blockquote><di=
v><br></div><div>It is pretty clear from the diagram on page 10 that for ea=
ch host candidate X:x, the agent will gather a srflx candidate X1&#39;:x1&#=
39; and relay candidate Y:y. This is also spelled out in the text underneat=
h the diagram:</div><div><br></div><div><div>=C2=A0 =C2=A0When the agent se=
nds the TURN Allocate request from IP address and</div><div>=C2=A0 =C2=A0po=
rt X:x, the NAT (assuming there is one) will create a binding</div><div>=C2=
=A0 =C2=A0X1&#39;:x1&#39;, mapping this server reflexive candidate to the h=
ost</div><div>=C2=A0 =C2=A0candidate X:x.=C2=A0</div></div><div><br></div><=
div>Ergo, one tries to gather srflx and relay candidates for each host cand=
idate X:x.</div></div></div></div>

--047d7b10c87181c0df051b1a4a2c--


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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Sat, 18 Jul 2015 01:34:52 +0200
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2015-07-18 1:09 GMT+02:00 Justin Uberti <juberti@google.com>:
> It is pretty clear from the diagram on page 10 that for each host candida=
te
> X:x, the agent will gather a srflx candidate X1':x1' and relay candidate
> Y:y. This is also spelled out in the text underneath the diagram:
>
>    When the agent sends the TURN Allocate request from IP address and
>    port X:x, the NAT (assuming there is one) will create a binding
>    X1':x1', mapping this server reflexive candidate to the host
>    candidate X:x.
>
> Ergo, one tries to gather srflx and relay candidates for each host candid=
ate
> X:x.

That's not what I understand.

What I understand is that the browser should bind on 0.0.0.0 (or ::0)
and send the STUN/TURN request, and the underlying OS would decide the
proper interface and source IP. So probably just one or two (IPv4 and
IPv6) STUN/TURN attempts would be performed during the gathering, and
when *that* happens the quoted text above applies (being X the IP of
the candidate matching the source IP chosen by the OS to route the
STUN/TURN request).

Said that, and omitting the above confusing text: what is the purpose
of sending STUN/TURN requests over all the interfaces? How is that
good at all? AFAIK in most cases that will just mean timeouts waiting
for STUN responses (which is bad if trickle ICE is not being used).

To be even more clear: STUN/TURN servers are typically in the public
Internet, although they may also be placed in local networks. Anyhow,
computers within that network have a single and already configured
route to reach the STUN/TURN server (usually the default route).
Sending STUN/TURN requests from other interfaces will just fail.


--=20
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<ibc@aliax.net>


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On 18/07/2015 1:09, Justin Uberti wrote:
>     Next, the agent uses STUN or TURN to obtain additional candidates.
>     These come in two flavors: translated addresses on the public side of
>     a NAT (SERVER REFLEXIVE CANDIDATES) and addresses on TURN servers
>     (RELAYED CANDIDATES).  When TURN servers are utilized, both types of
>     candidates are obtained from the TURN server.  If only STUN servers
>     are utilized, only server reflexive candidates are obtained from
>     them.  The relationship of these candidates to the host candidate is
>     shown in Figure 2.  In this figure, both types of candidates are
>     discovered using TURN.  In the figure, the notation X:x means IP
>     address X and UDP port x.
>
> In this one is talking about getting server reflexive and relay candidates, but in no way I see that the stun server should be contacted by all the interfaces or overriding default routing from OS.
>
> It is pretty clear from the diagram on page 10 that for each host 
> candidate X:x, the agent will gather a srflx candidate X1':x1' and 
> relay candidate Y:y. This is also spelled out in the text underneath 
> the diagram:
>
>    When the agent sends the TURN Allocate request from IP address and
>    port X:x, the NAT (assuming there is one) will create a binding
>    X1':x1', mapping this server reflexive candidate to the host
>    candidate X:x.
>
> Ergo, one tries to gather srflx and relay candidates for each host 
> candidate X:x.

Agree, for each HOST candidate, it should send a STUN request to the 
turn server from that IP:port. But shouldn't the VPN configuration 
prevent the non-VPN-host-candidate STUN request to be sent via the 
non-VPN-interface? (i.e. applying default route based on dest?)

Best regards
Sergio


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On Jul 17, 2015 4:35 PM, "I=C3=B1aki Baz Castillo" <ibc@aliax.net> wrote:
> What I understand is that the browser should bind on 0.0.0.0 (or ::0)
> and send the STUN/TURN request,

That works most of the time, but it is a cheat. What Chrome and Firefox do
- bind to reach local interface separately - is the only reliable way to do
this. If you bind to 0.0.0.0, you can't handle multiple interfaces
correctly, and that reduces the odds of completing ICE with the best result=
.

--001a11490eb81fac35051b1aba31
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<p dir=3D"ltr"><br>
On Jul 17, 2015 4:35 PM, &quot;I=C3=B1aki Baz Castillo&quot; &lt;<a href=3D=
"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt; wrote:<br>
&gt; What I understand is that the browser should bind on 0.0.0.0 (or ::0)<=
br>
&gt; and send the STUN/TURN request, </p>
<p dir=3D"ltr">That works most of the time, but it is a cheat. What Chrome =
and Firefox do - bind to reach local interface separately - is the only rel=
iable way to do this. If you bind to 0.0.0.0, you can&#39;t handle multiple=
 interfaces correctly, and that reduces the odds of completing ICE with the=
 best result.</p>

--001a11490eb81fac35051b1aba31--


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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Sat, 18 Jul 2015 01:47:47 +0200
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2015-07-18 1:41 GMT+02:00 Martin Thomson <martin.thomson@gmail.com>:
>
> On Jul 17, 2015 4:35 PM, "I=C3=B1aki Baz Castillo" <ibc@aliax.net> wrote:
>> What I understand is that the browser should bind on 0.0.0.0 (or ::0)
>> and send the STUN/TURN request,
>
> That works most of the time, but it is a cheat. What Chrome and Firefox d=
o -
> bind to reach local interface separately - is the only reliable way to do
> this. If you bind to 0.0.0.0, you can't handle multiple interfaces
> correctly, and that reduces the odds of completing ICE with the best resu=
lt.

If you bind to 0.0.0.0 the OS will find out the proper interface to
send the STUN/TURN request. The browser should then check the chosen
source IP to match the associated local candidate.

>  If you bind to 0.0.0.0, you can't handle multiple interfaces
> correctly, and that reduces the odds of completing ICE with the best resu=
lt.

Any real usecase in which that could be true? Why should the app
override the OS routing table?

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


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On Jul 17, 2015 4:48 PM, "I=C3=B1aki Baz Castillo" <ibc@aliax.net> wrote:
> If you bind to 0.0.0.0 the OS will find out the proper interface to
> send the STUN/TURN request.

No, that's the point, you need control too get this right.

> The browser should then check the chosen
> source IP to match the associated local candidate.

That doesn't always work either. The source address of the packet you send
isn't always reported correctly (not does it necessarily go out the
interface you choose, but that is another problem).

> >  If you bind to 0.0.0.0, you can't handle multiple interfaces
> > correctly, and that reduces the odds of completing ICE with the best
result.
>
> Any real usecase in which that could be true? Why should the app
> override the OS routing table?

Which use case? ICE

--001a11490eb87b5473051b1af16f
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<p dir=3D"ltr"><br>
On Jul 17, 2015 4:48 PM, &quot;I=C3=B1aki Baz Castillo&quot; &lt;<a href=3D=
"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt; wrote:<br>
&gt; If you bind to 0.0.0.0 the OS will find out the proper interface to<br=
>
&gt; send the STUN/TURN request.</p>
<p dir=3D"ltr">No, that&#39;s the point, you need control too get this righ=
t.</p>
<p dir=3D"ltr">&gt; The browser should then check the chosen<br>
&gt; source IP to match the associated local candidate.</p>
<p dir=3D"ltr">That doesn&#39;t always work either. The source address of t=
he packet you send isn&#39;t always reported correctly (not does it necessa=
rily go out the interface you choose, but that is another problem).</p>
<p dir=3D"ltr">&gt; &gt;=C2=A0 If you bind to 0.0.0.0, you can&#39;t handle=
 multiple interfaces<br>
&gt; &gt; correctly, and that reduces the odds of completing ICE with the b=
est result.<br>
&gt;<br>
&gt; Any real usecase in which that could be true? Why should the app<br>
&gt; override the OS routing table?</p>
<p dir=3D"ltr">Which use case? ICE</p>

--001a11490eb87b5473051b1af16f--


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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Sat, 18 Jul 2015 02:00:45 +0200
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Subject: Re: [rtcweb] Please require user consent for data channels
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2015-07-18 1:56 GMT+02:00 Martin Thomson <martin.thomson@gmail.com>:
>> The browser should then check the chosen
>> source IP to match the associated local candidate.
>
> That doesn't always work either. The source address of the packet you sen=
d
> isn't always reported correctly (not does it necessarily go out the
> interface you choose, but that is another problem).

The point is that you don't even choose the interface. The OS will do for y=
ou.


>> >  If you bind to 0.0.0.0, you can't handle multiple interfaces
>> > correctly, and that reduces the odds of completing ICE with the best
>> > result.
>>
>> Any real usecase in which that could be true? Why should the app
>> override the OS routing table?
>
> Which use case? ICE

I understand we are here discussing how ICE is supposed to work. What
I wonder is: in which case it is useful to send packets to the
STUN/TURN server on all the interfaces instead on relying on the
interface the OS would choose to reach that server?





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


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From: Justin Uberti <juberti@google.com>
Date: Fri, 17 Jul 2015 17:19:38 -0700
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--089e01183ebcbcf6d0051b1b431a
Content-Type: text/plain; charset=UTF-8

On Fri, Jul 17, 2015 at 4:35 PM, Sergio Garcia Murillo <
sergio.garcia.murillo@gmail.com> wrote:

> On 18/07/2015 1:09, Justin Uberti wrote:
>
>>     Next, the agent uses STUN or TURN to obtain additional candidates.
>>     These come in two flavors: translated addresses on the public side of
>>     a NAT (SERVER REFLEXIVE CANDIDATES) and addresses on TURN servers
>>     (RELAYED CANDIDATES).  When TURN servers are utilized, both types of
>>     candidates are obtained from the TURN server.  If only STUN servers
>>     are utilized, only server reflexive candidates are obtained from
>>     them.  The relationship of these candidates to the host candidate is
>>     shown in Figure 2.  In this figure, both types of candidates are
>>     discovered using TURN.  In the figure, the notation X:x means IP
>>     address X and UDP port x.
>>
>> In this one is talking about getting server reflexive and relay
>> candidates, but in no way I see that the stun server should be contacted by
>> all the interfaces or overriding default routing from OS.
>>
>> It is pretty clear from the diagram on page 10 that for each host
>> candidate X:x, the agent will gather a srflx candidate X1':x1' and relay
>> candidate Y:y. This is also spelled out in the text underneath the diagram:
>>
>>    When the agent sends the TURN Allocate request from IP address and
>>    port X:x, the NAT (assuming there is one) will create a binding
>>    X1':x1', mapping this server reflexive candidate to the host
>>    candidate X:x.
>>
>> Ergo, one tries to gather srflx and relay candidates for each host
>> candidate X:x.
>>
>
> Agree, for each HOST candidate, it should send a STUN request to the turn
> server from that IP:port. But shouldn't the VPN configuration prevent the
> non-VPN-host-candidate STUN request to be sent via the non-VPN-interface?
> (i.e. applying default route based on dest?)
>

Replying to Sergio:

Typically with a VPN, the non-VPN interface is locked down so it only has a
single route: to the VPN server. However, split tunnel VPNs allow the
non-VPN interface to route to other addresses (possibly, for performance
reasons), so in this case, the non-VPN-host-candidate STUN request will be
sent successfully from the non-VPN interface.

Sometimes this will be desirable: imagine someone working from home who
wants to connect to their corp VPN, but wants to have their video
conferencing traffic not go through the VPN. Sometimes this will not be
desirable, i.e. when the VPN is being used for privacy. This makes picking
a perfect default setting difficult, although we have some ideas that we
are exploring.

--089e01183ebcbcf6d0051b1b431a
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Fri, Jul 17, 2015 at 4:35 PM, Sergio Garcia Murillo <span dir=3D"ltr=
">&lt;<a href=3D"mailto:sergio.garcia.murillo@gmail.com" target=3D"_blank">=
sergio.garcia.murillo@gmail.com</a>&gt;</span> wrote:<br><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex"><span class=3D"">On 18/07/2015 1:09, Justin Uberti wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
=C2=A0 =C2=A0 Next, the agent uses STUN or TURN to obtain additional candid=
ates.<br>
=C2=A0 =C2=A0 These come in two flavors: translated addresses on the public=
 side of<br>
=C2=A0 =C2=A0 a NAT (SERVER REFLEXIVE CANDIDATES) and addresses on TURN ser=
vers<br>
=C2=A0 =C2=A0 (RELAYED CANDIDATES).=C2=A0 When TURN servers are utilized, b=
oth types of<br>
=C2=A0 =C2=A0 candidates are obtained from the TURN server.=C2=A0 If only S=
TUN servers<br>
=C2=A0 =C2=A0 are utilized, only server reflexive candidates are obtained f=
rom<br>
=C2=A0 =C2=A0 them.=C2=A0 The relationship of these candidates to the host =
candidate is<br>
=C2=A0 =C2=A0 shown in Figure 2.=C2=A0 In this figure, both types of candid=
ates are<br>
=C2=A0 =C2=A0 discovered using TURN.=C2=A0 In the figure, the notation X:x =
means IP<br>
=C2=A0 =C2=A0 address X and UDP port x.<br>
<br>
In this one is talking about getting server reflexive and relay candidates,=
 but in no way I see that the stun server should be contacted by all the in=
terfaces or overriding default routing from OS.<br>
<br>
It is pretty clear from the diagram on page 10 that for each host candidate=
 X:x, the agent will gather a srflx candidate X1&#39;:x1&#39; and relay can=
didate Y:y. This is also spelled out in the text underneath the diagram:<br=
>
<br>
=C2=A0 =C2=A0When the agent sends the TURN Allocate request from IP address=
 and<br>
=C2=A0 =C2=A0port X:x, the NAT (assuming there is one) will create a bindin=
g<br>
=C2=A0 =C2=A0X1&#39;:x1&#39;, mapping this server reflexive candidate to th=
e host<br>
=C2=A0 =C2=A0candidate X:x.<br>
<br>
Ergo, one tries to gather srflx and relay candidates for each host candidat=
e X:x.<br>
</blockquote>
<br></span>
Agree, for each HOST candidate, it should send a STUN request to the turn s=
erver from that IP:port. But shouldn&#39;t the VPN configuration prevent th=
e non-VPN-host-candidate STUN request to be sent via the non-VPN-interface?=
 (i.e. applying default route based on dest?)<br></blockquote><div><br></di=
v><div>Replying to Sergio:</div><div><br></div><div>Typically with a VPN, t=
he non-VPN interface is locked down so it only has a single route: to the V=
PN server. However, split tunnel VPNs allow the non-VPN interface to route =
to other addresses (possibly, for performance reasons), so in this case, th=
e non-VPN-host-candidate STUN request will be sent successfully from the no=
n-VPN interface.</div><div><br></div><div>Sometimes this will be desirable:=
 imagine someone working from home who wants to connect to their corp VPN, =
but wants to have their video conferencing traffic not go through the VPN. =
Sometimes this will not be desirable, i.e. when the VPN is being used for p=
rivacy. This makes picking a perfect default setting difficult, although we=
 have some ideas that we are exploring.</div></div></div></div>

--089e01183ebcbcf6d0051b1b431a--


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From: Martin Thomson <martin.thomson@gmail.com>
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Subject: Re: [rtcweb] Please require user consent for data channels
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On Jul 17, 2015 5:01 PM, "I=C3=B1aki Baz Castillo" <ibc@aliax.net> wrote:
>
> The point is that you don't even choose the interface. The OS will do for
you.

The OS can - and frequently does - get that wrong. The default route can
fail when another might succeed.

You can't allow that to happen if you care about connecting successfully.

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<p dir=3D"ltr"><br>
On Jul 17, 2015 5:01 PM, &quot;I=C3=B1aki Baz Castillo&quot; &lt;<a href=3D=
"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt; wrote:<br>
&gt;<br>
&gt; The point is that you don&#39;t even choose the interface. The OS will=
 do for you.</p>
<p dir=3D"ltr">The OS can - and frequently does - get that wrong. The defau=
lt route can fail when another might succeed.</p>
<p dir=3D"ltr">You can&#39;t allow that to happen if you care about connect=
ing successfully.</p>

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Subject: Re: [rtcweb] Please require user consent for data channels
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On 07/17/2015 08:22 PM, Daniel Roesler wrote:
> In fact, is WebRTC the only
> browser standard (i.e. not flash) that can fire off a DNS request
> that's not (at least at first) for an HTTP address[1]?

There's also the ftp: URL. And the gopher: URL. And the telnet: URL, if
anyone still remembers that one. In fact, every single URL type that has
an embedded hostname fires off a DNS query.

There is no such thing as a HTTP address. There are only A and AAAA recor=
ds.
And ws: URLs don't have to point to a server that serves HTTP.

So I'm not sure what the question is really asking, but for all the
interpretations I can come up with that make sense to me, the answer is
"no".

--=20
Surveillance is pervasive. Go Dark.



From nobody Sat Jul 18 01:37:08 2015
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To: Justin Uberti <juberti@google.com>
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From: Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com>
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On 18/07/2015 2:19, Justin Uberti wrote:
>
> On Fri, Jul 17, 2015 at 4:35 PM, Sergio Garcia Murillo 
> <sergio.garcia.murillo@gmail.com 
> <mailto:sergio.garcia.murillo@gmail.com>> wrote:
>
>     On 18/07/2015 1:09, Justin Uberti wrote:
>
>     Agree, for each HOST candidate, it should send a STUN request to
>     the turn server from that IP:port. But shouldn't the VPN
>     configuration prevent the non-VPN-host-candidate STUN request to
>     be sent via the non-VPN-interface? (i.e. applying default route
>     based on dest?)
>
>
> Replying to Sergio:
>
> Typically with a VPN, the non-VPN interface is locked down so it only 
> has a single route: to the VPN server. However, split tunnel VPNs 
> allow the non-VPN interface to route to other addresses (possibly, for 
> performance reasons), so in this case, the non-VPN-host-candidate STUN 
> request will be sent successfully from the non-VPN interface.
>
> Sometimes this will be desirable: imagine someone working from home 
> who wants to connect to their corp VPN, but wants to have their video 
> conferencing traffic not go through the VPN. Sometimes this will not 
> be desirable, i.e. when the VPN is being used for privacy. This makes 
> picking a perfect default setting difficult, although we have some 
> ideas that we are exploring.

Agree, but IMHO in the case of case of split tunnels, shouldn't that 
issue be solved better if the VPN is configured with the only the routes 
to the corporate system via the VPN and leave the default route for the 
non-VPN-interface for all the other traffic?

You know that I am all for fixing connectivity issues in corp networks 
(my business depends on that!) but in this case I think the privacy of 
non-WebRTC users that has a VPN configured correctly should be more 
important than solving the issues of someone willing to do WebRTC but 
not doing enough to make it work.

Could you share the ideas in the list? As Firefox and Edge might have 
the same behavior (and issues) I think it could be interesting if we can 
get to a good common solution and add it to the security draft (if you 
have not done it already ;)

Best regards
Sergio

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<html>
  <head>
    <meta content="text/html; charset=utf-8" http-equiv="Content-Type">
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  <body bgcolor="#FFFFFF" text="#000000">
    <div class="moz-cite-prefix">On 18/07/2015 2:19, Justin Uberti
      wrote:<br>
    </div>
    <blockquote
cite="mid:CAOJ7v-1cB47793yH+-mWrdU9rNqX=+HadZpbRf3PezXspqLm1A@mail.gmail.com"
      type="cite">
      <div dir="ltr"><br>
        <div class="gmail_extra">On Fri, Jul 17, 2015 at 4:35 PM, Sergio
          Garcia Murillo <span dir="ltr">&lt;<a moz-do-not-send="true"
              href="mailto:sergio.garcia.murillo@gmail.com"
              target="_blank">sergio.garcia.murillo@gmail.com</a>&gt;</span>
          wrote:<br>
          <div class="gmail_quote"><br>
            <blockquote class="gmail_quote" style="margin:0 0 0
              .8ex;border-left:1px #ccc solid;padding-left:1ex"><span
                class="">On 18/07/2015 1:09, Justin Uberti wrote:<br>
                <br>
              </span>
              Agree, for each HOST candidate, it should send a STUN
              request to the turn server from that IP:port. But
              shouldn't the VPN configuration prevent the
              non-VPN-host-candidate STUN request to be sent via the
              non-VPN-interface? (i.e. applying default route based on
              dest?)<br>
            </blockquote>
            <div><br>
            </div>
            <div>Replying to Sergio:</div>
            <div><br>
            </div>
            <div>Typically with a VPN, the non-VPN interface is locked
              down so it only has a single route: to the VPN server.
              However, split tunnel VPNs allow the non-VPN interface to
              route to other addresses (possibly, for performance
              reasons), so in this case, the non-VPN-host-candidate STUN
              request will be sent successfully from the non-VPN
              interface.</div>
            <div><br>
            </div>
            <div>Sometimes this will be desirable: imagine someone
              working from home who wants to connect to their corp VPN,
              but wants to have their video conferencing traffic not go
              through the VPN. Sometimes this will not be desirable,
              i.e. when the VPN is being used for privacy. This makes
              picking a perfect default setting difficult, although we
              have some ideas that we are exploring.</div>
          </div>
        </div>
      </div>
    </blockquote>
    <br>
    Agree, but IMHO in the case of case of split tunnels, shouldn't that
    issue be solved better if the VPN is configured with the only the
    routes to the corporate system via the VPN and leave the default
    route for the non-VPN-interface for all the other traffic?<br>
    <br>
    You know that I am all for fixing connectivity issues in corp
    networks (my business depends on that!) but in this case I think the
    privacy of non-WebRTC users that has a VPN configured correctly
    should be more important than solving the issues of someone willing
    to do WebRTC but not doing enough to make it work. <br>
    <br>
    Could you share the ideas in the list? As Firefox and Edge might
    have the same behavior (and issues) I think it could be interesting
    if we can get to a good common solution and add it to the security
    draft (if you have not done it already ;)<br>
    <br>
    Best regards<br>
    Sergio<br>
  </body>
</html>

--------------090907020106050703060103--


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From: "Black, David" <david.black@emc.com>
To: Harald Alvestrand <harald@alvestrand.no>, "tsvwg@ietf.org" <tsvwg@ietf.org>
Thread-Topic: [tsvwg] draft-ietf-tsvwg-rtcweb-qos-04: How many DSCPs?
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Cc: "Cullen Jennings \(fluffy\)" <fluffy@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] [tsvwg] draft-ietf-tsvwg-rtcweb-qos-04: How many DSCPs?
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Harald,
[+rtcweb list]

> This may or may not help, but should be considered in the discussion on
> Wednesday.

That should help, although I wonder about larger API concerns - my initial
take is that the first JavaScript application API call that sends data
traffic is likely to set the DSCP for the data channel for all data traffic
for the RTCWEB session (or that DSCP will be a browser implementation
constant, in which case, only DF appears to be defensible in full generalit=
y,
IMHO), especially as:

> RTCWEB considered the option of supporting multiple SCTP channels, then
> decided not to pursue that option - it just got a bit too messy.

In contrast, the DSCP for audio/video traffic can vary within an RTCWEB ses=
sion.

Meanwhile, on multiple SCTP channels, I've heard *exactly* the opposite fro=
m
Cullen.  I've already "volunteered" to show up in RTCWEB's second session o=
n
Friday to talk about the QoS draft (Paul, Cullen and I have talked - we thi=
nk
we know what has to be done, and hope to have draft text by Thu [and TSVWG
currently has the rtcweb-qos draft on our Thu agenda, not Wed]).

Could you and/or Cullen make sure that Diffserv/DSCP QoS in general and
this multiple SCTP channel concern get onto RTCWEB's Friday agenda when
that agenda gets bashed, please, as I really should be in TAPS during
RTCWEB's Thu session?

Thanks,
--David

> -----Original Message-----
> From: Harald Alvestrand [mailto:harald@alvestrand.no]
> Sent: Thursday, July 16, 2015 9:34 AM
> To: Black, David; tsvwg@ietf.org
> Subject: Re: [tsvwg] draft-ietf-tsvwg-rtcweb-qos-04: How many DSCPs?
>=20
> Den 07. juli 2015 20:10, skrev Black, David:
> > - For data, the WebRTC QoS draft uses four different sets of DSCPs
> > 	across which there are no constraints on reordering:
> >
> >    |            Data           |  CS1  |  DF  | AF1x (10,  | AF2x (18, =
|
> >    |                           |  (8)  | (0)  |  12, 14)   |  20, 22)  =
|
> >
> > 	This is asking for trouble if all the data involved flows over a
> > 	single SCTP association ... which appears to be exactly what WebRTC
> > 	is going to do:
> >
> > 	https://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-13#section-=
6
> >
> > 	Something is definitely wrong here, as the DART draft clearly advises
> > 	against more than one DSCP per SCTP association (from Section 6):
>=20
> David,
>=20
> draft-ietf-rtcweb-transport put in this language after discussing
> reordering:
>=20
>=20
>    Implementations SHOULD attempt to set QoS on the packets sent,
>    according to the guidelines in [I-D.ietf-tsvwg-rtcweb-qos].
>=20
> .....
>=20
>    All packets carrying data from the SCTP association supporting the
>    data channels MUST use a single DSCP code point.
>=20
>    All packets on one TCP connection, no matter what it carries, MUST
>    use a single DSCP code point.
>=20
>=20
> What codepoint to pick is a good question (the one reflecting the
> highest priority channel currently being sent is an obvious, but perhaps
> not optimal choice).
>=20
> RTCWEB considered the option of supporting multiple SCTP channels, then
> decided not to pursue that option - it just got a bit too messy.
>=20
> This may or may not help, but should be considered in the discussion on
> Wednesday.
>=20
> Harald


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From: Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com>
Message-ID: <55AA15AF.40908@gmail.com>
Date: Sat, 18 Jul 2015 11:00:31 +0200
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Subject: Re: [rtcweb] Please require user consent for data channels
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--------------030001080508050405060004
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On 18/07/2015 10:37, Sergio Garcia Murillo wrote:
> On 18/07/2015 2:19, Justin Uberti wrote:
>>
>> On Fri, Jul 17, 2015 at 4:35 PM, Sergio Garcia Murillo 
>> <sergio.garcia.murillo@gmail.com 
>> <mailto:sergio.garcia.murillo@gmail.com>> wrote:
>>
>>     On 18/07/2015 1:09, Justin Uberti wrote:
>>
>>     Agree, for each HOST candidate, it should send a STUN request to
>>     the turn server from that IP:port. But shouldn't the VPN
>>     configuration prevent the non-VPN-host-candidate STUN request to
>>     be sent via the non-VPN-interface? (i.e. applying default route
>>     based on dest?)
>>
>>
>> Replying to Sergio:
>>
>> Typically with a VPN, the non-VPN interface is locked down so it only 
>> has a single route: to the VPN server. However, split tunnel VPNs 
>> allow the non-VPN interface to route to other addresses (possibly, 
>> for performance reasons), so in this case, the non-VPN-host-candidate 
>> STUN request will be sent successfully from the non-VPN interface.
>>
>> Sometimes this will be desirable: imagine someone working from home 
>> who wants to connect to their corp VPN, but wants to have their video 
>> conferencing traffic not go through the VPN. Sometimes this will not 
>> be desirable, i.e. when the VPN is being used for privacy. This makes 
>> picking a perfect default setting difficult, although we have some 
>> ideas that we are exploring.
>
> Agree, but IMHO in the case of case of split tunnels, shouldn't that 
> issue be solved better if the VPN is configured with the only the 
> routes to the corporate system via the VPN and leave the default route 
> for the non-VPN-interface for all the other traffic?
>
> You know that I am all for fixing connectivity issues in corp networks 
> (my business depends on that!) but in this case I think the privacy of 
> non-WebRTC users that has a VPN configured correctly should be more 
> important than solving the issues of someone willing to do WebRTC but 
> not doing enough to make it work.
>
> Could you share the ideas in the list? As Firefox and Edge might have 
> the same behavior (and issues) I think it could be interesting if we 
> can get to a good common solution and add it to the security draft (if 
> you have not done it already ;)
>

BTW, shouldn't this conversation be crossposted/forwarded to the TRAM 
list as well?

Best regards
Sergio

--------------030001080508050405060004
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    <div class="moz-cite-prefix">On 18/07/2015 10:37, Sergio Garcia
      Murillo wrote:<br>
    </div>
    <blockquote cite="mid:55AA1039.7070403@gmail.com" type="cite">
      <meta content="text/html; charset=utf-8" http-equiv="Content-Type">
      <div class="moz-cite-prefix">On 18/07/2015 2:19, Justin Uberti
        wrote:<br>
      </div>
      <blockquote
cite="mid:CAOJ7v-1cB47793yH+-mWrdU9rNqX=+HadZpbRf3PezXspqLm1A@mail.gmail.com"
        type="cite">
        <div dir="ltr"><br>
          <div class="gmail_extra">On Fri, Jul 17, 2015 at 4:35 PM,
            Sergio Garcia Murillo <span dir="ltr">&lt;<a
                moz-do-not-send="true"
                href="mailto:sergio.garcia.murillo@gmail.com"
                target="_blank"><a class="moz-txt-link-abbreviated" href="mailto:sergio.garcia.murillo@gmail.com">sergio.garcia.murillo@gmail.com</a></a>&gt;</span>
            wrote:<br>
            <div class="gmail_quote"><br>
              <blockquote class="gmail_quote" style="margin:0 0 0
                .8ex;border-left:1px #ccc solid;padding-left:1ex"><span
                  class="">On 18/07/2015 1:09, Justin Uberti wrote:<br>
                  <br>
                </span> Agree, for each HOST candidate, it should send a
                STUN request to the turn server from that IP:port. But
                shouldn't the VPN configuration prevent the
                non-VPN-host-candidate STUN request to be sent via the
                non-VPN-interface? (i.e. applying default route based on
                dest?)<br>
              </blockquote>
              <div><br>
              </div>
              <div>Replying to Sergio:</div>
              <div><br>
              </div>
              <div>Typically with a VPN, the non-VPN interface is locked
                down so it only has a single route: to the VPN server.
                However, split tunnel VPNs allow the non-VPN interface
                to route to other addresses (possibly, for performance
                reasons), so in this case, the non-VPN-host-candidate
                STUN request will be sent successfully from the non-VPN
                interface.</div>
              <div><br>
              </div>
              <div>Sometimes this will be desirable: imagine someone
                working from home who wants to connect to their corp
                VPN, but wants to have their video conferencing traffic
                not go through the VPN. Sometimes this will not be
                desirable, i.e. when the VPN is being used for privacy.
                This makes picking a perfect default setting difficult,
                although we have some ideas that we are exploring.</div>
            </div>
          </div>
        </div>
      </blockquote>
      <br>
      Agree, but IMHO in the case of case of split tunnels, shouldn't
      that issue be solved better if the VPN is configured with the only
      the routes to the corporate system via the VPN and leave the
      default route for the non-VPN-interface for all the other traffic?<br>
      <br>
      You know that I am all for fixing connectivity issues in corp
      networks (my business depends on that!) but in this case I think
      the privacy of non-WebRTC users that has a VPN configured
      correctly should be more important than solving the issues of
      someone willing to do WebRTC but not doing enough to make it work.
      <br>
      <br>
      Could you share the ideas in the list? As Firefox and Edge might
      have the same behavior (and issues) I think it could be
      interesting if we can get to a good common solution and add it to
      the security draft (if you have not done it already ;)<br>
      <br>
    </blockquote>
    <br>
    BTW, shouldn't this conversation be crossposted/forwarded to the
    TRAM list as well? <br>
    <br>
    Best regards<br>
    Sergio<br>
  </body>
</html>

--------------030001080508050405060004--


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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Sat, 18 Jul 2015 11:22:59 +0200
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Subject: Re: [rtcweb] Please require user consent for data channels
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2015-07-18 3:31 GMT+02:00 Martin Thomson <martin.thomson@gmail.com>:
> The OS can - and frequently does - get that wrong. The default route can
> fail when another might succeed.
>
> You can't allow that to happen if you care about connecting successfully.

Are we talking about ICE gathering and STUN/TURN servers? or are we
talking about p2p ICE connectivity?


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


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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Sat, 18 Jul 2015 11:31:44 +0200
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2015-07-18 2:19 GMT+02:00 Justin Uberti <juberti@google.com>:
> Typically with a VPN, the non-VPN interface is locked down so it only has=
 a
> single route: to the VPN server. However, split tunnel VPNs allow the
> non-VPN interface to route to other addresses (possibly, for performance
> reasons), so in this case, the non-VPN-host-candidate STUN request will b=
e
> sent successfully from the non-VPN interface.

Is that really the most common VNC scenario? Never seen that. I
thought that usually the VNC client adds its own network routes to
reach other users/servers within the same VNC network and leaves the
default route (and others) intact.

My worries are about ICE gathering and the timeout it causes when a
STUN request is sent to the STUN/TURN server from all the local
interfaces. Security/privacy issues (if we consider that leaking all
the host and reflexive addresses is an issue) can happen during ICE
gathering and also when connecting the remote peer.


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


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To: Martin Thomson <martin.thomson@gmail.com>
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On Fri, Jul 17, 2015 at 9:31 PM, Martin Thomson <martin.thomson@gmail.com>
wrote:

> On Jul 17, 2015 5:01 PM, "I=C3=B1aki Baz Castillo" <ibc@aliax.net> wrote:
> >
> > The point is that you don't even choose the interface. The OS will do
> for you.
>
> The OS can - and frequently does - get that wrong. The default route can
> fail when another might succeed.
>
> You can't allow that to happen if you care about connecting successfully.
>
Sending data to a particular destination over an interface that is not
supposed to be used based on the local routing is definitely something that
goes against the local policy and should be discouraged. For instance on a
mobile device, not being able to reach a particular STUN/TURN server over
WiFi, should not give the browser an automatic permission to send media
over cell data interface and generate a huge bill for the customer.
Forcibly overwriting default route is something that should need user
consent. I understand that we want the call to succeed in as many cases as
possible, but blindly assuming that the local routing policy exists for no
reason whatsoever is somewhat dangerous.

Would it make sense to use the default route when sending BIND requests to
the STUN/TURN server and ICE connectivity checks? All the host interfaces,
including non-default, should, of cause, be included as targets for remote
connectivity checks. An extra option should enable using non-default
interfaces to send ICE requests and it should only be enabled if user
explicitly agrees to it. Vast majority of users will never need it anyway.

Regards,
_____________
Roman Shpount

--001a113f911c8a0807051b2311f5
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On F=
ri, Jul 17, 2015 at 9:31 PM, Martin Thomson <span dir=3D"ltr">&lt;<a href=
=3D"mailto:martin.thomson@gmail.com" target=3D"_blank">martin.thomson@gmail=
.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"ma=
rgin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,=
204);border-left-style:solid;padding-left:1ex"><span class=3D""><p dir=3D"l=
tr">On Jul 17, 2015 5:01 PM, &quot;I=C3=B1aki Baz Castillo&quot; &lt;<a hre=
f=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.net</a>&gt; wrote:<b=
r>
&gt;<br>
&gt; The point is that you don&#39;t even choose the interface. The OS will=
 do for you.</p>
</span><p dir=3D"ltr">The OS can - and frequently does - get that wrong. Th=
e default route can fail when another might succeed.</p>
<p dir=3D"ltr">You can&#39;t allow that to happen if you care about connect=
ing successfully.</p>
</blockquote></div></div><div class=3D"gmail_extra">Sending data to a parti=
cular destination over an interface that is not supposed to be used based o=
n the local routing is definitely something that goes against the local pol=
icy and should be discouraged. For instance on a mobile device, not being a=
ble to reach a particular STUN/TURN server over WiFi, should not give the b=
rowser an automatic permission to send media over cell data interface and g=
enerate a huge bill for the customer. Forcibly overwriting default route is=
 something that should need user consent. I understand that we want the cal=
l to succeed in as many cases as possible, but blindly assuming that the lo=
cal routing policy exists for no reason whatsoever is somewhat dangerous.</=
div><div class=3D"gmail_extra"><br></div><div class=3D"gmail_extra">Would i=
t make sense to use the default route when sending BIND requests to the STU=
N/TURN server and ICE connectivity checks? All the host interfaces, includi=
ng non-default, should, of cause, be included as targets for remote connect=
ivity checks. An extra option should enable using non-default interfaces to=
 send ICE requests and it should only be enabled if user explicitly agrees =
to it. Vast majority of users will never need it anyway.</div><div class=3D=
"gmail_extra"><br></div><div class=3D"gmail_extra">Regards,</div><div class=
=3D"gmail_extra"><div><div class=3D"gmail_signature">_____________<br>Roman=
 Shpount</div></div><div><br></div></div></div>

--001a113f911c8a0807051b2311f5--


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On 07/18/2015 10:38 AM, Black, David wrote:
> Harald,
> [+rtcweb list]
>
>> This may or may not help, but should be considered in the discussion o=
n
>> Wednesday.
> That should help, although I wonder about larger API concerns - my init=
ial
> take is that the first JavaScript application API call that sends data
> traffic is likely to set the DSCP for the data channel for all data tra=
ffic
> for the RTCWEB session (or that DSCP will be a browser implementation
> constant, in which case, only DF appears to be defensible in full gener=
ality,
> IMHO),

If we can agree on and document what browsers *should* do, it is likely
that this is what they'll end up doing. The implementors are in the
room, after all.

>  especially as:
>
>> RTCWEB considered the option of supporting multiple SCTP channels, the=
n
>> decided not to pursue that option - it just got a bit too messy.
> In contrast, the DSCP for audio/video traffic can vary within an RTCWEB=
 session.

Yep. Both because tsvw-rtcweb-qos specifies multiple DSCP code points
per priority level, and because the API as currently designed permits
changing the priority of a media track.

>
> Meanwhile, on multiple SCTP channels, I've heard *exactly* the opposite=
 from
> Cullen.  I've already "volunteered" to show up in RTCWEB's second sessi=
on on
> Friday to talk about the QoS draft (Paul, Cullen and I have talked - we=
 think
> we know what has to be done, and hope to have draft text by Thu [and TS=
VWG
> currently has the rtcweb-qos draft on our Thu agenda, not Wed]).

Thanks for the fix! (was wrong in my notes)
>
> Could you and/or Cullen make sure that Diffserv/DSCP QoS in general and=

> this multiple SCTP channel concern get onto RTCWEB's Friday agenda when=

> that agenda gets bashed, please, as I really should be in TAPS during
> RTCWEB's Thu session?

Cullen's one of the chairs, I'm just a WG member; I'll try to help get
it there.

>
> Thanks,
> --David
>
>> -----Original Message-----
>> From: Harald Alvestrand [mailto:harald@alvestrand.no]
>> Sent: Thursday, July 16, 2015 9:34 AM
>> To: Black, David; tsvwg@ietf.org
>> Subject: Re: [tsvwg] draft-ietf-tsvwg-rtcweb-qos-04: How many DSCPs?
>>
>> Den 07. juli 2015 20:10, skrev Black, David:
>>> - For data, the WebRTC QoS draft uses four different sets of DSCPs
>>> 	across which there are no constraints on reordering:
>>>
>>>    |            Data           |  CS1  |  DF  | AF1x (10,  | AF2x (18=
, |
>>>    |                           |  (8)  | (0)  |  12, 14)   |  20, 22)=
  |
>>>
>>> 	This is asking for trouble if all the data involved flows over a
>>> 	single SCTP association ... which appears to be exactly what WebRTC
>>> 	is going to do:
>>>
>>> 	https://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-13#sectio=
n-6
>>>
>>> 	Something is definitely wrong here, as the DART draft clearly advise=
s
>>> 	against more than one DSCP per SCTP association (from Section 6):
>> David,
>>
>> draft-ietf-rtcweb-transport put in this language after discussing
>> reordering:
>>
>>
>>    Implementations SHOULD attempt to set QoS on the packets sent,
>>    according to the guidelines in [I-D.ietf-tsvwg-rtcweb-qos].
>>
>> .....
>>
>>    All packets carrying data from the SCTP association supporting the
>>    data channels MUST use a single DSCP code point.
>>
>>    All packets on one TCP connection, no matter what it carries, MUST
>>    use a single DSCP code point.
>>
>>
>> What codepoint to pick is a good question (the one reflecting the
>> highest priority channel currently being sent is an obvious, but perha=
ps
>> not optimal choice).
>>
>> RTCWEB considered the option of supporting multiple SCTP channels, the=
n
>> decided not to pursue that option - it just got a bit too messy.
>>
>> This may or may not help, but should be considered in the discussion o=
n
>> Wednesday.
>>
>> Harald


--=20
Surveillance is pervasive. Go Dark.



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Subject: Re: [rtcweb] Please require user consent for data channels
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2015-07-18 11:38 GMT+02:00 Roman Shpount <roman@telurix.com>:
> Sending data to a particular destination over an interface that is not
> supposed to be used based on the local routing is definitely something th=
at
> goes against the local policy and should be discouraged. For instance on =
a
> mobile device, not being able to reach a particular STUN/TURN server over
> WiFi, should not give the browser an automatic permission to send media o=
ver
> cell data interface and generate a huge bill for the customer. Forcibly
> overwriting default route is something that should need user consent. I
> understand that we want the call to succeed in as many cases as possible,
> but blindly assuming that the local routing policy exists for no reason
> whatsoever is somewhat dangerous.

+1





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


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Le 2015-07-18 06:01, IÃ±aki Baz Castillo a Ã©crit :
> 2015-07-18 11:38 GMT+02:00 Roman Shpount <roman@telurix.com>:
>> Sending data to a particular destination over an interface that is not
>> supposed to be used based on the local routing is definitely something that
>> goes against the local policy and should be discouraged. For instance on a
>> mobile device, not being able to reach a particular STUN/TURN server over
>> WiFi, should not give the browser an automatic permission to send media over
>> cell data interface and generate a huge bill for the customer. Forcibly
>> overwriting default route is something that should need user consent. I
>> understand that we want the call to succeed in as many cases as possible,
>> but blindly assuming that the local routing policy exists for no reason
>> whatsoever is somewhat dangerous.
> 
> +1

-1

Justin and Martin are right.

This is a concept that was explained during the ICE tutorial that was
presented at IETF 92.

Imagine the following multihomed host with three interfaces:

- Wi-fi interface with default route.
- Wired interface with default route.
- VPN interface with default route.

Which one should you use to talk to the STUN/TURN server? If you bind to
0.0.0.0 then the kernel *will* pick one, but only because it *has to.
All three interfaces are equivalent: they all have a default route and
can a priori be used to talk to the server.

Being multihomed means multiple interfaces with a default route. There's
no "going against the local policy" in this case when sending packets on
any interface.

The point is that in this case, all three interfaces can potentially
yield different server-reflexive candidates, which is why they must be
gathered separately. As for relayed candidates, they also need to be
gathered separately because some interfaces may be faster for talking to
the TURN server and it is useful to compare the candidates gathered on
all 3 interfaces. Also nothing guarantees that you're allowed to reach
the server server on any interface that has a default route, so you need
to try them all.

Simon


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Subject: Re: [rtcweb] Please require user consent for data channels
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Subject: Re: [rtcweb] Please require user consent for data channels
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On 07/19/2015 08:06 AM, Tirumaleswar Reddy (tireddy) wrote:
>> Being multihomed means multiple interfaces with a default route. There's no
>> > "going against the local policy" in this case when sending packets on any
>> > interface.
>> > 
>> > The point is that in this case, all three interfaces can potentially yield different
>> > server-reflexive candidates, which is why they must be gathered separately. As
>> > for relayed candidates, they also need to be gathered separately because some
>> > interfaces may be faster for talking to the TURN server and it is useful to
>> > compare the candidates gathered on all 3 interfaces. Also nothing guarantees
>> > that you're allowed to reach the server server on any interface that has a
>> > default route, so you need to try them all.
> This will change in the future with multiple provisioning domains (PVD) defined in MIF WG, new socket API will be provided to allow applications to learn about each PVD and use them separately, and as discussed in https://tools.ietf.org/html/rfc7556#section-6.3 application may not override hard set selected by the user.
Rule #53 of the IETF: Things will change in the future, including
deployment of just-completed iETF standards, but we don't know when and how.

Solutions we engineer MUST work in the current Internet.

+1 to Simon - the way systems are designed, implemented and deployed
*now*, there may be multiple interfaces with multiple default addresses,
and there's nothing intrinsically illegal, immoral or fattening about an
application accessing them.

(I have personal opinions about the APIs we use to access those
interfaces, only some of which are suitable for repeating in polite
company, but that is a completely different topic.)

-- 
Surveillance is pervasive. Go Dark.


From nobody Sun Jul 19 00:42:02 2015
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Date: Sun, 19 Jul 2015 09:41:59 +0200
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From: Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com>
To: Justin Uberti <juberti@google.com>
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Subject: Re: [rtcweb] Please require user consent for data channels
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El 18/7/2015 2:19, "Justin Uberti" <juberti@google.com> escribi=C3=B3:
>
> Sometimes this will be desirable: imagine someone working from home who
wants to connect to their corp VPN, but wants to have their video
conferencing traffic not go through the VPN. Sometimes this will not be
desirable, i.e. when the VPN is being used for privacy. This makes picking
a perfect default setting difficult, although we have some ideas that we
are exploring.

Most probably you already have thought about it, but how about adding an
interface black/white list for webrtc in privacy settings? IIRC chrome was
already skipping some hardcoded virtual interface names.

Best regards
Sergio

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<p dir=3D"ltr"><br>
El 18/7/2015 2:19, &quot;Justin Uberti&quot; &lt;<a href=3D"mailto:juberti@=
google.com">juberti@google.com</a>&gt; escribi=C3=B3:<br>
&gt;<br>
&gt; Sometimes this will be desirable: imagine someone working from home wh=
o wants to connect to their corp VPN, but wants to have their video confere=
ncing traffic not go through the VPN. Sometimes this will not be desirable,=
 i.e. when the VPN is being used for privacy. This makes picking a perfect =
default setting difficult, although we have some ideas that we are explorin=
g.<br></p>
<p dir=3D"ltr">Most probably you already have thought about it, but how abo=
ut adding an interface black/white list for webrtc in privacy settings? IIR=
C chrome was already skipping some hardcoded virtual interface names.</p>
<p dir=3D"ltr">Best regards<br>
Sergio<br>
</p>

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Subject: Re: [rtcweb] draft-ietf-rtcweb-return-00.txt; Sealed, Configuration and TURN discovery:
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Benjamin =E2=80=93 your concern is valid =E2=80=93 I thought exo-sealed =
was easy to do in the browser.

=20

Below I list a firewall configuration assisted method to address the =
TURN server and bandwidth resource consumption problem for local WebRTC =
calls when using the sealed configuration. I believe my suggestion also =
will meet all security/trust/privacy perspectives and give secure =
auto-configuration, so mobile devices can to be used with high-quality =
secure WebRTC communication =E2=80=9Ceverywhere you can surf=E2=80=9D  =
through proper use of Border TURN servers - which must be the (almost =
already achieved) purpose of this RETURN draft.

=20

After some consideration, to achieve the preferred exo-sealed =
configuration, the RETURN drafts already prescribes the best default =
configuration: =E2=80=9Cleaky=E2=80=9D ! :-)=20

Explanations inline below -->

=20

Below are also some suggestions for inclusion in the drafts:

draft-ietf-tram-turn-server-discovery-03

draft-jennings-behave-rtcweb-firewall-00

=20

/Karl

=20

PS: Regarding the TURN Origin debate, I don=E2=80=99t have a clear view, =
but am I not right in the understanding that:

* The purpose of the TURN Origin draft  was to reduce the pain of WebRTC =
service providers having to deploy network hardware (the Application =
Provided TURN server) to get their services used, by allowing the TURN =
server to serve many applications (through a third party).=20

 - The introduction of Border TURN servers releases WebRTC service =
providers of that pain (when commonly deployed) (and also put security =
back in the right place) (Until deployed commonly, yes these SPs =
unfortunately have this pain to =E2=80=9Cfool their services=E2=80=9D =
through users security devices =E2=80=93 the firewalls (something which =
the discussed 3 drafts now finally can remedy (without the use of =
Origin))

* However, the Application Provided TURN server may also be for (from =
the RETURN draft: 3 Goals, 3.2 independent path control):=20

   o  improving quality of service by tunneling packets through a =
network that is faster than the public internet,

   +  monitoring the usage of UDP services,

   +  troubleshooting and debugging problematic services,

   +  logging connection metadata for legal or auditing reasons,

   +  recording the entire contents of all connections, or

   -  providing partial IP address anonymization (as described in =
[I-D.ietf-rtcweb-security] Section 4.2.4).

(+ and =E2=80=93 indicates the conflicting impact the use Origin will =
bring for the listed purposes)

=20

The RETURN draft honors the application=E2=80=99s efforts to enforce =
media through such Application Provided TURN server (without judging the =
purpose) (by use of the Recursively Encapsulation mechanism).

=20

My advice now is just to not introduce the Origin discussion in these =
drafts where we finally have to resolve the firewall traversal problem =
with real-time traffic. We need these RFCs in place as soon as ever =
possible. I see a risk in that draft-jennings-behave-rtcweb-firewall-00 =
starts discussing Origin/SNI in an effort to make a pretty secure WebRTC =
aware firewall somewhat more secure, and think the draft can advice =
firewall vendors how to build firewalls that are fully secure and allows =
real-time traffic and should also be able to provide a quality pipe for =
the real-time traffic. Origin/SNI is not required for that. The Border =
TURN server and this RETURN-draft will make that possible.

=20

-- > Don=E2=80=99t forget the inline below -- >

=20

Fr=C3=A5n: Benjamin Schwartz [mailto:bemasc@webrtc.org]=20
Skickat: den 14 juli 2015 03:16
Till: Karl Stahl
Kopia: rtcweb@ietf.org; tram@ietf.org
=C3=84mne: Re: [rtcweb] draft-ietf-rtcweb-return-00.txt; Sealed, =
Configuration and TURN discovery:

=20

On Mon, Jul 13, 2015 at 6:19 PM, Karl Stahl <karl.stahl@intertex.se> =
wrote:

We need this draft implemented in WebRTC browsers as soon as possible, =
so we
can get good WebRTC in difficult places (for firewall traversal and for
quality). Is there any implementation e.g. in Chrome that can be tried?

Now having stated that I am positive to this draft, I have some
concerns/questions/suggestions:

1) We need more levels than sealed/leaky

It is not acceptable that WebRTC traffic between parties on a typical =
LAN is
forced through a TURN server, just because of a sealed configuration
required for good WebRTC traffic over the WAN. This consequence is =
pointed
out under "5.11.  Reusing the same TURN server", but IS a problem even =
if it
"work as expected".

Such TURN resource consumption, waste of bandwidth and possible =
performance
degradation do not happen today, and can easily be avoided.

I realize this consequence of "sealed" is required to meet some of the
requirements under "3.2.  Independent Path Control", but we simply need =
an
"exo-sealed" level, in addition to the proposed strictly sealed and =
leaky,
allowing local endpoints to connect media directly, while external =
traffic
is forced to use the TURN server (rather than STUN through the default
gateway).

=20

[BS] How do you propose to tell what is "local"?  I don't know how to =
define that procedurally.

=20

[BS] I think my main concern here is that it requires us to "lie" to the =
remote party, by sending candidates that we may then refuse to use even =
if they work.  This seems very strange.

=20

[Karl] You are right. I assumed the preferred exo-sealed behavior could =
be achieved as easily as we do it =E2=80=9Chardware wise=E2=80=9D today =
on the inside of our open test-site https://webrtc.ingate.com where we =
have a border placed TURN server paralleling our firewall/default =
gateway. Our firewall looks for the magic STUN cookie and discards all =
STUN packets from the LAN side. TURN is then automatically enforced and =
today=E2=80=99s leaky WebRTC browsers automatically get the preferred =
exo-sealed behavior.

=20

(Just not using STUN in the browser, would have the same effect but not =
fully, if there are more subnets on the LAN side. Crossing those subnets =
would then take media through the dual trip via the TURN server that I =
want to avoid.)

=20

So, let=E2=80=99s enforce the preferred exo-sealed configuration by =
configuration of the firewall/default gateway (Security from this =
perspective is then also in the right hands - the network =
admin=E2=80=99s):=20

- Today=E2=80=99s RETURN-draft=E2=80=99s leaky default configuration is =
OK (since the firewalls discarding STUN automatically enforce the =
preferred exo-sealed behavior)

- Recommend in the RETURN-draft (and in =
draft-jennings-behave-rtcweb-firewall-00)  that the network admin =
discards STUN packets from the LAN side through the default gateway =
(consent STUNs will go OK through the TURN server), by looking for the =
magic STUN cookie, by closing port 3478 or by closing all UDP (the =
restrictive enterprise firewall can simply be left as it is) to achieve =
exo-sealed leakiness.=20

- The network admin offers a Border TURN server for its network users =
(LAN side only) =E2=80=93 no need for authentication (free to use from =
LAN side being a Border TURN server configuration)=20

- The Border TURN server is offered by the network you are using =
=E2=80=93 Just use it! (should the network provider have evil =
intentions, he needs no TURN server, he can simply close or destroy =
WebRTC in the firewall if he which,  which Cullen just exemplified in =
draft-jennings-behave-rtcweb-firewall-00)

=20

As you may have noticed, I stress that NO credentials should be needed =
for using an auto discovered border TURN server. That is also the goal =
of TRAM=E2=80=99s discovery draft, but I have asked TRAM  to clarify =
that achievement. If we don=E2=80=99t fix that, we have failed to make =
WebRTC usable on mobile devices =E2=80=93 or can anyone explain where we =
would get those credentials from when we run around and have Web access =
here and there and the goal of WebRTC was to use real-time =
communication, securely and with high quality, everywhere we can surf, =
wasn=E2=80=99t it?=20

=20

If anyone cries that we NEED AUTHENTICATION, they should remember the IP =
authentication (simply being on a certain network) is the most commonly =
used =E2=80=9Cauthentication method=E2=80=9D today and protecting huge =
values (almost the whole telco industry) and also achieves privacy =
(through the use of firewalls). No individual user credentials which are =
impossible to handle must be required.

=20

I will in next email propose this new variant of the Anycast method  to =
TRAM. It should work, shouldn=E2=80=99t it?

And does it provide the claimed advantages/features? =E2=80=93 please =
check.

=20

Implicit discovery by interception of TURN allocate requests at a =
default gateway

=20

An enterprise firewall or a network server provider=E2=80=99s router at =
a default gateway can intercept a TURN allocate request and respond with =
a 300 (Try Alternate) just as described in the Anycast method above. The =
following procedures and advantages are inherited from the anycast =
method.

=20

This variant adds the following features:

- no separate deployment of discovery mechanism is required

- no ANA allocated anycast address is required (works with any public ip =
address)=20

- the default gateway router/firewall in charge of the local network =
responds with the ip address of:

(i)   a Border TURN server (paralleling the default gateway), or=20

(ii)  a TURN server behind that firewall/default gateway and opens a =
path to that TURN server =20

- if the intercepted allocate request was intended for an application =
provided TURN server, the TURN client (e.g. WebRTC browser) on behalf of =
the user, can select to:

(iii) only use the auto discovered TURN server

(iv)  honor an allocation request by really sending the real-time =
traffic to the Application Provided TURN server, using the Recursively =
Encapsulated method, unless

(v)   the TURN  server connected in (ii) already is the Application =
Provided TURN server

=20

Security wise, the right party is in charge in all cases.

A TURN server pinpointed by a firewall/router at the default gateway =
should always be trusted (no explicit authentication with credentials is =
needed) and used. (We trust the default gateway with all our traffic =
anyway, and any evil intent to disturb real-time traffic can in such =
case be implemented in a default gateway by easier means.)

=20

=20

Such "exo-sealed" configuration should be the default configuration if a
network provided TURN server is made available for good (non-evil) =
purposes.


2) Mobile devices need to work on various networks, without individual
configuration

I am not clear about the thoughts behind the "configuration" of sealed =
level
and requirements for authentication of the turn server, but if there are
methods where an enterprise network (LAN) admin can provide credentials =
and
force a certain sealed level upon the WebRTC-browser when used at that =
LAN,
or if the user has the possibility to configure manually, I am OK with =
that
this can and should OVERRIDE any default configuration or other
configuration (on that particular network).

However, for mobile devices - Don't we all happily connect our browsers =
via
WiFi here and there? - there must be a way for a network provider with a
good intent to offer an exo-sealed TURN server that automatically should =
be
used, e.g. to achieve the purpose of offering better RTC quality than is
available through the default gateway WAN path on that network.

A network provided TURN-server can easily be made available ONLY to the
network users that it is serving, so there is no need for authentication =
and
credentials to protect against illegal usage of a valuable resource, =
like it
is with the application provided turn server. Network provided TURN =
servers
are simply deployed for the network users to be use without other
authentication than being granted an IP address and a default gateway to =
use
on that particular network in itself.

The same goes for trusting the TURN server with our real-time traffic - =
If
we would trust the same network provider with the same real-time traffic
through the default gateway - we should also use the offered TURN server
without other "authentication" than that it is available on our network:
Data through the default gateway, RTC through the TURN path is why the =
TURN
server is provided!

But do we know that it is not someone with the evil purpose of stealing =
our
(encrypted!) real time traffic that is offering the TURN server (instead =
of
the network provider), which leads us to my point 3). (If we really =
think
there is a risk of and danger with such attacks...)


3) Is it now good enough to simply use the auto discovered TURN server? =
Hope
so!

There has been some recent improvements to
draft-ietf-tram-turn-server-discovery-03 and hopefully that is =
sufficient
for assuming that the auto-discovered TURN server is offered by our =
network
provider (a service provider or the enterprise itself) that we trust =
anyway,
and the TURN server then simply shall be used and considered exo-sealed.
(The current RETURN draft says a non configured TURN server should be
leaky.)

=20

The security considerations in that document suggest that not all =
discovery mechanisms have equal additional exposure to local network =
interference.  I would appreciate clearer guidance on that front.  It =
seems to me that DHCP is the only listed mechanism that creates minimal =
additional exposure to network interference.

=20

--- I am addressing TRAM about this and believe that the variant of the =
Anycast method I suggest to be added, will fully resolve all security =
concerns. I am also suggesting that it is up to this RTCWEB draft to =
select from TRAM=E2=80=99s methods and advices for the WebRTC =
application and its use cases.

=20

--- See my deployment concerns regarding DHCP put in front of TRAM. I =
think that the Anycast method with a short TTL-limitation is good enough =
from the privacy aspect, when used to find Border TURN server.

=20

--- Even though the new variant I propose is more secure and even easier =
to deploy, I suggest that this RETURN draft mandates the use of the both =
the Anycast method and its new variant. The reason is that already =
today, all routers/firewalls at a default gateways can add the anycast =
route required for discovery, while intercepting the TURN  allocate =
request and responding with the Try Alternate server probably requires a =
software upgrade of installed  firewalls and routers with default =
gateways (a firewall in front of the default gateway can also do the =
interception and response).=20

=20

Some concerns regarding the DISCOVERY draft and whether these things =
should
be specified in the RETURN document:

- Is the above the case with all discovery methods proposed? Or shall =
the
RETURN document make one method mandatory to implement?

- Can that method then be the Anycast method (which by far seems to be =
the
most convenient to deploy and to implement and should work for all =
network
types and in all devices operating systems). I think what is said under
"9.3.  Anycast" under "9.  Security Considerations" is sufficient for
trusting such discovered TURN server. What should then go into the =
RETURN
draft - or should the DISCOVERY draft be more specific?

I don't fully understand the DISCOVERY draft's: "... three discovery
mechanisms. The reason for providing multiple mechanisms is to maximize =
the
opportunity for discovery,"

- Can we really put the burden upon all WebRTC devices to implement all
methods and I doubt network providers will deploy more than one method. =
That
seems risking rather than maximizing discovery...


It is high time to get this done - can we?

/Karl



-----Ursprungligt meddelande-----
Fr=C3=A5n: rtcweb [mailto:rtcweb-bounces@ietf.org] F=C3=B6r =
internet-drafts@ietf.org
Skickat: den 14 maj 2015 00:29
Till: i-d-announce@ietf.org
Kopia: rtcweb@ietf.org
=C3=84mne: [rtcweb] I-D Action: draft-ietf-rtcweb-return-00.txt


A New Internet-Draft is available from the on-line Internet-Drafts
directories.
 This draft is a work item of the Real-Time Communication in =
WEB-browsers
Working Group of the IETF.

        Title           : Recursively Encapsulated TURN (RETURN) for
Connectivity and Privacy in WebRTC
        Authors         : Benjamin M. Schwartz
                          Justin Uberti
        Filename        : draft-ietf-rtcweb-return-00.txt
        Pages           : 18
        Date            : 2015-05-13

Abstract:
   In the context of WebRTC, the concept of a local TURN proxy has been
   suggested, but not reviewed in detail.  WebRTC applications are
   already using TURN to enhance connectivity and privacy.  This
   document explains how local TURN proxies and WebRTC applications can
   work together.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-return/

There's also a htmlized version available at:
https://tools.ietf.org/html/draft-ietf-rtcweb-return-00


Please note that it may take a couple of minutes from the time of =
submission
until the htmlized version and diff are available at tools.ietf.org.

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

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

=20


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</o:shapelayout></xml><![endif]--></head><body lang=3DSV link=3Dblue =
vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Be=
njamin =E2=80=93 your concern is valid =E2=80=93 I thought exo-sealed =
was easy to do in the browser.<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Be=
low I list a firewall configuration assisted method to address the TURN =
server and bandwidth resource consumption problem for local WebRTC calls =
when using the sealed configuration. I believe my suggestion also will =
meet all security/trust/privacy perspectives and give secure =
auto-configuration, so mobile devices can to be used with high-quality =
secure WebRTC communication =E2=80=9Ceverywhere you can surf=E2=80=9D =
=C2=A0through proper use of Border TURN servers - which must be the =
(almost already achieved) purpose of this RETURN =
draft.<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Af=
ter some consideration, to achieve the preferred exo-sealed =
configuration, the RETURN drafts already prescribes the best default =
configuration: =E2=80=9Cleaky=E2=80=9D ! :-) <o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Ex=
planations inline below --&gt;<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Be=
low are also some suggestions for inclusion in the =
drafts:<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>draft-ietf-tr=
am-turn-server-discovery-03<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>draft-jenning=
s-behave-rtcweb-firewall-00<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>/K=
arl<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>PS=
: Regarding the TURN Origin debate, I don=E2=80=99t have a clear view, =
but am I not right in the understanding that:<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>* =
The purpose of the TURN Origin draft =C2=A0was to reduce the pain of =
WebRTC service providers having to deploy network hardware (the =
Application Provided TURN server) to get their services used, by =
allowing the TURN server to serve many applications (through a third =
party). <o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>=C2=
=A0- The introduction of Border TURN servers releases WebRTC service =
providers of that pain (when commonly deployed) (and also put security =
back in the right place) (Until deployed commonly, yes these SPs =
unfortunately have this pain to =E2=80=9Cfool their services=E2=80=9D =
through users security devices =E2=80=93 the firewalls (something which =
the discussed 3 drafts now finally can remedy (without the use of =
Origin))<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>* =
However, the Application Provided TURN server may also be for (from the =
RETURN draft: 3 Goals, 3.2 independent path control): =
<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>=C2=
=A0=C2=A0=C2=A0o=C2=A0 improving quality of service by tunneling packets =
through a network that is faster than the public =
internet,<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>=C2=
=A0=C2=A0 +=C2=A0 monitoring the usage of UDP =
services,<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>=C2=
=A0=C2=A0 +=C2=A0 troubleshooting and debugging problematic =
services,<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>=C2=
=A0=C2=A0 +=C2=A0 logging connection metadata for legal or auditing =
reasons,<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>=C2=
=A0=C2=A0 +=C2=A0 recording the entire contents of all connections, =
or<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>=C2=
=A0=C2=A0 -=C2=A0 providing partial IP address anonymization (as =
described in [I-D.ietf-rtcweb-security] Section =
4.2.4).<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(+=
 and =E2=80=93 indicates the conflicting impact the use Origin will =
bring for the listed purposes)<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Th=
e RETURN draft honors the application=E2=80=99s efforts to enforce media =
through such Application Provided TURN server (without judging the =
purpose) (by use of the Recursively Encapsulation =
mechanism).<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>My=
 advice now is just to not introduce the Origin discussion in these =
drafts where we finally have to resolve the firewall traversal problem =
with real-time traffic. We need these RFCs in place as soon as ever =
possible. I see a risk in that </span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>draft-jenning=
s-behave-rtcweb-firewall-00 s<span style=3D'color:blue'>tarts discussing =
Origin/SNI in an effort to make a pretty secure WebRTC aware firewall =
somewhat more secure, and think the draft can advice firewall vendors =
how to build firewalls that are fully secure and allows real-time =
traffic and should also be able to provide a quality pipe for the =
real-time traffic. Origin/SNI is not required for that. The Border TURN =
server and this RETURN-draft will make that =
possible.<o:p></o:p></span></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>--=
 &gt; Don=E2=80=99t forget the inline below -- =
&gt;<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><div style=3D'border:none;border-top:solid =
#B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm'><p class=3DMsoNormal><b><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Fr=C3=A5n:</=
span></b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Benjamin =
Schwartz [mailto:bemasc@webrtc.org] <br><b>Skickat:</b> den 14 juli 2015 =
03:16<br><b>Till:</b> Karl Stahl<br><b>Kopia:</b> rtcweb@ietf.org; =
tram@ietf.org<br><b>=C3=84mne:</b> Re: [rtcweb] =
draft-ietf-rtcweb-return-00.txt; Sealed, Configuration and TURN =
discovery:<o:p></o:p></span></p></div><p class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><div><div><div><p =
class=3DMsoNormal>On Mon, Jul 13, 2015 at 6:19 PM, Karl Stahl &lt;<a =
href=3D"mailto:karl.stahl@intertex.se" =
target=3D"_blank">karl.stahl@intertex.se</a>&gt; wrote:<o:p></o:p></p><p =
class=3DMsoNormal>We need this draft implemented in WebRTC browsers as =
soon as possible, so we<br>can get good WebRTC in difficult places (for =
firewall traversal and for<br>quality). Is there any implementation e.g. =
in Chrome that can be tried?<br><br>Now having stated that I am positive =
to this draft, I have some<br>concerns/questions/suggestions:<br><br>1) =
We need more levels than sealed/leaky<br><br>It is not acceptable that =
WebRTC traffic between parties on a typical LAN is<br>forced through a =
TURN server, just because of a sealed configuration<br>required for good =
WebRTC traffic over the WAN. This consequence is pointed<br>out under =
&quot;5.11.&nbsp; Reusing the same TURN server&quot;, but IS a problem =
even if it<br>&quot;work as expected&quot;.<br><br>Such TURN resource =
consumption, waste of bandwidth and possible performance<br>degradation =
do not happen today, and can easily be avoided.<br><br>I realize this =
consequence of &quot;sealed&quot; is required to meet some of =
the<br>requirements under &quot;3.2.&nbsp; Independent Path =
Control&quot;, but we simply need an<br>&quot;exo-sealed&quot; level, in =
addition to the proposed strictly sealed and leaky,<br>allowing local =
endpoints to connect media directly, while external traffic<br>is forced =
to use the TURN server (rather than STUN through the =
default<br>gateway).<o:p></o:p></p><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal><span lang=3DEN-US style=3D'color:blue'>[BS] =
</span><span lang=3DEN-US>How do you propose to tell what is =
&quot;local&quot;?&nbsp; </span>I don't know how to define that =
procedurally.<o:p></o:p></p></div><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal><span lang=3DEN-US style=3D'color:blue'>[BS] =
</span><span lang=3DEN-US>I think my main concern here is that it =
requires us to &quot;lie&quot; to the remote party, by sending =
candidates that we may then refuse to use even if they work.&nbsp; =
</span>This seems very strange.<o:p></o:p></p></div><div><p =
class=3DMsoNormal><span =
style=3D'color:blue'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>[K=
arl] You are right. I assumed the preferred exo-sealed behavior could be =
achieved as easily as we do it =E2=80=9Chardware wise=E2=80=9D today on =
the inside of our open test-site <a =
href=3D"https://webrtc.ingate.com">https://webrtc.ingate.com</a> where =
we have a border placed TURN server paralleling our firewall/default =
gateway. Our firewall looks for the magic STUN cookie and discards all =
STUN packets from the LAN side. TURN is then automatically enforced and =
today=E2=80=99s leaky WebRTC browsers automatically get the preferred =
exo-sealed behavior.<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(J=
ust not using STUN in the browser, would have the same effect but not =
fully, if there are more subnets on the LAN side. Crossing those subnets =
would then take media through the dual trip via the TURN server that I =
want to avoid.)<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>So=
, let=E2=80=99s enforce the preferred exo-sealed configuration by =
configuration of the firewall/default gateway (Security from this =
perspective is then also in the right hands - the network =
admin=E2=80=99s): <o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>- =
Today=E2=80=99s RETURN-draft=E2=80=99s leaky default configuration is OK =
(since the firewalls discarding STUN automatically enforce the preferred =
exo-sealed behavior)<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>- =
Recommend in the RETURN-draft (and in </span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>draft-jenning=
s-behave-rtcweb-firewall-00<span style=3D'color:blue'>) =C2=A0that the =
network admin discards STUN packets from the LAN side through the =
default gateway (consent STUNs will go OK through the TURN server), by =
looking for the magic STUN cookie, by closing port 3478 or by closing =
all UDP (the restrictive enterprise firewall can simply be left as it =
is) to achieve exo-sealed leakiness. <o:p></o:p></span></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>- =
The network admin offers a Border TURN server for its network users (LAN =
side only) =E2=80=93 no need for authentication (free to use from LAN =
side being a Border TURN server configuration) <o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>- =
The Border TURN server is offered by the network you are using =E2=80=93 =
Just use it! (should the network provider have evil intentions, he needs =
no TURN server, he can simply close or destroy WebRTC in the firewall if =
he which,=C2=A0 which Cullen just exemplified in </span><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>draft-jenning=
s-behave-rtcweb-firewall-00<span =
style=3D'color:blue'>)<o:p></o:p></span></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>As=
 you may have noticed, I stress that NO credentials should be needed for =
using an auto discovered border TURN server. That is also the goal of =
TRAM=E2=80=99s discovery draft, but I have asked TRAM =C2=A0to clarify =
that achievement. If we don=E2=80=99t fix that, we have failed to make =
WebRTC usable on mobile devices =E2=80=93 or can anyone explain where we =
would get those credentials from when we run around and have Web access =
here and there and the goal of WebRTC was to use real-time =
communication, securely and with high quality, everywhere we can surf, =
wasn=E2=80=99t it? <o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>If=
 anyone cries that we NEED AUTHENTICATION, they should remember the IP =
authentication (simply being on a certain network) is the most commonly =
used =E2=80=9Cauthentication method=E2=80=9D today and protecting huge =
values (almost the whole telco industry) and also achieves privacy =
(through the use of firewalls). No individual user credentials which are =
impossible to handle must be required.<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>I =
will in next email propose this new variant of the Anycast method =
=C2=A0to TRAM. It should work, shouldn=E2=80=99t =
it?<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>An=
d does it provide the claimed advantages/features? =E2=80=93 please =
check.<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoPlainText><u><span lang=3DEN-US =
style=3D'font-family:"Courier New"'>Implicit discovery by interception =
of TURN allocate requests at a default =
gateway<o:p></o:p></span></u></p><p class=3DMsoPlainText><span =
lang=3DEN-US style=3D'font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US style=3D'font-family:"Courier New"'>An enterprise firewall =
or a network server provider=E2=80=99s router at a default gateway can =
intercept a TURN allocate request and respond with a 300 (Try Alternate) =
just as described in the Anycast method above. The following procedures =
and advantages are inherited from the anycast =
method.<o:p></o:p></span></p><p class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-family:"Courier New"'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US style=3D'font-family:"Courier =
New"'>This variant adds the following features:<o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US style=3D'font-family:"Courier =
New"'>- no separate deployment of discovery mechanism is =
required<o:p></o:p></span></p><p class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-family:"Courier New"'>- no ANA allocated anycast address =
is required (works with any public ip address) <o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US style=3D'font-family:"Courier =
New"'>- the default gateway router/firewall in charge of the local =
network responds with the ip address of:<o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US style=3D'font-family:"Courier =
New"'>(i) =C2=A0=C2=A0a Border TURN server (paralleling the default =
gateway), or <o:p></o:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US style=3D'font-family:"Courier New"'>(ii) =C2=A0a TURN =
server behind that firewall/default gateway and opens a path to that =
TURN server =C2=A0<o:p></o:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US style=3D'font-family:"Courier New"'>- if the intercepted =
allocate request was intended for an application provided TURN server, =
the TURN client (e.g. WebRTC browser) on behalf of the user, can select =
to:<o:p></o:p></span></p><p class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-family:"Courier New"'>(iii) only use the auto discovered =
TURN server<o:p></o:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US style=3D'font-family:"Courier New"'>(iv) =C2=A0honor an =
allocation request by really sending the real-time traffic to the =
Application Provided TURN server, using the Recursively Encapsulated =
method, unless<o:p></o:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US style=3D'font-family:"Courier New"'>(v) =C2=A0=C2=A0the =
TURN=C2=A0 server connected in (ii) already is the Application Provided =
TURN server<o:p></o:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US style=3D'font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US style=3D'font-family:"Courier New"'>Security wise, the =
right party is in charge in all cases.<o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US style=3D'font-family:"Courier =
New"'>A TURN server pinpointed by a firewall/router at the default =
gateway should always be trusted (no explicit authentication with =
credentials is needed) and used. (We trust the default gateway with all =
our traffic anyway, and any evil intent to disturb real-time traffic can =
in such case be implemented in a default gateway by easier =
means.)<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p></div><blockquote =
style=3D'border:none;border-left:solid #CCCCCC 1.0pt;padding:0cm 0cm 0cm =
6.0pt;margin-left:4.8pt;margin-right:0cm'><p class=3DMsoNormal><span =
lang=3DEN-US>Such &quot;exo-sealed&quot; configuration should be the =
default configuration if a<br>network provided TURN server is made =
available for good (non-evil) purposes.<br><br><br></span>2) Mobile =
devices need to work on various networks, without =
individual<br>configuration<br><br>I am not clear about the thoughts =
behind the &quot;configuration&quot; of sealed level<br>and requirements =
for authentication of the turn server, but if there are<br>methods where =
an enterprise network (LAN) admin can provide credentials and<br>force a =
certain sealed level upon the WebRTC-browser when used at that =
LAN,<br>or if the user has the possibility to configure manually, I am =
OK with that<br>this can and should OVERRIDE any default configuration =
or other<br>configuration (on that particular network).<br><br>However, =
for mobile devices - Don't we all happily connect our browsers =
via<br>WiFi here and there? - there must be a way for a network provider =
with a<br>good intent to offer an exo-sealed TURN server that =
automatically should be<br>used, e.g. to achieve the purpose of offering =
better RTC quality than is<br>available through the default gateway WAN =
path on that network.<br><br>A network provided TURN-server can easily =
be made available ONLY to the<br>network users that it is serving, so =
there is no need for authentication and<br>credentials to protect =
against illegal usage of a valuable resource, like it<br>is with the =
application provided turn server. Network provided TURN servers<br>are =
simply deployed for the network users to be use without =
other<br>authentication than being granted an IP address and a default =
gateway to use<br>on that particular network in itself.<br><br>The same =
goes for trusting the TURN server with our real-time traffic - If<br>we =
would trust the same network provider with the same real-time =
traffic<br>through the default gateway - we should also use the offered =
TURN server<br>without other &quot;authentication&quot; than that it is =
available on our network:<br>Data through the default gateway, RTC =
through the TURN path is why the TURN<br>server is provided!<br><br>But =
do we know that it is not someone with the evil purpose of stealing =
our<br>(encrypted!) real time traffic that is offering the TURN server =
(instead of<br>the network provider), which leads us to my point 3). (If =
we really think<br>there is a risk of and danger with such =
attacks...)<br><br><br>3) Is it now good enough to simply use the auto =
discovered TURN server? Hope<br>so!<br><br>There has been some recent =
improvements to<br>draft-ietf-tram-turn-server-discovery-03 and =
hopefully that is sufficient<br>for assuming that the auto-discovered =
TURN server is offered by our network<br>provider (a service provider or =
the enterprise itself) that we trust anyway,<br>and the TURN server then =
simply shall be used and considered exo-sealed.<br>(The current RETURN =
draft says a non configured TURN server should =
be<br>leaky.)<o:p></o:p></p></blockquote><div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p =
class=3DMsoNormal>The security considerations in that document suggest =
that not all discovery mechanisms have equal additional exposure to =
local network interference.&nbsp; I would appreciate clearer guidance on =
that front.&nbsp; It seems to me that DHCP is the only listed mechanism =
that creates minimal additional exposure to network =
interference.<o:p></o:p></p><p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>--=
- I am addressing TRAM about this and believe that the variant of the =
Anycast method I suggest to be added, will fully resolve all security =
concerns. I am also suggesting that it is up to this RTCWEB draft to =
select from TRAM=E2=80=99s methods and advices for the WebRTC =
application and its use cases.<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>--=
- See my deployment concerns regarding DHCP put in front of TRAM. I =
think that the Anycast method with a short TTL-limitation is good enough =
from the privacy aspect, when used to find Border TURN =
server.<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>--=
- Even though the new variant I propose is more secure and even easier =
to deploy, I suggest that this RETURN draft mandates the use of the both =
the Anycast method and its new variant. The reason is that already =
today, all routers/firewalls at a default gateways can add the anycast =
route required for discovery, while intercepting the TURN=C2=A0 allocate =
request and responding with the Try Alternate server probably requires a =
software upgrade of installed =C2=A0firewalls and routers with default =
gateways (a firewall in front of the default gateway can also do the =
interception and response). <o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p></div><blockquote =
style=3D'border:none;border-left:solid #CCCCCC 1.0pt;padding:0cm 0cm 0cm =
6.0pt;margin-left:4.8pt;margin-right:0cm'><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'><span lang=3DEN-US>Some concerns =
regarding the DISCOVERY draft and whether these things should<br>be =
specified in the RETURN document:<br><br>- Is the above the case with =
all discovery methods proposed? </span>Or shall the<br>RETURN document =
make one method mandatory to implement?<br><br>- Can that method then be =
the Anycast method (which by far seems to be the<br>most convenient to =
deploy and to implement and should work for all network<br>types and in =
all devices operating systems). I think what is said =
under<br>&quot;9.3.&nbsp; Anycast&quot; under &quot;9.&nbsp; Security =
Considerations&quot; is sufficient for<br>trusting such discovered TURN =
server. What should then go into the RETURN<br>draft - or should the =
DISCOVERY draft be more specific?<br><br>I don't fully understand the =
DISCOVERY draft's: &quot;... three discovery<br>mechanisms. The reason =
for providing multiple mechanisms is to maximize the<br>opportunity for =
discovery,&quot;<br><br>- Can we really put the burden upon all WebRTC =
devices to implement all<br>methods and I doubt network providers will =
deploy more than one method. That<br>seems risking rather than =
maximizing discovery...<br><br><br>It is high time to get this done - =
can we?<br><br>/Karl<br><br><br><br>-----Ursprungligt =
meddelande-----<br>Fr=C3=A5n: rtcweb [mailto:<a =
href=3D"mailto:rtcweb-bounces@ietf.org" =
target=3D"_blank">rtcweb-bounces@ietf.org</a>] F=C3=B6r <a =
href=3D"mailto:internet-drafts@ietf.org" =
target=3D"_blank">internet-drafts@ietf.org</a><br>Skickat: den 14 maj =
2015 00:29<br>Till: <a href=3D"mailto:i-d-announce@ietf.org" =
target=3D"_blank">i-d-announce@ietf.org</a><br>Kopia: <a =
href=3D"mailto:rtcweb@ietf.org" =
target=3D"_blank">rtcweb@ietf.org</a><br>=C3=84mne: [rtcweb] I-D Action: =
draft-ietf-rtcweb-return-00.txt<br><br><br>A New Internet-Draft is =
available from the on-line Internet-Drafts<br>directories.<br>&nbsp;This =
draft is a work item of the Real-Time Communication in =
WEB-browsers<br>Working Group of the IETF.<br><br>&nbsp; &nbsp; &nbsp; =
&nbsp; Title&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;: Recursively =
Encapsulated TURN (RETURN) for<br>Connectivity and Privacy in =
WebRTC<br>&nbsp; &nbsp; &nbsp; &nbsp; Authors&nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp;: Benjamin M. Schwartz<br>&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; Justin =
Uberti<br>&nbsp; &nbsp; &nbsp; &nbsp; Filename&nbsp; &nbsp; &nbsp; =
&nbsp; : draft-ietf-rtcweb-return-00.txt<br>&nbsp; &nbsp; &nbsp; &nbsp; =
Pages&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;: 18<br>&nbsp; &nbsp; =
&nbsp; &nbsp; Date&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; : =
2015-05-13<br><br>Abstract:<br>&nbsp; &nbsp;In the context of WebRTC, =
the concept of a local TURN proxy has been<br>&nbsp; &nbsp;suggested, =
but not reviewed in detail.&nbsp; WebRTC applications are<br>&nbsp; =
&nbsp;already using TURN to enhance connectivity and privacy.&nbsp; =
This<br>&nbsp; &nbsp;document explains how local TURN proxies and WebRTC =
applications can<br>&nbsp; &nbsp;work together.<br><br><br>The IETF =
datatracker status page for this draft is:<br><a =
href=3D"https://datatracker.ietf.org/doc/draft-ietf-rtcweb-return/" =
target=3D"_blank">https://datatracker.ietf.org/doc/draft-ietf-rtcweb-retu=
rn/</a><br><br>There's also a htmlized version available at:<br><a =
href=3D"https://tools.ietf.org/html/draft-ietf-rtcweb-return-00" =
target=3D"_blank">https://tools.ietf.org/html/draft-ietf-rtcweb-return-00=
</a><br><br><br>Please note that it may take a couple of minutes from =
the time of submission<br>until the htmlized version and diff are =
available at <a href=3D"http://tools.ietf.org" =
target=3D"_blank">tools.ietf.org</a>.<br><br>Internet-Drafts are also =
available by anonymous FTP at:<br><a =
href=3D"ftp://ftp.ietf.org/internet-drafts/" =
target=3D"_blank">ftp://ftp.ietf.org/internet-drafts/</a><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><o:p></=
o:p></p></blockquote></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div></div></div></body></html>
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Mon, 20 Jul 2015 13:43:10 +0200
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Subject: Re: [rtcweb] Please require user consent for data channels
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2015-07-20 13:37 GMT+02:00 Tim Panton <thp@westhawk.co.uk>:

> E.g. I=E2=80=99m at home and my chromebook pixel (or firefox tablet) is o=
n wifi,
> but I=E2=80=99ve left LTE enabled.
> I (or the OS) is configured to prefer wifi wen available - but it happens
> that for a specific peer LTE completes first.
> So now my video call goes over LTE without my say-so and with no hint thi=
s
> is happening  - costing me real
> money. My only option is to completely disable LTE when I get home  (and
> lose SMS too) ?
>

Exactly.

Setting the default network route should be enough to avoid that and to
ensure a global network policy for the whole OS, unless an app (the
browser) plays to be God.


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

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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">2015=
-07-20 13:37 GMT+02:00 Tim Panton <span dir=3D"ltr">&lt;<a href=3D"mailto:t=
hp@westhawk.co.uk" target=3D"_blank">thp@westhawk.co.uk</a>&gt;</span>:<br>=
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div>E.g. I=E2=80=99m at home and my chromeb=
ook pixel (or firefox tablet) is on wifi, but I=E2=80=99ve left LTE enabled=
.</div><div>I (or the OS) is configured to prefer wifi wen available - but =
it happens that for a specific peer LTE completes first.</div><div>So now m=
y video call goes over LTE without my say-so and with no hint this is happe=
ning =C2=A0- costing me real</div><div>money. My only option is to complete=
ly disable LTE when I get home =C2=A0(and lose SMS too) ?</div></blockquote=
></div><br>Exactly.</div><div class=3D"gmail_extra"><br></div><div class=3D=
"gmail_extra">Setting the default network route should be enough to avoid t=
hat and to ensure a global network policy for the whole OS, unless an app (=
the browser) plays to be God.<br><br clear=3D"all"><div><br></div>-- <br><d=
iv class=3D"gmail_signature">I=C3=B1aki Baz Castillo<br>&lt;<a href=3D"mail=
to:ibc@aliax.net" target=3D"_blank">ibc@aliax.net</a>&gt;</div>
</div></div>

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From: Tim Panton <tim@phonefromhere.com>
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Date: Mon, 20 Jul 2015 13:23:08 +0100
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References: <55352067.6070007@nostrum.com> <20150421154657.GA4071@lo.psyced.org> <20150715123803.GA24976@lo.psyced.org> <2BCEF545-D322-4589-AC85-AA80E7F934D9@phonefromhere.com> <20150715164129.GA27105@lo.psyced.org> <AF6485D4-1579-45AF-937C-1682AC1AA645@phonefromhere.com> <20150716084225.GA29652@lo.psyced.org>
To: carlo von lynX <lynX@i.know.you.are.psyced.org>
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Subject: Re: [rtcweb] Prompt for mic, camera *and* network access
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> On 16 Jul 2015, at 09:42, carlo von lynX =
<lynX@i.know.you.are.psyced.org <mailto:lynX@i.know.you.are.psyced.org>> =
wrote:
>=20
> Yes, I dare to speak out of my 25 years of experience on the
> Internet even if I didn't have to use much VPN ever. I can
> imagine plenty of constellations where the VPN is NOT supposed
> to take over all communications on the system but rather offer
> a side route into a company's servers or suchlike. Therefore
> I worry if the software provider can always come up with a
> configuration that still does what the customers expect and
> at the same time deals with the new backdoor introduced by WebRTC.
> Of course anyone with better understanding can help clear up
> these worries.
>=20
>> That isn=E2=80=99t what I meant. I expect secure software providers =
to keep their
>> software abreast of new developments.=20

It turns out that I was completely wrong. The browsers are currently so=20=

thorough in their discovery of available interfaces and routes that no
amount of changes by the sys admin or VPN supplier will fix this =
problem.

Sorry about that.=20

>=20
> Uh, not sure if that is an established pattern in the industry.
> Usually somebody does whatever fits into their business model
> best, then there is a public outcry and possibly some bloodspill,
> then all the players look at how they can make a good impression
> by reducing the collateral damages.. but if that doesn't work out
> cheap and easy.. tant pis. Let's continue with business.
>=20
>> I happen to have the (no doubt under informed) opinion that
>> this isn=E2=80=99t as simple as it has been made out to be.

At least I was right here. - It is even more complex than I believed.

>=20
> Well, in case of doubt it is important to pick the conservative
> option until the issue is properly analyzed rather than hoping
> for the general public to forget about the problem.
>=20
>>> Also the "solution" you suggest AFAIU does not work for people
>>> who use Tor in a non-NATted environment, for example on a
>>> fixed system in a university library with a static IP address.
>>=20
>> That is an interesting case. In that instance hiding the local IP =
would be=20
>> valuable. I thought however those users would normally run a TOR os =
off=20
>> a usb stick in a VM. That would probably mask the local IP.
>=20
> Now that this is more clear, have another look at my original mail
> that describes how such a dissident would use any Windows OS system
> standing around. I doubt anything advanced like TAILS sticks are
> out there as merely possessing one could be life-threatening.


Here too I find that I was wrong and the current situation not what I =
assumed
and is unsatisfactory.

Most SOCKS proxies don't have support for ICE/STUN nor arbitrary UDP.
The current browser implementations (backed by the ICE spec) take the =
view that
if SOCKS doesn=E2=80=99t do UDP, then they should go directly and send =
UDP anyhow.

I=E2=80=99m not clear that this is what a user might expect.=20
An alternative might be to detect a proxy config _only_ offer TURN over =
TLS=20
candidates (and route the traffic over the proxy) - This would result in
poor audio/video but would be more in keeping with the spirit of the =
proxy config.

(a variant might be to inspect the proxy in detail and use the address =
whitelists
to filter allowed traffic somehow?)=20


>>=20
>> I totally agree. My solution is imho a better fix, since it
>> also protects against many other apps that might get started from
>> a browser page (email, dns, ipv6 tunnels, flash, mp3 players etc).


Turns out I was wrong - sigh - as I said the browsers are more =
aggressive
in finding routes than I assumed (or indeed than my ICE implementations =
are).

> Breaking the proxy principle is neither a Tor nor a VPN problem.
> It's just browser vendors breaking a promise given to users that
> configured proxies would be respected.
>=20
>> I=E2=80=99m SOCKs ignorant.
>=20
> So we all have our knowledge deficiencies...
> good to be honest about them.
>=20

And now that I=E2=80=99ve educated myself somewhat, I=E2=80=99m agreeing =
with you.


>>> And if local IP information is so super harmless, then why is it
>>> an obvious reason for TBB devs to disable the entire thing?
>>> I presume for the reasons I stated before.
>>=20
>> I have no idea, but I fear it is because no one has split these =
issues out.
>=20
> So let's dissect some more until we know what we are doing.

Which turns out to be the only useful contribution my <rant/> made.


>=20
>> I=E2=80=99m of the view that there are _useful_ instances of P2P =
crypto
>> in-browser - we will clearly have to disagree about that.
>=20
> I think we could improve the user's ability to ensure end-to-end
> authentication beyond server-based negotiation. Something like
> rendering persistent public keys into QR codes and having people
> exchange those QR codes at cocktail parties and hackathons.
>=20
> Servers would still be able to provide the signaling code, but users
> would be granted a guarantee that the E2E connection is authentic.
>=20

https://yopet.us <https://yopet.us/> :-)=20

Tim.=

--Apple-Mail=_627FF319-1132-431E-B675-8EA86B9B71BD
Content-Transfer-Encoding: quoted-printable
Content-Type: text/html;
	charset=utf-8

<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dutf-8"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" =
class=3D""><br class=3D""><div class=3D""><div class=3D"" =
style=3D"font-family: LucidaSansUnicode;"><blockquote type=3D"cite" =
class=3D""><div class=3D"">On 16 Jul 2015, at 09:42, carlo von lynX =
&lt;<a href=3D"mailto:lynX@i.know.you.are.psyced.org" =
class=3D"">lynX@i.know.you.are.psyced.org</a>&gt; wrote:</div><div =
class=3D""><br class=3D"">Yes, I dare to speak out of my 25 years of =
experience on the<br class=3D"">Internet even if I didn't have to use =
much VPN ever. I can<br class=3D"">imagine plenty of constellations =
where the VPN is NOT supposed<br class=3D"">to take over all =
communications on the system but rather offer<br class=3D"">a side route =
into a company's servers or suchlike. Therefore<br class=3D"">I worry if =
the software provider can always come up with a<br =
class=3D"">configuration that still does what the customers expect =
and<br class=3D"">at the same time deals with the new backdoor =
introduced by WebRTC.<br class=3D"">Of course anyone with better =
understanding can help clear up<br class=3D"">these worries.<br =
class=3D""><br class=3D""><blockquote type=3D"cite" class=3D"">That =
isn=E2=80=99t what I meant. I expect secure software providers to keep =
their<br class=3D"">software abreast of new developments.&nbsp;<br =
class=3D""></blockquote></div></blockquote><div class=3D""><br =
class=3D""></div><div class=3D"">It turns out that I was completely =
wrong. The browsers are currently so&nbsp;</div><div class=3D"">thorough =
in their discovery of available interfaces and routes that no</div><div =
class=3D"">amount of changes by the sys admin or VPN supplier will fix =
this problem.</div><div class=3D""><br class=3D""></div><div =
class=3D"">Sorry about that.&nbsp;</div><br class=3D""><blockquote =
type=3D"cite" class=3D""><div class=3D""><br class=3D"">Uh, not sure if =
that is an established pattern in the industry.<br class=3D"">Usually =
somebody does whatever fits into their business model<br class=3D"">best, =
then there is a public outcry and possibly some bloodspill,<br =
class=3D"">then all the players look at how they can make a good =
impression<br class=3D"">by reducing the collateral damages.. but if =
that doesn't work out<br class=3D"">cheap and easy.. tant pis. Let's =
continue with business.<br class=3D""><br class=3D""><blockquote =
type=3D"cite" class=3D"">I happen to have the (no doubt under informed) =
opinion that<br class=3D"">this isn=E2=80=99t as simple as it has been =
made out to be.<br class=3D""></blockquote></div></blockquote><div =
class=3D""><br class=3D""></div><div class=3D"">At least I was right =
here. - It is even more complex than I believed.</div><br =
class=3D""><blockquote type=3D"cite" class=3D""><div class=3D""><br =
class=3D"">Well, in case of doubt it is important to pick the =
conservative<br class=3D"">option until the issue is properly analyzed =
rather than hoping<br class=3D"">for the general public to forget about =
the problem.<br class=3D""><br class=3D""><blockquote type=3D"cite" =
class=3D""><blockquote type=3D"cite" class=3D"">Also the "solution" you =
suggest AFAIU does not work for people<br class=3D"">who use Tor in a =
non-NATted environment, for example on a<br class=3D"">fixed system in a =
university library with a static IP address.<br =
class=3D""></blockquote><br class=3D"">That is an interesting case. In =
that instance hiding the local IP would be&nbsp;<br class=3D"">valuable. =
I thought however those users would normally run a TOR os off&nbsp;<br =
class=3D"">a usb stick in a VM. That would probably mask the local =
IP.<br class=3D""></blockquote><br class=3D"">Now that this is more =
clear, have another look at my original mail<br class=3D"">that =
describes how such a dissident would use any Windows OS system<br =
class=3D"">standing around. I doubt anything advanced like TAILS sticks =
are<br class=3D"">out there as merely possessing one could be =
life-threatening.<br class=3D""></div></blockquote><div class=3D""><br =
class=3D""></div><div class=3D""><br class=3D""></div><div class=3D"">Here=
 too I find that I was wrong and the current situation not what I =
assumed</div><div class=3D"">and is unsatisfactory.</div><div =
class=3D""><br class=3D""></div><div class=3D"">Most SOCKS proxies don't =
have support for ICE/STUN nor arbitrary UDP.</div><div class=3D"">The =
current browser implementations (backed by the ICE spec) take the view =
that</div><div class=3D"">if SOCKS doesn=E2=80=99t do UDP, then they =
should go directly and send UDP anyhow.</div><div class=3D""><br =
class=3D""></div><div class=3D"">I=E2=80=99m not clear that this is what =
a user might expect.&nbsp;</div><div class=3D"">An alternative might be =
to detect a proxy config _only_ offer TURN over TLS&nbsp;</div><div =
class=3D"">candidates (and route the traffic over the proxy) - This =
would result in</div><div class=3D"">poor audio/video but would be more =
in keeping with the spirit of the proxy config.</div><div class=3D""><br =
class=3D""></div><div class=3D"">(a variant might be to inspect the =
proxy in detail and use the address whitelists</div><div class=3D"">to =
filter allowed traffic somehow?)&nbsp;</div><div class=3D""><br =
class=3D""></div><div class=3D""><br class=3D""></div><blockquote =
type=3D"cite" class=3D""><div class=3D""><blockquote type=3D"cite" =
class=3D""><br class=3D"">I totally agree. My solution is imho a better =
fix, since it<br class=3D"">also protects against many other apps that =
might get started from<br class=3D"">a browser page (email, dns, ipv6 =
tunnels, flash, mp3 players etc).<br =
class=3D""></blockquote></div></blockquote><div class=3D""><br =
class=3D""></div><div class=3D""><br class=3D""></div><div =
class=3D"">Turns out I was wrong - sigh - as I said the browsers are =
more aggressive</div><div class=3D"">in finding routes than I assumed =
(or indeed than my ICE implementations are).</div><br =
class=3D""><blockquote type=3D"cite" class=3D""><div class=3D"">Breaking =
the proxy principle is neither a Tor nor a VPN problem.<br class=3D"">It's=
 just browser vendors breaking a promise given to users that<br =
class=3D"">configured proxies would be respected.<br class=3D""><br =
class=3D""><blockquote type=3D"cite" class=3D"">I=E2=80=99m SOCKs =
ignorant.<br class=3D""></blockquote><br class=3D"">So we all have our =
knowledge deficiencies...<br class=3D"">good to be honest about them.<br =
class=3D""><br class=3D""></div></blockquote><div class=3D""><br =
class=3D""></div><div class=3D"">And now that I=E2=80=99ve educated =
myself somewhat, I=E2=80=99m agreeing with you.</div><div class=3D""><br =
class=3D""></div><br class=3D""><blockquote type=3D"cite" class=3D""><div =
class=3D""><blockquote type=3D"cite" class=3D""><blockquote type=3D"cite" =
class=3D"">And if local IP information is so super harmless, then why is =
it<br class=3D"">an obvious reason for TBB devs to disable the entire =
thing?<br class=3D"">I presume for the reasons I stated before.<br =
class=3D""></blockquote><br class=3D"">I have no idea, but I fear it is =
because no one has split these issues out.<br class=3D""></blockquote><br =
class=3D"">So let's dissect some more until we know what we are =
doing.<br class=3D""></div></blockquote><div class=3D""><br =
class=3D""></div><div class=3D"">Which turns out to be the only useful =
contribution my &lt;rant/&gt; made.</div><div class=3D""><br =
class=3D""></div><br class=3D""><blockquote type=3D"cite" class=3D""><div =
class=3D""><br class=3D""><blockquote type=3D"cite" class=3D"">I=E2=80=99m=
 of the view that there are _useful_ instances of P2P crypto<br =
class=3D"">in-browser - we will clearly have to disagree about that.<br =
class=3D""></blockquote><br class=3D"">I think we could improve the =
user's ability to ensure end-to-end<br class=3D"">authentication beyond =
server-based negotiation. Something like<br class=3D"">rendering =
persistent public keys into QR codes and having people<br =
class=3D"">exchange those QR codes at cocktail parties and =
hackathons.<br class=3D""><br class=3D"">Servers would still be able to =
provide the signaling code, but users<br class=3D"">would be granted a =
guarantee that the E2E connection is authentic.<br class=3D""><br =
class=3D""></div></blockquote><br class=3D""></div><div class=3D"" =
style=3D"font-family: LucidaSansUnicode;"><a href=3D"https://yopet.us/" =
class=3D"">https://yopet.us</a>&nbsp;:-)&nbsp;</div><div =
apple-content-edited=3D"true" class=3D""><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; border-spacing: 0px;"><div class=3D"" =
style=3D"font-family: Helvetica;"><br class=3D""></div><div =
class=3D""><font face=3D"LucidaSansUnicode" =
class=3D"">Tim.</font></div></span></div></div></body></html>=

--Apple-Mail=_627FF319-1132-431E-B675-8EA86B9B71BD--


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From: Tim Panton <tim@phonefromhere.com>
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Date: Mon, 20 Jul 2015 13:26:00 +0100
Message-Id: <7F818FAC-5559-4074-B1FC-EB9516A98FB7@phonefromhere.com>
References: <CA+65OspMD_PVjk0BXh7t4LtjmFDcDatoeNjFQOO_OVtC-Br+OA@mail.gmail.com> <CAOJ7v-0UBGtP0-atxP7X4OTj-H6Lost5o42aAS65mA6CEqcQsw@mail.gmail.com> <CA+65OsrhXHK+cRAFLCZFt+34vr8eRhj+CN3DgznUBfSwmWYggw@mail.gmail.com> <CAOJ7v-24VCW6kkn7LOLkqZzhYEU0r=nmd_F7Zns1rnyqKN6xAg@mail.gmail.com> <55A95364.2070806@gmail.com> <CAOJ7v-3t9BQabR2e4EHs4G0Sec4sU9DFC2aiSXXYrat+an+RYg@mail.gmail.com> <55A96DA3.1040907@gmail.com> <CAOJ7v-1ui7349NzK6NZNRHPbnHWZajctk4cDgMKqRZSv47EYdA@mail.gmail.com> <55A9860D.8030903@gmail.com> <CAOJ7v-3LGd32rnpFVW_U0s3+iVaJXsL4vt_YAo=cyp6YyOArdw@mail.gmail.com> <CALiegfmiS18Jux-kCgOhTKKiyGtMertj6xCegpFrox5NOf9EJg@mail.gmail.com> <CABkgnnW0Tmjqz823vKiF84_u6HasBJC7ERMYCO2HL_NPj5saTA@mail.gmail.com> <CALiegfkpbLy1QXxr-RRF0oOpVv1sWsFeab=vvC4iT4DnPtjKQw@mail.gmail.com> <CABkgnnVWcuhX2NjZgx87L+Uo6df6rEBWW73cxbaX3mu_VfHmCA@mail.gmail.com> <CALiegfkQWAn-jMrjhcDPA3rtowOPVk-S8z3c-jvjpNmjtf=3hA@mail.gmail.com> <CABkgnnWERM4oxozNCSvRf1o0Wm-d9Bjw=9B+xh_NJ+h6GfBJ6Q@mail.gmail.com>
To: Martin Thomson <martin.thomson@gmail.com>
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Subject: Re: [rtcweb] Please require user consent for data channels
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> On 18 Jul 2015, at 02:31, Martin Thomson <martin.thomson@gmail.com> =
wrote:
>=20
>=20
> On Jul 17, 2015 5:01 PM, "I=C3=B1aki Baz Castillo" <ibc@aliax.net =
<mailto:ibc@aliax.net>> wrote:
> >
> > The point is that you don't even choose the interface. The OS will =
do for you.
>=20
> The OS can - and frequently does - get that wrong. The default route =
can fail when another might succeed.
>=20
> You can't allow that to happen if you care about connecting =
successfully.
>=20
Gulp. Whilst I mostly see the logic - it is wholly unexpected behaviour =
to the average sys admin.=20
Certainly not what I expected.

It strikes me that binding to all interfaces might well give a vector =
for attackers to map out a company=E2=80=99s internal networks.
It also may restrict the user=E2=80=99s ability to manipulate which =
medium is used.=20

E.g. I=E2=80=99m at home and my chromebook pixel (or firefox tablet) is =
on wifi, but I=E2=80=99ve left LTE enabled.
I (or the OS) is configured to prefer wifi wen available - but it =
happens that for a specific peer LTE completes first.
So now my video call goes over LTE without my say-so and with no hint =
this is happening  - costing me real
money. My only option is to completely disable LTE when I get home  (and =
lose SMS too) ?

Perhaps we should default to binding to 0.0.0.0 and allow a user =
config=E2=80=99d preference for more exhaustive searching.

Tim.


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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dutf-8"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" =
class=3D""><br class=3D""><div><blockquote type=3D"cite" class=3D""><div =
class=3D"">On 18 Jul 2015, at 02:31, Martin Thomson &lt;<a =
href=3D"mailto:martin.thomson@gmail.com" =
class=3D"">martin.thomson@gmail.com</a>&gt; wrote:</div><br =
class=3D"Apple-interchange-newline"><div class=3D""><p dir=3D"ltr" =
class=3D""><br class=3D"">
On Jul 17, 2015 5:01 PM, "I=C3=B1aki Baz Castillo" &lt;<a =
href=3D"mailto:ibc@aliax.net" class=3D"">ibc@aliax.net</a>&gt; wrote:<br =
class=3D"">
&gt;<br class=3D"">
&gt; The point is that you don't even choose the interface. The OS will =
do for you.</p><p dir=3D"ltr" class=3D"">The OS can - and frequently =
does - get that wrong. The default route can fail when another might =
succeed.</p><p dir=3D"ltr" class=3D"">You can't allow that to happen if =
you care about connecting successfully.</p></div></blockquote><div =
class=3D"" style=3D"font-family: LucidaSansUnicode;">Gulp. Whilst I =
mostly see the logic - it is wholly unexpected behaviour to the average =
sys admin.&nbsp;</div><div class=3D"" style=3D"font-family: =
LucidaSansUnicode;">Certainly not what I expected.</div><div class=3D"" =
style=3D"font-family: LucidaSansUnicode;"><br class=3D""></div><div =
class=3D"" style=3D"font-family: LucidaSansUnicode;">It strikes me that =
binding to all interfaces might well give a vector for attackers to map =
out a company=E2=80=99s internal networks.</div><div class=3D"" =
style=3D"font-family: LucidaSansUnicode;">It also may restrict the =
user=E2=80=99s ability to manipulate which medium is =
used.&nbsp;</div><div class=3D"" style=3D"font-family: =
LucidaSansUnicode;"><br class=3D""></div><div class=3D"" =
style=3D"font-family: LucidaSansUnicode;">E.g. I=E2=80=99m at home and =
my chromebook pixel (or firefox tablet) is on wifi, but I=E2=80=99ve =
left LTE enabled.</div><div class=3D"" style=3D"font-family: =
LucidaSansUnicode;">I (or the OS) is configured to prefer wifi wen =
available - but it happens that for a specific peer LTE completes =
first.</div><div class=3D"" style=3D"font-family: LucidaSansUnicode;">So =
now my video call goes over LTE without my say-so and with no hint this =
is happening &nbsp;- costing me real</div><div class=3D"" =
style=3D"font-family: LucidaSansUnicode;">money. My only option is to =
completely disable LTE when I get home &nbsp;(and lose SMS too) =
?</div><div class=3D"" style=3D"font-family: LucidaSansUnicode;"><br =
class=3D""></div><div class=3D"" style=3D"font-family: =
LucidaSansUnicode;">Perhaps we should default to binding to 0.0.0.0 and =
allow a user config=E2=80=99d preference for more exhaustive =
searching.</div><div class=3D"" style=3D"font-family: =
LucidaSansUnicode;"><br class=3D""></div><div class=3D"" =
style=3D"font-family: LucidaSansUnicode;">Tim.</div></div><br =
class=3D""></body></html>=

--Apple-Mail=_74E349CF-115B-4059-9709-97857FCA9E63--


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Date: Mon, 20 Jul 2015 14:32:52 +0200
From: Simon Perreault <sperreault@jive.com>
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To: Tim Panton <tim@phonefromhere.com>,  Martin Thomson <martin.thomson@gmail.com>
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Subject: Re: [rtcweb] Please require user consent for data channels
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Le 2015-07-20 14:26, Tim Panton a écrit :
> Gulp. Whilst I mostly see the logic - it is wholly unexpected behaviour
> to the average sys admin. 
> Certainly not what I expected.

I don't know how you can expect ICE to work otherwise.

> It strikes me that binding to all interfaces might well give a vector
> for attackers to map out a company’s internal networks.

You are sending your IP address(es) as payload so that your peer may
connect to them. We've been doing this in various protocols for ages.
Just take FTP for example, where data and signalling are separate. How
is any of this different?

> It also may restrict the user’s ability to manipulate which medium is used. 
> E.g. I’m at home and my chromebook pixel (or firefox tablet) is on wifi,
> but I’ve left LTE enabled.
> I (or the OS) is configured to prefer wifi wen available - but it
> happens that for a specific peer LTE completes first.
> So now my video call goes over LTE without my say-so and with no hint
> this is happening  - costing me real
> money. My only option is to completely disable LTE when I get home  (and
> lose SMS too) ?

That is a valid concern, and the answer to it is MIF.

Simon


From nobody Mon Jul 20 05:35:38 2015
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From: "Hutton, Andrew" <andrew.hutton@unify.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: JSEP and -return- ICE Candidate Policy
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https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-11#section-3.4.3 talks a=
bout the application having control over the ICE Candidate Policy (RTCIceTr=
ansportPolicy) but it also contains the following statement regarding local=
 policy:

"In addition, administrators may also wish to control the set of ICE candid=
ates, and so the browser SHOULD also allow control via local policy, with t=
he most restrictive policy prevailing".

This seems to imply that the admin and application policies are controlling=
 the same functionality in the browser but is this really true?

https://tools.ietf.org/html/draft-ietf-rtcweb-return-00#section-4.3 also re=
fers to to the ICE candidate policy section of JSEP and indicates states th=
at:

"Proxy configuration, including leakiness, maybe set by local policy ([I-D.=
ietf-rtcweb-jsep], "ICE candidate policy") or other mechanisms".

It seems something is wrong here and probably JSEP should not be talking ab=
out the local policy as I assume this would not use the PeerConnection API =
and the -return- draft should not be referring to JSEP regarding this local=
 policy as again this will not use the PC API.

Andy


From nobody Mon Jul 20 05:58:56 2015
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> On 20 Jul 2015, at 13:32, Simon Perreault <sperreault@jive.com> wrote:
>=20
> Le 2015-07-20 14:26, Tim Panton a =E9crit :
>> Gulp. Whilst I mostly see the logic - it is wholly unexpected =
behaviour
>> to the average sys admin.=20
>> Certainly not what I expected.
>=20
> I don't know how you can expect ICE to work otherwise.
>=20
>> It strikes me that binding to all interfaces might well give a vector
>> for attackers to map out a company=92s internal networks.
>=20
> You are sending your IP address(es) as payload so that your peer may
> connect to them. We've been doing this in various protocols for ages.
> Just take FTP for example, where data and signalling are separate. How
> is any of this different?

I=92d hardly take FTP as a model citizen :-)

But what is different is that we have removed _all_ the sysadmin=92s =
levers
to prevent mapping of internal address spaces.

If I understand correctly we bind to each and every interface
that is up and send out packets to all remote candidates down _every_ =
interface,
ignoring any preferences set by the admin in the local routing table =
which
were set to ensure (say) only critical traffic is sent down certain =
interfaces.

I can think of some financial institutions who=92s IDSs will go off when =
packets
to public external addresses start turning up on internal LANS . =
(hopefully they=20
will get dropped at the egress router, but by then the IDS will have =
paged someone).

Likewise my fantastically expensive BGAN link which is used for
emergencies only will suddenly start carrying STUN requests irrespective =
of how I configure my
routes ?

>=20
>> It also may restrict the user=92s ability to manipulate which medium =
is used.=20
>> E.g. I=92m at home and my chromebook pixel (or firefox tablet) is on =
wifi,
>> but I=92ve left LTE enabled.
>> I (or the OS) is configured to prefer wifi wen available - but it
>> happens that for a specific peer LTE completes first.
>> So now my video call goes over LTE without my say-so and with no hint
>> this is happening  - costing me real
>> money. My only option is to completely disable LTE when I get home  =
(and
>> lose SMS too) ?
>=20
> That is a valid concern, and the answer to it is MIF.

And in the meanwhile ?

Tim.

>=20
> Simon


From nobody Mon Jul 20 06:13:28 2015
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Mon, 20 Jul 2015 15:13:05 +0200
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2015-07-20 14:32 GMT+02:00 Simon Perreault <sperreault@jive.com>:

> Le 2015-07-20 14:26, Tim Panton a =C3=A9crit :
> > Gulp. Whilst I mostly see the logic - it is wholly unexpected behaviour
> > to the average sys admin.
> > Certainly not what I expected.
>
> I don't know how you can expect ICE to work otherwise.
>

It would just work. The STUN/TURN server should be reachable using the OS
configured routes, isn't it?

Please, don't argument that "otherwise it would not work" because that is
not true at all.



>
> > It strikes me that binding to all interfaces might well give a vector
> > for attackers to map out a company=E2=80=99s internal networks.
>
> You are sending your IP address(es) as payload so that your peer may
> connect to them.


The fact that the client announces its (probably) private VPN addresses to
the remote peer does not mean that it also reveals the associated public
reflexive IP. That just happens if the client performs STUN procedures over
that interface, which is exactly the issue being discussed (at least right
now).



> We've been doing this in various protocols for ages.
> Just take FTP for example, where data and signalling are separate. How
> is any of this different?
>

The FTP client does not override the OS routes but instead contacts both
the FTP signaling address and the FTP data address using the OS configured
routes (rather than forcing local interfaces to reach them). So yes, it is
fully different IMHO.



>
> > It also may restrict the user=E2=80=99s ability to manipulate which med=
ium is
> used.
> > E.g. I=E2=80=99m at home and my chromebook pixel (or firefox tablet) is=
 on wifi,
> > but I=E2=80=99ve left LTE enabled.
> > I (or the OS) is configured to prefer wifi wen available - but it
> > happens that for a specific peer LTE completes first.
> > So now my video call goes over LTE without my say-so and with no hint
> > this is happening  - costing me real
> > money. My only option is to completely disable LTE when I get home  (an=
d
> > lose SMS too) ?
>
> That is a valid concern, and the answer to it is MIF.


Sorry, what does MIF stand for?



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

--001a114dcd807abd1c051b4e4d79
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<div dir=3D"ltr"><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">=
2015-07-20 14:32 GMT+02:00 Simon Perreault <span dir=3D"ltr">&lt;<a href=3D=
"mailto:sperreault@jive.com" target=3D"_blank">sperreault@jive.com</a>&gt;<=
/span>:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bor=
der-left:1px #ccc solid;padding-left:1ex"><span class=3D"">Le 2015-07-20 14=
:26, Tim Panton a =C3=A9crit :<br>
&gt; Gulp. Whilst I mostly see the logic - it is wholly unexpected behaviou=
r<br>
&gt; to the average sys admin.<br>
&gt; Certainly not what I expected.<br>
<br>
</span>I don&#39;t know how you can expect ICE to work otherwise.<br></bloc=
kquote><div><br></div><div>It would just work. The STUN/TURN server should =
be reachable using the OS configured routes, isn&#39;t it?</div><div><br></=
div><div>Please, don&#39;t argument that &quot;otherwise it would not work&=
quot; because that is not true at all.</div><div><br></div><div>=C2=A0</div=
><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1=
px #ccc solid;padding-left:1ex">
<span class=3D""><br>
&gt; It strikes me that binding to all interfaces might well give a vector<=
br>
&gt; for attackers to map out a company=E2=80=99s internal networks.<br>
<br>
</span>You are sending your IP address(es) as payload so that your peer may=
<br>
connect to them. </blockquote><div><br></div><div>The fact that the client =
announces its (probably) private VPN addresses to the remote peer does not =
mean that it also reveals the associated public reflexive IP. That just hap=
pens if the client performs STUN procedures over that interface, which is e=
xactly the issue being discussed (at least right now).</div><div><br></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">We&#39;ve been doing this i=
n various protocols for ages.<br>
Just take FTP for example, where data and signalling are separate. How<br>
is any of this different?<br></blockquote><div><br></div><div>The FTP clien=
t does not override the OS routes but instead contacts both the FTP signali=
ng address and the FTP data address using the OS configured routes (rather =
than forcing local interfaces to reach them). So yes, it is fully different=
 IMHO.</div><div><br></div><div>=C2=A0</div><blockquote class=3D"gmail_quot=
e" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<span class=3D""><br>
&gt; It also may restrict the user=E2=80=99s ability to manipulate which me=
dium is used.<br>
&gt; E.g. I=E2=80=99m at home and my chromebook pixel (or firefox tablet) i=
s on wifi,<br>
&gt; but I=E2=80=99ve left LTE enabled.<br>
&gt; I (or the OS) is configured to prefer wifi wen available - but it<br>
&gt; happens that for a specific peer LTE completes first.<br>
&gt; So now my video call goes over LTE without my say-so and with no hint<=
br>
&gt; this is happening=C2=A0 - costing me real<br>
&gt; money. My only option is to completely disable LTE when I get home=C2=
=A0 (and<br>
&gt; lose SMS too) ?<br>
<br>
</span>That is a valid concern, and the answer to it is MIF.</blockquote></=
div><br>Sorry, what does MIF stand for?</div><div class=3D"gmail_extra"><br=
><br clear=3D"all"><div><br></div>-- <br><div class=3D"gmail_signature">I=
=C3=B1aki Baz Castillo<br>&lt;<a href=3D"mailto:ibc@aliax.net" target=3D"_b=
lank">ibc@aliax.net</a>&gt;</div>
</div></div>

--001a114dcd807abd1c051b4e4d79--


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Le 2015-07-20 14:58, Tim Panton a écrit :
> If I understand correctly we bind to each and every interface
> that is up and send out packets to all remote candidates down _every_ interface,
> ignoring any preferences set by the admin in the local routing table which
> were set to ensure (say) only critical traffic is sent down certain interfaces.

I have difficulty understanding what it is that's bugging you. Is it the
signalling payload (i.e., sending the list of address), or is it the
connectivity checks that are done on all candidate pairs?

> I can think of some financial institutions who’s IDSs will go off when packets
> to public external addresses start turning up on internal LANS . (hopefully they 
> will get dropped at the egress router, but by then the IDS will have paged someone).

But that's not going to happen unless there's a route. I think there
could be a misunderstanding here. For example, if there's a VPN with a
10.0.0.0/8 route but no default route, the ICE client will *not* send a
request to a TURN server listening on 192.0.2.1 over the VPN interface.
That would make no sense whatsoever.

> Likewise my fantastically expensive BGAN link which is used for
> emergencies only will suddenly start carrying STUN requests irrespective of how I configure my
> routes ?

Only if it's possible to route the packet to the STUN server over the
BGAN link. There must be a route.

Simon


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Le 2015-07-20 15:13, IÃ±aki Baz Castillo a Ã©crit :
>     > Gulp. Whilst I mostly see the logic - it is wholly unexpected behaviour
>     > to the average sys admin.
>     > Certainly not what I expected.
> 
>     I don't know how you can expect ICE to work otherwise.
> 
> 
> It would just work. The STUN/TURN server should be reachable using the
> OS configured routes, isn't it?

I have no idea what you're talking about. Yes there must be a route. But
that's completely orthogonal to which interface you're binding to.

> Please, don't argument that "otherwise it would not work" because that
> is not true at all.

It is. But I'm not sure we're talking about the same "it" anymore.

>     > It strikes me that binding to all interfaces might well give a vector
>     > for attackers to map out a companyâ€™s internal networks.
> 
>     You are sending your IP address(es) as payload so that your peer may
>     connect to them. 
> 
> 
> The fact that the client announces its (probably) private VPN addresses
> to the remote peer does not mean that it also reveals the associated
> public reflexive IP. That just happens if the client performs STUN
> procedures over that interface, which is exactly the issue being
> discussed (at least right now).

If you can find a server-reflexive candidate on the VPN interface, it
means you can talk to the Internet using that interface, using that
server-reflexive IP address. How is that private knowledge at all? How
do you expect to keep your server-reflexive address private but still
use it for communication?

>     We've been doing this in various protocols for ages.
>     Just take FTP for example, where data and signalling are separate. How
>     is any of this different?
> 
> 
> The FTP client does not override the OS routes but instead contacts both
> the FTP signaling address and the FTP data address using the OS
> configured routes (rather than forcing local interfaces to reach them).
> So yes, it is fully different IMHO.

The routing table is *not* overridden by explicitly binding to an interface.

>     > It also may restrict the userâ€™s ability to manipulate which medium is used.
>     > E.g. Iâ€™m at home and my chromebook pixel (or firefox tablet) is on wifi,
>     > but Iâ€™ve left LTE enabled.
>     > I (or the OS) is configured to prefer wifi wen available - but it
>     > happens that for a specific peer LTE completes first.
>     > So now my video call goes over LTE without my say-so and with no hint
>     > this is happening  - costing me real
>     > money. My only option is to completely disable LTE when I get home  (and
>     > lose SMS too) ?
> 
>     That is a valid concern, and the answer to it is MIF.
> 
> 
> Sorry, what does MIF stand for?

https://datatracker.ietf.org/wg/mif/documents/

The MIF WG is working on exactly this kind of problem. They are
developing an API for solving precisely this kind of interface
preference problem. IMHO ICE could perhaps be extended to make use of
this API. But I see it as "just" an enhancement.

Simon


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2015-07-20 15:26 GMT+02:00 Simon Perreault <sperreault@jive.com>:

> Le 2015-07-20 14:58, Tim Panton a =C3=A9crit :
> > If I understand correctly we bind to each and every interface
> > that is up and send out packets to all remote candidates down _every_
> interface,
> > ignoring any preferences set by the admin in the local routing table
> which
> > were set to ensure (say) only critical traffic is sent down certain
> interfaces.
>
> I have difficulty understanding what it is that's bugging you. Is it the
> signalling payload (i.e., sending the list of address), or is it the
> connectivity checks that are done on all candidate pairs?
>

Unfortunately we are mixing two probably separate concerns:

1) ICE gathering and possible sensitive IP leak when connecting the
STUN/TURN server (decided by the web app) from all the interfaces (those
that of course have a route to the server).

2) Same issue when the IPs are announced in the SDP and ICE connectivity
checks with the remote peer are performed.

We should focus :)




>
> > Likewise my fantastically expensive BGAN link which is used for
> > emergencies only will suddenly start carrying STUN requests irrespectiv=
e
> of how I configure my
> > routes ?
>
> Only if it's possible to route the packet to the STUN server over the
> BGAN link. There must be a route.


 Yes, but the point is that one may have enabled a secondary route (a
default route via eth2) but prefer to use another route as default (eth1).
The browser will not respect that because it will try both eth1 and eth2.




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

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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">2015=
-07-20 15:26 GMT+02:00 Simon Perreault <span dir=3D"ltr">&lt;<a href=3D"mai=
lto:sperreault@jive.com" target=3D"_blank">sperreault@jive.com</a>&gt;</spa=
n>:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-=
left:1px #ccc solid;padding-left:1ex"><span class=3D"">Le 2015-07-20 14:58,=
 Tim Panton a =C3=A9crit :<br>
&gt; If I understand correctly we bind to each and every interface<br>
&gt; that is up and send out packets to all remote candidates down _every_ =
interface,<br>
&gt; ignoring any preferences set by the admin in the local routing table w=
hich<br>
&gt; were set to ensure (say) only critical traffic is sent down certain in=
terfaces.<br>
<br>
</span>I have difficulty understanding what it is that&#39;s bugging you. I=
s it the<br>
signalling payload (i.e., sending the list of address), or is it the<br>
connectivity checks that are done on all candidate pairs?<br></blockquote><=
div><br></div><div>Unfortunately we are mixing two probably separate concer=
ns:</div><div><br></div><div>1) ICE gathering and possible sensitive IP lea=
k when connecting the STUN/TURN server (decided by the web app) from all th=
e interfaces (those that of course have a route to the server).</div><div><=
br></div><div>2) Same issue when the IPs are announced in the SDP and ICE c=
onnectivity checks with the remote peer are performed.</div><div><br></div>=
<div>We should focus :)</div><div><br></div><div><br></div><div>=C2=A0</div=
><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1=
px #ccc solid;padding-left:1ex"><br></blockquote><blockquote class=3D"gmail=
_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:=
1ex"><span class=3D"">
&gt; Likewise my fantastically expensive BGAN link which is used for<br>
&gt; emergencies only will suddenly start carrying STUN requests irrespecti=
ve of how I configure my<br>
&gt; routes ?<br>
<br>
</span>Only if it&#39;s possible to route the packet to the STUN server ove=
r the<br>
BGAN link. There must be a route.</blockquote><div><br></div><div>=C2=A0Yes=
, but the point is that one may have enabled a secondary route (a default r=
oute via eth2) but prefer to use another route as default (eth1). The brows=
er will not respect that because it will try both eth1 and eth2.</div></div=
><br><br><br clear=3D"all"><div><br></div>-- <br><div class=3D"gmail_signat=
ure">I=C3=B1aki Baz Castillo<br>&lt;<a href=3D"mailto:ibc@aliax.net" target=
=3D"_blank">ibc@aliax.net</a>&gt;</div>
</div></div>

--001a1140c552d0808b051b4ebf8f--


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> On 20 Jul 2015, at 14:26, Simon Perreault <sperreault@jive.com> wrote:
>=20
> Le 2015-07-20 14:58, Tim Panton a =E9crit :
>> If I understand correctly we bind to each and every interface
>> that is up and send out packets to all remote candidates down _every_ =
interface,
>> ignoring any preferences set by the admin in the local routing table =
which
>> were set to ensure (say) only critical traffic is sent down certain =
interfaces.
>=20
> I have difficulty understanding what it is that's bugging you. Is it =
the
> signalling payload (i.e., sending the list of address), or is it the
> connectivity checks that are done on all candidate pairs?

I=92m not worried about the signalling payload (i.e. the list of local =
interface addresses -=20
except of course the mac-based ipv6 auto conf one).
I am worried about what can be learnt from /caused by sending packets =
down interfaces
when the sysadmin isn=92t expecting it.

(In the end some of those learnings may end up in signalling as =
candidates too).

>=20
>> I can think of some financial institutions who=92s IDSs will go off =
when packets
>> to public external addresses start turning up on internal LANS . =
(hopefully they=20
>> will get dropped at the egress router, but by then the IDS will have =
paged someone).
>=20
> But that's not going to happen unless there's a route. I think there
> could be a misunderstanding here. For example, if there's a VPN with a
> 10.0.0.0/8 route but no default route, the ICE client will *not* send =
a
> request to a TURN server listening on 192.0.2.1 over the VPN =
interface.
> That would make no sense whatsoever.

And yet that=92s what I read Martin to say :=20
"> The point is that you don't even choose the interface. The OS will do =
for you.
The OS can - and frequently does - get that wrong. The default route can =
fail when another might succeed."

I read that as 'another interface=92 - but if he means  =91another =
route=92 then I=92m happier - I think.
Can @justin or @martin confirm what the browsers currently do ?


>=20
>> Likewise my fantastically expensive BGAN link which is used for
>> emergencies only will suddenly start carrying STUN requests =
irrespective of how I configure my
>> routes ?
>=20
> Only if it's possible to route the packet to the STUN server over the
> BGAN link. There must be a route.

There must be a _locally_ configured route - It doesn=92t matter if the =
egress router at the far end of the BGAN drops=20
the STUN packet I=92ve still wasted my money :-(

>=20
> Simon

Tim.


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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Mon, 20 Jul 2015 15:51:02 +0200
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To: Simon Perreault <sperreault@jive.com>
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2015-07-20 15:32 GMT+02:00 Simon Perreault <sperreault@jive.com>:

> > The fact that the client announces its (probably) private VPN addresses
> > to the remote peer does not mean that it also reveals the associated
> > public reflexive IP. That just happens if the client performs STUN
> > procedures over that interface, which is exactly the issue being
> > discussed (at least right now).
>
> If you can find a server-reflexive candidate on the VPN interface, it
> means you can talk to the Internet using that interface, using that
> server-reflexive IP address. How is that private knowledge at all? How
> do you expect to keep your server-reflexive address private but still
> use it for communication?
>

Ok, let's simplify this subject:

- Computer with two default routes (A and B) and a private network route
(C).
- A is the preferred default route one so has priority in the routing table=
.
- Every packet whose destination is not C network should be sent over A.
...
- or not? should a specific app override it?

 This seems more a moral issue.




> The routing table is *not* overridden by explicitly binding to an
> interface.
>

I meant "using a secondary default route". Sorry for the confusion.


> >     That is a valid concern, and the answer to it is MIF.
> >
> >
> > Sorry, what does MIF stand for?
>
> https://datatracker.ietf.org/wg/mif/documents/
>
> The MIF WG is working on exactly this kind of problem. They are
> developing an API for solving precisely this kind of interface
> preference problem. IMHO ICE could perhaps be extended to make use of
> this API. But I see it as "just" an enhancement.


Good to know. Thanks.



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

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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">2015=
-07-20 15:32 GMT+02:00 Simon Perreault <span dir=3D"ltr">&lt;<a href=3D"mai=
lto:sperreault@jive.com" target=3D"_blank">sperreault@jive.com</a>&gt;</spa=
n>:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-=
left:1px #ccc solid;padding-left:1ex"><span class=3D"">&gt; The fact that t=
he client announces its (probably) private VPN addresses<br>
&gt; to the remote peer does not mean that it also reveals the associated<b=
r>
&gt; public reflexive IP. That just happens if the client performs STUN<br>
&gt; procedures over that interface, which is exactly the issue being<br>
&gt; discussed (at least right now).<br>
<br>
</span>If you can find a server-reflexive candidate on the VPN interface, i=
t<br>
means you can talk to the Internet using that interface, using that<br>
server-reflexive IP address. How is that private knowledge at all? How<br>
do you expect to keep your server-reflexive address private but still<br>
use it for communication?<br></blockquote><div><br></div><div>Ok, let&#39;s=
 simplify this subject:</div><div><br></div><div>- Computer with two defaul=
t routes (A and B) and a private network route (C).</div><div>- A is the pr=
eferred default route one so has priority in the routing table.</div><div>-=
 Every packet whose destination is not C network should be sent over A.</di=
v><div>...</div><div>- or not? should a specific app override it?</div><div=
><br></div><div>=C2=A0This seems more a moral issue.</div><div><br></div><d=
iv><br></div><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"ma=
rgin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">The routing ta=
ble is *not* overridden by explicitly binding to an interface.<br></blockqu=
ote><div><br></div><div>I meant &quot;using a secondary default route&quot;=
. Sorry for the confusion.</div><div><br></div><blockquote class=3D"gmail_q=
uote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1e=
x"><span class=3D""><br>&gt;=C2=A0 =C2=A0 =C2=A0That is a valid concern, an=
d the answer to it is MIF.<br>
&gt;<br>
&gt;<br>
&gt; Sorry, what does MIF stand for?<br>
<br>
</span><a href=3D"https://datatracker.ietf.org/wg/mif/documents/" rel=3D"no=
referrer" target=3D"_blank">https://datatracker.ietf.org/wg/mif/documents/<=
/a><br>
<br>
The MIF WG is working on exactly this kind of problem. They are<br>
developing an API for solving precisely this kind of interface<br>
preference problem. IMHO ICE could perhaps be extended to make use of<br>
this API. But I see it as &quot;just&quot; an enhancement.</blockquote><div=
><br></div><div>Good to know. Thanks.=C2=A0</div></div><br><br clear=3D"all=
"><div><br></div>-- <br><div class=3D"gmail_signature">I=C3=B1aki Baz Casti=
llo<br>&lt;<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.net=
</a>&gt;</div>
</div></div>

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Mon, 20 Jul 2015 15:56:10 +0200
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To: Tim Panton <tim@phonefromhere.com>
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2015-07-20 15:45 GMT+02:00 Tim Panton <tim@phonefromhere.com>:

> And yet that=E2=80=99s what I read Martin to say :
> "> The point is that you don't even choose the interface. The OS will do
> for you.
> The OS can - and frequently does - get that wrong. The default route can
> fail when another might succeed."
>
> I read that as 'another interface=E2=80=99 - but if he means  =E2=80=98an=
other route=E2=80=99 then
> I=E2=80=99m happier - I think.
> Can @justin or @martin confirm what the browsers currently do ?
>

I can confirm (OSX) that during ICE gathering STUN requests to the STUN
server are not sent over interfaces not belonging to any network route that
can reach the STUN/TURN server. This is, network routes are "respected".
The issue is that all the existing routes capables of reaching the STUN
server are tested (instead of the first one).

Note that we should avoid the term "default route" here as it stands for
0.0.0.0. It may happen that the webrtc app uses a private STUN server which
is just reachable via a VPN interface (and its specific route to reach the
VPN network in which the STUN server is placed) so in this case the default
route should not be tested because there is a route with higher priority to
reach the server.


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

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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">2015=
-07-20 15:45 GMT+02:00 Tim Panton <span dir=3D"ltr">&lt;<a href=3D"mailto:t=
im@phonefromhere.com" target=3D"_blank">tim@phonefromhere.com</a>&gt;</span=
>:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-l=
eft:1px #ccc solid;padding-left:1ex"><div id=3D":k6y" class=3D"a3s" style=
=3D"overflow:hidden">And yet that=E2=80=99s what I read Martin to say :<br>
<span class=3D"">&quot;&gt; The point is that you don&#39;t even choose the=
 interface. The OS will do for you.<br>
The OS can - and frequently does - get that wrong. The default route can fa=
il when another might succeed.&quot;<br>
<br>
</span>I read that as &#39;another interface=E2=80=99 - but if he means=C2=
=A0 =E2=80=98another route=E2=80=99 then I=E2=80=99m happier - I think.<br>
Can @justin or @martin confirm what the browsers currently do ?<br></div></=
blockquote></div><br>I can confirm (OSX) that during ICE gathering STUN req=
uests to the STUN server are not sent over interfaces not belonging to any =
network route that can reach the STUN/TURN server. This is, network routes =
are &quot;respected&quot;. The issue is that all the existing routes capabl=
es of reaching the STUN server are tested (instead of the first one).</div>=
<div class=3D"gmail_extra"><br></div><div class=3D"gmail_extra">Note that w=
e should avoid the term &quot;default route&quot; here as it stands for 0.0=
.0.0. It may happen that the webrtc app uses a private STUN server which is=
 just reachable via a VPN interface (and its specific route to reach the VP=
N network in which the STUN server is placed) so in this case the default r=
oute should not be tested because there is a route with higher priority to =
reach the server.<br><br clear=3D"all"><div><br></div>-- <br><div class=3D"=
gmail_signature">I=C3=B1aki Baz Castillo<br>&lt;<a href=3D"mailto:ibc@aliax=
.net" target=3D"_blank">ibc@aliax.net</a>&gt;</div>
</div></div>

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Date: Mon, 20 Jul 2015 16:00:30 +0200
From: Simon Perreault <sperreault@jive.com>
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To: Tim Panton <tim@phonefromhere.com>
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Subject: Re: [rtcweb] Please require user consent for data channels
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Le 2015-07-20 15:26, Simon Perreault a écrit :
>> I can think of some financial institutions who’s IDSs will go off when packets
>> > to public external addresses start turning up on internal LANS . (hopefully they 
>> > will get dropped at the egress router, but by then the IDS will have paged someone).
> But that's not going to happen unless there's a route. I think there
> could be a misunderstanding here. For example, if there's a VPN with a
> 10.0.0.0/8 route but no default route, the ICE client will *not* send a
> request to a TURN server listening on 192.0.2.1 over the VPN interface.
> That would make no sense whatsoever.

Just to be sure I am not going crazy, I just wrote this test:

> #include <sys/socket.h>
> #include <sys/types.h>
> 
> #include <netinet/in.h>
> #include <string.h>
> 
> int
> main(int argc, char *argv[])
> {
>         int                      s;
>         struct sockaddr_in       sin;
>         char                     foo[] = "bar";
> 
>         s = socket(AF_INET, SOCK_DGRAM, IPPROTO_UDP);
> 
>         sin.sin_family = AF_INET;
>         sin.sin_addr.s_addr = inet_addr("192.168.2.128");
>         sin.sin_port = htons(1234);
>         memset(&sin.sin_zero, 0, sizeof(sin.sin_zero));
>         bind(s, (struct sockaddr *)&sin, sizeof(sin));
> 
>         sin.sin_family = AF_INET;
>         sin.sin_addr.s_addr = inet_addr("192.168.2.200");
>         sin.sin_port = htons(53);
>         memset(&sin.sin_zero, 0, sizeof(sin.sin_zero));
>         sendto(s, foo, sizeof(foo), 0, (struct sockaddr *)&sin, sizeof(sin));
> 
>         sin.sin_family = AF_INET;
>         sin.sin_addr.s_addr = inet_addr("8.8.8.8");
>         sin.sin_port = htons(53);
>         memset(&sin.sin_zero, 0, sizeof(sin.sin_zero));
>         sendto(s, foo, sizeof(foo), 0, (struct sockaddr *)&sin, sizeof(sin));
> 
>         return (0);
> }

I ran it on this system:

> sperreault@debian:~$ ip a
> 1: lo: <LOOPBACK,UP,LOWER_UP> mtu 16436 qdisc noqueue state UNKNOWN
>     link/loopback 00:00:00:00:00:00 brd 00:00:00:00:00:00
>     inet 127.0.0.1/8 scope host lo
>     inet6 ::1/128 scope host
>        valid_lft forever preferred_lft forever
> 2: eth0: <BROADCAST,MULTICAST,UP,LOWER_UP> mtu 1500 qdisc pfifo_fast state UP qlen 1000
>     link/ether 00:0c:29:66:b9:2c brd ff:ff:ff:ff:ff:ff
>     inet 192.168.2.128/24 brd 192.168.2.255 scope global eth0
>     inet6 fe80::20c:29ff:fe66:b92c/64 scope link
>        valid_lft forever preferred_lft forever
> sperreault@debian:~$ ip r
> 192.168.2.0/24 dev eth0  proto kernel  scope link  src 192.168.2.128
> sperreault@debian:~$

Strace's output is:

> bind(3, {sa_family=AF_INET, sin_port=htons(1234), sin_addr=inet_addr("192.168.2.128")}, 16) = 0
> sendto(3, "bar\0", 4, 0, {sa_family=AF_INET, sin_port=htons(53), sin_addr=inet_addr("192.168.2.200")}, 16) = 4
> sendto(3, "bar\0", 4, 0, {sa_family=AF_INET, sin_port=htons(53), sin_addr=inet_addr("8.8.8.8")}, 16) = -1 ENETUNREACH (Network is unreachable)

This confirms that bind() does not override the routing table.

Simon


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From: Daniel Roesler <diafygi@gmail.com>
To: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
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On Jul 20, 2015 6:45 AM, "I=C3=B1aki Baz Castillo" <ibc@aliax.net> wrote:
>
> Unfortunately we are mixing two probably separate concerns:
>
> 1) ICE gathering and possible sensitive IP leak when connecting the
STUN/TURN server (decided by the web app) from all the interfaces (those
that of course have a route to the server).
>

A gentle reminder to stop referring to this as a "possible" sensitive IP
leak. It *is currently* a sensitive IP leak, and is already being exploited
in the wild[1].

[1]: https://webrtchacks.com/dear-ny-times/

Daniel

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<p dir=3D"ltr"><br>
On Jul 20, 2015 6:45 AM, &quot;I=C3=B1aki Baz Castillo&quot; &lt;<a href=3D=
"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt; wrote:<br>
&gt;<br>
&gt; Unfortunately we are mixing two probably separate concerns:<br>
&gt;<br>
&gt; 1) ICE gathering and possible sensitive IP leak when connecting the ST=
UN/TURN server (decided by the web app) from all the interfaces (those that=
 of course have a route to the server).<br>
&gt;</p>
<p dir=3D"ltr">A gentle reminder to stop referring to this as a &quot;possi=
ble&quot; sensitive IP leak. It *is currently* a sensitive IP leak, and is =
already being exploited in the wild[1].</p>
<p dir=3D"ltr">[1]: <a href=3D"https://webrtchacks.com/dear-ny-times/">http=
s://webrtchacks.com/dear-ny-times/</a></p>
<p dir=3D"ltr">Daniel<br>
</p>

--001a11353652a5090b051b4f1180--


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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Mon, 20 Jul 2015 16:09:27 +0200
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2015-07-20 16:00 GMT+02:00 Simon Perreault <sperreault@jive.com>:

> > bind(3, {sa_family=3DAF_INET, sin_port=3Dhtons(1234),
> sin_addr=3Dinet_addr("192.168.2.128")}, 16) =3D 0
> > sendto(3, "bar\0", 4, 0, {sa_family=3DAF_INET, sin_port=3Dhtons(53),
> sin_addr=3Dinet_addr("192.168.2.200")}, 16) =3D 4
> > sendto(3, "bar\0", 4, 0, {sa_family=3DAF_INET, sin_port=3Dhtons(53),
> sin_addr=3Dinet_addr("8.8.8.8")}, 16) =3D -1 ENETUNREACH (Network is
> unreachable)
>
> This confirms that bind() does not override the routing table.
>

Yes sure, I also confirmed that with some traces when Chrome performs ICE
gathering having my OSX computer a VPN interface with no route to reach the
STUN server.

So we are talking about whether the browser should just try the OS
preferred route capable of reaching the STUN server or all the capable ones=
.


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

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<div dir=3D"ltr"><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">=
2015-07-20 16:00 GMT+02:00 Simon Perreault <span dir=3D"ltr">&lt;<a href=3D=
"mailto:sperreault@jive.com" target=3D"_blank">sperreault@jive.com</a>&gt;<=
/span>:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bor=
der-left:1px #ccc solid;padding-left:1ex"><div id=3D":jy0" class=3D"a3s" st=
yle=3D"overflow:hidden">&gt; bind(3, {sa_family=3DAF_INET, sin_port=3Dhtons=
(1234), sin_addr=3Dinet_addr(&quot;192.168.2.128&quot;)}, 16) =3D 0<br>
&gt; sendto(3, &quot;bar\0&quot;, 4, 0, {sa_family=3DAF_INET, sin_port=3Dht=
ons(53), sin_addr=3Dinet_addr(&quot;192.168.2.200&quot;)}, 16) =3D 4<br>
&gt; sendto(3, &quot;bar\0&quot;, 4, 0, {sa_family=3DAF_INET, sin_port=3Dht=
ons(53), sin_addr=3Dinet_addr(&quot;8.8.8.8&quot;)}, 16) =3D -1 ENETUNREACH=
 (Network is unreachable)<br>
<br>
This confirms that bind() does not override the routing table.</div></block=
quote></div><br>Yes sure, I also confirmed that with some traces when Chrom=
e performs ICE gathering having my OSX computer a VPN interface with no rou=
te to reach the STUN server.</div><div class=3D"gmail_extra"><br></div><div=
 class=3D"gmail_extra">So we are talking about whether the browser should j=
ust try the OS preferred route capable of reaching the STUN server or all t=
he capable ones.<br><br clear=3D"all"><div><br></div>-- <br><div class=3D"g=
mail_signature">I=C3=B1aki Baz Castillo<br>&lt;<a href=3D"mailto:ibc@aliax.=
net" target=3D"_blank">ibc@aliax.net</a>&gt;</div>
</div></div>

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Mon, 20 Jul 2015 16:11:19 +0200
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To: Daniel Roesler <diafygi@gmail.com>
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2015-07-20 16:08 GMT+02:00 Daniel Roesler <diafygi@gmail.com>:

> > 1) ICE gathering and possible sensitive IP leak when connecting the
> STUN/TURN server (decided by the web app) from all the interfaces (those
> that of course have a route to the server).
> >
>
> A gentle reminder to stop referring to this as a "possible" sensitive IP
> leak. It *is currently* a sensitive IP leak, and is already being exploit=
ed
> in the wild[1].
>
> [1]: https://webrtchacks.com/dear-ny-times/
>

I just added "possible" because that is the original subject discussed in
this thread, and I didn't want to take part on it :)



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

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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">2015=
-07-20 16:08 GMT+02:00 Daniel Roesler <span dir=3D"ltr">&lt;<a href=3D"mail=
to:diafygi@gmail.com" target=3D"_blank">diafygi@gmail.com</a>&gt;</span>:<b=
r><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:=
1px #ccc solid;padding-left:1ex"><span class=3D""><p dir=3D"ltr">&gt; 1) IC=
E gathering and possible sensitive IP leak when connecting the STUN/TURN se=
rver (decided by the web app) from all the interfaces (those that of course=
 have a route to the server).<br>
&gt;</p>
</span><p dir=3D"ltr">A gentle reminder to stop referring to this as a &quo=
t;possible&quot; sensitive IP leak. It *is currently* a sensitive IP leak, =
and is already being exploited in the wild[1].</p>
<p dir=3D"ltr">[1]: <a href=3D"https://webrtchacks.com/dear-ny-times/" targ=
et=3D"_blank">https://webrtchacks.com/dear-ny-times/</a></p></blockquote></=
div><br>I just added &quot;possible&quot; because that is the original subj=
ect discussed in this thread, and I didn&#39;t want to take part on it :)</=
div><div class=3D"gmail_extra"><br><br clear=3D"all"><div><br></div>-- <br>=
<div class=3D"gmail_signature">I=C3=B1aki Baz Castillo<br>&lt;<a href=3D"ma=
ilto:ibc@aliax.net" target=3D"_blank">ibc@aliax.net</a>&gt;</div>
</div></div>

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Le 2015-07-20 16:09, IÃ±aki Baz Castillo a Ã©crit :
> So we are talking about whether the browser should just try the OS
> preferred route capable of reaching the STUN server or all the capable ones.

Correct. Thanks for re-focusing the discussion.

As for this precise question, I don't have an opinion. My intuition is
it would perhaps make sense to reflect routing priorities into ICE
prioritization or nomination criteria. But that would be very
implementation-specific.

Simon


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From: Roman Shpount <roman@telurix.com>
To: Simon Perreault <sperreault@jive.com>
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Subject: Re: [rtcweb] Please require user consent for data channels
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--047d7bd752e4552ce1051b4f2d2d
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Just to be completely clear, we are talking about the metric parameter in
the route. If I set a lower metric on the route to some destination (or if
OS sets a lower metric on the route to some destination), I expect this
route to be used. On some operating systems, this can be overwritten by
explicitly binding to a particular IP address (interface). This breaks
things and is somewhat unexpected to the system administrator. This is also
something that only happens in case of ICE. No other browser based protocol
would send data over a route with higher metric.

Regards,
_____________
Roman Shpount

--047d7bd752e4552ce1051b4f2d2d
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<div dir=3D"ltr">Just to be completely clear, we are talking about the metr=
ic parameter in the route. If I set a lower metric on the route to some des=
tination (or if OS sets a lower metric on the route to some destination), I=
 expect this route to be used. On some operating systems, this can be overw=
ritten by explicitly binding to a particular IP address (interface). This b=
reaks things and is somewhat unexpected to the system administrator. This i=
s also something that only happens in case of ICE. No other browser based p=
rotocol would send data over a route with higher metric.<div><br></div><div=
>Regards,<div class=3D"gmail_extra"><div><div class=3D"gmail_signature">___=
__________<br>Roman Shpount</div></div>
<br></div></div></div>

--047d7bd752e4552ce1051b4f2d2d--


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From: Jonathan Lennox <jonathan@vidyo.com>
To: Roman Shpount <roman@telurix.com>
Thread-Topic: [rtcweb] Please require user consent for data channels
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References: <CA+65OspMD_PVjk0BXh7t4LtjmFDcDatoeNjFQOO_OVtC-Br+OA@mail.gmail.com> <CAOJ7v-24VCW6kkn7LOLkqZzhYEU0r=nmd_F7Zns1rnyqKN6xAg@mail.gmail.com> <55A95364.2070806@gmail.com> <CAOJ7v-3t9BQabR2e4EHs4G0Sec4sU9DFC2aiSXXYrat+an+RYg@mail.gmail.com> <55A96DA3.1040907@gmail.com> <CAOJ7v-1ui7349NzK6NZNRHPbnHWZajctk4cDgMKqRZSv47EYdA@mail.gmail.com> <55A9860D.8030903@gmail.com> <CAOJ7v-3LGd32rnpFVW_U0s3+iVaJXsL4vt_YAo=cyp6YyOArdw@mail.gmail.com> <CALiegfmiS18Jux-kCgOhTKKiyGtMertj6xCegpFrox5NOf9EJg@mail.gmail.com> <CABkgnnW0Tmjqz823vKiF84_u6HasBJC7ERMYCO2HL_NPj5saTA@mail.gmail.com> <CALiegfkpbLy1QXxr-RRF0oOpVv1sWsFeab=vvC4iT4DnPtjKQw@mail.gmail.com> <CABkgnnVWcuhX2NjZgx87L+Uo6df6rEBWW73cxbaX3mu_VfHmCA@mail.gmail.com> <CALiegfkQWAn-jMrjhcDPA3rtowOPVk-S8z3c-jvjpNmjtf=3hA@mail.gmail.com> <CABkgnnWERM4oxozNCSvRf1o0Wm-d9Bjw=9B+xh_NJ+h6GfBJ6Q@mail.gmail.com> <CAD5OKxt-DZCGFECv2UYH8g0fD6EAk39cdpWD-CsN_ED6L4hfKg@mail.gmail.com> <CALiegf=zM54gNjgj65VYV5H3tS-iV5Kg0PrBeF7svYRYZ-JLPw@mail.gmail.com> <55AABE28.8070105@jive.com>
Date: Mon, 20 Jul 2015 10:28:59 -0400
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From: Roman Shpount <roman@telurix.com>
To: Simon Perreault <sperreault@jive.com>
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Subject: Re: [rtcweb] Please require user consent for data channels
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--001a113f911cca7a25051b4f5ba5
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On Sat, Jul 18, 2015 at 4:59 PM, Simon Perreault <sperreault@jive.com>
wrote:

> This is a concept that was explained during the ICE tutorial that was
> presented at IETF 92.
>
> Imagine the following multihomed host with three interfaces:
>
> - Wi-fi interface with default route.
> - Wired interface with default route.
> - VPN interface with default route.
>

On the multihomed host, all the default routes will have different metrics.
In some cases these metrics are automatically calculated by OS. In some
cases, they are set by the system administrator. In either case, ICE
ignores the metric value and tries to route over all interfaces. This is
unexpected and can be considered a violation of the local policy (i.e. if
administrator added a route with a lower metric, it expects it to be used).
No other browser protocol does this.


> Which one should you use to talk to the STUN/TURN server? If you bind to
> 0.0.0.0 then the kernel *will* pick one, but only because it *has to.
> All three interfaces are equivalent: they all have a default route and
> can a priori be used to talk to the server.
>

See above, this is not the case. All interfaces have a default route but
they are not equal. If they were equal, OS will round robin among them.

Being multihomed means multiple interfaces with a default route. There's
> no "going against the local policy" in this case when sending packets on
> any interface.
>

Once again, this is not the case. Metric value on the interface is local
policy and should be respected.


> The point is that in this case, all three interfaces can potentially
> yield different server-reflexive candidates, which is why they must be
> gathered separately. As for relayed candidates, they also need to be
> gathered separately because some interfaces may be faster for talking to
> the TURN server and it is useful to compare the candidates gathered on
> all 3 interfaces. Also nothing guarantees that you're allowed to reach
> the server server on any interface that has a default route, so you need
> to try them all.
>

This is a policy decision on the local computer. As a network administrator
I should be able to specified the preferred default route. It does not mean
that I should remove the default routes for all the other interfaces.
Setting a metric should be enough. On the other hand, if I prefer the
browser to pick the best connection on its own, I should be able to allow
this as well, but this is going to be a much less common case. Taking into
account potential privacy or financial implications of this choice, I think
the browser should error on the side of caution.

_____________
Roman Shpount

--001a113f911cca7a25051b4f5ba5
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On S=
at, Jul 18, 2015 at 4:59 PM, Simon Perreault <span dir=3D"ltr">&lt;<a href=
=3D"mailto:sperreault@jive.com" target=3D"_blank">sperreault@jive.com</a>&g=
t;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0px 0=
px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);borde=
r-left-style:solid;padding-left:1ex">This is a concept that was explained d=
uring the ICE tutorial that was<br>
presented at IETF 92.<br>
<br>
Imagine the following multihomed host with three interfaces:<br>
<br>
- Wi-fi interface with default route.<br>
- Wired interface with default route.<br>
- VPN interface with default route.<br></blockquote><div><br></div><div>On =
the multihomed host, all the default routes will have different metrics. In=
 some cases these metrics are automatically calculated by OS. In some cases=
, they are set by the system administrator. In either case, ICE ignores the=
 metric value and tries to route over all interfaces. This is unexpected an=
d can be considered a violation of the local policy (i.e. if administrator =
added a route with a lower metric, it expects it to be used). No other brow=
ser protocol does this.</div><div>=C2=A0</div><blockquote class=3D"gmail_qu=
ote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-co=
lor:rgb(204,204,204);border-left-style:solid;padding-left:1ex">Which one sh=
ould you use to talk to the STUN/TURN server? If you bind to<br>
0.0.0.0 then the kernel *will* pick one, but only because it *has to.<br>
All three interfaces are equivalent: they all have a default route and<br>
can a priori be used to talk to the server.<br></blockquote><div><br></div>=
<div>See above, this is not the case. All interfaces have a default route b=
ut they are not equal. If they were equal, OS will round robin among them.<=
/div><div><br></div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0=
px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);borde=
r-left-style:solid;padding-left:1ex">Being multihomed means multiple interf=
aces with a default route. There&#39;s<br>
no &quot;going against the local policy&quot; in this case when sending pac=
kets on<br>
any interface.<br></blockquote><div><br></div><div>Once again, this is not =
the case. Metric value on the interface is local policy and should be respe=
cted.</div><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"marg=
in:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,20=
4);border-left-style:solid;padding-left:1ex">The point is that in this case=
, all three interfaces can potentially<br>
yield different server-reflexive candidates, which is why they must be<br>
gathered separately. As for relayed candidates, they also need to be<br>
gathered separately because some interfaces may be faster for talking to<br=
>
the TURN server and it is useful to compare the candidates gathered on<br>
all 3 interfaces. Also nothing guarantees that you&#39;re allowed to reach<=
br>
the server server on any interface that has a default route, so you need<br=
>
to try them all.<br></blockquote><div><br></div><div>This is a policy decis=
ion on the local computer. As a network administrator I should be able to s=
pecified the preferred default route. It does not mean that I should remove=
 the default routes for all the other interfaces. Setting a metric should b=
e enough. On the other hand, if I prefer the browser to pick the best conne=
ction on its own, I should be able to allow this as well, but this is going=
 to be a much less common case. Taking into account potential privacy or fi=
nancial implications of this choice, I think the browser should error on th=
e side of caution.</div><div><br></div><div><div class=3D"gmail_signature">=
_____________<br>Roman Shpount</div></div><div>=C2=A0</div></div></div></di=
v>

--001a113f911cca7a25051b4f5ba5--


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References: <CA+65OspMD_PVjk0BXh7t4LtjmFDcDatoeNjFQOO_OVtC-Br+OA@mail.gmail.com> <CAOJ7v-3t9BQabR2e4EHs4G0Sec4sU9DFC2aiSXXYrat+an+RYg@mail.gmail.com> <55A96DA3.1040907@gmail.com> <CAOJ7v-1ui7349NzK6NZNRHPbnHWZajctk4cDgMKqRZSv47EYdA@mail.gmail.com> <55A9860D.8030903@gmail.com> <CAOJ7v-3LGd32rnpFVW_U0s3+iVaJXsL4vt_YAo=cyp6YyOArdw@mail.gmail.com> <CALiegfmiS18Jux-kCgOhTKKiyGtMertj6xCegpFrox5NOf9EJg@mail.gmail.com> <CABkgnnW0Tmjqz823vKiF84_u6HasBJC7ERMYCO2HL_NPj5saTA@mail.gmail.com> <CALiegfkpbLy1QXxr-RRF0oOpVv1sWsFeab=vvC4iT4DnPtjKQw@mail.gmail.com> <CABkgnnVWcuhX2NjZgx87L+Uo6df6rEBWW73cxbaX3mu_VfHmCA@mail.gmail.com> <CALiegfkQWAn-jMrjhcDPA3rtowOPVk-S8z3c-jvjpNmjtf=3hA@mail.gmail.com> <CABkgnnWERM4oxozNCSvRf1o0Wm-d9Bjw=9B+xh_NJ+h6GfBJ6Q@mail.gmail.com> <CAD5OKxt-DZCGFECv2UYH8g0fD6EAk39cdpWD-CsN_ED6L4hfKg@mail.gmail.com> <CALiegf=zM54gNjgj65VYV5H3tS-iV5Kg0PrBeF7svYRYZ-JLPw@mail.gmail.com> <55AABE28.8070105@jive.com> <913383AAA69FF945B8F946018B75898A47894E34@xmb-rcd-x10.cisco.com> <55AB4FEC.7050805@alvestrand.no>
Date: Mon, 20 Jul 2015 10:49:59 -0400
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From: Roman Shpount <roman@telurix.com>
To: Harald Alvestrand <harald@alvestrand.no>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Please require user consent for data channels
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--001a11c3b908e8380a051b4fa6a6
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On Sun, Jul 19, 2015 at 3:21 AM, Harald Alvestrand <harald@alvestrand.no>
wrote:

> Solutions we engineer MUST work in the current Internet.
>

I agree completely.


> +1 to Simon - the way systems are designed, implemented and deployed
> *now*, there may be multiple interfaces with multiple default addresses,
> and there's nothing intrinsically illegal, immoral or fattening about an
> application accessing them.
>

If a local system administrator set a lower metric to a particular default
(or any other route) vs all the other routes to a certain destination, this
route should be used to reach this destination. This is the normal behavior
for all other network communications from the browser (HTTP. WebSockets, or
even FTP or gopher). With ICE, browsers depart from this model. I
understand that this can enable browsers to set up a WebRTC media
connection where it would not be able to do so otherwise. On other hand
using secondary routes can produce some unexpected results, such as
exposing reflexive addresses on secondary interfaces or sending data over
LTE network when WiFi is available. I think using secondary routes should
only be enabled after user consent or if an appropriate browser setting is
enabled. In wast majority of cases, primary default route is sufficient to
set up the WebRTC connection. My assumption is that using secondary default
routes only improves the chances of connection in extremely small number of
cases. Based on the systems I have worked with, we are talking about less
then 1/10 of the percent of end users.

Regards,
_____________
Roman Shpount

--001a11c3b908e8380a051b4fa6a6
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_extra"><div><div class=3D"gmail_signat=
ure">On Sun, Jul 19, 2015 at 3:21 AM, Harald Alvestrand <span dir=3D"ltr">&=
lt;<a href=3D"mailto:harald@alvestrand.no" target=3D"_blank">harald@alvestr=
and.no</a>&gt;</span> wrote:<br></div></div><div class=3D"gmail_quote"><blo=
ckquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left=
-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;paddi=
ng-left:1ex">Solutions we engineer MUST work in the current Internet.<br></=
blockquote><div><br></div><div>I agree completely.</div><div>=C2=A0</div><b=
lockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-le=
ft-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;pad=
ding-left:1ex">+1 to Simon - the way systems are designed, implemented and =
deployed<br>
*now*, there may be multiple interfaces with multiple default addresses,<br=
>
and there&#39;s nothing intrinsically illegal, immoral or fattening about a=
n<br>
application accessing them.<br></blockquote><div><br></div><div>If a local =
system administrator set a lower metric to a particular default (or any oth=
er route) vs all the other routes to a certain destination, this route shou=
ld be used to reach this destination. This is the normal behavior for all o=
ther network communications from the browser (HTTP. WebSockets, or even FTP=
 or gopher). With ICE, browsers depart from this model. I understand that t=
his can enable browsers to set up a WebRTC media connection where it would =
not be able to do so otherwise. On other hand using secondary routes can pr=
oduce some unexpected results, such as exposing reflexive addresses on seco=
ndary interfaces or sending data over LTE network when WiFi is available. I=
 think using secondary routes should only be enabled after user consent or =
if an appropriate browser setting is enabled. In wast majority of cases, pr=
imary default route is sufficient to set up the WebRTC connection. My assum=
ption is that using secondary default routes only improves the chances of c=
onnection in extremely small number of cases. Based on the systems I have w=
orked with, we are talking about less then 1/10 of the percent of end users=
.</div><div><br></div><div>Regards,</div><div><div class=3D"gmail_signature=
">_____________<br>Roman Shpount</div></div><div>=C2=A0</div></div></div></=
div>

--001a11c3b908e8380a051b4fa6a6--


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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Mon, 20 Jul 2015 16:58:33 +0200
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To: Roman Shpount <roman@telurix.com>
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Subject: Re: [rtcweb] Please require user consent for data channels
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2015-07-20 16:49 GMT+02:00 Roman Shpount <roman@telurix.com>:

> If a local system administrator set a lower metric to a particular defaul=
t
> (or any other route) vs all the other routes to a certain destination, th=
is
> route should be used to reach this destination. This is the normal behavi=
or
> for all other network communications from the browser (HTTP. WebSockets, =
or
> even FTP or gopher). With ICE, browsers depart from this model. I
> understand that this can enable browsers to set up a WebRTC media
> connection where it would not be able to do so otherwise. On other hand
> using secondary routes can produce some unexpected results, such as
> exposing reflexive addresses on secondary interfaces or sending data over
> LTE network when WiFi is available. I think using secondary routes should
> only be enabled after user consent or if an appropriate browser setting i=
s
> enabled. In wast majority of cases, primary default route is sufficient t=
o
> set up the WebRTC connection. My assumption is that using secondary defau=
lt
> routes only improves the chances of connection in extremely small number =
of
> cases. Based on the systems I have worked with, we are talking about less
> then 1/10 of the percent of end users.
>

Agreed. Important concerns here are:

1) The app (browser) should respect the routing local policy. It means that
it should consider the metric/priority of the capable routes (as the
administrator or OS decided), or that it should just use the best one (but
this may also cause issues as stated in next bullet).

2) If the browser also uses a secondary capable route it may cause real mon=
ey
(if for example the secondary route goes through a LTE interface while the
primary route is WiFi based).


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

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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">2015=
-07-20 16:49 GMT+02:00 Roman Shpount <span dir=3D"ltr">&lt;<a href=3D"mailt=
o:roman@telurix.com" target=3D"_blank">roman@telurix.com</a>&gt;</span>:<br=
><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border=
-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;=
padding-left:1ex"><div>If a local system administrator set a lower metric t=
o a particular default (or any other route) vs all the other routes to a ce=
rtain destination, this route should be used to reach this destination. Thi=
s is the normal behavior for all other network communications from the brow=
ser (HTTP. WebSockets, or even FTP or gopher). With ICE, browsers depart fr=
om this model. I understand that this can enable browsers to set up a WebRT=
C media connection where it would not be able to do so otherwise. On other =
hand using secondary routes can produce some unexpected results, such as ex=
posing reflexive addresses on secondary interfaces or sending data over LTE=
 network when WiFi is available. I think using secondary routes should only=
 be enabled after user consent or if an appropriate browser setting is enab=
led. In wast majority of cases, primary default route is sufficient to set =
up the WebRTC connection. My assumption is that using secondary default rou=
tes only improves the chances of connection in extremely small number of ca=
ses. Based on the systems I have worked with, we are talking about less the=
n 1/10 of the percent of end users.</div><div></div></blockquote></div><br>=
Agreed. Important concerns here are:</div><div class=3D"gmail_extra"><br></=
div><div class=3D"gmail_extra">1) The app (browser) should respect the rout=
ing local policy. It means that it should consider the metric/priority of t=
he capable routes (as the administrator or OS decided), or that it should j=
ust use the best one (but this may also cause issues as stated in next bull=
et).</div><div class=3D"gmail_extra"><br></div><div class=3D"gmail_extra">2=
) If the browser also uses a secondary capable route it may cause=C2=A0<spa=
n style=3D"font-size:12.8000001907349px">real=C2=A0</span><span style=3D"fo=
nt-size:12.8000001907349px">money (if for example the secondary route goes =
through a LTE interface while the primary route is WiFi based).</span><span=
 style=3D"font-size:12.8000001907349px"><br></span><br clear=3D"all"><div><=
br></div>-- <br><div class=3D"gmail_signature">I=C3=B1aki Baz Castillo<br>&=
lt;<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.net</a>&gt;=
</div>
</div></div>

--94eb2c0888f6b08d9b051b4fc6ce--


From nobody Tue Jul 21 01:12:10 2015
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From: Bernard Aboba <bernard.aboba@gmail.com>
Date: Tue, 21 Jul 2015 10:11:46 +0200
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To: "Hutton, Andrew" <andrew.hutton@unify.com>
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Subject: Re: [rtcweb] JSEP and -return- ICE Candidate Policy
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Andrew said:

"It seems something is wrong here "

[BA] I do not believe that is true.  With respect to privacy, we have had
two parallel discussions, reflecting two distinct threats.  The original
discussion centered around API-based control of ICE candidate policy, but
the most recent one has been about malevolent JS that requires mitigation
outside of the API, such as browser settings that could be considered
"local".

On Mon, Jul 20, 2015 at 2:35 PM, Hutton, Andrew <andrew.hutton@unify.com>
wrote:

> https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-11#section-3.4.3 talks
> about the application having control over the ICE Candidate Policy
> (RTCIceTransportPolicy) but it also contains the following statement
> regarding local policy:
>
> "In addition, administrators may also wish to control the set of ICE
> candidates, and so the browser SHOULD also allow control via local policy,
> with the most restrictive policy prevailing".
>
> This seems to imply that the admin and application policies are
> controlling the same functionality in the browser but is this really true?
>
> https://tools.ietf.org/html/draft-ietf-rtcweb-return-00#section-4.3 also
> refers to to the ICE candidate policy section of JSEP and indicates states
> that:
>
> "Proxy configuration, including leakiness, maybe set by local policy
> ([I-D.ietf-rtcweb-jsep], "ICE candidate policy") or other mechanisms".
>
> It seems something is wrong here and probably JSEP should not be talking
> about the local policy as I assume this would not use the PeerConnection
> API and the -return- draft should not be referring to JSEP regarding this
> local policy as again this will not use the PC API.
>
> Andy
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<div dir=3D"ltr"><div>Andrew said: </div><div><br></div><div>&quot;It seems=
 something is wrong here &quot;</div><div><br></div><div>[BA] I do not beli=
eve that is true.=C2=A0 With respect to privacy, we have had two parallel d=
iscussions, reflecting two distinct threats.=C2=A0 The original discussion =
centered around API-based control of ICE candidate policy, but the most rec=
ent one has been about malevolent JS that requires mitigation outside of th=
e API, such as browser settings that could be considered &quot;local&quot;.=
=C2=A0 </div></div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote=
">On Mon, Jul 20, 2015 at 2:35 PM, Hutton, Andrew <span dir=3D"ltr">&lt;<a =
href=3D"mailto:andrew.hutton@unify.com" target=3D"_blank">andrew.hutton@uni=
fy.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"><a href=3D"h=
ttps://tools.ietf.org/html/draft-ietf-rtcweb-jsep-11#section-3.4.3" target=
=3D"_blank" rel=3D"noreferrer">https://tools.ietf.org/html/draft-ietf-rtcwe=
b-jsep-11#section-3.4.3</a> talks about the application having control over=
 the ICE Candidate Policy (RTCIceTransportPolicy) but it also contains the =
following statement regarding local policy:<br>
<br>
&quot;In addition, administrators may also wish to control the set of ICE c=
andidates, and so the browser SHOULD also allow control via local policy, w=
ith the most restrictive policy prevailing&quot;.<br>
<br>
This seems to imply that the admin and application policies are controlling=
 the same functionality in the browser but is this really true?<br>
<br>
<a href=3D"https://tools.ietf.org/html/draft-ietf-rtcweb-return-00#section-=
4.3" target=3D"_blank" rel=3D"noreferrer">https://tools.ietf.org/html/draft=
-ietf-rtcweb-return-00#section-4.3</a> also refers to to the ICE candidate =
policy section of JSEP and indicates states that:<br>
<br>
&quot;Proxy configuration, including leakiness, maybe set by local policy (=
[I-D.ietf-rtcweb-jsep], &quot;ICE candidate policy&quot;) or other mechanis=
ms&quot;.<br>
<br>
It seems something is wrong here and probably JSEP should not be talking ab=
out the local policy as I assume this would not use the PeerConnection API =
and the -return- draft should not be referring to JSEP regarding this local=
 policy as again this will not use the PC API.<br>
<br>
Andy<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" =
rel=3D"noreferrer">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div><br></div>

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To: Bernard Aboba <bernard.aboba@gmail.com>
Thread-Topic: [rtcweb] JSEP and -return- ICE Candidate Policy
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From: Justin Uberti <juberti@google.com>
Date: Tue, 21 Jul 2015 06:36:23 -0700
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To: "Hutton, Andrew" <andrew.hutton@unify.com>
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Subject: Re: [rtcweb] JSEP and -return- ICE Candidate Policy
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I think the confusion is regarding the use of RTCIceTransportPolicy.
Clearly that doesn't make sense in the RETURN context.

The point though is that the browser may have additional local policy,
separate from RTCIceTransportPolicy, that governs what candidates it
exposes.

On Tue, Jul 21, 2015 at 1:27 AM, Hutton, Andrew <andrew.hutton@unify.com>
wrote:

> On: 21 July 2015 10:12 Bernard Aboba Wrote:
> >
> > Andrew said:
> >
> > "It seems something is wrong here "
> >
> > [BA] I do not believe that is true.  With respect to privacy, we have
> > had two parallel discussions, reflecting two distinct threats.  The
> > original discussion centered around API-based control of ICE candidate
> > policy, but the most recent one has been about malevolent JS that
> > requires mitigation outside of the API, such as browser settings that
> > could be considered "local".
> >
>
> Understood the only thing that I think is wrong is that the -return-
> drafts refers to the ICE Candidate Policy in JSEP as the mechanism for
> controlling the local policy which as you say should be something outside
> of the API.
>
> Unless I misunderstood something we are in agreement.
>
> Andy
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<div dir=3D"ltr">I think the confusion is regarding the use of=C2=A0<span s=
tyle=3D"font-size:12.8px">RTCIceTransportPolicy. Clearly that doesn&#39;t m=
ake sense in the RETURN context.</span><div><span style=3D"font-size:12.8px=
"><br></span></div><div><span style=3D"font-size:12.8px">The point though i=
s that the browser may have additional local policy, separate from RTCIceTr=
ansportPolicy, that governs what candidates it exposes.</span></div></div><=
div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Tue, Jul 21, 20=
15 at 1:27 AM, Hutton, Andrew <span dir=3D"ltr">&lt;<a href=3D"mailto:andre=
w.hutton@unify.com" target=3D"_blank">andrew.hutton@unify.com</a>&gt;</span=
> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bo=
rder-left:1px #ccc solid;padding-left:1ex"><span class=3D"">On: 21 July 201=
5 10:12 Bernard Aboba Wrote:<br>
&gt;<br>
&gt; Andrew said:<br>
&gt;<br>
&gt; &quot;It seems something is wrong here &quot;<br>
&gt;<br>
&gt; [BA] I do not believe that is true.=C2=A0 With respect to privacy, we =
have<br>
&gt; had two parallel discussions, reflecting two distinct threats.=C2=A0 T=
he<br>
&gt; original discussion centered around API-based control of ICE candidate=
<br>
&gt; policy, but the most recent one has been about malevolent JS that<br>
&gt; requires mitigation outside of the API, such as browser settings that<=
br>
&gt; could be considered &quot;local&quot;.<br>
&gt;<br>
<br>
</span>Understood the only thing that I think is wrong is that the -return-=
 drafts refers to the ICE Candidate Policy in JSEP as the mechanism for con=
trolling the local policy which as you say should be something outside of t=
he API.<br>
<br>
Unless I misunderstood something we are in agreement.<br>
<span class=3D""><br>
Andy<br>
_______________________________________________<br>
rtcweb mailing list<br>
</span><a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div><br></div>

--001a113f2782ace79c051b62bebe--


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On Tue, Jul 21, 2015 at 7:11 AM, Hutton, Andrew <andrew.hutton@unify.com>
wrote:

> On: 21 July 2015 15:36 Justin Uberti Wrote:
> >
> > I think the confusion is regarding the use of RTCIceTransportPolicy.
> > Clearly that doesn't make sense in the RETURN context.
>
> Exactly the current -return- draft refers to RTCIceTransportPolicy,
>

Well, not exactly - it refers to the "ICE Candidate Policy" section in
JSEP, which says that there may be additional local policy. But I agree
that this is confusing and could be better explained in JSEP.


>
> >
> > The point though is that the browser may have additional local policy,
> > separate from RTCIceTransportPolicy, that governs what candidates it
> > exposes.
>
> Agreed.
>
> Andy
>

--001a1140c4263886e1051b635ca4
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Tue, Jul 21, 2015 at 7:11 AM, Hutton, Andrew <span dir=3D"ltr">&lt;<=
a href=3D"mailto:andrew.hutton@unify.com" target=3D"_blank">andrew.hutton@u=
nify.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"><span cl=
ass=3D"">On: 21 July 2015 15:36 Justin Uberti Wrote:<br>
&gt;<br>
&gt; I think the confusion is regarding the use of=C2=A0RTCIceTransportPoli=
cy.<br>
&gt; Clearly that doesn&#39;t make sense in the RETURN context.<br>
<br>
</span>Exactly the current -return- draft refers to RTCIceTransportPolicy,<=
br></blockquote><div><br></div><div>Well, not exactly - it refers to the &q=
uot;ICE Candidate Policy&quot; section in JSEP, which says that there may b=
e additional local policy. But I agree that this is confusing and could be =
better explained in JSEP.</div><div>=C2=A0</div><blockquote class=3D"gmail_=
quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1=
ex">
<span class=3D""><br>
&gt;<br>
&gt; The point though is that the browser may have additional local policy,=
<br>
&gt; separate from RTCIceTransportPolicy, that governs what candidates it<b=
r>
&gt; exposes.<br>
<br>
</span>Agreed.<br>
<br>
Andy<br>
</blockquote></div><br></div></div>

--001a1140c4263886e1051b635ca4--


From nobody Thu Jul 23 01:38:23 2015
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RTCWEB WG Participants,

If you are planning to participate in the RTCWEB sessions here at IETF =
93  =97 either locally in Prague or as a remote participant =97 we want =
to make sure that you are aware that the IETF is providing remote =
participants with a new way to ask questions or make comments. In =
addition to using the Jabber room, for the RTCWEB session there is also =
the opportunity for remote participants to enter a virtual queue and ask =
questions directly into the meeting room.=20

This experimental queue was used in a few sessions at IETF 92, so you =
may have already seen it in action. Some improvements have been made =
since the last meeting, but the concept is the same. There will be two =
queues for the RTCWEB sessions =97 a virtual queue and an actual =
(in-room) queue.  Remote attendees will log into the Meetecho platform =
and will have a virtual mic line that they can enter if they have a =
question or comment. In-room participants will continue to use normal =
mic lines.=20

Instructions for remote participants are at =
http://ietf93.conf.meetecho.com/index.php/Remote_Participation.

Join the Meetecho session for RTCWEB at 13:00 today using this link: =
http://www.meetecho.com/ietf93/rtcweb

Join the Meetecho session for RTCWEB at 11:50 tomorrow using this link: =
http://www.meetecho.com/ietf93/rtcweb_II

Regards,
Alexa



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The Third International Conference on Green Computing, Technology and 
Innovation (ICGCTI2015)

Universiti Putra Malaysia, Selangor, Malaysia
8-10 December 2015
http://sdiwc.net/conferences/icgcti2015/

The proposed conference on the above theme will held over three days, 
with presentations delivered by researchers from the international 
community, including presentations from keynote speakers and 
state-of-the-art lectures. ICGCTI2015 aims to enable researchers build 
connections between different digital applications.

The conference welcomes papers on the following (but not limited to) 
research topics:

- Benefits of, and barriers to, adopting greener IT practices
- Carbon metering and user feedback
- Climate and ecosystem monitoring
- Energy harvesting, storage, and recycling
- Energy-aware high performance computing and applications
- Energy-aware software
- Energy-efficient network services and operations
- Green IT metrics, maturity models, standards, and regulations
- Green computing models, methodologies and paradigms
- Green networking and communication
- Life-cycle analysis of IT equipment
- Management and profiling tools for energy efficient systems
- Modeling-representations, simulation and validation for energy 
consumption optimization problems
- Online dynamic optimization for energy efficient systems
- Power-aware algorithms and protocols
- Power-efficient delivery and cooling
- Renewable energy models and prediction
- Smart buildings and urban development
- Smart homes, buildings, offices, streets
- Stability of smart energy systems
- Using IT to reduce carbon emissions
- Carbon management policies and ecology- related issues with ICT
- Characterization, metrics, and modeling
- Creating green awareness using IT
- Energy-aware computing
- Energy-aware large scale distributed systems, such as Grids, Clouds 
and service computing
- Energy-efficient mass data storage and processing
- Governmentsâ€™ roles in fostering and enforcing green initiatives
- Green business process reengineering and management
- Green design, manufacture, use, disposal, and recycling of computers 
and communication systems
- Green software engineering
- Low-power electronics and systems
- Matching energy supply and demand
- Network design optimization
- Optimization of energy-efficient protocols
- Power-aware software and hardware
- Reliability, thermal behavior and control
- Robustness and performance guarantees
- Smart grid and microgrids
- Smart transportation and manufacturing
- Sustainable computing

Researchers are encouraged to submit their work electronically. All 
papers will be fully refereed by a minimum of two specialized referees. 
Before final acceptance, all referees comments must be considered.

Important Dates
==============
Submission Deadline	: November 8, 2015
Notification of Acceptance : 2-4 weeks from the submission date
Final Notification	: November 20, 2015
Camera Ready Deadline	: November 28, 2015
Registration Deadline	: November 28, 2015
Conference Dates	: December 8-10, 2015

Drop us an email at icgcti15@sdiwc.net


From nobody Thu Jul 23 09:15:50 2015
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Subject: [rtcweb] Assigning MediaStreamTracks to media sections (with LS)
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Warning: This may be pretty incomprehensible to anyone who wasn't in the
room for the discussion of the LS handling in Prague.

Initial offer handling is pretty trivial. The one who creates the offer
creates the LS groups (or not).
The below is the algorithm for the initial answer. The algorithm for
subsequent answers will have to take into account what assignments are
already in effect (this is the same as for the before-LS case).

When generating an initial answer, execute the following algorithm:

A - Make a list of all the LS groups in the offer. If any media section
is shared by two LS groups, merge the groups (they are all
synchronized). If there are media sections not in any LS group, put them
in the last group in the list.

B - Make a list of all the MediaStreamTracks the answererer wants to
send, grouped by MediaStream the MediaStreamTrack belongs to. If any
track occurs in multiple MediaStreams, merge the groups.

C - For each group "SG" in the MediaStreamTrack list:
   For each group "LSG" in the LS group list:
      If all MediaStreamTracks in SG can be sent using the media
sections in LSG:
          Mark "SG" as being sent using "LSG"
          Mark all media sections that are assigned for use by "SG" as
"in use".
          If all media sections in LSG are "in use":
             Remove "LSG" from the list of LS groups
          endif
      endif
   endfor
   If "SG" is not marked as being sendable
      set "must negotiate" flag
   endif
endfor

D - Generate the answer using the assignment above. All LS groups in the
answer are
the same as those in the offer.

If the "must negotiate" flag is set:
   When the PC reaches a stable negotiation state, fire a
"negotiationneeded" event.
endif

Note: When there are no LS groups, the algorithm trivially reduces to
what we have already (there is only one group in that case). We still
have to figure out what to do with leftover tracks, which may cause the
need to send "negotiationneeded". So the only added complexity is the
consideration of multiple LS groups.

Note: When there is only one LS group consisting of an audio stream and
a video stream, and the answerer only wants to send an audio stream and
a video stream, this algorithm produces the right result. This may be
its most important feature.

Note: This algorithm doesn't generate LS groups for stuff that is sent
using non-grouped
media sections from the offer. If that's desired, we need to set the
"must negotiate" flag
when using media sections from that group.

Note: For subsequent offers, if the offerer starts changing the LS
groups, this will invalidate the assignment of MediaStreamTracks to
media sections - something we have not contemplated in any other
context. I suggest that we forbid changing LS groups at the offerer (he
can only add new ones), and reject all modified LS groups if a
subsequent offer from a non-WebRTC sender starts changing LS groups on
us. This will keep media flowing, but will lose synchronization
signalling - this is (hopefully) an edge case.

Note: My preference is still to not do LS groups. But if we use them, we
must define how.
This is the least complicated algorithm I can see working.

--=20
Surveillance is pervasive. Go Dark.



From nobody Thu Jul 23 12:00:08 2015
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>, "rtcweb-chairs@tools.ietf.org" <rtcweb-chairs@tools.ietf.org>
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Hi,

Below are my notes from the 1st RTCWEB session.

Regards,

Christer

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


Topic:           JSEP

Presenter:       Eric Rescorla & Justin Uberti

Draft:           draft-ietf-rtcweb-jsep



The presenters went through the list of open issues in the draft, and prese=
nted suggestions on how to solve the issues.



#162: LS



Proposal: edge case - document that this is currently unsupported. The call=
ee can accept what caller wanted or reject the call.



There were two hums. The first hum was about who understands the problem, w=
ith very little response. The second hum was whether to NOT agree to the su=
ggested proposal, with a single objection.



Outcome: The issue will be re-visited on Friday. Harald Alvestrand will pro=
vide pseudo code for an alternative solution.





#158: (imageattr and CVO)



Proposal: always use a single SDP imageattr attribute.



Outcome: The proposal was agreed.







#122: AS->TIAS conversion



Proposal: The following formula for calculating the TIAS from the AS was pr=
oposed:



TIAS =3D AS * 0.95 - packet_rate * hfr_size

packet_rate =3D 50

hdr_size =3D 40

If more accurace is wanted, use TIAS instead of AS.



Outcome: The proposal as such was agreed, but the formula will be fixed, so=
 that the same units are used for all variables.







#125: BUNDLE and SDP mangling



Proposal: Do not support this



Outcome: The proposal was agreed.





#155: Selecting between multiple aspect ratios



Proposal: Implementations SHOULD pick imageattr with highest q=3D value tha=
t doesn't require upscaling.



Outcome: The proposal was agreed, but SHOUDLD will be changed to MUST.







#143: Can you call CreateOffer in remote-pranswer



Proposal: Simplest option - just allow this.



Outcome: The proposal was agreed.





#149: SSRC and clock rate



RFC 7160 prohibits changing of the clock rate within a given SSRC. Currentl=
y only a single SSRC value is provided per m- line, but an m- line can cont=
ain multiple codes, with different clock rates.





Proposal: document the limitation ad move on

- note that when changing between codes with different clock rates, you wil=
l get wacky RTP timestamps

- can revisit moving away from usage of SSRC in the future once MID and ESI=
D behaviours are better understood



It was discussed whether we should signal multiple SSRCs (one for each cloc=
k rate), and allow re-usage of SSRCs that have been previously used.



HUM:



      1) You must not switch between clock rates

      2) You are allowed to switch, but the impacts must be described



Not a clear consensus.



Outcome: Author is requested to provide some text, which will be provided t=
o the mailing list.





#169 multiple a=3Dfingerprint lines



Proposal:



- Generate a=3Dfingerprint line for each certificate

- Ensure received certification matches a=3Dfingerprint line with correspon=
ding hash



Outcome: The proposal was agreed.





#15 Early Transport Warmup



Randall J indicated that he is willing to provide an alternative solution.



Outcome: Will be re-visited on Friday. Randall J will provide some details =
on his alternative solution.


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<p class=3D"MsoNormal">Hi,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Below are my notes from the 1<sup>st</sup> RTCWEB se=
ssion.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Regards,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Christer<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">--------------------------------------------------<o=
:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">Topic:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; JSEP<=
o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">Presenter: &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Eric Rescorla &amp; Justin Ube=
rti<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">Draft:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; draft=
-ietf-rtcweb-jsep<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">The presenters went through the list of open issues in the draft, and pr=
esented suggestions on how to solve the issues.<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">#162: LS<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">Proposal: edge case - document that this is currently unsupported. The c=
allee can accept what caller wanted or reject the call.<o:p></o:p></span></=
p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">There were two hums. The first hum was about who understands the problem=
, with very little response. The second hum was whether to NOT agree to the=
 suggested proposal, with a single objection.<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">Outcome: The issue will be re-visited on Friday. Harald Alvestrand will =
provide pseudo code for an alternative solution.<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">#158: (imageattr and CVO)<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">Proposal: always use a single SDP imageattr attribute.<o:p></o:p></span>=
</p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">Outcome: The proposal was agreed.<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">#122: AS-&gt;TIAS conversion<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">Proposal: The following formula for calculating the TIAS from the AS was=
 proposed:
<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">TIAS =3D AS * 0.95 - packet_rate * hfr_size<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">packet_rate =3D 50<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">hdr_size =3D 40<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">If more accurace is wanted, use TIAS instead of AS.<o:p></o:p></span></p=
>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">Outcome: The proposal as such was agreed, but the formula will be fixed,=
 so that the same units are used for all variables.<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">#125: BUNDLE and SDP mangling<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">Proposal: Do not support this<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">Outcome: The proposal was agreed.<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">#155: Selecting between multiple aspect ratios<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">Proposal: Implementations SHOULD pick imageattr with highest q=3D value =
that doesn't require upscaling.<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">Outcome: The proposal was agreed, but SHOUDLD will be changed to MUST.<o=
:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">#143: Can you call CreateOffer in remote-pranswer<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">Proposal: Simplest option - just allow this.<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">Outcome: The proposal was agreed.<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">#149: SSRC and clock rate<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">RFC 7160 prohibits changing of the clock rate within a given SSRC. Curre=
ntly only a single SSRC value is provided per m- line, but an m- line can c=
ontain multiple codes, with different clock rates.<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">Proposal: document the limitation ad move on<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">- note that when changing between codes with different clock rates, you =
will get wacky RTP timestamps<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">- can revisit moving away from usage of SSRC in the future once MID and =
ESID behaviours are better understood<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">It was discussed whether we should signal multiple SSRCs (one for each c=
lock rate), and allow re-usage of SSRCs that have been previously used.<o:p=
></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">HUM:<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 1) You must not switch between clock rate=
s<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 2) You are allowed to switch, but the imp=
acts must be described<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">Not a clear consensus.&nbsp;
<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">Outcome: Author is requested to provide some text, which will be provide=
d to the mailing list.<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">#169 multiple a=3Dfingerprint lines<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">Proposal:<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">- Generate a=3Dfingerprint line for each certificate<o:p></o:p></span></=
p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">- Ensure received certification matches a=3Dfingerprint line with corres=
ponding hash
<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">Outcome: The proposal was agreed.<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">#15 Early Transport Warmup<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">Randall J indicated that he is willing to provide an alternative solutio=
n.<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span style=3D"font-family:&quot;Courier New&quot=
;">Outcome: Will be re-visited on Friday. Randall J will provide some detai=
ls on his alternative solution.&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
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From: Randell Jesup <randell-ietf@jesup.org>
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Subject: [rtcweb] Options for DTLS warmup and directionality
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To expand on my idea at the mic today:

The current proposal assumes prior agreement to persistent permissions 
to work well (and may 'light up' the camera/mic while the call is still 
waiting for the user to accept, which would be concerning to users).

We should not require persistent permissions in order to "warm up" DTLS 
connections before "Accept" from the user.  We also must NOT spin up the 
microphone or camera before the user has agreed to accept the call.  We 
also want to avoid a=inactive and a=recvonly answers.

Proposed solution: leverage 'dummy' tracks and replaceTrack (at the JS 
API level) to avoid renegotiation.

We need to attach some sort of track to each one we expect to send on in 
order to make use of the warmed-up connection when Accept happens.  
Without persistent permissions, we need alternative streams.  We have a 
few options:

1) WebAudio + canvas.captureStream()
     * Generate a silent WebAudio stream and generate a 
MediaStream/MediaStreamTrack (s)
     * Generate a blank (or not!) MediaStream/MediaStreamTrack from 
canvas.captureStream() (s)
        Note you'd tell it to use low-or-no-framerate (see below for 
another option).
     * Use MediaStream constructors to combine the audio and video track(s)
     * Attach to PeerConnection, and createOffer() (will typically be 
sendrecv (or sendonly if the other side is recvonly)).
     * On Accept by the user, use RTPSender.replaceTrack() on all 
accepted tracks, and sending will begin immediately; no renegotiation 
needed if all are accepted.
     * No new APIs needed, no W3 work.  Requires implementing agreed-on 
APIs.

2) Add a new way to getUserMedia/whatever to create fake streams, ala 
fake:true in Firefox
     * Note that we'd want to modify Firefox's impl and not use 
fake:true (probably), since we generate a specific test pattern for video.
     * This is a W3 Media Capture issue, not IETF, but we should realize 
it requires aligning with them

3) Allow some form of fake track without creating one (perhaps related 
to Peter's comment at the mic), and then replaceTrack to null tracks.
     * Does not allow spinning up congestion.
     * Won't spin up other stuff (decoders, jitter buffers, etc). Some 
spec work required; may be dragons here.

I prefer option 1.  It requires no new APIs or protocol decisions, and 
no action by W3 WGs that are trying to finalize 1.0.   We can do this as 
soon as people implement what we've already agreed to.

As a side benefit, if you actually push video via the canvas stream via 
larger resolutions & framerates, it would spin up congestion control 
ahead of time as well - under control of the application. This could be 
a significant win.

-- 
Randell Jesup -- rjesup a t mozilla d o t com
Please please please don't email randell-ietf@jesup.org!  Way oo much spam


From nobody Thu Jul 23 15:32:38 2015
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References: <55B1624D.4050908@jesup.org>
From: Peter Thatcher <pthatcher@google.com>
Date: Thu, 23 Jul 2015 15:31:53 -0700
Message-ID: <CAJrXDUFUsR8=qmompXzDJ4twBXOjH35ks69wT6r9d0vjUqu2Fw@mail.gmail.com>
To: Randell Jesup <randell-ietf@jesup.org>
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Cc: Cullen Jennings <fluffy@cisco.com>, Eric Rescorla <ekr@mozilla.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Options for DTLS warmup and directionality
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--001a113495b2a31d13051b92763d
Content-Type: text/plain; charset=UTF-8

Here is my idea:


PeerConnection.addTrack(null) means "don't send anything until
PeerConnection.replaceTrack is called with a non-null track".


Then the code for the callee would something like the following (warning:
it's late at night; I may have goofed up something):

var pc = new PeerConnection(...);
var sender = pc.addTrack(null);

pc.setLocalDescriptioon(offer);
var answer = pc.createAnswer(offer);
pc.setRemoteDescription(answer);
// send answer

// getUserMedia and ring in parallel
// get a track when the user accepts
sender.setTrack(track);


Then you don't need any fake things.

It could spin up congestion and other stuff if we want it to.  There is an
RtpSender object there, after all.  But I'm not sure we actually want to.


On Thu, Jul 23, 2015 at 2:53 PM, Randell Jesup <randell-ietf@jesup.org>
wrote:

> To expand on my idea at the mic today:
>
> The current proposal assumes prior agreement to persistent permissions to
> work well (and may 'light up' the camera/mic while the call is still
> waiting for the user to accept, which would be concerning to users).
>
> We should not require persistent permissions in order to "warm up" DTLS
> connections before "Accept" from the user.  We also must NOT spin up the
> microphone or camera before the user has agreed to accept the call.  We
> also want to avoid a=inactive and a=recvonly answers.
>
> Proposed solution: leverage 'dummy' tracks and replaceTrack (at the JS API
> level) to avoid renegotiation.
>
> We need to attach some sort of track to each one we expect to send on in
> order to make use of the warmed-up connection when Accept happens.  Without
> persistent permissions, we need alternative streams.  We have a few options:
>
> 1) WebAudio + canvas.captureStream()
>     * Generate a silent WebAudio stream and generate a
> MediaStream/MediaStreamTrack (s)
>     * Generate a blank (or not!) MediaStream/MediaStreamTrack from
> canvas.captureStream() (s)
>        Note you'd tell it to use low-or-no-framerate (see below for
> another option).
>     * Use MediaStream constructors to combine the audio and video track(s)
>     * Attach to PeerConnection, and createOffer() (will typically be
> sendrecv (or sendonly if the other side is recvonly)).
>     * On Accept by the user, use RTPSender.replaceTrack() on all accepted
> tracks, and sending will begin immediately; no renegotiation needed if all
> are accepted.
>     * No new APIs needed, no W3 work.  Requires implementing agreed-on
> APIs.
>
> 2) Add a new way to getUserMedia/whatever to create fake streams, ala
> fake:true in Firefox
>     * Note that we'd want to modify Firefox's impl and not use fake:true
> (probably), since we generate a specific test pattern for video.
>     * This is a W3 Media Capture issue, not IETF, but we should realize it
> requires aligning with them
>
> 3) Allow some form of fake track without creating one (perhaps related to
> Peter's comment at the mic), and then replaceTrack to null tracks.
>     * Does not allow spinning up congestion.
>     * Won't spin up other stuff (decoders, jitter buffers, etc). Some spec
> work required; may be dragons here.
>
> I prefer option 1.  It requires no new APIs or protocol decisions, and no
> action by W3 WGs that are trying to finalize 1.0.   We can do this as soon
> as people implement what we've already agreed to.
>
> As a side benefit, if you actually push video via the canvas stream via
> larger resolutions & framerates, it would spin up congestion control ahead
> of time as well - under control of the application. This could be a
> significant win.
>
> --
> Randell Jesup -- rjesup a t mozilla d o t com
> Please please please don't email randell-ietf@jesup.org!  Way oo much spam
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--001a113495b2a31d13051b92763d
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:arial,he=
lvetica,sans-serif">Here is my idea:</div><div class=3D"gmail_default" styl=
e=3D"font-family:arial,helvetica,sans-serif"><br></div><div class=3D"gmail_=
default" style=3D"font-family:arial,helvetica,sans-serif"><br></div><div cl=
ass=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-serif">Peer=
Connection.addTrack(null) means &quot;don&#39;t send anything until PeerCon=
nection.replaceTrack is called with a non-null track&quot;.</div><div class=
=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-serif"><br></d=
iv><div class=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-s=
erif"><br></div><div class=3D"gmail_default" style=3D"font-family:arial,hel=
vetica,sans-serif">Then the code for the callee would something like the fo=
llowing (warning: it&#39;s late at night; I may have goofed up something):<=
/div><div class=3D"gmail_default" style=3D"font-family:arial,helvetica,sans=
-serif"><br></div><div class=3D"gmail_default" style=3D"font-family:arial,h=
elvetica,sans-serif">var pc =3D new PeerConnection(...);</div><div class=3D=
"gmail_default" style=3D"font-family:arial,helvetica,sans-serif">var sender=
 =3D pc.addTrack(null);</div><div class=3D"gmail_default" style=3D"font-fam=
ily:arial,helvetica,sans-serif"><br></div><div class=3D"gmail_default" styl=
e=3D"font-family:arial,helvetica,sans-serif">pc.setLocalDescriptioon(offer)=
;<br></div><div class=3D"gmail_default" style=3D"font-family:arial,helvetic=
a,sans-serif">var answer =3D pc.createAnswer(offer);</div><div class=3D"gma=
il_default" style=3D"font-family:arial,helvetica,sans-serif">pc.setRemoteDe=
scription(answer);</div><div class=3D"gmail_default" style=3D"font-family:a=
rial,helvetica,sans-serif">// send answer</div><div class=3D"gmail_default"=
 style=3D"font-family:arial,helvetica,sans-serif"><br></div><div class=3D"g=
mail_default" style=3D"font-family:arial,helvetica,sans-serif">// getUserMe=
dia and ring in parallel</div><div class=3D"gmail_default" style=3D"font-fa=
mily:arial,helvetica,sans-serif">// get a track when the user accepts</div>=
<div class=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-seri=
f">sender.setTrack(track);<br></div><div class=3D"gmail_default" style=3D"f=
ont-family:arial,helvetica,sans-serif"><br></div><div class=3D"gmail_defaul=
t" style=3D"font-family:arial,helvetica,sans-serif"><br></div><div class=3D=
"gmail_default" style=3D"font-family:arial,helvetica,sans-serif">Then you d=
on&#39;t need any fake things. =C2=A0</div><div class=3D"gmail_default" sty=
le=3D"font-family:arial,helvetica,sans-serif"><br></div><div class=3D"gmail=
_default" style=3D"font-family:arial,helvetica,sans-serif">It could spin up=
 congestion and other stuff if we want it to.=C2=A0 There is an RtpSender o=
bject there, after all.=C2=A0 But I&#39;m not sure we actually want to. =C2=
=A0</div><div class=3D"gmail_default" style=3D"font-family:arial,helvetica,=
sans-serif"><br></div><div class=3D"gmail_extra"><br><div class=3D"gmail_qu=
ote">On Thu, Jul 23, 2015 at 2:53 PM, Randell Jesup <span dir=3D"ltr">&lt;<=
a href=3D"mailto:randell-ietf@jesup.org" target=3D"_blank">randell-ietf@jes=
up.org</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"=
margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,20=
4,204);border-left-style:solid;padding-left:1ex">To expand on my idea at th=
e mic today:<br>
<br>
The current proposal assumes prior agreement to persistent permissions to w=
ork well (and may &#39;light up&#39; the camera/mic while the call is still=
 waiting for the user to accept, which would be concerning to users).<br>
<br>
We should not require persistent permissions in order to &quot;warm up&quot=
; DTLS connections before &quot;Accept&quot; from the user.=C2=A0 We also m=
ust NOT spin up the microphone or camera before the user has agreed to acce=
pt the call.=C2=A0 We also want to avoid a=3Dinactive and a=3Drecvonly answ=
ers.<br>
<br>
Proposed solution: leverage &#39;dummy&#39; tracks and replaceTrack (at the=
 JS API level) to avoid renegotiation.<br>
<br>
We need to attach some sort of track to each one we expect to send on in or=
der to make use of the warmed-up connection when Accept happens.=C2=A0 With=
out persistent permissions, we need alternative streams.=C2=A0 We have a fe=
w options:<br>
<br>
1) WebAudio + canvas.captureStream()<br>
=C2=A0 =C2=A0 * Generate a silent WebAudio stream and generate a MediaStrea=
m/MediaStreamTrack (s)<br>
=C2=A0 =C2=A0 * Generate a blank (or not!) MediaStream/MediaStreamTrack fro=
m canvas.captureStream() (s)<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0Note you&#39;d tell it to use low-or-no-framerat=
e (see below for another option).<br>
=C2=A0 =C2=A0 * Use MediaStream constructors to combine the audio and video=
 track(s)<br>
=C2=A0 =C2=A0 * Attach to PeerConnection, and createOffer() (will typically=
 be sendrecv (or sendonly if the other side is recvonly)).<br>
=C2=A0 =C2=A0 * On Accept by the user, use RTPSender.replaceTrack() on all =
accepted tracks, and sending will begin immediately; no renegotiation neede=
d if all are accepted.<br>
=C2=A0 =C2=A0 * No new APIs needed, no W3 work.=C2=A0 Requires implementing=
 agreed-on APIs.<br>
<br>
2) Add a new way to getUserMedia/whatever to create fake streams, ala fake:=
true in Firefox<br>
=C2=A0 =C2=A0 * Note that we&#39;d want to modify Firefox&#39;s impl and no=
t use fake:true (probably), since we generate a specific test pattern for v=
ideo.<br>
=C2=A0 =C2=A0 * This is a W3 Media Capture issue, not IETF, but we should r=
ealize it requires aligning with them<br>
<br>
3) Allow some form of fake track without creating one (perhaps related to P=
eter&#39;s comment at the mic), and then replaceTrack to null tracks.<br>
=C2=A0 =C2=A0 * Does not allow spinning up congestion.<br>
=C2=A0 =C2=A0 * Won&#39;t spin up other stuff (decoders, jitter buffers, et=
c). Some spec work required; may be dragons here.<br>
<br>
I prefer option 1.=C2=A0 It requires no new APIs or protocol decisions, and=
 no action by W3 WGs that are trying to finalize 1.0.=C2=A0 =C2=A0We can do=
 this as soon as people implement what we&#39;ve already agreed to.<br>
<br>
As a side benefit, if you actually push video via the canvas stream via lar=
ger resolutions &amp; framerates, it would spin up congestion control ahead=
 of time as well - under control of the application. This could be a signif=
icant win.<span><font color=3D"#888888"><br>
<br>
-- <br>
Randell Jesup -- rjesup a t mozilla d o t com<br>
Please please please don&#39;t email <a href=3D"mailto:randell-ietf@jesup.o=
rg" target=3D"_blank">randell-ietf@jesup.org</a>!=C2=A0 Way oo much spam<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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</font></span></blockquote></div><br></div></div>

--001a113495b2a31d13051b92763d--


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From: Justin Uberti <juberti@google.com>
Date: Fri, 24 Jul 2015 00:41:49 +0200
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Subject: [rtcweb] Fuller proposal for early ICE/DTLS warmup
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--089e01184874f18812051b9298f5
Content-Type: text/plain; charset=UTF-8

*Graphical overview:*
*Caller                     Callee*
  |--- offer 1 (sendrecv) --->|
  |<-- answer 1 (sendonly) ---|
  |<-----ICE/DTLS start------>|

  .time passes, caller accepts.

  |<-- offer 2 (sendrecv) ----|
  |<---- callee media!!! -----|
  |--- answer 2 (sendrecv) -->|
  |----- caller media!!! ---->|

*New APIs needed:*
- create dummy sender using addTrack(null)
- RTCRtpReceiver.active property, which controls whether a receiver
indicates that it actually wants to receive media (by setting a=sendonly
instead of a=sendrecv)

*Annotated code sample:*
*- caller places call -*

*- caller creates audio track and offers it -*
track = gUM(audio)
sender = addTrack(track)
createOffer.then().setLD(offer).then().signal(offer)
*// offer is a=sendrecv*

*- callee gets offer, sets receiver as not active, adds dummy send track,
answers -*
offer = receive()
setRD(offer)
receiver = getReceivers()[0]
receiver.active = false
sender = addTrack(null)
createAnswer.then().setLD(answer).then().signal(answer)

*- answer is a=sendonly. Ensures caller doesn't send media during ring -*

*- caller gets answer, o/a 1 complete -*
answer = receive()
setRD(answer)

*- ICE and DTLS start -*

*- time passes - *

*- callee accepts call -*

*- callee creates audio track, replaces dummy track, toggles receiver,
offers -*
track = gUM(audio)
sender.setTrack(track)
receiver.active = true
createOffer.then().setLD(offer).then().signal(offer) // offer is a=sendrecv

*- callee -> caller media flows -*

*- callee gets offer, replies with normal answer - *
offer = receive();
setRD(offer)
createAnswer.then().setLD(answer).then().signal(answer) // answer is
a=sendrecv

*- caller -> callee media flows -*

*- callee gets answer, o/a 2 complete -*
answer = receive();
setRD(answer);

--089e01184874f18812051b9298f5
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div><b>Graphical overview:</b></div><div><b><font face=3D=
"monospace, monospace">Caller =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 =C2=A0 Callee</font></b></div><div><font face=3D"monospac=
e, monospace">=C2=A0 |--- offer 1 (sendrecv) ---&gt;|</font></div><div><fon=
t face=3D"monospace, monospace">=C2=A0 |&lt;-- answer 1 (sendonly) ---|</fo=
nt></div><div><font face=3D"monospace, monospace">=C2=A0 |&lt;-----ICE/DTLS=
 start------&gt;|</font></div><div><font face=3D"monospace, monospace"><br>=
</font></div><div><font face=3D"monospace, monospace">=C2=A0 .time passes, =
caller accepts.</font></div><div><font face=3D"monospace, monospace"><br></=
font></div><div><font face=3D"monospace, monospace">=C2=A0 |&lt;-- offer 2 =
(sendrecv) ----|</font></div><div><span style=3D"font-family:monospace,mono=
space">=C2=A0 |&lt;---- callee media!!! -----|</span><font face=3D"monospac=
e, monospace"><br></font></div><div><font face=3D"monospace, monospace">=C2=
=A0 |--- answer 2 (sendrecv) --&gt;|</font></div><div><span style=3D"font-f=
amily:monospace,monospace">=C2=A0 |----- caller media!!! ----&gt;|</span><f=
ont face=3D"monospace, monospace"><br></font></div><div><b><br></b></div><d=
iv><b>New APIs needed:</b></div><div>- create dummy sender using addTrack(n=
ull)</div><div>- RTCRtpReceiver.active property, which controls whether a r=
eceiver indicates that it actually wants to receive media (by setting a=3Ds=
endonly instead of a=3Dsendrecv)</div><div><br></div><div><b>Annotated code=
 sample:</b></div><div><i>- caller places call -</i></div><div><br></div><d=
iv><i>- caller creates audio track and offers it -</i></div><div>track =3D =
gUM(audio)=C2=A0</div><div>sender =3D addTrack(track)=C2=A0</div><div>creat=
eOffer.then().setLD(offer).then().signal(offer)=C2=A0</div><div><i>// offer=
 is a=3Dsendrecv</i></div><div><br></div><div><i>- callee gets offer, sets =
receiver as not active, adds dummy send track, answers -</i></div><div>offe=
r =3D receive()</div><div>setRD(offer)</div><div>receiver =3D getReceivers(=
)[0]</div><div>receiver.active =3D false</div><div>sender =3D addTrack(null=
)</div><div>createAnswer.then().setLD(answer).then().signal(answer)</div><d=
iv><i>- answer is a=3Dsendonly. Ensures caller doesn&#39;t send media durin=
g ring -<br></i></div><div><br></div><div><i>- caller gets answer, o/a 1 co=
mplete -</i></div><div>answer =3D receive()</div><div>setRD(answer)</div><d=
iv><br></div><div><i>- ICE and DTLS start -</i></div><div><br></div><div><i=
>- time passes -=C2=A0</i></div><div><br></div><div><i>- callee accepts cal=
l -</i></div><div><br></div><div><i>- callee creates audio track, replaces =
dummy track, toggles receiver, offers -</i></div><div>track =3D gUM(audio)<=
/div><div>sender.setTrack(track)</div><div>receiver.active =3D true</div><d=
iv>createOffer.then().setLD(offer).then().signal(offer) // offer is a=3Dsen=
drecv</div><div><br></div><div><i>- callee -&gt; caller media flows -</i></=
div><div><br></div><div><i>- callee gets offer, replies with normal answer =
-=C2=A0</i></div><div>offer =3D receive();</div><div>setRD(offer)</div><div=
>createAnswer.then().setLD(answer).then().signal(answer) // answer is a=3Ds=
endrecv</div><div><br></div><div><i>- caller -&gt; callee media flows -</i>=
</div><div><br></div><div><i>- callee gets answer, o/a 2 complete -</i></di=
v><div>answer =3D receive();</div><div>setRD(answer);</div><div><br></div><=
div><br></div><div><br></div><div><br></div><div><br></div></div>

--089e01184874f18812051b9298f5--


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From: Justin Uberti <juberti@google.com>
Date: Fri, 24 Jul 2015 00:44:09 +0200
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To: Peter Thatcher <pthatcher@google.com>
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Cc: Randell Jesup <randell-ietf@jesup.org>, Cullen Jennings <fluffy@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, Eric Rescorla <ekr@mozilla.com>
Subject: Re: [rtcweb] Options for DTLS warmup and directionality
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--001a113ec94e537e4b051b92a19d
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Peter's proposal pretty much matches what I just posted, but I think a
RTCRtpReceiver property is also needed to prevent caller->callee media
during ringing, which at a minimum would violate user expectations.

On Fri, Jul 24, 2015 at 12:31 AM, Peter Thatcher <pthatcher@google.com>
wrote:

> Here is my idea:
>
>
> PeerConnection.addTrack(null) means "don't send anything until
> PeerConnection.replaceTrack is called with a non-null track".
>
>
> Then the code for the callee would something like the following (warning:
> it's late at night; I may have goofed up something):
>
> var pc = new PeerConnection(...);
> var sender = pc.addTrack(null);
>
> pc.setLocalDescriptioon(offer);
> var answer = pc.createAnswer(offer);
> pc.setRemoteDescription(answer);
> // send answer
>
> // getUserMedia and ring in parallel
> // get a track when the user accepts
> sender.setTrack(track);
>
>
> Then you don't need any fake things.
>
> It could spin up congestion and other stuff if we want it to.  There is an
> RtpSender object there, after all.  But I'm not sure we actually want to.
>
>
> On Thu, Jul 23, 2015 at 2:53 PM, Randell Jesup <randell-ietf@jesup.org>
> wrote:
>
>> To expand on my idea at the mic today:
>>
>> The current proposal assumes prior agreement to persistent permissions to
>> work well (and may 'light up' the camera/mic while the call is still
>> waiting for the user to accept, which would be concerning to users).
>>
>> We should not require persistent permissions in order to "warm up" DTLS
>> connections before "Accept" from the user.  We also must NOT spin up the
>> microphone or camera before the user has agreed to accept the call.  We
>> also want to avoid a=inactive and a=recvonly answers.
>>
>> Proposed solution: leverage 'dummy' tracks and replaceTrack (at the JS
>> API level) to avoid renegotiation.
>>
>> We need to attach some sort of track to each one we expect to send on in
>> order to make use of the warmed-up connection when Accept happens.  Without
>> persistent permissions, we need alternative streams.  We have a few options:
>>
>> 1) WebAudio + canvas.captureStream()
>>     * Generate a silent WebAudio stream and generate a
>> MediaStream/MediaStreamTrack (s)
>>     * Generate a blank (or not!) MediaStream/MediaStreamTrack from
>> canvas.captureStream() (s)
>>        Note you'd tell it to use low-or-no-framerate (see below for
>> another option).
>>     * Use MediaStream constructors to combine the audio and video track(s)
>>     * Attach to PeerConnection, and createOffer() (will typically be
>> sendrecv (or sendonly if the other side is recvonly)).
>>     * On Accept by the user, use RTPSender.replaceTrack() on all accepted
>> tracks, and sending will begin immediately; no renegotiation needed if all
>> are accepted.
>>     * No new APIs needed, no W3 work.  Requires implementing agreed-on
>> APIs.
>>
>> 2) Add a new way to getUserMedia/whatever to create fake streams, ala
>> fake:true in Firefox
>>     * Note that we'd want to modify Firefox's impl and not use fake:true
>> (probably), since we generate a specific test pattern for video.
>>     * This is a W3 Media Capture issue, not IETF, but we should realize
>> it requires aligning with them
>>
>> 3) Allow some form of fake track without creating one (perhaps related to
>> Peter's comment at the mic), and then replaceTrack to null tracks.
>>     * Does not allow spinning up congestion.
>>     * Won't spin up other stuff (decoders, jitter buffers, etc). Some
>> spec work required; may be dragons here.
>>
>> I prefer option 1.  It requires no new APIs or protocol decisions, and no
>> action by W3 WGs that are trying to finalize 1.0.   We can do this as soon
>> as people implement what we've already agreed to.
>>
>> As a side benefit, if you actually push video via the canvas stream via
>> larger resolutions & framerates, it would spin up congestion control ahead
>> of time as well - under control of the application. This could be a
>> significant win.
>>
>>
>> --
>> Randell Jesup -- rjesup a t mozilla d o t com
>> Please please please don't email randell-ietf@jesup.org!  Way oo much
>> spam
>>
>> _______________________________________________
>> 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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<div dir=3D"ltr">Peter&#39;s proposal pretty much matches what I just poste=
d, but I think a RTCRtpReceiver property is also needed to prevent caller-&=
gt;callee media during ringing, which at a minimum would violate user expec=
tations.<div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Fri, J=
ul 24, 2015 at 12:31 AM, Peter Thatcher <span dir=3D"ltr">&lt;<a href=3D"ma=
ilto:pthatcher@google.com" target=3D"_blank">pthatcher@google.com</a>&gt;</=
span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8e=
x;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div class=
=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-serif">Here is=
 my idea:</div><div class=3D"gmail_default" style=3D"font-family:arial,helv=
etica,sans-serif"><br></div><div class=3D"gmail_default" style=3D"font-fami=
ly:arial,helvetica,sans-serif"><br></div><div class=3D"gmail_default" style=
=3D"font-family:arial,helvetica,sans-serif">PeerConnection.addTrack(null) m=
eans &quot;don&#39;t send anything until PeerConnection.replaceTrack is cal=
led with a non-null track&quot;.</div><div class=3D"gmail_default" style=3D=
"font-family:arial,helvetica,sans-serif"><br></div><div class=3D"gmail_defa=
ult" style=3D"font-family:arial,helvetica,sans-serif"><br></div><div class=
=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-serif">Then th=
e code for the callee would something like the following (warning: it&#39;s=
 late at night; I may have goofed up something):</div><div class=3D"gmail_d=
efault" style=3D"font-family:arial,helvetica,sans-serif"><br></div><div cla=
ss=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-serif">var p=
c =3D new PeerConnection(...);</div><div class=3D"gmail_default" style=3D"f=
ont-family:arial,helvetica,sans-serif">var sender =3D pc.addTrack(null);</d=
iv><div class=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-s=
erif"><br></div><div class=3D"gmail_default" style=3D"font-family:arial,hel=
vetica,sans-serif">pc.setLocalDescriptioon(offer);<br></div><div class=3D"g=
mail_default" style=3D"font-family:arial,helvetica,sans-serif">var answer =
=3D pc.createAnswer(offer);</div><div class=3D"gmail_default" style=3D"font=
-family:arial,helvetica,sans-serif">pc.setRemoteDescription(answer);</div><=
div class=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-serif=
">// send answer</div><div class=3D"gmail_default" style=3D"font-family:ari=
al,helvetica,sans-serif"><br></div><div class=3D"gmail_default" style=3D"fo=
nt-family:arial,helvetica,sans-serif">// getUserMedia and ring in parallel<=
/div><div class=3D"gmail_default" style=3D"font-family:arial,helvetica,sans=
-serif">// get a track when the user accepts</div><div class=3D"gmail_defau=
lt" style=3D"font-family:arial,helvetica,sans-serif">sender.setTrack(track)=
;<br></div><div class=3D"gmail_default" style=3D"font-family:arial,helvetic=
a,sans-serif"><br></div><div class=3D"gmail_default" style=3D"font-family:a=
rial,helvetica,sans-serif"><br></div><div class=3D"gmail_default" style=3D"=
font-family:arial,helvetica,sans-serif">Then you don&#39;t need any fake th=
ings. =C2=A0</div><div class=3D"gmail_default" style=3D"font-family:arial,h=
elvetica,sans-serif"><br></div><div class=3D"gmail_default" style=3D"font-f=
amily:arial,helvetica,sans-serif">It could spin up congestion and other stu=
ff if we want it to.=C2=A0 There is an RtpSender object there, after all.=
=C2=A0 But I&#39;m not sure we actually want to. =C2=A0</div><div class=3D"=
gmail_default" style=3D"font-family:arial,helvetica,sans-serif"><br></div><=
div class=3D"gmail_extra"><br><div class=3D"gmail_quote"><div><div>On Thu, =
Jul 23, 2015 at 2:53 PM, Randell Jesup <span dir=3D"ltr">&lt;<a href=3D"mai=
lto:randell-ietf@jesup.org" target=3D"_blank">randell-ietf@jesup.org</a>&gt=
;</span> wrote:<br></div></div><blockquote class=3D"gmail_quote" style=3D"m=
argin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204=
,204);border-left-style:solid;padding-left:1ex"><div><div>To expand on my i=
dea at the mic today:<br>
<br>
The current proposal assumes prior agreement to persistent permissions to w=
ork well (and may &#39;light up&#39; the camera/mic while the call is still=
 waiting for the user to accept, which would be concerning to users).<br>
<br>
We should not require persistent permissions in order to &quot;warm up&quot=
; DTLS connections before &quot;Accept&quot; from the user.=C2=A0 We also m=
ust NOT spin up the microphone or camera before the user has agreed to acce=
pt the call.=C2=A0 We also want to avoid a=3Dinactive and a=3Drecvonly answ=
ers.<br>
<br>
Proposed solution: leverage &#39;dummy&#39; tracks and replaceTrack (at the=
 JS API level) to avoid renegotiation.<br>
<br>
We need to attach some sort of track to each one we expect to send on in or=
der to make use of the warmed-up connection when Accept happens.=C2=A0 With=
out persistent permissions, we need alternative streams.=C2=A0 We have a fe=
w options:<br>
<br>
1) WebAudio + canvas.captureStream()<br>
=C2=A0 =C2=A0 * Generate a silent WebAudio stream and generate a MediaStrea=
m/MediaStreamTrack (s)<br>
=C2=A0 =C2=A0 * Generate a blank (or not!) MediaStream/MediaStreamTrack fro=
m canvas.captureStream() (s)<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0Note you&#39;d tell it to use low-or-no-framerat=
e (see below for another option).<br>
=C2=A0 =C2=A0 * Use MediaStream constructors to combine the audio and video=
 track(s)<br>
=C2=A0 =C2=A0 * Attach to PeerConnection, and createOffer() (will typically=
 be sendrecv (or sendonly if the other side is recvonly)).<br>
=C2=A0 =C2=A0 * On Accept by the user, use RTPSender.replaceTrack() on all =
accepted tracks, and sending will begin immediately; no renegotiation neede=
d if all are accepted.<br>
=C2=A0 =C2=A0 * No new APIs needed, no W3 work.=C2=A0 Requires implementing=
 agreed-on APIs.<br>
<br>
2) Add a new way to getUserMedia/whatever to create fake streams, ala fake:=
true in Firefox<br>
=C2=A0 =C2=A0 * Note that we&#39;d want to modify Firefox&#39;s impl and no=
t use fake:true (probably), since we generate a specific test pattern for v=
ideo.<br>
=C2=A0 =C2=A0 * This is a W3 Media Capture issue, not IETF, but we should r=
ealize it requires aligning with them<br>
<br>
3) Allow some form of fake track without creating one (perhaps related to P=
eter&#39;s comment at the mic), and then replaceTrack to null tracks.<br>
=C2=A0 =C2=A0 * Does not allow spinning up congestion.<br>
=C2=A0 =C2=A0 * Won&#39;t spin up other stuff (decoders, jitter buffers, et=
c). Some spec work required; may be dragons here.<br>
<br>
I prefer option 1.=C2=A0 It requires no new APIs or protocol decisions, and=
 no action by W3 WGs that are trying to finalize 1.0.=C2=A0 =C2=A0We can do=
 this as soon as people implement what we&#39;ve already agreed to.<br>
<br>
As a side benefit, if you actually push video via the canvas stream via lar=
ger resolutions &amp; framerates, it would spin up congestion control ahead=
 of time as well - under control of the application. This could be a signif=
icant win.</div></div><span><font color=3D"#888888"><div><div><br>
<br>
-- <br>
Randell Jesup -- rjesup a t mozilla d o t com<br>
Please please please don&#39;t email <a href=3D"mailto:randell-ietf@jesup.o=
rg" target=3D"_blank">randell-ietf@jesup.org</a>!=C2=A0 Way oo much spam<br=
>
<br></div></div>
_______________________________________________<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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</font></span></blockquote></div><br></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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div></div>

--001a113ec94e537e4b051b92a19d--


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Date: Fri, 24 Jul 2015 01:17:08 -0400
From: Randell Jesup <randell-ietf@jesup.org>
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Subject: [rtcweb] Congestion warmup
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I'd love to be able to warm up congestion control, and not just DTLS.  
That requires sending data and allowing it to increase to probe the 
connection and open the congestion window, and preferably sending in 
both directions (i.e. a=sendrecv in the warm-up answer).

We'd need to find a way to generically pad RTP packets to a target size 
(since encoding a static/black image won't use enough bytes).  The 
obvious codec-agnostic mechanism would be a header extension.  Since 
they can be any 4-byte-multiple size, that would pretty much work up to 
~384K (1500 byte packets, 1/frame @30fps). However, we could run the 
fake video framerate arbitrarily high to hit any bandwidth target we need.

Alternatively (and *much* better) we could use a "fake" codec 
(a=rtpmap:111 fake/90000) which doesn't actually include any real data, 
just padding.  Advantages: no CPU code to encode/decode.  No dealing 
with header extensions.  Clear indication if the other side wants to do 
spinup (they can reject the codec if they don't want to).  No 
encoding/decoding CPU use.

The offer would include it as a least-preferred codec to indicate warmup 
support.  If answerer answers with fake as the accepted codec (and real 
codecs following), it's a warmup answer.  Both sides will send and 
receive fake data to spin up the congestion windows (and also the jitter 
buffers, etc).  On actual answer, the fake codec would be dropped (or 
dropped to bottom priority).   The answerer would start sending in a 
real codec, which would work since the warmup answer would have included 
that codec.  The offerer would see the incoming real-codec packets, and 
decode them.   When the real answer arrives, it will show the accepted 
codec has changed, and the offerer will install that answer and thus and 
start sending (real) media.

We still need a way to tell the system at the answerer side "I want to 
accept but it's a warm-up accept" (and perhaps if we want to spin up 
congestion control or not while waiting), and a way to say on the 
offerer side "I want to offer/use warmups".  For those API points see 
the previous options from myself and Peter - or simply a way to say 
createAnswer of a warmup answer, and have that agree to all tracks with 
fake codecs being primary.

I'll note that all of these methods would benefit from a mechanism to 
indicate the desired bitrate targets for each fake track; for solutions 
that use some sort of fake video input (canvas.captureStream/etc) it 
inherently allows the application to define the size and framerate, 
which are the primary knobs used by the browser for setting 'max total 
bitrate' for the track.   You don't want the application to set the 
bandwidth directly, you want it to define the parameters of the video 
stream so the system can set "reasonable" limits for that resolution & 
situation.


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On 7/23/2015 6:44 PM, Justin Uberti wrote:
> Peter's proposal pretty much matches what I just posted, but I think a 
> RTCRtpReceiver property is also needed to prevent caller->callee media 
> during ringing, which at a minimum would violate user expectations.

If we want to answer a=sendrev offers with a=sendonly (regardless of 
this use-case), we need something like that.  However, we may not care 
much outside of this usecase.

I'd love to be able to just answer a=sendrecv in the warm-up answer, and 
in any case the answerer must indicate somehow in conjunction with the 
warmup answer that it's not a real user answer (I do not consider 
parsing all the directions for a=sendonly a reasonable way to know).  
Since the answer comes along with "this is warmup" (somehow), the 
offerer can see that and replaceTrack with fake (or null) tracks before 
SetRemoteDescription, and so not send media (or send only fake 
congestion-spinup data).

> On Fri, Jul 24, 2015 at 12:31 AM, Peter Thatcher <pthatcher@google.com 
> <mailto:pthatcher@google.com>> wrote:
>
>     Here is my idea:
>
>
>     PeerConnection.addTrack(null) means "don't send anything until
>     PeerConnection.replaceTrack is called with a non-null track".
>

Note that this would be problematic on the offerer side (is it audio or 
video?)  We'd have to figure out when this is valid and what it does 
when it id valid (in W3 land).  Also, perhaps we could use 
addTrack(null) as an indicator to accept a 'fake' codec as the primary 
codec (see previous email on congestion spinup).  This would enable (but 
not require) congestion spinup.

>
>
>     Then the code for the callee would something like the following
>     (warning: it's late at night; I may have goofed up something):
>
>     var pc = new PeerConnection(...);
>     var sender = pc.addTrack(null);
>
>     pc.setLocalDescriptioon(offer);
>     var answer = pc.createAnswer(offer);
>     pc.setRemoteDescription(answer);
>     // send answer
>
>     // getUserMedia and ring in parallel
>     // get a track when the user accepts
>     sender.setTrack(track);
>

Another advantage of this (Peter's addTrack(null)) is that it doesn't 
send muted data (from webaudio or canvas.captureStream). This means when 
the user *does* answer, you swap to real sources and start encoding, so 
the offerer can key off incoming data to know it should start showing 
the data (before the final answer arrives). Note that the rtcpmap fake 
codec/payload idea would also work (the "decoder" would eat all data and 
never generate a frame of video, and would generate no audio or silent 
depending on how it works in the stack - then when the real answer 
happens, we switch to using a real encoder to encode real data, and it 
will end up decoding and showing).

>
>
>     Then you don't need any fake things.
>
>     It could spin up congestion and other stuff if we want it to. 
>     There is an RtpSender object there, after all. But I'm not sure we
>     actually want to.
>

Spinning congestion with 'null' tracks would be more complex - more API 
points probably (and needs a=sendrecv really, but per above we could).  
See previous message about how you'd do it.

-- 
Randell Jesup -- rjesup a t mozilla d o t com
Please please please don't email randell-ietf@jesup.org!  Way too much spam


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From: Cullen Jennings <fluffy@cisco.com>
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Date: Fri, 24 Jul 2015 07:47:08 +0200
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Subject: Re: [rtcweb] Options for DTLS warmup and directionality
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My initial reaction to this was I can see how it would work but I sort =
of hope we could get something that was easier for web developers and =
that did not use bandwidth to send the media before there was real =
media.  Good food for thought.=20


> On Jul 23, 2015, at 11:53 PM, Randell Jesup <randell-ietf@jesup.org> =
wrote:
>=20
> To expand on my idea at the mic today:
>=20
> The current proposal assumes prior agreement to persistent permissions =
to work well (and may 'light up' the camera/mic while the call is still =
waiting for the user to accept, which would be concerning to users).
>=20
> We should not require persistent permissions in order to "warm up" =
DTLS connections before "Accept" from the user.  We also must NOT spin =
up the microphone or camera before the user has agreed to accept the =
call.  We also want to avoid a=3Dinactive and a=3Drecvonly answers.
>=20
> Proposed solution: leverage 'dummy' tracks and replaceTrack (at the JS =
API level) to avoid renegotiation.
>=20
> We need to attach some sort of track to each one we expect to send on =
in order to make use of the warmed-up connection when Accept happens.  =
Without persistent permissions, we need alternative streams.  We have a =
few options:
>=20
> 1) WebAudio + canvas.captureStream()
>    * Generate a silent WebAudio stream and generate a =
MediaStream/MediaStreamTrack (s)
>    * Generate a blank (or not!) MediaStream/MediaStreamTrack from =
canvas.captureStream() (s)
>       Note you'd tell it to use low-or-no-framerate (see below for =
another option).
>    * Use MediaStream constructors to combine the audio and video =
track(s)
>    * Attach to PeerConnection, and createOffer() (will typically be =
sendrecv (or sendonly if the other side is recvonly)).
>    * On Accept by the user, use RTPSender.replaceTrack() on all =
accepted tracks, and sending will begin immediately; no renegotiation =
needed if all are accepted.
>    * No new APIs needed, no W3 work.  Requires implementing agreed-on =
APIs.
>=20
> 2) Add a new way to getUserMedia/whatever to create fake streams, ala =
fake:true in Firefox
>    * Note that we'd want to modify Firefox's impl and not use =
fake:true (probably), since we generate a specific test pattern for =
video.
>    * This is a W3 Media Capture issue, not IETF, but we should realize =
it requires aligning with them
>=20
> 3) Allow some form of fake track without creating one (perhaps related =
to Peter's comment at the mic), and then replaceTrack to null tracks.
>    * Does not allow spinning up congestion.
>    * Won't spin up other stuff (decoders, jitter buffers, etc). Some =
spec work required; may be dragons here.
>=20
> I prefer option 1.  It requires no new APIs or protocol decisions, and =
no action by W3 WGs that are trying to finalize 1.0.   We can do this as =
soon as people implement what we've already agreed to.
>=20
> As a side benefit, if you actually push video via the canvas stream =
via larger resolutions & framerates, it would spin up congestion control =
ahead of time as well - under control of the application. This could be =
a significant win.
>=20
> --=20
> Randell Jesup -- rjesup a t mozilla d o t com
> Please please please don't email randell-ietf@jesup.org!  Way oo much =
spam
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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Subject: Re: [rtcweb] Options for DTLS warmup and directionality
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> On Jul 24, 2015, at 12:31 AM, Peter Thatcher <pthatcher@google.com> =
wrote:
>=20
> PeerConnection.addTrack(null) means "don't send anything until =
PeerConnection.replaceTrack is called with a non-null track".
>=20

Would we need extra info to know if this was audio, video etc. I'm just =
thinking about the issues where the PC creates an answer without knowing =
anything about what is in the track then when the PC see the real track =
(after answer is sent), I'm wondering if we could end up with a case =
where what was in the track was not consistent with the answer that got =
sent. I don't have a concrete example of a problem here but I have a =
nagging feeling there might be problems here - particularly on things =
where the encoding is done in hardware.=20=


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Subject: Re: [rtcweb] Fuller proposal for early ICE/DTLS warmup
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This seems generally reasonable, but share Cullen's concern about how the
answerer PC knows the characteristics of the dummy track
(this is something that would be addressed by Jesup's more explicit dummy
track example).

-Ekr


On Fri, Jul 24, 2015 at 12:41 AM, Justin Uberti <juberti@google.com> wrote:

> *Graphical overview:*
> *Caller                     Callee*
>   |--- offer 1 (sendrecv) --->|
>   |<-- answer 1 (sendonly) ---|
>   |<-----ICE/DTLS start------>|
>
>   .time passes, caller accepts.
>
>   |<-- offer 2 (sendrecv) ----|
>   |<---- callee media!!! -----|
>   |--- answer 2 (sendrecv) -->|
>   |----- caller media!!! ---->|
>
> *New APIs needed:*
> - create dummy sender using addTrack(null)
> - RTCRtpReceiver.active property, which controls whether a receiver
> indicates that it actually wants to receive media (by setting a=sendonly
> instead of a=sendrecv)
>
> *Annotated code sample:*
> *- caller places call -*
>
> *- caller creates audio track and offers it -*
> track = gUM(audio)
> sender = addTrack(track)
> createOffer.then().setLD(offer).then().signal(offer)
> *// offer is a=sendrecv*
>
> *- callee gets offer, sets receiver as not active, adds dummy send track,
> answers -*
> offer = receive()
> setRD(offer)
> receiver = getReceivers()[0]
> receiver.active = false
> sender = addTrack(null)
> createAnswer.then().setLD(answer).then().signal(answer)
>
> *- answer is a=sendonly. Ensures caller doesn't send media during ring -*
>
> *- caller gets answer, o/a 1 complete -*
> answer = receive()
> setRD(answer)
>
> *- ICE and DTLS start -*
>
> *- time passes - *
>
> *- callee accepts call -*
>
> *- callee creates audio track, replaces dummy track, toggles receiver,
> offers -*
> track = gUM(audio)
> sender.setTrack(track)
> receiver.active = true
> createOffer.then().setLD(offer).then().signal(offer) // offer is a=sendrecv
>
> *- callee -> caller media flows -*
>
> *- callee gets offer, replies with normal answer - *
> offer = receive();
> setRD(offer)
> createAnswer.then().setLD(answer).then().signal(answer) // answer is
> a=sendrecv
>
> *- caller -> callee media flows -*
>
> *- callee gets answer, o/a 2 complete -*
> answer = receive();
> setRD(answer);
>
>
>
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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<div dir=3D"ltr">This seems generally reasonable, but share Cullen&#39;s co=
ncern about how the<div>answerer PC knows the characteristics of the dummy =
track</div><div>(this is something that would be addressed by Jesup&#39;s m=
ore explicit dummy</div><div>track example).<br><div><br></div><div><div><d=
iv><div><div>-Ekr</div><div><br></div></div></div></div></div></div></div><=
div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Fri, Jul 24, 20=
15 at 12:41 AM, Justin Uberti <span dir=3D"ltr">&lt;<a href=3D"mailto:juber=
ti@google.com" target=3D"_blank">juberti@google.com</a>&gt;</span> wrote:<b=
r><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:=
1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div><b>Graphical overvie=
w:</b></div><div><b><font face=3D"monospace, monospace">Caller =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 Callee</font></=
b></div><div><font face=3D"monospace, monospace">=C2=A0 |--- offer 1 (sendr=
ecv) ---&gt;|</font></div><div><font face=3D"monospace, monospace">=C2=A0 |=
&lt;-- answer 1 (sendonly) ---|</font></div><div><font face=3D"monospace, m=
onospace">=C2=A0 |&lt;-----ICE/DTLS start------&gt;|</font></div><div><font=
 face=3D"monospace, monospace"><br></font></div><div><font face=3D"monospac=
e, monospace">=C2=A0 .time passes, caller accepts.</font></div><div><font f=
ace=3D"monospace, monospace"><br></font></div><div><font face=3D"monospace,=
 monospace">=C2=A0 |&lt;-- offer 2 (sendrecv) ----|</font></div><div><span =
style=3D"font-family:monospace,monospace">=C2=A0 |&lt;---- callee media!!! =
-----|</span><font face=3D"monospace, monospace"><br></font></div><div><fon=
t face=3D"monospace, monospace">=C2=A0 |--- answer 2 (sendrecv) --&gt;|</fo=
nt></div><div><span style=3D"font-family:monospace,monospace">=C2=A0 |-----=
 caller media!!! ----&gt;|</span><font face=3D"monospace, monospace"><br></=
font></div><div><b><br></b></div><div><b>New APIs needed:</b></div><div>- c=
reate dummy sender using addTrack(null)</div><div>- RTCRtpReceiver.active p=
roperty, which controls whether a receiver indicates that it actually wants=
 to receive media (by setting a=3Dsendonly instead of a=3Dsendrecv)</div><d=
iv><br></div><div><b>Annotated code sample:</b></div><div><i>- caller place=
s call -</i></div><div><br></div><div><i>- caller creates audio track and o=
ffers it -</i></div><div>track =3D gUM(audio)=C2=A0</div><div>sender =3D ad=
dTrack(track)=C2=A0</div><div>createOffer.then().setLD(offer).then().signal=
(offer)=C2=A0</div><div><i>// offer is a=3Dsendrecv</i></div><div><br></div=
><div><i>- callee gets offer, sets receiver as not active, adds dummy send =
track, answers -</i></div><div>offer =3D receive()</div><div>setRD(offer)</=
div><div>receiver =3D getReceivers()[0]</div><div>receiver.active =3D false=
</div><div>sender =3D addTrack(null)</div><div>createAnswer.then().setLD(an=
swer).then().signal(answer)</div><div><i>- answer is a=3Dsendonly. Ensures =
caller doesn&#39;t send media during ring -<br></i></div><div><br></div><di=
v><i>- caller gets answer, o/a 1 complete -</i></div><div>answer =3D receiv=
e()</div><div>setRD(answer)</div><div><br></div><div><i>- ICE and DTLS star=
t -</i></div><div><br></div><div><i>- time passes -=C2=A0</i></div><div><br=
></div><div><i>- callee accepts call -</i></div><div><br></div><div><i>- ca=
llee creates audio track, replaces dummy track, toggles receiver, offers -<=
/i></div><div>track =3D gUM(audio)</div><div>sender.setTrack(track)</div><d=
iv>receiver.active =3D true</div><div>createOffer.then().setLD(offer).then(=
).signal(offer) // offer is a=3Dsendrecv</div><div><br></div><div><i>- call=
ee -&gt; caller media flows -</i></div><div><br></div><div><i>- callee gets=
 offer, replies with normal answer -=C2=A0</i></div><div>offer =3D receive(=
);</div><div>setRD(offer)</div><div>createAnswer.then().setLD(answer).then(=
).signal(answer) // answer is a=3Dsendrecv</div><div><br></div><div><i>- ca=
ller -&gt; callee media flows -</i></div><div><br></div><div><i>- callee ge=
ts answer, o/a 2 complete -</i></div><div>answer =3D receive();</div><div>s=
etRD(answer);</div><div><br></div><div><br></div><div><br></div><div><br></=
div><div><br></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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>

--f46d04440196627ab2051b9a0456--


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Subject: Re: [rtcweb] Options for DTLS warmup and directionality
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On 24 July 2015 at 00:31, Peter Thatcher <pthatcher@google.com> wrote:
> PeerConnection.addTrack(null)

I'm not entirely comfortable with the notion that we allow null
arguments.  Doing that could be a footgun.  Especially since it is
only to support what I consider to be a corner case (even though it
might be an important one).


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From: Peter Thatcher <pthatcher@google.com>
Date: Fri, 24 Jul 2015 01:54:14 -0700
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Subject: Re: [rtcweb] Congestion warmup
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--047d7bfd05085acbfa051b9b28ba
Content-Type: text/plain; charset=UTF-8

Could the new flexfec payload type be used for padding?

On Thu, Jul 23, 2015 at 10:17 PM, Randell Jesup <randell-ietf@jesup.org>
wrote:

> I'd love to be able to warm up congestion control, and not just DTLS.
> That requires sending data and allowing it to increase to probe the
> connection and open the congestion window, and preferably sending in both
> directions (i.e. a=sendrecv in the warm-up answer).
>
> We'd need to find a way to generically pad RTP packets to a target size
> (since encoding a static/black image won't use enough bytes).  The obvious
> codec-agnostic mechanism would be a header extension.  Since they can be
> any 4-byte-multiple size, that would pretty much work up to ~384K (1500
> byte packets, 1/frame @30fps). However, we could run the fake video
> framerate arbitrarily high to hit any bandwidth target we need.
>
> Alternatively (and *much* better) we could use a "fake" codec
> (a=rtpmap:111 fake/90000) which doesn't actually include any real data,
> just padding.  Advantages: no CPU code to encode/decode.  No dealing with
> header extensions.  Clear indication if the other side wants to do spinup
> (they can reject the codec if they don't want to).  No encoding/decoding
> CPU use.
>
> The offer would include it as a least-preferred codec to indicate warmup
> support.  If answerer answers with fake as the accepted codec (and real
> codecs following), it's a warmup answer.  Both sides will send and receive
> fake data to spin up the congestion windows (and also the jitter buffers,
> etc).  On actual answer, the fake codec would be dropped (or dropped to
> bottom priority).   The answerer would start sending in a real codec, which
> would work since the warmup answer would have included that codec.  The
> offerer would see the incoming real-codec packets, and decode them.   When
> the real answer arrives, it will show the accepted codec has changed, and
> the offerer will install that answer and thus and start sending (real)
> media.
>
> We still need a way to tell the system at the answerer side "I want to
> accept but it's a warm-up accept" (and perhaps if we want to spin up
> congestion control or not while waiting), and a way to say on the offerer
> side "I want to offer/use warmups".  For those API points see the previous
> options from myself and Peter - or simply a way to say createAnswer of a
> warmup answer, and have that agree to all tracks with fake codecs being
> primary.
>
> I'll note that all of these methods would benefit from a mechanism to
> indicate the desired bitrate targets for each fake track; for solutions
> that use some sort of fake video input (canvas.captureStream/etc) it
> inherently allows the application to define the size and framerate, which
> are the primary knobs used by the browser for setting 'max total bitrate'
> for the track.   You don't want the application to set the bandwidth
> directly, you want it to define the parameters of the video stream so the
> system can set "reasonable" limits for that resolution & situation.
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--047d7bfd05085acbfa051b9b28ba
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:arial,he=
lvetica,sans-serif">Could the new flexfec payload type be used for padding?=
 =C2=A0</div></div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote=
">On Thu, Jul 23, 2015 at 10:17 PM, Randell Jesup <span dir=3D"ltr">&lt;<a =
href=3D"mailto:randell-ietf@jesup.org" target=3D"_blank">randell-ietf@jesup=
.org</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">I&#39;d love t=
o be able to warm up congestion control, and not just DTLS.=C2=A0 That requ=
ires sending data and allowing it to increase to probe the connection and o=
pen the congestion window, and preferably sending in both directions (i.e. =
a=3Dsendrecv in the warm-up answer).<br>
<br>
We&#39;d need to find a way to generically pad RTP packets to a target size=
 (since encoding a static/black image won&#39;t use enough bytes).=C2=A0 Th=
e obvious codec-agnostic mechanism would be a header extension.=C2=A0 Since=
 they can be any 4-byte-multiple size, that would pretty much work up to ~3=
84K (1500 byte packets, 1/frame @30fps). However, we could run the fake vid=
eo framerate arbitrarily high to hit any bandwidth target we need.<br>
<br>
Alternatively (and *much* better) we could use a &quot;fake&quot; codec (a=
=3Drtpmap:111 fake/90000) which doesn&#39;t actually include any real data,=
 just padding.=C2=A0 Advantages: no CPU code to encode/decode.=C2=A0 No dea=
ling with header extensions.=C2=A0 Clear indication if the other side wants=
 to do spinup (they can reject the codec if they don&#39;t want to).=C2=A0 =
No encoding/decoding CPU use.<br>
<br>
The offer would include it as a least-preferred codec to indicate warmup su=
pport.=C2=A0 If answerer answers with fake as the accepted codec (and real =
codecs following), it&#39;s a warmup answer.=C2=A0 Both sides will send and=
 receive fake data to spin up the congestion windows (and also the jitter b=
uffers, etc).=C2=A0 On actual answer, the fake codec would be dropped (or d=
ropped to bottom priority).=C2=A0 =C2=A0The answerer would start sending in=
 a real codec, which would work since the warmup answer would have included=
 that codec.=C2=A0 The offerer would see the incoming real-codec packets, a=
nd decode them.=C2=A0 =C2=A0When the real answer arrives, it will show the =
accepted codec has changed, and the offerer will install that answer and th=
us and start sending (real) media.<br>
<br>
We still need a way to tell the system at the answerer side &quot;I want to=
 accept but it&#39;s a warm-up accept&quot; (and perhaps if we want to spin=
 up congestion control or not while waiting), and a way to say on the offer=
er side &quot;I want to offer/use warmups&quot;.=C2=A0 For those API points=
 see the previous options from myself and Peter - or simply a way to say cr=
eateAnswer of a warmup answer, and have that agree to all tracks with fake =
codecs being primary.<br>
<br>
I&#39;ll note that all of these methods would benefit from a mechanism to i=
ndicate the desired bitrate targets for each fake track; for solutions that=
 use some sort of fake video input (canvas.captureStream/etc) it inherently=
 allows the application to define the size and framerate, which are the pri=
mary knobs used by the browser for setting &#39;max total bitrate&#39; for =
the track.=C2=A0 =C2=A0You don&#39;t want the application to set the bandwi=
dth directly, you want it to define the parameters of the video stream so t=
he system can set &quot;reasonable&quot; limits for that resolution &amp; s=
ituation.<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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div><br></div>

--047d7bfd05085acbfa051b9b28ba--


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From: Peter Thatcher <pthatcher@google.com>
Date: Fri, 24 Jul 2015 01:57:03 -0700
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Subject: Re: [rtcweb] Options for DTLS warmup and directionality
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Would you be comfortable with either addTrack("audio") or
createRtpSender("audio")?

On Fri, Jul 24, 2015 at 12:50 AM, Martin Thomson <martin.thomson@gmail.com>
wrote:

> On 24 July 2015 at 00:31, Peter Thatcher <pthatcher@google.com> wrote:
> > PeerConnection.addTrack(null)
>
> I'm not entirely comfortable with the notion that we allow null
> arguments.  Doing that could be a footgun.  Especially since it is
> only to support what I consider to be a corner case (even though it
> might be an important one).
>

--001a11331df061837b051b9b3215
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<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:arial,he=
lvetica,sans-serif">Would you be comfortable with either addTrack(&quot;aud=
io&quot;) or createRtpSender(&quot;audio&quot;)?</div></div><div class=3D"g=
mail_extra"><br><div class=3D"gmail_quote">On Fri, Jul 24, 2015 at 12:50 AM=
, Martin Thomson <span dir=3D"ltr">&lt;<a href=3D"mailto:martin.thomson@gma=
il.com" target=3D"_blank">martin.thomson@gmail.com</a>&gt;</span> wrote:<br=
><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1=
px #ccc solid;padding-left:1ex"><span class=3D"">On 24 July 2015 at 00:31, =
Peter Thatcher &lt;<a href=3D"mailto:pthatcher@google.com">pthatcher@google=
.com</a>&gt; wrote:<br>
&gt; PeerConnection.addTrack(null)<br>
<br>
</span>I&#39;m not entirely comfortable with the notion that we allow null<=
br>
arguments.=C2=A0 Doing that could be a footgun.=C2=A0 Especially since it i=
s<br>
only to support what I consider to be a corner case (even though it<br>
might be an important one).<br>
</blockquote></div><br></div>

--001a11331df061837b051b9b3215--


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Subject: [rtcweb] Day 2 updated agenda and slides up
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I=92ve updated/uploaded the agenda to =
http://datatracker.ietf.org/meeting/93/materials.html

spt=


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On Fri, Jul 24, 2015 at 10:57 AM, Peter Thatcher <pthatcher@google.com>
wrote:

> Would you be comfortable with either addTrack("audio")
>

This seems awful.



> or createRtpSender("audio")?
>

This seems OKish

-Ekr


> On Fri, Jul 24, 2015 at 12:50 AM, Martin Thomson <martin.thomson@gmail.com
> > wrote:
>
>> On 24 July 2015 at 00:31, Peter Thatcher <pthatcher@google.com> wrote:
>> > PeerConnection.addTrack(null)
>>
>> I'm not entirely comfortable with the notion that we allow null
>> arguments.  Doing that could be a footgun.  Especially since it is
>> only to support what I consider to be a corner case (even though it
>> might be an important one).
>>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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<div dir=3D"ltr"><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">=
On Fri, Jul 24, 2015 at 10:57 AM, Peter Thatcher <span dir=3D"ltr">&lt;<a h=
ref=3D"mailto:pthatcher@google.com" target=3D"_blank">pthatcher@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"><div dir=3D"ltr"><d=
iv style=3D"font-family:arial,helvetica,sans-serif">Would you be comfortabl=
e with either addTrack(&quot;audio&quot;)</div></div></blockquote><div><br>=
</div><div>This seems awful.</div><div><br></div><div>=C2=A0</div><blockquo=
te class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc so=
lid;padding-left:1ex"><div dir=3D"ltr"><div style=3D"font-family:arial,helv=
etica,sans-serif"> or createRtpSender(&quot;audio&quot;)?</div></div></bloc=
kquote><div><br></div><div>This seems OKish</div><div><br></div><div>-Ekr</=
div><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 =
0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div class=3D"gmail_ext=
ra"><div class=3D"gmail_quote">On Fri, Jul 24, 2015 at 12:50 AM, Martin Tho=
mson <span dir=3D"ltr">&lt;<a href=3D"mailto:martin.thomson@gmail.com" targ=
et=3D"_blank">martin.thomson@gmail.com</a>&gt;</span> wrote:<span class=3D"=
"><br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-l=
eft:1px #ccc solid;padding-left:1ex"><span>On 24 July 2015 at 00:31, Peter =
Thatcher &lt;<a href=3D"mailto:pthatcher@google.com" target=3D"_blank">ptha=
tcher@google.com</a>&gt; wrote:<br>
&gt; PeerConnection.addTrack(null)<br>
<br>
</span>I&#39;m not entirely comfortable with the notion that we allow null<=
br>
arguments.=C2=A0 Doing that could be a footgun.=C2=A0 Especially since it i=
s<br>
only to support what I consider to be a corner case (even though it<br>
might be an important one).<br>
</blockquote></span></div><br></div>
<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div></div>

--001a11c37eb82b2dba051b9b40ee--


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Cc: Randell Jesup <randell-ietf@jesup.org>, "<rtcweb@ietf.org>" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Congestion warmup
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This is an interesting possibility. Using some kind of fec as padding =
could make possible to avoid using a fake codec allowing receivers to =
behave normally and without requiring implementing a different behavior =
for the fake codec data. In addition, this protection of the encoded =
static/black image may avoid PLIs in case of severe packet loss when =
hitting a BW during the probe. This could be even used as a probe =
mechanism for BW estimation with the call fully open, but this is =
another story.
El 24/07/2015, a las 10:54, Peter Thatcher <pthatcher@google.com> =
escribi=F3:

> Could the new flexfec payload type be used for padding? =20
>=20
> On Thu, Jul 23, 2015 at 10:17 PM, Randell Jesup =
<randell-ietf@jesup.org> wrote:
> I'd love to be able to warm up congestion control, and not just DTLS.  =
That requires sending data and allowing it to increase to probe the =
connection and open the congestion window, and preferably sending in =
both directions (i.e. a=3Dsendrecv in the warm-up answer).
>=20
> We'd need to find a way to generically pad RTP packets to a target =
size (since encoding a static/black image won't use enough bytes).  The =
obvious codec-agnostic mechanism would be a header extension.  Since =
they can be any 4-byte-multiple size, that would pretty much work up to =
~384K (1500 byte packets, 1/frame @30fps). However, we could run the =
fake video framerate arbitrarily high to hit any bandwidth target we =
need.
>=20
> Alternatively (and *much* better) we could use a "fake" codec =
(a=3Drtpmap:111 fake/90000) which doesn't actually include any real =
data, just padding.  Advantages: no CPU code to encode/decode.  No =
dealing with header extensions.  Clear indication if the other side =
wants to do spinup (they can reject the codec if they don't want to).  =
No encoding/decoding CPU use.
>=20
> The offer would include it as a least-preferred codec to indicate =
warmup support.  If answerer answers with fake as the accepted codec =
(and real codecs following), it's a warmup answer.  Both sides will send =
and receive fake data to spin up the congestion windows (and also the =
jitter buffers, etc).  On actual answer, the fake codec would be dropped =
(or dropped to bottom priority).   The answerer would start sending in a =
real codec, which would work since the warmup answer would have included =
that codec.  The offerer would see the incoming real-codec packets, and =
decode them.   When the real answer arrives, it will show the accepted =
codec has changed, and the offerer will install that answer and thus and =
start sending (real) media.
>=20
> We still need a way to tell the system at the answerer side "I want to =
accept but it's a warm-up accept" (and perhaps if we want to spin up =
congestion control or not while waiting), and a way to say on the =
offerer side "I want to offer/use warmups".  For those API points see =
the previous options from myself and Peter - or simply a way to say =
createAnswer of a warmup answer, and have that agree to all tracks with =
fake codecs being primary.
>=20
> I'll note that all of these methods would benefit from a mechanism to =
indicate the desired bitrate targets for each fake track; for solutions =
that use some sort of fake video input (canvas.captureStream/etc) it =
inherently allows the application to define the size and framerate, =
which are the primary knobs used by the browser for setting 'max total =
bitrate' for the track.   You don't want the application to set the =
bandwidth directly, you want it to define the parameters of the video =
stream so the system can set "reasonable" limits for that resolution & =
situation.
>=20
> _______________________________________________
> 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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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Diso-8859-1"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; "><div =
apple-content-edited=3D"true">This is an interesting possibility. Using =
some kind of fec as padding could make possible to avoid using a fake =
codec allowing receivers to behave normally and without requiring =
implementing a different behavior for the fake codec data. In addition, =
this protection of the encoded static/black image may avoid PLIs in case =
of severe packet loss when hitting a BW during the probe. This could be =
even used as a probe mechanism for BW estimation with the call fully =
open, but this is another story.
</div>
<br><div><div>El 24/07/2015, a las 10:54, Peter Thatcher &lt;<a =
href=3D"mailto:pthatcher@google.com">pthatcher@google.com</a>&gt; =
escribi=F3:</div><br class=3D"Apple-interchange-newline"><blockquote =
type=3D"cite"><meta http-equiv=3D"Content-Type" content=3D"text/html; =
charset=3Dutf-8"><div dir=3D"ltr"><div class=3D"gmail_default" =
style=3D"font-family:arial,helvetica,sans-serif">Could the new flexfec =
payload type be used for padding? &nbsp;</div></div><div =
class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Thu, Jul 23, =
2015 at 10:17 PM, Randell Jesup <span dir=3D"ltr">&lt;<a =
href=3D"mailto:randell-ietf@jesup.org" =
target=3D"_blank">randell-ietf@jesup.org</a>&gt;</span> =
wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 =
.8ex;border-left:1px #ccc solid;padding-left:1ex">I'd love to be able to =
warm up congestion control, and not just DTLS.&nbsp; That requires =
sending data and allowing it to increase to probe the connection and =
open the congestion window, and preferably sending in both directions =
(i.e. a=3Dsendrecv in the warm-up answer).<br>
<br>
We'd need to find a way to generically pad RTP packets to a target size =
(since encoding a static/black image won't use enough bytes).&nbsp; The =
obvious codec-agnostic mechanism would be a header extension.&nbsp; =
Since they can be any 4-byte-multiple size, that would pretty much work =
up to ~384K (1500 byte packets, 1/frame @30fps). However, we could run =
the fake video framerate arbitrarily high to hit any bandwidth target we =
need.<br>
<br>
Alternatively (and *much* better) we could use a "fake" codec =
(a=3Drtpmap:111 fake/90000) which doesn't actually include any real =
data, just padding.&nbsp; Advantages: no CPU code to =
encode/decode.&nbsp; No dealing with header extensions.&nbsp; Clear =
indication if the other side wants to do spinup (they can reject the =
codec if they don't want to).&nbsp; No encoding/decoding CPU use.<br>
<br>
The offer would include it as a least-preferred codec to indicate warmup =
support.&nbsp; If answerer answers with fake as the accepted codec (and =
real codecs following), it's a warmup answer.&nbsp; Both sides will send =
and receive fake data to spin up the congestion windows (and also the =
jitter buffers, etc).&nbsp; On actual answer, the fake codec would be =
dropped (or dropped to bottom priority).&nbsp; &nbsp;The answerer would =
start sending in a real codec, which would work since the warmup answer =
would have included that codec.&nbsp; The offerer would see the incoming =
real-codec packets, and decode them.&nbsp; &nbsp;When the real answer =
arrives, it will show the accepted codec has changed, and the offerer =
will install that answer and thus and start sending (real) media.<br>
<br>
We still need a way to tell the system at the answerer side "I want to =
accept but it's a warm-up accept" (and perhaps if we want to spin up =
congestion control or not while waiting), and a way to say on the =
offerer side "I want to offer/use warmups".&nbsp; For those API points =
see the previous options from myself and Peter - or simply a way to say =
createAnswer of a warmup answer, and have that agree to all tracks with =
fake codecs being primary.<br>
<br>
I'll note that all of these methods would benefit from a mechanism to =
indicate the desired bitrate targets for each fake track; for solutions =
that use some sort of fake video input (canvas.captureStream/etc) it =
inherently allows the application to define the size and framerate, =
which are the primary knobs used by the browser for setting 'max total =
bitrate' for the track.&nbsp; &nbsp;You don't want the application to =
set the bandwidth directly, you want it to define the parameters of the =
video stream so the system can set "reasonable" limits for that =
resolution &amp; situation.<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" =
rel=3D"noreferrer" =
target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div><br></div>
_______________________________________________<br>rtcweb mailing =
list<br><a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>https://www.ietf.or=
g/mailman/listinfo/rtcweb<br></blockquote></div><br></body></html>=

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From: Justin Uberti <juberti@google.com>
Date: Fri, 24 Jul 2015 11:51:54 +0200
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Cc: Randell Jesup <randell-ietf@jesup.org>, "<rtcweb@ietf.org>" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Congestion warmup
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I tend to think BWE warmup is out of scope for this discussion. It's not
clear that the callee wants to receive perhaps substantial amounts of
traffic prior to having accepted the call.

On Fri, Jul 24, 2015 at 11:47 AM, Luis L=C3=B3pez Fern=C3=A1ndez <luis.lope=
z@urjc.es>
wrote:

> This is an interesting possibility. Using some kind of fec as padding
> could make possible to avoid using a fake codec allowing receivers to
> behave normally and without requiring implementing a different behavior f=
or
> the fake codec data. In addition, this protection of the encoded
> static/black image may avoid PLIs in case of severe packet loss when
> hitting a BW during the probe. This could be even used as a probe mechani=
sm
> for BW estimation with the call fully open, but this is another story.
>
> El 24/07/2015, a las 10:54, Peter Thatcher <pthatcher@google.com>
> escribi=C3=B3:
>
> Could the new flexfec payload type be used for padding?
>
> On Thu, Jul 23, 2015 at 10:17 PM, Randell Jesup <randell-ietf@jesup.org>
> wrote:
>
>> I'd love to be able to warm up congestion control, and not just DTLS.
>> That requires sending data and allowing it to increase to probe the
>> connection and open the congestion window, and preferably sending in bot=
h
>> directions (i.e. a=3Dsendrecv in the warm-up answer).
>>
>> We'd need to find a way to generically pad RTP packets to a target size
>> (since encoding a static/black image won't use enough bytes).  The obvio=
us
>> codec-agnostic mechanism would be a header extension.  Since they can be
>> any 4-byte-multiple size, that would pretty much work up to ~384K (1500
>> byte packets, 1/frame @30fps). However, we could run the fake video
>> framerate arbitrarily high to hit any bandwidth target we need.
>>
>> Alternatively (and *much* better) we could use a "fake" codec
>> (a=3Drtpmap:111 fake/90000) which doesn't actually include any real data=
,
>> just padding.  Advantages: no CPU code to encode/decode.  No dealing wit=
h
>> header extensions.  Clear indication if the other side wants to do spinu=
p
>> (they can reject the codec if they don't want to).  No encoding/decoding
>> CPU use.
>>
>> The offer would include it as a least-preferred codec to indicate warmup
>> support.  If answerer answers with fake as the accepted codec (and real
>> codecs following), it's a warmup answer.  Both sides will send and recei=
ve
>> fake data to spin up the congestion windows (and also the jitter buffers=
,
>> etc).  On actual answer, the fake codec would be dropped (or dropped to
>> bottom priority).   The answerer would start sending in a real codec, wh=
ich
>> would work since the warmup answer would have included that codec.  The
>> offerer would see the incoming real-codec packets, and decode them.   Wh=
en
>> the real answer arrives, it will show the accepted codec has changed, an=
d
>> the offerer will install that answer and thus and start sending (real)
>> media.
>>
>> We still need a way to tell the system at the answerer side "I want to
>> accept but it's a warm-up accept" (and perhaps if we want to spin up
>> congestion control or not while waiting), and a way to say on the offere=
r
>> side "I want to offer/use warmups".  For those API points see the previo=
us
>> options from myself and Peter - or simply a way to say createAnswer of a
>> warmup answer, and have that agree to all tracks with fake codecs being
>> primary.
>>
>> I'll note that all of these methods would benefit from a mechanism to
>> indicate the desired bitrate targets for each fake track; for solutions
>> that use some sort of fake video input (canvas.captureStream/etc) it
>> inherently allows the application to define the size and framerate, whic=
h
>> are the primary knobs used by the browser for setting 'max total bitrate=
'
>> for the track.   You don't want the application to set the bandwidth
>> directly, you want it to define the parameters of the video stream so th=
e
>> system can set "reasonable" limits for that resolution & situation.
>>
>> _______________________________________________
>> 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
>
>

--047d7b10c8715f09af051b9bf5e3
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">I tend to think BWE warmup is out of scope for this discus=
sion. It&#39;s not clear that the callee wants to receive perhaps substanti=
al amounts of traffic prior to having accepted the call.</div><div class=3D=
"gmail_extra"><br><div class=3D"gmail_quote">On Fri, Jul 24, 2015 at 11:47 =
AM, Luis L=C3=B3pez Fern=C3=A1ndez <span dir=3D"ltr">&lt;<a href=3D"mailto:=
luis.lopez@urjc.es" target=3D"_blank">luis.lopez@urjc.es</a>&gt;</span> wro=
te:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-=
left:1px #ccc solid;padding-left:1ex"><div style=3D"word-wrap:break-word"><=
div>This is an interesting possibility. Using some kind of fec as padding c=
ould make possible to avoid using a fake codec allowing receivers to behave=
 normally and without requiring implementing a different behavior for the f=
ake codec data. In addition, this protection of the encoded static/black im=
age may avoid PLIs in case of severe packet loss when hitting a BW during t=
he probe. This could be even used as a probe mechanism for BW estimation wi=
th the call fully open, but this is another story.
</div><div><div class=3D"h5">
<br><div><div>El 24/07/2015, a las 10:54, Peter Thatcher &lt;<a href=3D"mai=
lto:pthatcher@google.com" target=3D"_blank">pthatcher@google.com</a>&gt; es=
cribi=C3=B3:</div><br><blockquote type=3D"cite"><div dir=3D"ltr"><div class=
=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-serif">Could t=
he new flexfec payload type be used for padding? =C2=A0</div></div><div cla=
ss=3D"gmail_extra"><br><div class=3D"gmail_quote">On Thu, Jul 23, 2015 at 1=
0:17 PM, Randell Jesup <span dir=3D"ltr">&lt;<a href=3D"mailto:randell-ietf=
@jesup.org" target=3D"_blank">randell-ietf@jesup.org</a>&gt;</span> wrote:<=
br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left=
:1px #ccc solid;padding-left:1ex">I&#39;d love to be able to warm up conges=
tion control, and not just DTLS.=C2=A0 That requires sending data and allow=
ing it to increase to probe the connection and open the congestion window, =
and preferably sending in both directions (i.e. a=3Dsendrecv in the warm-up=
 answer).<br>
<br>
We&#39;d need to find a way to generically pad RTP packets to a target size=
 (since encoding a static/black image won&#39;t use enough bytes).=C2=A0 Th=
e obvious codec-agnostic mechanism would be a header extension.=C2=A0 Since=
 they can be any 4-byte-multiple size, that would pretty much work up to ~3=
84K (1500 byte packets, 1/frame @30fps). However, we could run the fake vid=
eo framerate arbitrarily high to hit any bandwidth target we need.<br>
<br>
Alternatively (and *much* better) we could use a &quot;fake&quot; codec (a=
=3Drtpmap:111 fake/90000) which doesn&#39;t actually include any real data,=
 just padding.=C2=A0 Advantages: no CPU code to encode/decode.=C2=A0 No dea=
ling with header extensions.=C2=A0 Clear indication if the other side wants=
 to do spinup (they can reject the codec if they don&#39;t want to).=C2=A0 =
No encoding/decoding CPU use.<br>
<br>
The offer would include it as a least-preferred codec to indicate warmup su=
pport.=C2=A0 If answerer answers with fake as the accepted codec (and real =
codecs following), it&#39;s a warmup answer.=C2=A0 Both sides will send and=
 receive fake data to spin up the congestion windows (and also the jitter b=
uffers, etc).=C2=A0 On actual answer, the fake codec would be dropped (or d=
ropped to bottom priority).=C2=A0 =C2=A0The answerer would start sending in=
 a real codec, which would work since the warmup answer would have included=
 that codec.=C2=A0 The offerer would see the incoming real-codec packets, a=
nd decode them.=C2=A0 =C2=A0When the real answer arrives, it will show the =
accepted codec has changed, and the offerer will install that answer and th=
us and start sending (real) media.<br>
<br>
We still need a way to tell the system at the answerer side &quot;I want to=
 accept but it&#39;s a warm-up accept&quot; (and perhaps if we want to spin=
 up congestion control or not while waiting), and a way to say on the offer=
er side &quot;I want to offer/use warmups&quot;.=C2=A0 For those API points=
 see the previous options from myself and Peter - or simply a way to say cr=
eateAnswer of a warmup answer, and have that agree to all tracks with fake =
codecs being primary.<br>
<br>
I&#39;ll note that all of these methods would benefit from a mechanism to i=
ndicate the desired bitrate targets for each fake track; for solutions that=
 use some sort of fake video input (canvas.captureStream/etc) it inherently=
 allows the application to define the size and framerate, which are the pri=
mary knobs used by the browser for setting &#39;max total bitrate&#39; for =
the track.=C2=A0 =C2=A0You don&#39;t want the application to set the bandwi=
dth directly, you want it to define the parameters of the video stream so t=
he system can set &quot;reasonable&quot; limits for that resolution &amp; s=
ituation.<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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div><br></div>
_______________________________________________<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></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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>

--047d7b10c8715f09af051b9bf5e3--


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From: Justin Uberti <juberti@google.com>
Date: Fri, 24 Jul 2015 11:50:41 +0200
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To: Eric Rescorla <ekr@rtfm.com>
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Cc: Randell Jesup <randell-ietf@jesup.org>, Cullen Jennings <fluffy@cisco.com>, Eric Rescorla <ekr@mozilla.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Options for DTLS warmup and directionality
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--001a1140c426279403051b9bf1fd
Content-Type: text/plain; charset=UTF-8

How about:

track = new MediaStreamTrack("audio")
addTrack(track)

or

track = new MediaStreamTrack();
track.kind = "audio";
addTrack(track)

On Fri, Jul 24, 2015 at 11:00 AM, Eric Rescorla <ekr@rtfm.com> wrote:

>
> On Fri, Jul 24, 2015 at 10:57 AM, Peter Thatcher <pthatcher@google.com>
> wrote:
>
>> Would you be comfortable with either addTrack("audio")
>>
>
> This seems awful.
>
>
>
>> or createRtpSender("audio")?
>>
>
> This seems OKish
>
> -Ekr
>
>
>> On Fri, Jul 24, 2015 at 12:50 AM, Martin Thomson <
>> martin.thomson@gmail.com> wrote:
>>
>>> On 24 July 2015 at 00:31, Peter Thatcher <pthatcher@google.com> wrote:
>>> > PeerConnection.addTrack(null)
>>>
>>> I'm not entirely comfortable with the notion that we allow null
>>> arguments.  Doing that could be a footgun.  Especially since it is
>>> only to support what I consider to be a corner case (even though it
>>> might be an important one).
>>>
>>
>>
>> _______________________________________________
>> 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
>
>

--001a1140c426279403051b9bf1fd
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<div dir=3D"ltr">How about:<div><br></div><div>track =3D new MediaStreamTra=
ck(&quot;audio&quot;)</div><div>addTrack(track)</div><div><br></div><div>or=
=C2=A0</div><div><br></div><div>track =3D new MediaStreamTrack();</div><div=
>track.kind =3D &quot;audio&quot;;</div><div>addTrack(track)<br></div></div=
><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Fri, Jul 24, =
2015 at 11:00 AM, Eric Rescorla <span dir=3D"ltr">&lt;<a href=3D"mailto:ekr=
@rtfm.com" target=3D"_blank">ekr@rtfm.com</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"><div dir=3D"ltr"><div class=3D"gmail_extra"><br><div=
 class=3D"gmail_quote"><span class=3D"">On Fri, Jul 24, 2015 at 10:57 AM, P=
eter Thatcher <span dir=3D"ltr">&lt;<a href=3D"mailto:pthatcher@google.com"=
 target=3D"_blank">pthatcher@google.com</a>&gt;</span> wrote:<br><blockquot=
e class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc sol=
id;padding-left:1ex"><div dir=3D"ltr"><div style=3D"font-family:arial,helve=
tica,sans-serif">Would you be comfortable with either addTrack(&quot;audio&=
quot;)</div></div></blockquote><div><br></div></span><div>This seems awful.=
</div><div><br></div><div>=C2=A0</div><blockquote class=3D"gmail_quote" sty=
le=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div d=
ir=3D"ltr"><div style=3D"font-family:arial,helvetica,sans-serif"> or create=
RtpSender(&quot;audio&quot;)?</div></div></blockquote><div><br></div><div>T=
his seems OKish</div><div><br></div><div>-Ekr</div><div>=C2=A0</div><blockq=
uote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc =
solid;padding-left:1ex"><span class=3D""><div class=3D"gmail_extra"><div cl=
ass=3D"gmail_quote">On Fri, Jul 24, 2015 at 12:50 AM, Martin Thomson <span =
dir=3D"ltr">&lt;<a href=3D"mailto:martin.thomson@gmail.com" target=3D"_blan=
k">martin.thomson@gmail.com</a>&gt;</span> wrote:<span><br><blockquote clas=
s=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;pad=
ding-left:1ex"><span>On 24 July 2015 at 00:31, Peter Thatcher &lt;<a href=
=3D"mailto:pthatcher@google.com" target=3D"_blank">pthatcher@google.com</a>=
&gt; wrote:<br>
&gt; PeerConnection.addTrack(null)<br>
<br>
</span>I&#39;m not entirely comfortable with the notion that we allow null<=
br>
arguments.=C2=A0 Doing that could be a footgun.=C2=A0 Especially since it i=
s<br>
only to support what I consider to be a corner case (even though it<br>
might be an important one).<br>
</blockquote></span></div><br></div>
<br></span><span class=3D"">_______________________________________________=
<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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></span></blockquote></div><br></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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>

--001a1140c426279403051b9bf1fd--


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From: Karen Elisabeth Egede Nielsen <karen.nielsen@tieto.com>
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Subject: [rtcweb] diffserv marking for rtcwb data channel
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--001a113f9656913488051b995788
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HI,



I don=E2=80=99t think that it makes much difference (though see further bel=
ow=E2=80=A6)
whether one use

different sctp associations for multiple data channels with different

diffserv code points or have this feature be built into SCTP.



The differences I can think of are:

=C2=B7         end-point performance (only significant if cache and cpu
constrained)

=C2=B7         flow control handling =E2=80=93 can be both good and bad to =
have separate
receiver windows perhaps



I do question, however, whether it is a =E2=80=9Cgood=E2=80=9D choice to us=
e the diffserv
marking and sctp association split for

prioritization of the traffic instead of SCTP scheduling differentiation
within one and the same association over a non-marked

or homogeneously marked network.

Especially the two have very different characteristics from a (coupled)
congestion control perspective (including loss recovery).



But if both possibilities go in, I suppose that we will learn much more
about this in the future.

But one should perhaps do well  to provide some starting ground rules for
when one or the other is considered

most appropriate to use =E2=80=93 possible try to describe the conditions u=
nder
which one would like to use one or

the other.



Finally there was a comment at the tsvwg meeting that the same 5-tuple with
different diffserv markings would not get

through the network =E2=80=93 if that generally is correct,  then the solut=
ion is
very easy: one MUST use different associations to get port (5-tuple)
differentiation.

I think that if this is correct, one also wants to ask what happens if the
diffserv is changed for the same 5-tuple =E2=80=93 i.e., is the network

robust to this change even if it is not robust to concurrent usage of
different markings for the same 5-tuple.



BR, Karen

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class=3D"WordSection1"><p class=3D"MsoNormal"><span lang=3D"DA">HI,</span><=
/p><p class=3D"MsoNormal"><span lang=3D"DA">=C2=A0</span></p><p class=3D"Ms=
oNormal">I don=E2=80=99t think that it makes much difference (though see fu=
rther below=E2=80=A6) whether one use </p><p class=3D"MsoNormal">different =
sctp associations for multiple data channels with different </p><p class=3D=
"MsoNormal">diffserv code points or have this feature be built into SCTP.</=
p><p class=3D"MsoNormal">=C2=A0</p><p class=3D"MsoNormal">The differences I=
 can think of are:</p><p class=3D"MsoListParagraph" style=3D"text-indent:-1=
8.0pt;mso-list:l0 level1 lfo1"><span style=3D"font-family:Symbol"><span sty=
le=3D"mso-list:Ignore">=C2=B7<span style=3D"font:7.0pt &quot;Times New Roma=
n&quot;">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 </span></span></s=
pan>end-point performance (only significant if cache and cpu constrained)</=
p><p class=3D"MsoListParagraph" style=3D"text-indent:-18.0pt;mso-list:l0 le=
vel1 lfo1"><span style=3D"font-family:Symbol"><span style=3D"mso-list:Ignor=
e">=C2=B7<span style=3D"font:7.0pt &quot;Times New Roman&quot;">=C2=A0=C2=
=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 </span></span></span>flow control h=
andling =E2=80=93 can be both good and bad to have separate receiver window=
s perhaps</p><p class=3D"MsoNormal">=C2=A0</p><p class=3D"MsoNormal">I do q=
uestion, however, whether it is a =E2=80=9Cgood=E2=80=9D choice to use the =
diffserv marking and sctp association split for</p><p class=3D"MsoNormal">p=
rioritization of the traffic instead of SCTP scheduling differentiation wit=
hin one and the same association over a non-marked </p><p class=3D"MsoNorma=
l">or homogeneously marked network.</p><p class=3D"MsoNormal">Especially th=
e two have very different characteristics from a (coupled) congestion contr=
ol perspective (including loss recovery).</p><p class=3D"MsoNormal">=C2=A0<=
/p><p class=3D"MsoNormal">But if both possibilities go in, I suppose that w=
e will learn much more about this in the future.</p><p class=3D"MsoNormal">=
But one should perhaps do well =C2=A0to provide some starting ground rules =
for when one or the other is considered</p><p class=3D"MsoNormal">most appr=
opriate to use =E2=80=93 possible try to describe the conditions under whic=
h one would like to use one or</p><p class=3D"MsoNormal">the other. =C2=A0<=
/p><p class=3D"MsoNormal">=C2=A0</p><p class=3D"MsoNormal">Finally there wa=
s a comment at the tsvwg meeting that the same 5-tuple with different diffs=
erv markings would not get</p><p class=3D"MsoNormal">through the network =
=E2=80=93 if that generally is correct, =C2=A0then the solution is very eas=
y: one MUST use different associations to get port (5-tuple) differentiatio=
n.</p><p class=3D"MsoNormal">I think that if this is correct, one also want=
s to ask what happens if the diffserv is changed for the same 5-tuple =E2=
=80=93 i.e., is the network </p><p class=3D"MsoNormal">robust to this chang=
e even if it is not robust to concurrent usage of different markings for th=
e same 5-tuple.</p><p class=3D"MsoNormal">=C2=A0</p><p class=3D"MsoNormal">=
BR, Karen =C2=A0</p><p class=3D"MsoNormal" style=3D"margin-left:18.0pt">=C2=
=A0</p><p class=3D"MsoNormal" style=3D"margin-left:18.0pt">=C2=A0</p><p cla=
ss=3D"MsoNormal">=C2=A0</p></div></body></html>

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From: Karen Elisabeth Egede Nielsen <karen.nielsen@tieto.com>
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Subject: Re: [rtcweb] [tsvwg] diffserv marking for rtcwb data channel
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Hi Ruediger,



Thanks a lot.



So I think that you=E2=80=99re saying that even if one made sure that diffe=
rent
diff serv markings were

split on different 5-tuples, this setting likely would be bleached at the
network.



Is that correctly understood ?



If so, then (it sounds to me =E2=80=93 not knowing too much about this) tha=
t from a
coupling congestion control perspective it

would be preferable to  use SCTP scheduling prioritization (within one and
the same association) for fine granular differentiation in between

multiple data channels in between the same end-points rather than
differentiation by means of diff serv marking.



When/if one get a more consistent and forceful ECN operation in AQMs and
SCTP end-points, perhaps the coupling congestion

principles achieved by means of sctp stream scheduling get to be less
important.

On the other hand, again as said before, having the one and the same, or
multiple, flow control (receiver buffer) contexts may be

a strength (simplicity) or a weakness.



BR, Karen



*From:* Ruediger.Geib@telekom.de [mailto:Ruediger.Geib@telekom.de]
*Sent:* Friday, July 24, 2015 10:00 AM
*To:* karen.nielsen@tieto.com
*Cc:* rtcweb@ietf.org; tsvwg@ietf.org
*Subject:* AW: [tsvwg] diffserv marking for rtcwb data channel



Karen,



      Finally there was a comment at the tsvwg meeting that the same
5-tuple with

       different diffserv markings would not get through the network =E2=80=
=93 if
that generally

       is correct,  then the solution is very easy: one MUST use different
associations to

       get port (5-tuple) differentiation.

       I think that if this is correct, one also wants to ask what happens
if the diffserv is

       changed for the same 5-tuple =E2=80=93 i.e., is the network robust t=
o this
change even if

       it is not robust to concurrent usage of different markings for the
same 5-tuple.



Talking about the policies of the company I work for only:



Possibilities to enforce QoS policies at a provider BRAS or BNG terminating
wired

consumer accesses are limited. QoS policies based on full 5 tuple are very

conservative and hard to maintain. I wouldn=E2=80=99t expect full 5 tuple a=
s basis
for a QoS

policy then. Trust in the correct information on which a BRAS/BNG can

base QoS classification is an important issue. In any case, I=E2=80=99d not=
 expect

more than one DSCP is assigned based on such a policy for the time

being. I=E2=80=99m not aware of any actual requirement to support e.g. an A=
Fn

class by all three PHBs.



At interconnection points, any unexpected DSCP is bleached. This must

be done to protect backbone QoS production. QoS transport can be

negotiated for a set of defined DSCPs.



A study of the EU found DPI to be more common in wireless communication
than in wired communication. I=E2=80=99d not expect the QoS policies in tha=
t area
to be more liberal than for fixed access networks for the time being.



Regards,



Ruediger

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--></style></head><body lang=3D"EN-US" link=3D"blue" vlink=3D"purple"><div =
class=3D"WordSection1"><p class=3D"MsoNormal"><span style=3D"color:#1f497d"=
>Hi Ruediger,</span></p><p class=3D"MsoNormal"><span style=3D"color:#1f497d=
">=C2=A0</span></p><p class=3D"MsoNormal"><span style=3D"color:#1f497d">Tha=
nks a lot.</span></p><p class=3D"MsoNormal"><span style=3D"color:#1f497d">=
=C2=A0</span></p><p class=3D"MsoNormal"><span style=3D"color:#1f497d">So I =
think that you=E2=80=99re saying that even if one made sure that different =
diff serv markings were</span></p><p class=3D"MsoNormal"><span style=3D"col=
or:#1f497d">split on different 5-tuples, this setting likely would be bleac=
hed at the network.</span></p><p class=3D"MsoNormal"><span style=3D"color:#=
1f497d">=C2=A0</span></p><p class=3D"MsoNormal"><span style=3D"color:#1f497=
d">Is that correctly understood ?</span></p><p class=3D"MsoNormal"><span st=
yle=3D"color:#1f497d">=C2=A0</span></p><p class=3D"MsoNormal"><span style=
=3D"color:#1f497d">If so, then (it sounds to me =E2=80=93 not knowing too m=
uch about this) that from a coupling congestion control perspective it </sp=
an></p><p class=3D"MsoNormal"><span style=3D"color:#1f497d">would be prefer=
able to=C2=A0 use SCTP scheduling prioritization (within one and the same a=
ssociation) for fine granular differentiation in between</span></p><p class=
=3D"MsoNormal"><span style=3D"color:#1f497d">multiple data channels in betw=
een the same end-points rather than differentiation by means of diff serv m=
arking.</span></p><p class=3D"MsoNormal"><span style=3D"color:#1f497d">=C2=
=A0</span></p><p class=3D"MsoNormal"><span style=3D"color:#1f497d">When/if =
one get a more consistent and forceful ECN operation in AQMs and SCTP end-p=
oints, perhaps the coupling congestion</span></p><p class=3D"MsoNormal"><sp=
an style=3D"color:#1f497d">principles achieved by means of sctp stream sche=
duling get to be less important. </span></p><p class=3D"MsoNormal"><span st=
yle=3D"color:#1f497d">On the other hand, again as said before, having the o=
ne and the same, or multiple, flow control (receiver buffer) contexts may b=
e</span></p><p class=3D"MsoNormal"><span style=3D"color:#1f497d">a strength=
 (simplicity) or a weakness. =C2=A0=C2=A0=C2=A0</span></p><p class=3D"MsoNo=
rmal"><span style=3D"color:#1f497d">=C2=A0</span></p><p class=3D"MsoNormal"=
><span style=3D"color:#1f497d">BR, Karen </span></p><p class=3D"MsoNormal">=
<span style=3D"color:#1f497d">=C2=A0</span></p><div style=3D"border:none;bo=
rder-left:solid blue 1.5pt;padding:0cm 0cm 0cm 4.0pt"><div><div style=3D"bo=
rder:none;border-top:solid #b5c4df 1.0pt;padding:3.0pt 0cm 0cm 0cm"><p clas=
s=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot;Tahoma=
&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-size:10.=
0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> <a href=3D"mail=
to:Ruediger.Geib@telekom.de">Ruediger.Geib@telekom.de</a> [mailto:<a href=
=3D"mailto:Ruediger.Geib@telekom.de">Ruediger.Geib@telekom.de</a>] <br><b>S=
ent:</b> Friday, July 24, 2015 10:00 AM<br><b>To:</b> <a href=3D"mailto:kar=
en.nielsen@tieto.com">karen.nielsen@tieto.com</a><br><b>Cc:</b> <a href=3D"=
mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>; <a href=3D"mailto:tsvwg@ietf.o=
rg">tsvwg@ietf.org</a><br><b>Subject:</b> AW: [tsvwg] diffserv marking for =
rtcwb data channel</span></p></div></div><p class=3D"MsoNormal">=C2=A0</p><=
p class=3D"MsoNormal"><span style=3D"color:#1f497d">Karen,</span></p><p cla=
ss=3D"MsoNormal"><span style=3D"color:#1f497d">=C2=A0</span></p><p class=3D=
"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quo=
t;,&quot;sans-serif&quot;;color:#1f497d">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 </s=
pan></b>Finally there was a comment at the tsvwg meeting that the same 5-tu=
ple with <span style=3D"color:#1f497d"></span></p><p class=3D"MsoNormal"><s=
pan style=3D"color:#1f497d">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0</spa=
n>different diffserv markings would not get<span style=3D"color:#1f497d"> <=
/span>through the network =E2=80=93 if that generally <span style=3D"color:=
#1f497d"></span></p><p class=3D"MsoNormal"><span style=3D"color:#1f497d">=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0</span>is correct, =C2=A0then the=
 solution is very easy: one MUST use different associations to <span style=
=3D"color:#1f497d"></span></p><p class=3D"MsoNormal"><span style=3D"color:#=
1f497d">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0</span>get port (5-tuple)=
 differentiation.</p><p class=3D"MsoNormal"><span style=3D"color:#1f497d">=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 </span>I think that if this is correct=
, one also wants to ask what happens if the diffserv is <span style=3D"colo=
r:#1f497d"></span></p><p class=3D"MsoNormal"><span style=3D"color:#1f497d">=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0</span>changed for the same 5-tup=
le =E2=80=93 i.e., is the network robust to this change even if <span style=
=3D"color:#1f497d"></span></p><p class=3D"MsoNormal"><span style=3D"color:#=
1f497d">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0</span>it is not robust t=
o concurrent usage of different markings for the same 5-tuple.</p><p class=
=3D"MsoNormal">=C2=A0<span style=3D"color:#1f497d"></span></p><p class=3D"M=
soNormal"><span style=3D"color:#1f497d">Talking about the policies of the c=
ompany I work for only:</span></p><p class=3D"MsoNormal"><span style=3D"col=
or:#1f497d">=C2=A0</span></p><p class=3D"MsoNormal"><span style=3D"color:#1=
f497d">Possibilities to enforce QoS policies at a provider BRAS or</span>=
=C2=A0<span style=3D"color:#1f497d">BNG terminating wired </span></p><p cla=
ss=3D"MsoNormal"><span style=3D"color:#1f497d">consumer accesses are limite=
d. QoS policies based on full 5 tuple are very </span></p><p class=3D"MsoNo=
rmal"><span style=3D"color:#1f497d">conservative and hard to maintain. I wo=
uldn=E2=80=99t expect full 5 tuple as basis for a QoS </span></p><p class=
=3D"MsoNormal"><span style=3D"color:#1f497d">policy then. Trust in the corr=
ect information on which a BRAS/BNG can </span></p><p class=3D"MsoNormal"><=
span style=3D"color:#1f497d">base QoS classification is an important issue.=
 In any case, I=E2=80=99d not expect </span></p><p class=3D"MsoNormal"><spa=
n style=3D"color:#1f497d">more than one DSCP is assigned based on such a po=
licy for the time </span></p><p class=3D"MsoNormal"><span style=3D"color:#1=
f497d">being. I=E2=80=99m not aware of any actual requirement to support e.=
g. an AFn </span></p><p class=3D"MsoNormal"><span style=3D"color:#1f497d">c=
lass by all three PHBs.</span></p><p class=3D"MsoNormal"><span style=3D"col=
or:#1f497d">=C2=A0</span></p><p class=3D"MsoNormal"><span style=3D"color:#1=
f497d">At interconnection points, any unexpected DSCP is bleached. This mus=
t </span></p><p class=3D"MsoNormal"><span style=3D"color:#1f497d">be done t=
o protect backbone QoS production. QoS transport can be </span></p><p class=
=3D"MsoNormal"><span style=3D"color:#1f497d">negotiated for a set of define=
d DSCPs. </span></p><p class=3D"MsoNormal"><span style=3D"color:#1f497d">=
=C2=A0</span></p><p class=3D"MsoNormal"><span style=3D"color:#1f497d">A stu=
dy of the EU found DPI to be more common in wireless communication than in =
wired communication. I=E2=80=99d not expect the QoS policies in that area t=
o be more liberal than for fixed access networks for the time being.</span>=
</p><p class=3D"MsoNormal"><span style=3D"color:#1f497d">=C2=A0</span></p><=
p class=3D"MsoNormal"><span style=3D"color:#1f497d">Regards,</span></p><p c=
lass=3D"MsoNormal"><span style=3D"color:#1f497d">=C2=A0</span></p><p class=
=3D"MsoNormal"><span style=3D"color:#1f497d">Ruediger</span></p><p class=3D=
"MsoNormal"><span style=3D"color:#1f497d">=C2=A0</span></p><p class=3D"MsoN=
ormal"><span style=3D"color:#1f497d">=C2=A0</span></p><p class=3D"MsoNormal=
" style=3D"margin-left:18.0pt">=C2=A0</p><p class=3D"MsoNormal">=C2=A0</p><=
/div></div></body></html>

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To: <karen.nielsen@tieto.com>
Date: Fri, 24 Jul 2015 11:32:33 +0200
Thread-Topic: [tsvwg] diffserv marking for rtcwb data channel
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From: Peter Thatcher <pthatcher@google.com>
Date: Fri, 24 Jul 2015 03:11:54 -0700
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Subject: [rtcweb] Comments on draft-nandakumar-rtcweb-sdp
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In the WG meeting today, we were asked to
read draft-nandakumar-rtcweb-sdp.  So I read it.  Here are my comments:

1.  The annotated examples with ASCII art tables are really nice.  I like
that part.  Detailed, readable, and thorough.

2.  The diagrams with offer/answer diagrams imply certain models of
signalling which might not be in use by WebRTC applications.  I think
instead they should be modeled as calls to createOffer, createAnswer,
setLocalDescription and setRemoteDescription.  There are applications
already that basically build custom SDP with JS to use it as an API and
doesn't send the SDP over the signalling.   Developers doing such things
would find examples of SDP useful, but the diagrams which assume signalling
wouldn't fit.

In other words, instead of this document being "Examples of SDP that is
signalled", I think it should be "Examples of SDP that can be put into
setLocalDescription and setRemoteDescription, and returned by createOffer
and createAnswer".

I think Harald said something similar in the WG meeting.

--047d7bd76692172299051b9c3ed6
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<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:arial,he=
lvetica,sans-serif">In the WG meeting today, we were asked to read=C2=A0dra=
ft-nandakumar-rtcweb-sdp.=C2=A0 So I read it.=C2=A0 Here are my comments:</=
div><div class=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-=
serif"><br></div><div class=3D"gmail_default" style=3D"font-family:arial,he=
lvetica,sans-serif">1.=C2=A0 The annotated examples with ASCII art tables a=
re really nice.=C2=A0 I like that part.=C2=A0 Detailed, readable, and thoro=
ugh.=C2=A0</div><div class=3D"gmail_default" style=3D"font-family:arial,hel=
vetica,sans-serif"><br></div><div class=3D"gmail_default" style=3D"font-fam=
ily:arial,helvetica,sans-serif">2.=C2=A0 The diagrams with offer/answer dia=
grams imply certain models of signalling which might not be in use by WebRT=
C applications.=C2=A0 I think instead they should be modeled as calls to cr=
eateOffer, createAnswer, setLocalDescription and setRemoteDescription.=C2=
=A0 There are applications already that basically build custom SDP with JS =
to use it as an API and doesn&#39;t send the SDP over the signalling. =C2=
=A0 Developers doing such things would find examples of SDP useful, but the=
 diagrams which assume signalling wouldn&#39;t fit. =C2=A0</div><div class=
=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-serif"><br></d=
iv><div class=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-s=
erif">In other words, instead of this document being &quot;Examples of SDP =
that is signalled&quot;, I think it should be &quot;Examples of SDP that ca=
n be put into setLocalDescription and setRemoteDescription, and returned by=
 createOffer and createAnswer&quot;.</div><div class=3D"gmail_default" styl=
e=3D"font-family:arial,helvetica,sans-serif"><br></div><div class=3D"gmail_=
default" style=3D"font-family:arial,helvetica,sans-serif">I think Harald sa=
id something similar in the WG meeting.</div></div>

--047d7bd76692172299051b9c3ed6--


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Subject: Re: [rtcweb] Options for DTLS warmup and directionality
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On 24 July 2015 at 11:50, Justin Uberti <juberti@google.com> wrote:
> track = new MediaStreamTrack();
> track.kind = "audio";
> addTrack(track)


I would like to keep track.kind readonly.


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From: Peter Thatcher <pthatcher@google.com>
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Subject: Re: [rtcweb] Options for DTLS warmup and directionality
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--001a114266aad8697b051b9c4947
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I prefer a blank RtpSender over a blank Track, and thus prefer
createRtpSender("audio"), but I could live with "new
MediaStreamTrack('audio')".  I wouldn't want the second one where you set
the kind because there's a time period in which you have a track with no
kind, which is strange, even for a fake one.

On Fri, Jul 24, 2015 at 2:50 AM, Justin Uberti <juberti@google.com> wrote:

> How about:
>
> track = new MediaStreamTrack("audio")
> addTrack(track)
>
> or
>
> track = new MediaStreamTrack();
> track.kind = "audio";
> addTrack(track)
>
> On Fri, Jul 24, 2015 at 11:00 AM, Eric Rescorla <ekr@rtfm.com> wrote:
>
>>
>> On Fri, Jul 24, 2015 at 10:57 AM, Peter Thatcher <pthatcher@google.com>
>> wrote:
>>
>>> Would you be comfortable with either addTrack("audio")
>>>
>>
>> This seems awful.
>>
>>
>>
>>> or createRtpSender("audio")?
>>>
>>
>> This seems OKish
>>
>> -Ekr
>>
>>
>>> On Fri, Jul 24, 2015 at 12:50 AM, Martin Thomson <
>>> martin.thomson@gmail.com> wrote:
>>>
>>>> On 24 July 2015 at 00:31, Peter Thatcher <pthatcher@google.com> wrote:
>>>> > PeerConnection.addTrack(null)
>>>>
>>>> I'm not entirely comfortable with the notion that we allow null
>>>> arguments.  Doing that could be a footgun.  Especially since it is
>>>> only to support what I consider to be a corner case (even though it
>>>> might be an important one).
>>>>
>>>
>>>
>>> _______________________________________________
>>> 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
>>
>>
>

--001a114266aad8697b051b9c4947
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<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:arial,he=
lvetica,sans-serif">I prefer a blank RtpSender over a blank Track, and thus=
 prefer createRtpSender(&quot;audio&quot;), but I could live with &quot;new=
 MediaStreamTrack(&#39;audio&#39;)&quot;.=C2=A0 I wouldn&#39;t want the sec=
ond one where you set the kind because there&#39;s a time period in which y=
ou have a track with no kind, which is strange, even for a fake one.</div><=
/div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Fri, Jul =
24, 2015 at 2:50 AM, Justin Uberti <span dir=3D"ltr">&lt;<a href=3D"mailto:=
juberti@google.com" target=3D"_blank">juberti@google.com</a>&gt;</span> wro=
te:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-=
left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr">How about:<div><br><=
/div><div>track =3D new MediaStreamTrack(&quot;audio&quot;)</div><div>addTr=
ack(track)</div><div><br></div><div>or=C2=A0</div><div><br></div><div>track=
 =3D new MediaStreamTrack();</div><div>track.kind =3D &quot;audio&quot;;</d=
iv><div>addTrack(track)<br></div></div><div class=3D"HOEnZb"><div class=3D"=
h5"><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Fri, Jul 2=
4, 2015 at 11:00 AM, Eric Rescorla <span dir=3D"ltr">&lt;<a href=3D"mailto:=
ekr@rtfm.com" target=3D"_blank">ekr@rtfm.com</a>&gt;</span> wrote:<br><bloc=
kquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #cc=
c solid;padding-left:1ex"><div dir=3D"ltr"><div class=3D"gmail_extra"><br><=
div class=3D"gmail_quote"><span>On Fri, Jul 24, 2015 at 10:57 AM, Peter Tha=
tcher <span dir=3D"ltr">&lt;<a href=3D"mailto:pthatcher@google.com" target=
=3D"_blank">pthatcher@google.com</a>&gt;</span> wrote:<br><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex"><div dir=3D"ltr"><div style=3D"font-family:arial,helvetica,sa=
ns-serif">Would you be comfortable with either addTrack(&quot;audio&quot;)<=
/div></div></blockquote><div><br></div></span><div>This seems awful.</div><=
div><br></div><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"m=
argin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"l=
tr"><div style=3D"font-family:arial,helvetica,sans-serif"> or createRtpSend=
er(&quot;audio&quot;)?</div></div></blockquote><div><br></div><div>This see=
ms OKish</div><div><br></div><div>-Ekr</div><div>=C2=A0</div><blockquote cl=
ass=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;p=
adding-left:1ex"><span><div class=3D"gmail_extra"><div class=3D"gmail_quote=
">On Fri, Jul 24, 2015 at 12:50 AM, Martin Thomson <span dir=3D"ltr">&lt;<a=
 href=3D"mailto:martin.thomson@gmail.com" target=3D"_blank">martin.thomson@=
gmail.com</a>&gt;</span> wrote:<span><br><blockquote class=3D"gmail_quote" =
style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><sp=
an>On 24 July 2015 at 00:31, Peter Thatcher &lt;<a href=3D"mailto:pthatcher=
@google.com" target=3D"_blank">pthatcher@google.com</a>&gt; wrote:<br>
&gt; PeerConnection.addTrack(null)<br>
<br>
</span>I&#39;m not entirely comfortable with the notion that we allow null<=
br>
arguments.=C2=A0 Doing that could be a footgun.=C2=A0 Especially since it i=
s<br>
only to support what I consider to be a corner case (even though it<br>
might be an important one).<br>
</blockquote></span></div><br></div>
<br></span><span>_______________________________________________<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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></span></blockquote></div><br></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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>
</div></div></blockquote></div><br></div>

--001a114266aad8697b051b9c4947--


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On 7/24/2015 6:15 AM, Peter Thatcher wrote:
> I prefer a blank RtpSender over a blank Track, and thus prefer 
> createRtpSender("audio"), but I could live with "new 
> MediaStreamTrack('audio')".  I wouldn't want the second one where you 
> set the kind because there's a time period in which you have a track 
> with no kind, which is strange, even for a fake one.

If you have new MediaStreamTrack("audio"), you need to define what this 
means for *other* sinks/etc for MediaStreamTracks...  media elements, 
webaudio, combined in MediaStream constructors, etc... which is an 
argument for createRtpSender("audio").

Note that from JS's point of view, track = new MediaStreamTrack(); 
track.kind = "audio" is all in the same 'run' between stable states, so 
it's not really supposed to be "observable".  However, what if the JS 
doesn't do that... you can have a track that's neither audio nor video...

>
> On Fri, Jul 24, 2015 at 2:50 AM, Justin Uberti <juberti@google.com 
> <mailto:juberti@google.com>> wrote:
>
>     How about:
>
>     track = new MediaStreamTrack("audio")
>     addTrack(track)
>
>     or
>
>     track = new MediaStreamTrack();
>     track.kind = "audio";
>     addTrack(track)
>
>     On Fri, Jul 24, 2015 at 11:00 AM, Eric Rescorla <ekr@rtfm.com
>     <mailto:ekr@rtfm.com>> wrote:
>
>
>         On Fri, Jul 24, 2015 at 10:57 AM, Peter Thatcher
>         <pthatcher@google.com <mailto:pthatcher@google.com>> wrote:
>
>             Would you be comfortable with either addTrack("audio")
>
>
>         This seems awful.
>
>             or createRtpSender("audio")?
>
>
>         This seems OKish
>
>         -Ekr
>

-- 
Randell Jesup -- rjesup a t mozilla d o t com
Please please please don't email randell-ietf@jesup.org!  Way too much spam


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<html>
  <head>
    <meta content=3D"text/html; charset=3Dwindows-1252"
      http-equiv=3D"Content-Type">
  </head>
  <body bgcolor=3D"#FFFFFF" text=3D"#000000">
    <div class=3D"moz-cite-prefix">On 7/24/2015 6:15 AM, Peter Thatcher
      wrote:<br>
    </div>
    <blockquote
cite=3D"mid:CAJrXDUEgj2RPdRm11y6DJXkZ1uF-JcHU9h59MbZGN5WDL8jE5g@mail.gmai=
l.com"
      type=3D"cite">
      <div dir=3D"ltr">
        <div class=3D"gmail_default"
          style=3D"font-family:arial,helvetica,sans-serif">I prefer a
          blank RtpSender over a blank Track, and thus prefer
          createRtpSender("audio"), but I could live with "new
          MediaStreamTrack('audio')".=A0 I wouldn't want the second one
          where you set the kind because there's a time period in which
          you have a track with no kind, which is strange, even for a
          fake one.</div>
      </div>
    </blockquote>
    <br>
    If you have new MediaStreamTrack("audio"), you need to define what
    this means for *other* sinks/etc for MediaStreamTracks...=A0 media
    elements, webaudio, combined in MediaStream constructors, etc...
    which is an argument for createRtpSender("audio"). <br>
    <br>
    Note that from JS's point of view, track =3D new MediaStreamTrack();
    track.kind =3D "audio" is all in the same 'run' between stable states=
,
    so it's not really supposed to be "observable".=A0 However, what if
    the JS doesn't do that... you can have a track that's neither audio
    nor video...<br>
    <br>
    <blockquote
cite=3D"mid:CAJrXDUEgj2RPdRm11y6DJXkZ1uF-JcHU9h59MbZGN5WDL8jE5g@mail.gmai=
l.com"
      type=3D"cite">
      <div class=3D"gmail_extra"><br>
        <div class=3D"gmail_quote">On Fri, Jul 24, 2015 at 2:50 AM, Justi=
n
          Uberti <span dir=3D"ltr">&lt;<a moz-do-not-send=3D"true"
              href=3D"mailto:juberti@google.com" target=3D"_blank">jubert=
i@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">
            <div dir=3D"ltr">How about:
              <div><br>
              </div>
              <div>track =3D new MediaStreamTrack("audio")</div>
              <div>addTrack(track)</div>
              <div><br>
              </div>
              <div>or=A0</div>
              <div><br>
              </div>
              <div>track =3D new MediaStreamTrack();</div>
              <div>track.kind =3D "audio";</div>
              <div>addTrack(track)<br>
              </div>
            </div>
            <div class=3D"HOEnZb">
              <div class=3D"h5">
                <div class=3D"gmail_extra"><br>
                  <div class=3D"gmail_quote">On Fri, Jul 24, 2015 at 11:0=
0
                    AM, Eric Rescorla <span dir=3D"ltr">&lt;<a
                        moz-do-not-send=3D"true"
                        href=3D"mailto:ekr@rtfm.com" target=3D"_blank">ek=
r@rtfm.com</a>&gt;</span>
                    wrote:<br>
                    <blockquote class=3D"gmail_quote" style=3D"margin:0 0=
 0
                      .8ex;border-left:1px #ccc solid;padding-left:1ex">
                      <div dir=3D"ltr">
                        <div class=3D"gmail_extra"><br>
                          <div class=3D"gmail_quote"><span>On Fri, Jul 24=
,
                              2015 at 10:57 AM, Peter Thatcher <span
                                dir=3D"ltr">&lt;<a moz-do-not-send=3D"tru=
e"
                                  href=3D"mailto:pthatcher@google.com"
                                  target=3D"_blank">pthatcher@google.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 dir=3D"ltr">
                                  <div
                                    style=3D"font-family:arial,helvetica,=
sans-serif">Would
                                    you be comfortable with either
                                    addTrack("audio")</div>
                                </div>
                              </blockquote>
                              <div><br>
                              </div>
                            </span>
                            <div>This seems awful.</div>
                            <div><br>
                            </div>
                            <div>=A0</div>
                            <blockquote class=3D"gmail_quote"
                              style=3D"margin:0 0 0 .8ex;border-left:1px
                              #ccc solid;padding-left:1ex">
                              <div dir=3D"ltr">
                                <div
                                  style=3D"font-family:arial,helvetica,sa=
ns-serif">
                                  or createRtpSender("audio")?</div>
                              </div>
                            </blockquote>
                            <div><br>
                            </div>
                            <div>This seems OKish</div>
                            <div><br>
                            </div>
                            <div>-Ekr</div>
                            <div>=A0</div>
                            <br>
                          </div>
                        </div>
                      </div>
                    </blockquote>
                  </div>
                </div>
              </div>
            </div>
          </blockquote>
        </div>
      </div>
    </blockquote>
    <br>
    <pre class=3D"moz-signature" cols=3D"72">--=20
Randell Jesup -- rjesup a t mozilla d o t com
Please please please don't email <a class=3D"moz-txt-link-abbreviated" hr=
ef=3D"mailto:randell-ietf@jesup.org">randell-ietf@jesup.org</a>!  Way too=
 much spam
</pre>
  </body>
</html>

--------------040001040800040407070308--


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> On 18 Jul 2015, at 00:41, Martin Thomson <martin.thomson@gmail.com =
<mailto:martin.thomson@gmail.com>> wrote:
>=20
>=20
> On Jul 17, 2015 4:35 PM, "I=C3=B1aki Baz Castillo" <ibc@aliax.net =
<mailto:ibc@aliax.net>> wrote:
> > What I understand is that the browser should bind on 0.0.0.0 (or =
::0)
> > and send the STUN/TURN request,
>=20
> That works most of the time, but it is a cheat. What Chrome and =
Firefox do - bind to reach local interface separately - is the only =
reliable way to do this. If you bind to 0.0.0.0, you can't handle =
multiple interfaces correctly, and that reduces the odds of completing =
ICE with the best result.
>=20
>=20

Gulp. Whilst I mostly see the logic - it is wholly unexpected behaviour =
to the average sys admin.=20
Certainly not what I expected.

It strikes me that binding to all interfaces might well give a vector =
for attackers to map out a company=E2=80=99s internal networks.
It also may restrict the user=E2=80=99s ability to manipulate which =
medium is used.=20

E.g. I=E2=80=99m at home and my chromebook pixel (or firefox tablet) is =
on wifi, but I=E2=80=99ve left LTE enabled.
I (or the OS) is configured to prefer wifi wen available - but it =
happens that for a specific peer LTE completes first.
So now my video call goes over LTE without my say-so and with no hint =
this is happening  - costing me real
money. My only option is to completely disable LTE when I get home  (and =
lose SMS too) ?

Perhaps we should default to binding to 0.0.0.0 and allow a user =
config=E2=80=99d preference for more exhaustive searching.

Tim.

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

Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk <http://www.westhawk.co.uk/>




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class=3D""><div class=3D"">On 18 Jul 2015, at 00:41, Martin Thomson =
&lt;<a href=3D"mailto:martin.thomson@gmail.com" =
class=3D"">martin.thomson@gmail.com</a>&gt; wrote:</div><br =
class=3D"Apple-interchange-newline"><div class=3D""><p dir=3D"ltr" =
class=3D""><br class=3D"">
On Jul 17, 2015 4:35 PM, "I=C3=B1aki Baz Castillo" &lt;<a =
href=3D"mailto:ibc@aliax.net" class=3D"">ibc@aliax.net</a>&gt; wrote:<br =
class=3D"">
&gt; What I understand is that the browser should bind on 0.0.0.0 (or =
::0)<br class=3D"">
&gt; and send the STUN/TURN request, </p><p dir=3D"ltr" class=3D"">That =
works most of the time, but it is a cheat. What Chrome and Firefox do - =
bind to reach local interface separately - is the only reliable way to =
do this. If you bind to 0.0.0.0, you can't handle multiple interfaces =
correctly, and that reduces the odds of completing ICE with the best =
result.</p><div class=3D""><br class=3D""></div></div></blockquote><div =
class=3D""><br class=3D""></div>Gulp. Whilst I mostly see the logic - it =
is wholly unexpected behaviour to the average sys admin.&nbsp;</div><div =
class=3D"">Certainly not what I expected.</div><div class=3D""><br =
class=3D""></div><div class=3D"">It strikes me that binding to all =
interfaces might well give a vector for attackers to map out a =
company=E2=80=99s internal networks.</div><div class=3D"">It also may =
restrict the user=E2=80=99s ability to manipulate which medium is =
used.&nbsp;</div><div class=3D""><br class=3D""></div><div class=3D"">E.g.=
 I=E2=80=99m at home and my chromebook pixel (or firefox tablet) is on =
wifi, but I=E2=80=99ve left LTE enabled.</div><div class=3D"">I (or the =
OS) is configured to prefer wifi wen available - but it happens that for =
a specific peer LTE completes first.</div><div class=3D"">So now my =
video call goes over LTE without my say-so and with no hint this is =
happening &nbsp;- costing me real</div><div class=3D"">money. My only =
option is to completely disable LTE when I get home &nbsp;(and lose SMS =
too) ?</div><div class=3D""><br class=3D""></div><div class=3D"">Perhaps =
we should default to binding to 0.0.0.0 and allow a user config=E2=80=99d =
preference for more exhaustive searching.</div><div class=3D""><br =
class=3D""></div><div class=3D"">Tim.<br class=3D""><br =
class=3D""><blockquote type=3D"cite" class=3D""><div class=3D"">
_______________________________________________<br class=3D"">rtcweb =
mailing list<br class=3D""><a href=3D"mailto:rtcweb@ietf.org" =
class=3D"">rtcweb@ietf.org</a><br class=3D""><a =
href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" =
class=3D"">https://www.ietf.org/mailman/listinfo/rtcweb</a><br =
class=3D""></div></blockquote></div><br class=3D""><div =
apple-content-edited=3D"true" class=3D"">
<span class=3D"Apple-style-span" style=3D"border-collapse: separate; =
color: rgb(0, 0, 0); font-family: Helvetica; font-size: 12px; =
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; "><div class=3D"">Tim Panton - =
Web/VoIP consultant and implementor</div><div class=3D""><a =
href=3D"http://www.westhawk.co.uk" =
class=3D"">www.westhawk.co.uk</a></div><div class=3D""><br =
class=3D""></div></span><br class=3D"Apple-interchange-newline">

</div>
<br class=3D""></body></html>=

--Apple-Mail=_E1137B76-A6CB-4289-9321-42DF1A31DD08--


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From: Tim Panton <thp@westhawk.co.uk>
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> On 16 Jul 2015, at 09:42, carlo von lynX =
<lynX@i.know.you.are.psyced.org> wrote:
>=20
> Yes, I dare to speak out of my 25 years of experience on the
> Internet even if I didn't have to use much VPN ever. I can
> imagine plenty of constellations where the VPN is NOT supposed
> to take over all communications on the system but rather offer
> a side route into a company's servers or suchlike. Therefore
> I worry if the software provider can always come up with a
> configuration that still does what the customers expect and
> at the same time deals with the new backdoor introduced by WebRTC.
> Of course anyone with better understanding can help clear up
> these worries.
>=20
>> That isn=E2=80=99t what I meant. I expect secure software providers =
to keep their
>> software abreast of new developments.=20

It turns out that I was completely wrong. The browsers are currently so=20=

thorough in their discovery of available interfaces and routes that no
amount of changes by the sys admin or VPN supplier will fix this =
problem.

Sorry about that.=20

>=20
> Uh, not sure if that is an established pattern in the industry.
> Usually somebody does whatever fits into their business model
> best, then there is a public outcry and possibly some bloodspill,
> then all the players look at how they can make a good impression
> by reducing the collateral damages.. but if that doesn't work out
> cheap and easy.. tant pis. Let's continue with business.
>=20
>> I happen to have the (no doubt under informed) opinion that
>> this isn=E2=80=99t as simple as it has been made out to be.

At least I was right here. - It is even more complex than I believed.

>=20
> Well, in case of doubt it is important to pick the conservative
> option until the issue is properly analyzed rather than hoping
> for the general public to forget about the problem.
>=20
>>> Also the "solution" you suggest AFAIU does not work for people
>>> who use Tor in a non-NATted environment, for example on a
>>> fixed system in a university library with a static IP address.
>>=20
>> That is an interesting case. In that instance hiding the local IP =
would be=20
>> valuable. I thought however those users would normally run a TOR os =
off=20
>> a usb stick in a VM. That would probably mask the local IP.
>=20
> Now that this is more clear, have another look at my original mail
> that describes how such a dissident would use any Windows OS system
> standing around. I doubt anything advanced like TAILS sticks are
> out there as merely possessing one could be life-threatening.


Here too I find that I was wrong and the current situation not what I =
assumed
and is unsatisfactory.

Most SOCKS proxies don't have support for ICE/STUN nor arbitrary UDP.
The current browser implementations (backed by the ICE spec) take the =
view that
if SOCKS doesn=E2=80=99t do UDP, then they should go directly and send =
UDP anyhow.

I=E2=80=99m not clear that this is what a user might expect.=20
An alternative might be to detect a proxy config _only_ offer TURN over =
TLS=20
candidates (and route the traffic over the proxy) - This would result in
poor audio/video but would be more in keeping with the spirit of the =
proxy config.

(a variant might be to inspect the proxy in detail and use the address =
whitelists
to filter allowed traffic somehow?)=20


>>=20
>> I totally agree. My solution is imho a better fix, since it
>> also protects against many other apps that might get started from
>> a browser page (email, dns, ipv6 tunnels, flash, mp3 players etc).


Turns out I was wrong - sigh - as I said the browsers are more =
aggressive
in finding routes than I assumed (or indeed than my ICE implementations =
are).

> Breaking the proxy principle is neither a Tor nor a VPN problem.
> It's just browser vendors breaking a promise given to users that
> configured proxies would be respected.
>=20
>> I=E2=80=99m SOCKs ignorant.
>=20
> So we all have our knowledge deficiencies...
> good to be honest about them.
>=20

And now that I=E2=80=99ve educated myself somewhat, I=E2=80=99m agreeing =
with you.


>>> And if local IP information is so super harmless, then why is it
>>> an obvious reason for TBB devs to disable the entire thing?
>>> I presume for the reasons I stated before.
>>=20
>> I have no idea, but I fear it is because no one has split these =
issues out.
>=20
> So let's dissect some more until we know what we are doing.

Which turns out to be the only useful contribution my <rant/> made.


>=20
>> I=E2=80=99m of the view that there are _useful_ instances of P2P =
crypto
>> in-browser - we will clearly have to disagree about that.
>=20
> I think we could improve the user's ability to ensure end-to-end
> authentication beyond server-based negotiation. Something like
> rendering persistent public keys into QR codes and having people
> exchange those QR codes at cocktail parties and hackathons.
>=20
> Servers would still be able to provide the signaling code, but users
> would be granted a guarantee that the E2E connection is authentic.
>=20

https://yopet.us <https://yopet.us/> :-)=20

Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk




--Apple-Mail=_4FABBCD0-4254-42BB-BB84-DBFEC1E4DAA7
Content-Transfer-Encoding: quoted-printable
Content-Type: text/html;
	charset=utf-8

<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dutf-8"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" =
class=3D""><br class=3D""><div><blockquote type=3D"cite" class=3D""><div =
class=3D"">On 16 Jul 2015, at 09:42, carlo von lynX &lt;<a =
href=3D"mailto:lynX@i.know.you.are.psyced.org" =
class=3D"">lynX@i.know.you.are.psyced.org</a>&gt; wrote:</div><div =
class=3D""><br class=3D"">Yes, I dare to speak out of my 25 years of =
experience on the<br class=3D"">Internet even if I didn't have to use =
much VPN ever. I can<br class=3D"">imagine plenty of constellations =
where the VPN is NOT supposed<br class=3D"">to take over all =
communications on the system but rather offer<br class=3D"">a side route =
into a company's servers or suchlike. Therefore<br class=3D"">I worry if =
the software provider can always come up with a<br =
class=3D"">configuration that still does what the customers expect =
and<br class=3D"">at the same time deals with the new backdoor =
introduced by WebRTC.<br class=3D"">Of course anyone with better =
understanding can help clear up<br class=3D"">these worries.<br =
class=3D""><br class=3D""><blockquote type=3D"cite" class=3D"">That =
isn=E2=80=99t what I meant. I expect secure software providers to keep =
their<br class=3D"">software abreast of new developments. <br =
class=3D""></blockquote></div></blockquote><div><br =
class=3D""></div><div>It turns out that I was completely wrong. The =
browsers are currently so&nbsp;</div><div>thorough in their discovery of =
available interfaces and routes that no</div><div>amount of changes by =
the sys admin or VPN supplier will fix this problem.</div><div><br =
class=3D""></div><div>Sorry about that.&nbsp;</div><br =
class=3D""><blockquote type=3D"cite" class=3D""><div class=3D""><br =
class=3D"">Uh, not sure if that is an established pattern in the =
industry.<br class=3D"">Usually somebody does whatever fits into their =
business model<br class=3D"">best, then there is a public outcry and =
possibly some bloodspill,<br class=3D"">then all the players look at how =
they can make a good impression<br class=3D"">by reducing the collateral =
damages.. but if that doesn't work out<br class=3D"">cheap and easy.. =
tant pis. Let's continue with business.<br class=3D""><br =
class=3D""><blockquote type=3D"cite" class=3D"">I happen to have the (no =
doubt under informed) opinion that<br class=3D"">this isn=E2=80=99t as =
simple as it has been made out to be.<br =
class=3D""></blockquote></div></blockquote><div><br =
class=3D""></div><div>At least I was right here. - It is even more =
complex than I believed.</div><br class=3D""><blockquote type=3D"cite" =
class=3D""><div class=3D""><br class=3D"">Well, in case of doubt it is =
important to pick the conservative<br class=3D"">option until the issue =
is properly analyzed rather than hoping<br class=3D"">for the general =
public to forget about the problem.<br class=3D""><br =
class=3D""><blockquote type=3D"cite" class=3D""><blockquote type=3D"cite" =
class=3D"">Also the "solution" you suggest AFAIU does not work for =
people<br class=3D"">who use Tor in a non-NATted environment, for =
example on a<br class=3D"">fixed system in a university library with a =
static IP address.<br class=3D""></blockquote><br class=3D"">That is an =
interesting case. In that instance hiding the local IP would be <br =
class=3D"">valuable. I thought however those users would normally run a =
TOR os off <br class=3D"">a usb stick in a VM. That would probably mask =
the local IP.<br class=3D""></blockquote><br class=3D"">Now that this is =
more clear, have another look at my original mail<br class=3D"">that =
describes how such a dissident would use any Windows OS system<br =
class=3D"">standing around. I doubt anything advanced like TAILS sticks =
are<br class=3D"">out there as merely possessing one could be =
life-threatening.<br class=3D""></div></blockquote><div><br =
class=3D""></div><div><br class=3D""></div><div>Here too I find that I =
was wrong and the current situation not what I assumed</div><div>and is =
unsatisfactory.</div><div><br class=3D""></div><div>Most SOCKS proxies =
don't have support for ICE/STUN nor arbitrary UDP.</div><div>The current =
browser implementations (backed by the ICE spec) take the view =
that</div><div>if SOCKS doesn=E2=80=99t do UDP, then they should go =
directly and send UDP anyhow.</div><div><br class=3D""></div><div>I=E2=80=99=
m not clear that this is what a user might expect.&nbsp;</div><div>An =
alternative might be to detect a proxy config _only_ offer TURN over =
TLS&nbsp;</div><div>candidates (and route the traffic over the proxy) - =
This would result in</div><div>poor audio/video but would be more in =
keeping with the spirit of the proxy config.</div><div><br =
class=3D""></div><div>(a variant might be to inspect the proxy in detail =
and use the address whitelists</div><div>to filter allowed traffic =
somehow?)&nbsp;</div><div><br class=3D""></div><div><br =
class=3D""></div><blockquote type=3D"cite" class=3D""><div =
class=3D""><blockquote type=3D"cite" class=3D""><br class=3D"">I totally =
agree. My solution is imho a better fix, since it<br class=3D"">also =
protects against many other apps that might get started from<br =
class=3D"">a browser page (email, dns, ipv6 tunnels, flash, mp3 players =
etc).<br class=3D""></blockquote></div></blockquote><div><br =
class=3D""></div><div><br class=3D""></div><div>Turns out I was wrong - =
sigh - as I said the browsers are more aggressive</div><div>in finding =
routes than I assumed (or indeed than my ICE implementations =
are).</div><br class=3D""><blockquote type=3D"cite" class=3D""><div =
class=3D"">Breaking the proxy principle is neither a Tor nor a VPN =
problem.<br class=3D"">It's just browser vendors breaking a promise =
given to users that<br class=3D"">configured proxies would be =
respected.<br class=3D""><br class=3D""><blockquote type=3D"cite" =
class=3D"">I=E2=80=99m SOCKs ignorant.<br class=3D""></blockquote><br =
class=3D"">So we all have our knowledge deficiencies...<br class=3D"">good=
 to be honest about them.<br class=3D""><br =
class=3D""></div></blockquote><div><br class=3D""></div><div>And now =
that I=E2=80=99ve educated myself somewhat, I=E2=80=99m agreeing with =
you.</div><div><br class=3D""></div><br class=3D""><blockquote =
type=3D"cite" class=3D""><div class=3D""><blockquote type=3D"cite" =
class=3D""><blockquote type=3D"cite" class=3D"">And if local IP =
information is so super harmless, then why is it<br class=3D"">an =
obvious reason for TBB devs to disable the entire thing?<br class=3D"">I =
presume for the reasons I stated before.<br class=3D""></blockquote><br =
class=3D"">I have no idea, but I fear it is because no one has split =
these issues out.<br class=3D""></blockquote><br class=3D"">So let's =
dissect some more until we know what we are doing.<br =
class=3D""></div></blockquote><div><br class=3D""></div><div>Which turns =
out to be the only useful contribution my &lt;rant/&gt; =
made.</div><div><br class=3D""></div><br class=3D""><blockquote =
type=3D"cite" class=3D""><div class=3D""><br class=3D""><blockquote =
type=3D"cite" class=3D"">I=E2=80=99m of the view that there are _useful_ =
instances of P2P crypto<br class=3D"">in-browser - we will clearly have =
to disagree about that.<br class=3D""></blockquote><br class=3D"">I =
think we could improve the user's ability to ensure end-to-end<br =
class=3D"">authentication beyond server-based negotiation. Something =
like<br class=3D"">rendering persistent public keys into QR codes and =
having people<br class=3D"">exchange those QR codes at cocktail parties =
and hackathons.<br class=3D""><br class=3D"">Servers would still be able =
to provide the signaling code, but users<br class=3D"">would be granted =
a guarantee that the E2E connection is authentic.<br class=3D""><br =
class=3D""></div></blockquote><br class=3D""></div><div><a =
href=3D"https://yopet.us" =
class=3D"">https://yopet.us</a>&nbsp;:-)&nbsp;</div><br class=3D""><div =
apple-content-edited=3D"true" class=3D"">
<span class=3D"Apple-style-span" style=3D"border-collapse: separate; =
color: rgb(0, 0, 0); font-family: Helvetica; font-size: 12px; =
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; "><div class=3D"">Tim Panton - =
Web/VoIP consultant and implementor</div><div class=3D""><a =
href=3D"http://www.westhawk.co.uk" =
class=3D"">www.westhawk.co.uk</a></div><div class=3D""><br =
class=3D""></div></span><br class=3D"Apple-interchange-newline">

</div>
<br class=3D""></body></html>=

--Apple-Mail=_4FABBCD0-4254-42BB-BB84-DBFEC1E4DAA7--


From nobody Fri Jul 24 03:31:32 2015
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From: Tim Panton <thp@westhawk.co.uk>
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References: <55352067.6070007@nostrum.com> <20150421154657.GA4071@lo.psyced.org> <20150715123803.GA24976@lo.psyced.org> <2BCEF545-D322-4589-AC85-AA80E7F934D9@phonefromhere.com> <20150715164129.GA27105@lo.psyced.org> <AF6485D4-1579-45AF-937C-1682AC1AA645@phonefromhere.com> <20150716084225.GA29652@lo.psyced.org>
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> On 16 Jul 2015, at 09:42, carlo von lynX =
<lynX@i.know.you.are.psyced.org <mailto:lynX@i.know.you.are.psyced.org>> =
wrote:
>=20
> Yes, I dare to speak out of my 25 years of experience on the
> Internet even if I didn't have to use much VPN ever. I can
> imagine plenty of constellations where the VPN is NOT supposed
> to take over all communications on the system but rather offer
> a side route into a company's servers or suchlike. Therefore
> I worry if the software provider can always come up with a
> configuration that still does what the customers expect and
> at the same time deals with the new backdoor introduced by WebRTC.
> Of course anyone with better understanding can help clear up
> these worries.
>=20
>> That isn=E2=80=99t what I meant. I expect secure software providers =
to keep their
>> software abreast of new developments.=20

It turns out that I was completely wrong. The browsers are currently so=20=

thorough in their discovery of available interfaces and routes that no
amount of changes by the sys admin or VPN supplier will fix this =
problem.

Sorry about that.=20

>=20
> Uh, not sure if that is an established pattern in the industry.
> Usually somebody does whatever fits into their business model
> best, then there is a public outcry and possibly some bloodspill,
> then all the players look at how they can make a good impression
> by reducing the collateral damages.. but if that doesn't work out
> cheap and easy.. tant pis. Let's continue with business.
>=20
>> I happen to have the (no doubt under informed) opinion that
>> this isn=E2=80=99t as simple as it has been made out to be.

At least I was right here. - It is even more complex than I believed.

>=20
> Well, in case of doubt it is important to pick the conservative
> option until the issue is properly analyzed rather than hoping
> for the general public to forget about the problem.
>=20
>>> Also the "solution" you suggest AFAIU does not work for people
>>> who use Tor in a non-NATted environment, for example on a
>>> fixed system in a university library with a static IP address.
>>=20
>> That is an interesting case. In that instance hiding the local IP =
would be=20
>> valuable. I thought however those users would normally run a TOR os =
off=20
>> a usb stick in a VM. That would probably mask the local IP.
>=20
> Now that this is more clear, have another look at my original mail
> that describes how such a dissident would use any Windows OS system
> standing around. I doubt anything advanced like TAILS sticks are
> out there as merely possessing one could be life-threatening.


Here too I find that I was wrong and the current situation not what I =
assumed
and is unsatisfactory.

Most SOCKS proxies don't have support for ICE/STUN nor arbitrary UDP.
The current browser implementations (backed by the ICE spec) take the =
view that
if SOCKS doesn=E2=80=99t do UDP, then they should go directly and send =
UDP anyhow.

I=E2=80=99m not clear that this is what a user might expect.=20
An alternative might be to detect a proxy config _only_ offer TURN over =
TLS=20
candidates (and route the traffic over the proxy) - This would result in
poor audio/video but would be more in keeping with the spirit of the =
proxy config.

(a variant might be to inspect the proxy in detail and use the address =
whitelists
to filter allowed traffic somehow?)=20


>>=20
>> I totally agree. My solution is imho a better fix, since it
>> also protects against many other apps that might get started from
>> a browser page (email, dns, ipv6 tunnels, flash, mp3 players etc).


Turns out I was wrong - sigh - as I said the browsers are more =
aggressive
in finding routes than I assumed (or indeed than my ICE implementations =
are).

> Breaking the proxy principle is neither a Tor nor a VPN problem.
> It's just browser vendors breaking a promise given to users that
> configured proxies would be respected.
>=20
>> I=E2=80=99m SOCKs ignorant.
>=20
> So we all have our knowledge deficiencies...
> good to be honest about them.
>=20

And now that I=E2=80=99ve educated myself somewhat, I=E2=80=99m agreeing =
with you.


>>> And if local IP information is so super harmless, then why is it
>>> an obvious reason for TBB devs to disable the entire thing?
>>> I presume for the reasons I stated before.
>>=20
>> I have no idea, but I fear it is because no one has split these =
issues out.
>=20
> So let's dissect some more until we know what we are doing.

Which turns out to be the only useful contribution my <rant/> made.


>=20
>> I=E2=80=99m of the view that there are _useful_ instances of P2P =
crypto
>> in-browser - we will clearly have to disagree about that.
>=20
> I think we could improve the user's ability to ensure end-to-end
> authentication beyond server-based negotiation. Something like
> rendering persistent public keys into QR codes and having people
> exchange those QR codes at cocktail parties and hackathons.
>=20
> Servers would still be able to provide the signaling code, but users
> would be granted a guarantee that the E2E connection is authentic.
>=20

https://yopet.us <https://yopet.us/> :-)=20

Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk <http://www.westhawk.co.uk/>




--Apple-Mail=_6282B1E8-A6F9-4743-9129-C807A72FF543
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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dutf-8"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" =
class=3D""><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dutf-8" class=3D""><div style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" =
class=3D""><br class=3D""><div class=3D""><blockquote type=3D"cite" =
class=3D""><div class=3D"">On 16 Jul 2015, at 09:42, carlo von lynX =
&lt;<a href=3D"mailto:lynX@i.know.you.are.psyced.org" =
class=3D"">lynX@i.know.you.are.psyced.org</a>&gt; wrote:</div><div =
class=3D""><br class=3D"">Yes, I dare to speak out of my 25 years of =
experience on the<br class=3D"">Internet even if I didn't have to use =
much VPN ever. I can<br class=3D"">imagine plenty of constellations =
where the VPN is NOT supposed<br class=3D"">to take over all =
communications on the system but rather offer<br class=3D"">a side route =
into a company's servers or suchlike. Therefore<br class=3D"">I worry if =
the software provider can always come up with a<br =
class=3D"">configuration that still does what the customers expect =
and<br class=3D"">at the same time deals with the new backdoor =
introduced by WebRTC.<br class=3D"">Of course anyone with better =
understanding can help clear up<br class=3D"">these worries.<br =
class=3D""><br class=3D""><blockquote type=3D"cite" class=3D"">That =
isn=E2=80=99t what I meant. I expect secure software providers to keep =
their<br class=3D"">software abreast of new developments. <br =
class=3D""></blockquote></div></blockquote><div class=3D""><br =
class=3D""></div><div class=3D"">It turns out that I was completely =
wrong. The browsers are currently so&nbsp;</div><div class=3D"">thorough =
in their discovery of available interfaces and routes that no</div><div =
class=3D"">amount of changes by the sys admin or VPN supplier will fix =
this problem.</div><div class=3D""><br class=3D""></div><div =
class=3D"">Sorry about that.&nbsp;</div><br class=3D""><blockquote =
type=3D"cite" class=3D""><div class=3D""><br class=3D"">Uh, not sure if =
that is an established pattern in the industry.<br class=3D"">Usually =
somebody does whatever fits into their business model<br class=3D"">best, =
then there is a public outcry and possibly some bloodspill,<br =
class=3D"">then all the players look at how they can make a good =
impression<br class=3D"">by reducing the collateral damages.. but if =
that doesn't work out<br class=3D"">cheap and easy.. tant pis. Let's =
continue with business.<br class=3D""><br class=3D""><blockquote =
type=3D"cite" class=3D"">I happen to have the (no doubt under informed) =
opinion that<br class=3D"">this isn=E2=80=99t as simple as it has been =
made out to be.<br class=3D""></blockquote></div></blockquote><div =
class=3D""><br class=3D""></div><div class=3D"">At least I was right =
here. - It is even more complex than I believed.</div><br =
class=3D""><blockquote type=3D"cite" class=3D""><div class=3D""><br =
class=3D"">Well, in case of doubt it is important to pick the =
conservative<br class=3D"">option until the issue is properly analyzed =
rather than hoping<br class=3D"">for the general public to forget about =
the problem.<br class=3D""><br class=3D""><blockquote type=3D"cite" =
class=3D""><blockquote type=3D"cite" class=3D"">Also the "solution" you =
suggest AFAIU does not work for people<br class=3D"">who use Tor in a =
non-NATted environment, for example on a<br class=3D"">fixed system in a =
university library with a static IP address.<br =
class=3D""></blockquote><br class=3D"">That is an interesting case. In =
that instance hiding the local IP would be <br class=3D"">valuable. I =
thought however those users would normally run a TOR os off <br =
class=3D"">a usb stick in a VM. That would probably mask the local =
IP.<br class=3D""></blockquote><br class=3D"">Now that this is more =
clear, have another look at my original mail<br class=3D"">that =
describes how such a dissident would use any Windows OS system<br =
class=3D"">standing around. I doubt anything advanced like TAILS sticks =
are<br class=3D"">out there as merely possessing one could be =
life-threatening.<br class=3D""></div></blockquote><div class=3D""><br =
class=3D""></div><div class=3D""><br class=3D""></div><div class=3D"">Here=
 too I find that I was wrong and the current situation not what I =
assumed</div><div class=3D"">and is unsatisfactory.</div><div =
class=3D""><br class=3D""></div><div class=3D"">Most SOCKS proxies don't =
have support for ICE/STUN nor arbitrary UDP.</div><div class=3D"">The =
current browser implementations (backed by the ICE spec) take the view =
that</div><div class=3D"">if SOCKS doesn=E2=80=99t do UDP, then they =
should go directly and send UDP anyhow.</div><div class=3D""><br =
class=3D""></div><div class=3D"">I=E2=80=99m not clear that this is what =
a user might expect.&nbsp;</div><div class=3D"">An alternative might be =
to detect a proxy config _only_ offer TURN over TLS&nbsp;</div><div =
class=3D"">candidates (and route the traffic over the proxy) - This =
would result in</div><div class=3D"">poor audio/video but would be more =
in keeping with the spirit of the proxy config.</div><div class=3D""><br =
class=3D""></div><div class=3D"">(a variant might be to inspect the =
proxy in detail and use the address whitelists</div><div class=3D"">to =
filter allowed traffic somehow?)&nbsp;</div><div class=3D""><br =
class=3D""></div><div class=3D""><br class=3D""></div><blockquote =
type=3D"cite" class=3D""><div class=3D""><blockquote type=3D"cite" =
class=3D""><br class=3D"">I totally agree. My solution is imho a better =
fix, since it<br class=3D"">also protects against many other apps that =
might get started from<br class=3D"">a browser page (email, dns, ipv6 =
tunnels, flash, mp3 players etc).<br =
class=3D""></blockquote></div></blockquote><div class=3D""><br =
class=3D""></div><div class=3D""><br class=3D""></div><div =
class=3D"">Turns out I was wrong - sigh - as I said the browsers are =
more aggressive</div><div class=3D"">in finding routes than I assumed =
(or indeed than my ICE implementations are).</div><br =
class=3D""><blockquote type=3D"cite" class=3D""><div class=3D"">Breaking =
the proxy principle is neither a Tor nor a VPN problem.<br class=3D"">It's=
 just browser vendors breaking a promise given to users that<br =
class=3D"">configured proxies would be respected.<br class=3D""><br =
class=3D""><blockquote type=3D"cite" class=3D"">I=E2=80=99m SOCKs =
ignorant.<br class=3D""></blockquote><br class=3D"">So we all have our =
knowledge deficiencies...<br class=3D"">good to be honest about them.<br =
class=3D""><br class=3D""></div></blockquote><div class=3D""><br =
class=3D""></div><div class=3D"">And now that I=E2=80=99ve educated =
myself somewhat, I=E2=80=99m agreeing with you.</div><div class=3D""><br =
class=3D""></div><br class=3D""><blockquote type=3D"cite" class=3D""><div =
class=3D""><blockquote type=3D"cite" class=3D""><blockquote type=3D"cite" =
class=3D"">And if local IP information is so super harmless, then why is =
it<br class=3D"">an obvious reason for TBB devs to disable the entire =
thing?<br class=3D"">I presume for the reasons I stated before.<br =
class=3D""></blockquote><br class=3D"">I have no idea, but I fear it is =
because no one has split these issues out.<br class=3D""></blockquote><br =
class=3D"">So let's dissect some more until we know what we are =
doing.<br class=3D""></div></blockquote><div class=3D""><br =
class=3D""></div><div class=3D"">Which turns out to be the only useful =
contribution my &lt;rant/&gt; made.</div><div class=3D""><br =
class=3D""></div><br class=3D""><blockquote type=3D"cite" class=3D""><div =
class=3D""><br class=3D""><blockquote type=3D"cite" class=3D"">I=E2=80=99m=
 of the view that there are _useful_ instances of P2P crypto<br =
class=3D"">in-browser - we will clearly have to disagree about that.<br =
class=3D""></blockquote><br class=3D"">I think we could improve the =
user's ability to ensure end-to-end<br class=3D"">authentication beyond =
server-based negotiation. Something like<br class=3D"">rendering =
persistent public keys into QR codes and having people<br =
class=3D"">exchange those QR codes at cocktail parties and =
hackathons.<br class=3D""><br class=3D"">Servers would still be able to =
provide the signaling code, but users<br class=3D"">would be granted a =
guarantee that the E2E connection is authentic.<br class=3D""><br =
class=3D""></div></blockquote><br class=3D""></div><div class=3D""><a =
href=3D"https://yopet.us/" =
class=3D"">https://yopet.us</a>&nbsp;:-)&nbsp;</div><br class=3D""><div =
apple-content-edited=3D"true" class=3D"">
<span class=3D"Apple-style-span" style=3D"border-collapse: separate; =
font-family: Helvetica; font-size: 12px; 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; border-spacing: 0px; =
-webkit-text-decorations-in-effect: none; -webkit-text-stroke-width: =
0px;"><div class=3D"">Tim Panton - Web/VoIP consultant and =
implementor</div><div class=3D""><a href=3D"http://www.westhawk.co.uk/" =
class=3D"">www.westhawk.co.uk</a></div><div class=3D""><br =
class=3D""></div></span><br class=3D"Apple-interchange-newline">

</div>
<br class=3D""></div></body></html>=

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Date: Mon, 20 Jul 2015 12:53:59 +0100
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References: <55352067.6070007@nostrum.com> <20150421154657.GA4071@lo.psyced.org> <20150715123803.GA24976@lo.psyced.org> <2BCEF545-D322-4589-AC85-AA80E7F934D9@phonefromhere.com> <20150715164129.GA27105@lo.psyced.org> <AF6485D4-1579-45AF-937C-1682AC1AA645@phonefromhere.com> <20150716084225.GA29652@lo.psyced.org>
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> On 16 Jul 2015, at 09:42, carlo von lynX =
<lynX@i.know.you.are.psyced.org <mailto:lynX@i.know.you.are.psyced.org>> =
wrote:
>=20
> Yes, I dare to speak out of my 25 years of experience on the
> Internet even if I didn't have to use much VPN ever. I can
> imagine plenty of constellations where the VPN is NOT supposed
> to take over all communications on the system but rather offer
> a side route into a company's servers or suchlike. Therefore
> I worry if the software provider can always come up with a
> configuration that still does what the customers expect and
> at the same time deals with the new backdoor introduced by WebRTC.
> Of course anyone with better understanding can help clear up
> these worries.
>=20
>> That isn=E2=80=99t what I meant. I expect secure software providers =
to keep their
>> software abreast of new developments.=20

It turns out that I was completely wrong. The browsers are currently so=20=

thorough in their discovery of available interfaces and routes that no
amount of changes by the sys admin or VPN supplier will fix this =
problem.

Sorry about that.=20

>=20
> Uh, not sure if that is an established pattern in the industry.
> Usually somebody does whatever fits into their business model
> best, then there is a public outcry and possibly some bloodspill,
> then all the players look at how they can make a good impression
> by reducing the collateral damages.. but if that doesn't work out
> cheap and easy.. tant pis. Let's continue with business.
>=20
>> I happen to have the (no doubt under informed) opinion that
>> this isn=E2=80=99t as simple as it has been made out to be.

At least I was right here. - It is even more complex than I believed.

>=20
> Well, in case of doubt it is important to pick the conservative
> option until the issue is properly analyzed rather than hoping
> for the general public to forget about the problem.
>=20
>>> Also the "solution" you suggest AFAIU does not work for people
>>> who use Tor in a non-NATted environment, for example on a
>>> fixed system in a university library with a static IP address.
>>=20
>> That is an interesting case. In that instance hiding the local IP =
would be=20
>> valuable. I thought however those users would normally run a TOR os =
off=20
>> a usb stick in a VM. That would probably mask the local IP.
>=20
> Now that this is more clear, have another look at my original mail
> that describes how such a dissident would use any Windows OS system
> standing around. I doubt anything advanced like TAILS sticks are
> out there as merely possessing one could be life-threatening.


Here too I find that I was wrong and the current situation not what I =
assumed
and is unsatisfactory.

Most SOCKS proxies don't have support for ICE/STUN nor arbitrary UDP.
The current browser implementations (backed by the ICE spec) take the =
view that
if SOCKS doesn=E2=80=99t do UDP, then they should go directly and send =
UDP anyhow.

I=E2=80=99m not clear that this is what a user might expect.=20
An alternative might be to detect a proxy config _only_ offer TURN over =
TLS=20
candidates (and route the traffic over the proxy) - This would result in
poor audio/video but would be more in keeping with the spirit of the =
proxy config.

(a variant might be to inspect the proxy in detail and use the address =
whitelists
to filter allowed traffic somehow?)=20


>>=20
>> I totally agree. My solution is imho a better fix, since it
>> also protects against many other apps that might get started from
>> a browser page (email, dns, ipv6 tunnels, flash, mp3 players etc).


Turns out I was wrong - sigh - as I said the browsers are more =
aggressive
in finding routes than I assumed (or indeed than my ICE implementations =
are).

> Breaking the proxy principle is neither a Tor nor a VPN problem.
> It's just browser vendors breaking a promise given to users that
> configured proxies would be respected.
>=20
>> I=E2=80=99m SOCKs ignorant.
>=20
> So we all have our knowledge deficiencies...
> good to be honest about them.
>=20

And now that I=E2=80=99ve educated myself somewhat, I=E2=80=99m agreeing =
with you.


>>> And if local IP information is so super harmless, then why is it
>>> an obvious reason for TBB devs to disable the entire thing?
>>> I presume for the reasons I stated before.
>>=20
>> I have no idea, but I fear it is because no one has split these =
issues out.
>=20
> So let's dissect some more until we know what we are doing.

Which turns out to be the only useful contribution my <rant/> made.


>=20
>> I=E2=80=99m of the view that there are _useful_ instances of P2P =
crypto
>> in-browser - we will clearly have to disagree about that.
>=20
> I think we could improve the user's ability to ensure end-to-end
> authentication beyond server-based negotiation. Something like
> rendering persistent public keys into QR codes and having people
> exchange those QR codes at cocktail parties and hackathons.
>=20
> Servers would still be able to provide the signaling code, but users
> would be granted a guarantee that the E2E connection is authentic.
>=20

https://yopet.us <https://yopet.us/> :-)=20

Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk <http://www.westhawk.co.uk/>




--Apple-Mail=_2788FB8A-88E5-4452-A3BA-9B612E2E9AFD
Content-Transfer-Encoding: quoted-printable
Content-Type: text/html;
	charset=utf-8

<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dutf-8"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" =
class=3D""><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dutf-8" class=3D""><div style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" =
class=3D""><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dutf-8" class=3D""><div style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" =
class=3D""><br class=3D""><div class=3D""><blockquote type=3D"cite" =
class=3D""><div class=3D"">On 16 Jul 2015, at 09:42, carlo von lynX =
&lt;<a href=3D"mailto:lynX@i.know.you.are.psyced.org" =
class=3D"">lynX@i.know.you.are.psyced.org</a>&gt; wrote:</div><div =
class=3D""><br class=3D"">Yes, I dare to speak out of my 25 years of =
experience on the<br class=3D"">Internet even if I didn't have to use =
much VPN ever. I can<br class=3D"">imagine plenty of constellations =
where the VPN is NOT supposed<br class=3D"">to take over all =
communications on the system but rather offer<br class=3D"">a side route =
into a company's servers or suchlike. Therefore<br class=3D"">I worry if =
the software provider can always come up with a<br =
class=3D"">configuration that still does what the customers expect =
and<br class=3D"">at the same time deals with the new backdoor =
introduced by WebRTC.<br class=3D"">Of course anyone with better =
understanding can help clear up<br class=3D"">these worries.<br =
class=3D""><br class=3D""><blockquote type=3D"cite" class=3D"">That =
isn=E2=80=99t what I meant. I expect secure software providers to keep =
their<br class=3D"">software abreast of new developments. <br =
class=3D""></blockquote></div></blockquote><div class=3D""><br =
class=3D""></div><div class=3D"">It turns out that I was completely =
wrong. The browsers are currently so&nbsp;</div><div class=3D"">thorough =
in their discovery of available interfaces and routes that no</div><div =
class=3D"">amount of changes by the sys admin or VPN supplier will fix =
this problem.</div><div class=3D""><br class=3D""></div><div =
class=3D"">Sorry about that.&nbsp;</div><br class=3D""><blockquote =
type=3D"cite" class=3D""><div class=3D""><br class=3D"">Uh, not sure if =
that is an established pattern in the industry.<br class=3D"">Usually =
somebody does whatever fits into their business model<br class=3D"">best, =
then there is a public outcry and possibly some bloodspill,<br =
class=3D"">then all the players look at how they can make a good =
impression<br class=3D"">by reducing the collateral damages.. but if =
that doesn't work out<br class=3D"">cheap and easy.. tant pis. Let's =
continue with business.<br class=3D""><br class=3D""><blockquote =
type=3D"cite" class=3D"">I happen to have the (no doubt under informed) =
opinion that<br class=3D"">this isn=E2=80=99t as simple as it has been =
made out to be.<br class=3D""></blockquote></div></blockquote><div =
class=3D""><br class=3D""></div><div class=3D"">At least I was right =
here. - It is even more complex than I believed.</div><br =
class=3D""><blockquote type=3D"cite" class=3D""><div class=3D""><br =
class=3D"">Well, in case of doubt it is important to pick the =
conservative<br class=3D"">option until the issue is properly analyzed =
rather than hoping<br class=3D"">for the general public to forget about =
the problem.<br class=3D""><br class=3D""><blockquote type=3D"cite" =
class=3D""><blockquote type=3D"cite" class=3D"">Also the "solution" you =
suggest AFAIU does not work for people<br class=3D"">who use Tor in a =
non-NATted environment, for example on a<br class=3D"">fixed system in a =
university library with a static IP address.<br =
class=3D""></blockquote><br class=3D"">That is an interesting case. In =
that instance hiding the local IP would be <br class=3D"">valuable. I =
thought however those users would normally run a TOR os off <br =
class=3D"">a usb stick in a VM. That would probably mask the local =
IP.<br class=3D""></blockquote><br class=3D"">Now that this is more =
clear, have another look at my original mail<br class=3D"">that =
describes how such a dissident would use any Windows OS system<br =
class=3D"">standing around. I doubt anything advanced like TAILS sticks =
are<br class=3D"">out there as merely possessing one could be =
life-threatening.<br class=3D""></div></blockquote><div class=3D""><br =
class=3D""></div><div class=3D""><br class=3D""></div><div class=3D"">Here=
 too I find that I was wrong and the current situation not what I =
assumed</div><div class=3D"">and is unsatisfactory.</div><div =
class=3D""><br class=3D""></div><div class=3D"">Most SOCKS proxies don't =
have support for ICE/STUN nor arbitrary UDP.</div><div class=3D"">The =
current browser implementations (backed by the ICE spec) take the view =
that</div><div class=3D"">if SOCKS doesn=E2=80=99t do UDP, then they =
should go directly and send UDP anyhow.</div><div class=3D""><br =
class=3D""></div><div class=3D"">I=E2=80=99m not clear that this is what =
a user might expect.&nbsp;</div><div class=3D"">An alternative might be =
to detect a proxy config _only_ offer TURN over TLS&nbsp;</div><div =
class=3D"">candidates (and route the traffic over the proxy) - This =
would result in</div><div class=3D"">poor audio/video but would be more =
in keeping with the spirit of the proxy config.</div><div class=3D""><br =
class=3D""></div><div class=3D"">(a variant might be to inspect the =
proxy in detail and use the address whitelists</div><div class=3D"">to =
filter allowed traffic somehow?)&nbsp;</div><div class=3D""><br =
class=3D""></div><div class=3D""><br class=3D""></div><blockquote =
type=3D"cite" class=3D""><div class=3D""><blockquote type=3D"cite" =
class=3D""><br class=3D"">I totally agree. My solution is imho a better =
fix, since it<br class=3D"">also protects against many other apps that =
might get started from<br class=3D"">a browser page (email, dns, ipv6 =
tunnels, flash, mp3 players etc).<br =
class=3D""></blockquote></div></blockquote><div class=3D""><br =
class=3D""></div><div class=3D""><br class=3D""></div><div =
class=3D"">Turns out I was wrong - sigh - as I said the browsers are =
more aggressive</div><div class=3D"">in finding routes than I assumed =
(or indeed than my ICE implementations are).</div><br =
class=3D""><blockquote type=3D"cite" class=3D""><div class=3D"">Breaking =
the proxy principle is neither a Tor nor a VPN problem.<br class=3D"">It's=
 just browser vendors breaking a promise given to users that<br =
class=3D"">configured proxies would be respected.<br class=3D""><br =
class=3D""><blockquote type=3D"cite" class=3D"">I=E2=80=99m SOCKs =
ignorant.<br class=3D""></blockquote><br class=3D"">So we all have our =
knowledge deficiencies...<br class=3D"">good to be honest about them.<br =
class=3D""><br class=3D""></div></blockquote><div class=3D""><br =
class=3D""></div><div class=3D"">And now that I=E2=80=99ve educated =
myself somewhat, I=E2=80=99m agreeing with you.</div><div class=3D""><br =
class=3D""></div><br class=3D""><blockquote type=3D"cite" class=3D""><div =
class=3D""><blockquote type=3D"cite" class=3D""><blockquote type=3D"cite" =
class=3D"">And if local IP information is so super harmless, then why is =
it<br class=3D"">an obvious reason for TBB devs to disable the entire =
thing?<br class=3D"">I presume for the reasons I stated before.<br =
class=3D""></blockquote><br class=3D"">I have no idea, but I fear it is =
because no one has split these issues out.<br class=3D""></blockquote><br =
class=3D"">So let's dissect some more until we know what we are =
doing.<br class=3D""></div></blockquote><div class=3D""><br =
class=3D""></div><div class=3D"">Which turns out to be the only useful =
contribution my &lt;rant/&gt; made.</div><div class=3D""><br =
class=3D""></div><br class=3D""><blockquote type=3D"cite" class=3D""><div =
class=3D""><br class=3D""><blockquote type=3D"cite" class=3D"">I=E2=80=99m=
 of the view that there are _useful_ instances of P2P crypto<br =
class=3D"">in-browser - we will clearly have to disagree about that.<br =
class=3D""></blockquote><br class=3D"">I think we could improve the =
user's ability to ensure end-to-end<br class=3D"">authentication beyond =
server-based negotiation. Something like<br class=3D"">rendering =
persistent public keys into QR codes and having people<br =
class=3D"">exchange those QR codes at cocktail parties and =
hackathons.<br class=3D""><br class=3D"">Servers would still be able to =
provide the signaling code, but users<br class=3D"">would be granted a =
guarantee that the E2E connection is authentic.<br class=3D""><br =
class=3D""></div></blockquote><br class=3D""></div><div class=3D""><a =
href=3D"https://yopet.us/" =
class=3D"">https://yopet.us</a>&nbsp;:-)&nbsp;</div><br class=3D""><div =
apple-content-edited=3D"true" class=3D"">
<span class=3D"Apple-style-span" style=3D"border-collapse: separate; =
font-family: Helvetica; font-size: 12px; 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; border-spacing: 0px; =
-webkit-text-decorations-in-effect: none; -webkit-text-stroke-width: =
0px;"><div class=3D"">Tim Panton - Web/VoIP consultant and =
implementor</div><div class=3D""><a href=3D"http://www.westhawk.co.uk/" =
class=3D"">www.westhawk.co.uk</a></div><div class=3D""><br =
class=3D""></div></span><br class=3D"Apple-interchange-newline">

</div>
<br class=3D""></div></div></body></html>=

--Apple-Mail=_2788FB8A-88E5-4452-A3BA-9B612E2E9AFD--


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--Apple-Mail=_22967C1B-2449-4FEB-AF05-79B520E714E4
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> On 16 Jul 2015, at 09:42, carlo von lynX =
<lynX@i.know.you.are.psyced.org <mailto:lynX@i.know.you.are.psyced.org>> =
wrote:
>=20
> Yes, I dare to speak out of my 25 years of experience on the
> Internet even if I didn't have to use much VPN ever. I can
> imagine plenty of constellations where the VPN is NOT supposed
> to take over all communications on the system but rather offer
> a side route into a company's servers or suchlike. Therefore
> I worry if the software provider can always come up with a
> configuration that still does what the customers expect and
> at the same time deals with the new backdoor introduced by WebRTC.
> Of course anyone with better understanding can help clear up
> these worries.
>=20
>> That isn=E2=80=99t what I meant. I expect secure software providers =
to keep their
>> software abreast of new developments.=20

It turns out that I was completely wrong. The browsers are currently so=20=

thorough in their discovery of available interfaces and routes that no
amount of changes by the sys admin or VPN supplier will fix this =
problem.

Sorry about that.=20

>=20
> Uh, not sure if that is an established pattern in the industry.
> Usually somebody does whatever fits into their business model
> best, then there is a public outcry and possibly some bloodspill,
> then all the players look at how they can make a good impression
> by reducing the collateral damages.. but if that doesn't work out
> cheap and easy.. tant pis. Let's continue with business.
>=20
>> I happen to have the (no doubt under informed) opinion that
>> this isn=E2=80=99t as simple as it has been made out to be.

At least I was right here. - It is even more complex than I believed.

>=20
> Well, in case of doubt it is important to pick the conservative
> option until the issue is properly analyzed rather than hoping
> for the general public to forget about the problem.
>=20
>>> Also the "solution" you suggest AFAIU does not work for people
>>> who use Tor in a non-NATted environment, for example on a
>>> fixed system in a university library with a static IP address.
>>=20
>> That is an interesting case. In that instance hiding the local IP =
would be=20
>> valuable. I thought however those users would normally run a TOR os =
off=20
>> a usb stick in a VM. That would probably mask the local IP.
>=20
> Now that this is more clear, have another look at my original mail
> that describes how such a dissident would use any Windows OS system
> standing around. I doubt anything advanced like TAILS sticks are
> out there as merely possessing one could be life-threatening.


Here too I find that I was wrong and the current situation not what I =
assumed
and is unsatisfactory.

Most SOCKS proxies don't have support for ICE/STUN nor arbitrary UDP.
The current browser implementations (backed by the ICE spec) take the =
view that
if SOCKS doesn=E2=80=99t do UDP, then they should go directly and send =
UDP anyhow.

I=E2=80=99m not clear that this is what a user might expect.=20
An alternative might be to detect a proxy config _only_ offer TURN over =
TLS=20
candidates (and route the traffic over the proxy) - This would result in
poor audio/video but would be more in keeping with the spirit of the =
proxy config.

(a variant might be to inspect the proxy in detail and use the address =
whitelists
to filter allowed traffic somehow?)=20


>>=20
>> I totally agree. My solution is imho a better fix, since it
>> also protects against many other apps that might get started from
>> a browser page (email, dns, ipv6 tunnels, flash, mp3 players etc).


Turns out I was wrong - sigh - as I said the browsers are more =
aggressive
in finding routes than I assumed (or indeed than my ICE implementations =
are).

> Breaking the proxy principle is neither a Tor nor a VPN problem.
> It's just browser vendors breaking a promise given to users that
> configured proxies would be respected.
>=20
>> I=E2=80=99m SOCKs ignorant.
>=20
> So we all have our knowledge deficiencies...
> good to be honest about them.
>=20

And now that I=E2=80=99ve educated myself somewhat, I=E2=80=99m agreeing =
with you.


>>> And if local IP information is so super harmless, then why is it
>>> an obvious reason for TBB devs to disable the entire thing?
>>> I presume for the reasons I stated before.
>>=20
>> I have no idea, but I fear it is because no one has split these =
issues out.
>=20
> So let's dissect some more until we know what we are doing.

Which turns out to be the only useful contribution my <rant/> made.


>=20
>> I=E2=80=99m of the view that there are _useful_ instances of P2P =
crypto
>> in-browser - we will clearly have to disagree about that.
>=20
> I think we could improve the user's ability to ensure end-to-end
> authentication beyond server-based negotiation. Something like
> rendering persistent public keys into QR codes and having people
> exchange those QR codes at cocktail parties and hackathons.
>=20
> Servers would still be able to provide the signaling code, but users
> would be granted a guarantee that the E2E connection is authentic.
>=20

https://yopet.us <https://yopet.us/> :-)=20


--Apple-Mail=_22967C1B-2449-4FEB-AF05-79B520E714E4
Content-Transfer-Encoding: quoted-printable
Content-Type: text/html;
	charset=utf-8

<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dutf-8"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" =
class=3D""><br class=3D""><div class=3D""><div class=3D"" =
style=3D"font-family: LucidaSansUnicode;"><blockquote type=3D"cite" =
class=3D""><div class=3D"">On 16 Jul 2015, at 09:42, carlo von lynX =
&lt;<a href=3D"mailto:lynX@i.know.you.are.psyced.org" =
class=3D"">lynX@i.know.you.are.psyced.org</a>&gt; wrote:</div><div =
class=3D""><br class=3D"">Yes, I dare to speak out of my 25 years of =
experience on the<br class=3D"">Internet even if I didn't have to use =
much VPN ever. I can<br class=3D"">imagine plenty of constellations =
where the VPN is NOT supposed<br class=3D"">to take over all =
communications on the system but rather offer<br class=3D"">a side route =
into a company's servers or suchlike. Therefore<br class=3D"">I worry if =
the software provider can always come up with a<br =
class=3D"">configuration that still does what the customers expect =
and<br class=3D"">at the same time deals with the new backdoor =
introduced by WebRTC.<br class=3D"">Of course anyone with better =
understanding can help clear up<br class=3D"">these worries.<br =
class=3D""><br class=3D""><blockquote type=3D"cite" class=3D"">That =
isn=E2=80=99t what I meant. I expect secure software providers to keep =
their<br class=3D"">software abreast of new developments.&nbsp;<br =
class=3D""></blockquote></div></blockquote><div class=3D""><br =
class=3D""></div><div class=3D"">It turns out that I was completely =
wrong. The browsers are currently so&nbsp;</div><div class=3D"">thorough =
in their discovery of available interfaces and routes that no</div><div =
class=3D"">amount of changes by the sys admin or VPN supplier will fix =
this problem.</div><div class=3D""><br class=3D""></div><div =
class=3D"">Sorry about that.&nbsp;</div><br class=3D""><blockquote =
type=3D"cite" class=3D""><div class=3D""><br class=3D"">Uh, not sure if =
that is an established pattern in the industry.<br class=3D"">Usually =
somebody does whatever fits into their business model<br class=3D"">best, =
then there is a public outcry and possibly some bloodspill,<br =
class=3D"">then all the players look at how they can make a good =
impression<br class=3D"">by reducing the collateral damages.. but if =
that doesn't work out<br class=3D"">cheap and easy.. tant pis. Let's =
continue with business.<br class=3D""><br class=3D""><blockquote =
type=3D"cite" class=3D"">I happen to have the (no doubt under informed) =
opinion that<br class=3D"">this isn=E2=80=99t as simple as it has been =
made out to be.<br class=3D""></blockquote></div></blockquote><div =
class=3D""><br class=3D""></div><div class=3D"">At least I was right =
here. - It is even more complex than I believed.</div><br =
class=3D""><blockquote type=3D"cite" class=3D""><div class=3D""><br =
class=3D"">Well, in case of doubt it is important to pick the =
conservative<br class=3D"">option until the issue is properly analyzed =
rather than hoping<br class=3D"">for the general public to forget about =
the problem.<br class=3D""><br class=3D""><blockquote type=3D"cite" =
class=3D""><blockquote type=3D"cite" class=3D"">Also the "solution" you =
suggest AFAIU does not work for people<br class=3D"">who use Tor in a =
non-NATted environment, for example on a<br class=3D"">fixed system in a =
university library with a static IP address.<br =
class=3D""></blockquote><br class=3D"">That is an interesting case. In =
that instance hiding the local IP would be&nbsp;<br class=3D"">valuable. =
I thought however those users would normally run a TOR os off&nbsp;<br =
class=3D"">a usb stick in a VM. That would probably mask the local =
IP.<br class=3D""></blockquote><br class=3D"">Now that this is more =
clear, have another look at my original mail<br class=3D"">that =
describes how such a dissident would use any Windows OS system<br =
class=3D"">standing around. I doubt anything advanced like TAILS sticks =
are<br class=3D"">out there as merely possessing one could be =
life-threatening.<br class=3D""></div></blockquote><div class=3D""><br =
class=3D""></div><div class=3D""><br class=3D""></div><div class=3D"">Here=
 too I find that I was wrong and the current situation not what I =
assumed</div><div class=3D"">and is unsatisfactory.</div><div =
class=3D""><br class=3D""></div><div class=3D"">Most SOCKS proxies don't =
have support for ICE/STUN nor arbitrary UDP.</div><div class=3D"">The =
current browser implementations (backed by the ICE spec) take the view =
that</div><div class=3D"">if SOCKS doesn=E2=80=99t do UDP, then they =
should go directly and send UDP anyhow.</div><div class=3D""><br =
class=3D""></div><div class=3D"">I=E2=80=99m not clear that this is what =
a user might expect.&nbsp;</div><div class=3D"">An alternative might be =
to detect a proxy config _only_ offer TURN over TLS&nbsp;</div><div =
class=3D"">candidates (and route the traffic over the proxy) - This =
would result in</div><div class=3D"">poor audio/video but would be more =
in keeping with the spirit of the proxy config.</div><div class=3D""><br =
class=3D""></div><div class=3D"">(a variant might be to inspect the =
proxy in detail and use the address whitelists</div><div class=3D"">to =
filter allowed traffic somehow?)&nbsp;</div><div class=3D""><br =
class=3D""></div><div class=3D""><br class=3D""></div><blockquote =
type=3D"cite" class=3D""><div class=3D""><blockquote type=3D"cite" =
class=3D""><br class=3D"">I totally agree. My solution is imho a better =
fix, since it<br class=3D"">also protects against many other apps that =
might get started from<br class=3D"">a browser page (email, dns, ipv6 =
tunnels, flash, mp3 players etc).<br =
class=3D""></blockquote></div></blockquote><div class=3D""><br =
class=3D""></div><div class=3D""><br class=3D""></div><div =
class=3D"">Turns out I was wrong - sigh - as I said the browsers are =
more aggressive</div><div class=3D"">in finding routes than I assumed =
(or indeed than my ICE implementations are).</div><br =
class=3D""><blockquote type=3D"cite" class=3D""><div class=3D"">Breaking =
the proxy principle is neither a Tor nor a VPN problem.<br class=3D"">It's=
 just browser vendors breaking a promise given to users that<br =
class=3D"">configured proxies would be respected.<br class=3D""><br =
class=3D""><blockquote type=3D"cite" class=3D"">I=E2=80=99m SOCKs =
ignorant.<br class=3D""></blockquote><br class=3D"">So we all have our =
knowledge deficiencies...<br class=3D"">good to be honest about them.<br =
class=3D""><br class=3D""></div></blockquote><div class=3D""><br =
class=3D""></div><div class=3D"">And now that I=E2=80=99ve educated =
myself somewhat, I=E2=80=99m agreeing with you.</div><div class=3D""><br =
class=3D""></div><br class=3D""><blockquote type=3D"cite" class=3D""><div =
class=3D""><blockquote type=3D"cite" class=3D""><blockquote type=3D"cite" =
class=3D"">And if local IP information is so super harmless, then why is =
it<br class=3D"">an obvious reason for TBB devs to disable the entire =
thing?<br class=3D"">I presume for the reasons I stated before.<br =
class=3D""></blockquote><br class=3D"">I have no idea, but I fear it is =
because no one has split these issues out.<br class=3D""></blockquote><br =
class=3D"">So let's dissect some more until we know what we are =
doing.<br class=3D""></div></blockquote><div class=3D""><br =
class=3D""></div><div class=3D"">Which turns out to be the only useful =
contribution my &lt;rant/&gt; made.</div><div class=3D""><br =
class=3D""></div><br class=3D""><blockquote type=3D"cite" class=3D""><div =
class=3D""><br class=3D""><blockquote type=3D"cite" class=3D"">I=E2=80=99m=
 of the view that there are _useful_ instances of P2P crypto<br =
class=3D"">in-browser - we will clearly have to disagree about that.<br =
class=3D""></blockquote><br class=3D"">I think we could improve the =
user's ability to ensure end-to-end<br class=3D"">authentication beyond =
server-based negotiation. Something like<br class=3D"">rendering =
persistent public keys into QR codes and having people<br =
class=3D"">exchange those QR codes at cocktail parties and =
hackathons.<br class=3D""><br class=3D"">Servers would still be able to =
provide the signaling code, but users<br class=3D"">would be granted a =
guarantee that the E2E connection is authentic.<br class=3D""><br =
class=3D""></div></blockquote><br class=3D""></div><div class=3D"" =
style=3D"font-family: LucidaSansUnicode;"><a href=3D"https://yopet.us/" =
class=3D"">https://yopet.us</a>&nbsp;:-)&nbsp;</div><div =
apple-content-edited=3D"true" class=3D"" style=3D"font-family: =
LucidaSansUnicode;"><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; font-family: Helvetica; =
border-spacing: 0px;"><div class=3D""><br =
class=3D""></div></span></div></div></body></html>=

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From: Stefan Holmer <holmer@google.com>
Date: Fri, 24 Jul 2015 10:51:58 +0000
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Subject: Re: [rtcweb] Congestion warmup
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Luis, for what you are talking about there exists a draft in the rmcat
working group: https://tools.ietf.org/html/draft-singh-rmcat-adaptive-fec-0=
1

I tend to agree that it may not be a good idea to blast packets to the
other end before accepting the call.

/Stefan

On Fri, Jul 24, 2015 at 11:52 AM Justin Uberti <juberti@google.com> wrote:

> I tend to think BWE warmup is out of scope for this discussion. It's not
> clear that the callee wants to receive perhaps substantial amounts of
> traffic prior to having accepted the call.
>
> On Fri, Jul 24, 2015 at 11:47 AM, Luis L=C3=B3pez Fern=C3=A1ndez <luis.lo=
pez@urjc.es
> > wrote:
>
>> This is an interesting possibility. Using some kind of fec as padding
>> could make possible to avoid using a fake codec allowing receivers to
>> behave normally and without requiring implementing a different behavior =
for
>> the fake codec data. In addition, this protection of the encoded
>> static/black image may avoid PLIs in case of severe packet loss when
>> hitting a BW during the probe. This could be even used as a probe mechan=
ism
>> for BW estimation with the call fully open, but this is another story.
>>
>> El 24/07/2015, a las 10:54, Peter Thatcher <pthatcher@google.com>
>> escribi=C3=B3:
>>
>> Could the new flexfec payload type be used for padding?
>>
>> On Thu, Jul 23, 2015 at 10:17 PM, Randell Jesup <randell-ietf@jesup.org>
>> wrote:
>>
>>> I'd love to be able to warm up congestion control, and not just DTLS.
>>> That requires sending data and allowing it to increase to probe the
>>> connection and open the congestion window, and preferably sending in bo=
th
>>> directions (i.e. a=3Dsendrecv in the warm-up answer).
>>>
>>> We'd need to find a way to generically pad RTP packets to a target size
>>> (since encoding a static/black image won't use enough bytes).  The obvi=
ous
>>> codec-agnostic mechanism would be a header extension.  Since they can b=
e
>>> any 4-byte-multiple size, that would pretty much work up to ~384K (1500
>>> byte packets, 1/frame @30fps). However, we could run the fake video
>>> framerate arbitrarily high to hit any bandwidth target we need.
>>>
>>> Alternatively (and *much* better) we could use a "fake" codec
>>> (a=3Drtpmap:111 fake/90000) which doesn't actually include any real dat=
a,
>>> just padding.  Advantages: no CPU code to encode/decode.  No dealing wi=
th
>>> header extensions.  Clear indication if the other side wants to do spin=
up
>>> (they can reject the codec if they don't want to).  No encoding/decodin=
g
>>> CPU use.
>>>
>>> The offer would include it as a least-preferred codec to indicate warmu=
p
>>> support.  If answerer answers with fake as the accepted codec (and real
>>> codecs following), it's a warmup answer.  Both sides will send and rece=
ive
>>> fake data to spin up the congestion windows (and also the jitter buffer=
s,
>>> etc).  On actual answer, the fake codec would be dropped (or dropped to
>>> bottom priority).   The answerer would start sending in a real codec, w=
hich
>>> would work since the warmup answer would have included that codec.  The
>>> offerer would see the incoming real-codec packets, and decode them.   W=
hen
>>> the real answer arrives, it will show the accepted codec has changed, a=
nd
>>> the offerer will install that answer and thus and start sending (real)
>>> media.
>>>
>>> We still need a way to tell the system at the answerer side "I want to
>>> accept but it's a warm-up accept" (and perhaps if we want to spin up
>>> congestion control or not while waiting), and a way to say on the offer=
er
>>> side "I want to offer/use warmups".  For those API points see the previ=
ous
>>> options from myself and Peter - or simply a way to say createAnswer of =
a
>>> warmup answer, and have that agree to all tracks with fake codecs being
>>> primary.
>>>
>>> I'll note that all of these methods would benefit from a mechanism to
>>> indicate the desired bitrate targets for each fake track; for solutions
>>> that use some sort of fake video input (canvas.captureStream/etc) it
>>> inherently allows the application to define the size and framerate, whi=
ch
>>> are the primary knobs used by the browser for setting 'max total bitrat=
e'
>>> for the track.   You don't want the application to set the bandwidth
>>> directly, you want it to define the parameters of the video stream so t=
he
>>> system can set "reasonable" limits for that resolution & situation.
>>>
>>> _______________________________________________
>>> 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
>

--047d7bdc189a948ffc051b9ccb09
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<div dir=3D"ltr">Luis, for what you are talking about there exists a draft =
in the rmcat working group:=C2=A0<a href=3D"https://tools.ietf.org/html/dra=
ft-singh-rmcat-adaptive-fec-01">https://tools.ietf.org/html/draft-singh-rmc=
at-adaptive-fec-01</a><div><br></div><div>I tend to agree that it may not b=
e a good idea to blast packets to the other end before accepting the call.<=
/div><div><div><div><br></div><div>/Stefan</div></div></div></div><br><div =
class=3D"gmail_quote"><div dir=3D"ltr">On Fri, Jul 24, 2015 at 11:52 AM Jus=
tin Uberti &lt;<a href=3D"mailto:juberti@google.com">juberti@google.com</a>=
&gt; wrote:<br></div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 =
0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr">I tend=
 to think BWE warmup is out of scope for this discussion. It&#39;s not clea=
r that the callee wants to receive perhaps substantial amounts of traffic p=
rior to having accepted the call.</div><div class=3D"gmail_extra"><br><div =
class=3D"gmail_quote">On Fri, Jul 24, 2015 at 11:47 AM, Luis L=C3=B3pez Fer=
n=C3=A1ndez <span dir=3D"ltr">&lt;<a href=3D"mailto:luis.lopez@urjc.es" tar=
get=3D"_blank">luis.lopez@urjc.es</a>&gt;</span> wrote:<br><blockquote clas=
s=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;pad=
ding-left:1ex"><div style=3D"word-wrap:break-word"><div>This is an interest=
ing possibility. Using some kind of fec as padding could make possible to a=
void using a fake codec allowing receivers to behave normally and without r=
equiring implementing a different behavior for the fake codec data. In addi=
tion, this protection of the encoded static/black image may avoid PLIs in c=
ase of severe packet loss when hitting a BW during the probe. This could be=
 even used as a probe mechanism for BW estimation with the call fully open,=
 but this is another story.
</div><div><div>
<br><div><div>El 24/07/2015, a las 10:54, Peter Thatcher &lt;<a href=3D"mai=
lto:pthatcher@google.com" target=3D"_blank">pthatcher@google.com</a>&gt; es=
cribi=C3=B3:</div><br><blockquote type=3D"cite"><div dir=3D"ltr"><div class=
=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-serif">Could t=
he new flexfec payload type be used for padding? =C2=A0</div></div><div cla=
ss=3D"gmail_extra"><br><div class=3D"gmail_quote">On Thu, Jul 23, 2015 at 1=
0:17 PM, Randell Jesup <span dir=3D"ltr">&lt;<a href=3D"mailto:randell-ietf=
@jesup.org" target=3D"_blank">randell-ietf@jesup.org</a>&gt;</span> wrote:<=
br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left=
:1px #ccc solid;padding-left:1ex">I&#39;d love to be able to warm up conges=
tion control, and not just DTLS.=C2=A0 That requires sending data and allow=
ing it to increase to probe the connection and open the congestion window, =
and preferably sending in both directions (i.e. a=3Dsendrecv in the warm-up=
 answer).<br>
<br>
We&#39;d need to find a way to generically pad RTP packets to a target size=
 (since encoding a static/black image won&#39;t use enough bytes).=C2=A0 Th=
e obvious codec-agnostic mechanism would be a header extension.=C2=A0 Since=
 they can be any 4-byte-multiple size, that would pretty much work up to ~3=
84K (1500 byte packets, 1/frame @30fps). However, we could run the fake vid=
eo framerate arbitrarily high to hit any bandwidth target we need.<br>
<br>
Alternatively (and *much* better) we could use a &quot;fake&quot; codec (a=
=3Drtpmap:111 fake/90000) which doesn&#39;t actually include any real data,=
 just padding.=C2=A0 Advantages: no CPU code to encode/decode.=C2=A0 No dea=
ling with header extensions.=C2=A0 Clear indication if the other side wants=
 to do spinup (they can reject the codec if they don&#39;t want to).=C2=A0 =
No encoding/decoding CPU use.<br>
<br>
The offer would include it as a least-preferred codec to indicate warmup su=
pport.=C2=A0 If answerer answers with fake as the accepted codec (and real =
codecs following), it&#39;s a warmup answer.=C2=A0 Both sides will send and=
 receive fake data to spin up the congestion windows (and also the jitter b=
uffers, etc).=C2=A0 On actual answer, the fake codec would be dropped (or d=
ropped to bottom priority).=C2=A0 =C2=A0The answerer would start sending in=
 a real codec, which would work since the warmup answer would have included=
 that codec.=C2=A0 The offerer would see the incoming real-codec packets, a=
nd decode them.=C2=A0 =C2=A0When the real answer arrives, it will show the =
accepted codec has changed, and the offerer will install that answer and th=
us and start sending (real) media.<br>
<br>
We still need a way to tell the system at the answerer side &quot;I want to=
 accept but it&#39;s a warm-up accept&quot; (and perhaps if we want to spin=
 up congestion control or not while waiting), and a way to say on the offer=
er side &quot;I want to offer/use warmups&quot;.=C2=A0 For those API points=
 see the previous options from myself and Peter - or simply a way to say cr=
eateAnswer of a warmup answer, and have that agree to all tracks with fake =
codecs being primary.<br>
<br>
I&#39;ll note that all of these methods would benefit from a mechanism to i=
ndicate the desired bitrate targets for each fake track; for solutions that=
 use some sort of fake video input (canvas.captureStream/etc) it inherently=
 allows the application to define the size and framerate, which are the pri=
mary knobs used by the browser for setting &#39;max total bitrate&#39; for =
the track.=C2=A0 =C2=A0You don&#39;t want the application to set the bandwi=
dth directly, you want it to define the parameters of the video stream so t=
he system can set &quot;reasonable&quot; limits for that resolution &amp; s=
ituation.<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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div><br></div>
_______________________________________________<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></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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>
_______________________________________________<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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div>

--047d7bdc189a948ffc051b9ccb09--


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Subject: Re: [rtcweb] Options for DTLS warmup and directionality
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I have added this bug:

https://github.com/w3c/mediacapture-main/issues/221

to the W3C github in order to track creating "naked" MediaStreamTracks.
Please comment there, or on public-mediacapture@w3.org.



From nobody Fri Jul 24 04:20:45 2015
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Subject: Re: [rtcweb] Options for DTLS warmup and directionality
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--001a11c345c06b69ca051b9d31ed
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I sent an email to the W3C list initiating discussion about what API points
we need.  I think we should continue all discussion there.

And I think the bug is more than just "Add naked tracks", and that title is
already assuming a particular solution to the problem.

On Fri, Jul 24, 2015 at 4:05 AM, Harald Alvestrand <harald@alvestrand.no>
wrote:

> I have added this bug:
>
> https://github.com/w3c/mediacapture-main/issues/221
>
> to the W3C github in order to track creating "naked" MediaStreamTracks.
> Please comment there, or on public-mediacapture@w3.org.
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--001a11c345c06b69ca051b9d31ed
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<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:arial,he=
lvetica,sans-serif">I sent an email to the W3C list initiating discussion a=
bout what API points we need.=C2=A0 I think we should continue all discussi=
on there. =C2=A0</div><div class=3D"gmail_default" style=3D"font-family:ari=
al,helvetica,sans-serif"><br></div><div class=3D"gmail_default" style=3D"fo=
nt-family:arial,helvetica,sans-serif">And I think the bug is more than just=
 &quot;Add naked tracks&quot;, and that title is already assuming a particu=
lar solution to the problem.</div></div><div class=3D"gmail_extra"><br><div=
 class=3D"gmail_quote">On Fri, Jul 24, 2015 at 4:05 AM, Harald Alvestrand <=
span dir=3D"ltr">&lt;<a href=3D"mailto:harald@alvestrand.no" target=3D"_bla=
nk">harald@alvestrand.no</a>&gt;</span> wrote:<br><blockquote class=3D"gmai=
l_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left=
:1ex">I have added this bug:<br>
<br>
<a href=3D"https://github.com/w3c/mediacapture-main/issues/221" rel=3D"nore=
ferrer" target=3D"_blank">https://github.com/w3c/mediacapture-main/issues/2=
21</a><br>
<br>
to the W3C github in order to track creating &quot;naked&quot; MediaStreamT=
racks.<br>
Please comment there, or on <a href=3D"mailto:public-mediacapture@w3.org">p=
ublic-mediacapture@w3.org</a>.<br>
<div class=3D"HOEnZb"><div class=3D"h5"><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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div>

--001a11c345c06b69ca051b9d31ed--


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From: "Black, David" <david.black@emc.com>
To: "Ruediger.Geib@telekom.de" <Ruediger.Geib@telekom.de>, "karen.nielsen@tieto.com" <karen.nielsen@tieto.com>
Thread-Topic: [tsvwg] diffserv marking for rtcweb data channel
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Cc: "Black, David" <david.black@emc.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, "tsvwg@ietf.org" <tsvwg@ietf.org>
Subject: Re: [rtcweb] [tsvwg] diffserv marking for rtcweb data channel
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From nobody Fri Jul 24 06:12:42 2015
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Fri, 24 Jul 2015 15:12:19 +0200
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Subject: Re: [rtcweb] Comments on draft-nandakumar-rtcweb-sdp
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2015-07-24 12:11 GMT+02:00 Peter Thatcher <pthatcher@google.com>:
> There are applications already that basically build custom SDP with JS to
> use it as an API and doesn't send the SDP over the signalling.

The fact that developers are capable of building SDP monsters for
local use based on custom signaled data, and still must feed the
RTCPeerConnection with them, makes me unhappy. Typically the JS app
parses data sent by the peer/server via HTTP or WebSocket, but when it
comes to WebRTC 1.0 the JS app must re-build a blob for the browser to
parse it. The more I think about it the less sense it has.


>  instead of this document being "Examples of SDP that is signalled", I th=
ink it should be "Examples of SDP that can be put into setLocalDescription =
and setRemoteDescription, and returned by createOffer and createAnswer".

If the draft was about inputs for setLocalDescription, etc, it should
be part of the W3C WebRTC specification, am I wrong?


Said that, the draft looks very well (given what we have: the SDP).


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


From nobody Fri Jul 24 23:01:22 2015
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From: <richard.vandet@kpn.com>
To: <david.black@emc.com>
Thread-Topic: [rtcweb] [tsvwg] diffserv marking for rtcweb data channel
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--_000_E4FD72E43B1C46B9B4E4624BDC4085ECkpncom_
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Sometimes legislation prohibits the QoS model over open internet, all traff=
ic must be treated the same.

A VPN may be a solution, based on a SPs services model like Voice. Setting =
up a VPN on the fly once the connection is established may suit the needs o=
f a developer.

Richard

Op 24 jul. 2015 om 14:29 heeft Black, David <david.black@emc.com<mailto:dav=
id.black@emc.com>> het volgende geschreven:

<WG chair hat OFF>

> All DSCPs, which have not been negotiated between access and interconnect=
ion are possibly bleached, especially if a network
> provider operates QoS within the network.

=93possibly=94 -> =93probably=94 -> =93almost certainly=94 depending on the=
 network (YMMV).

> I=92d like to read a little more about how WEBrtc figures out which QoS r=
esources are configured and which of these are available
> for such a flow at an access when required. How and where are the resourc=
es consumed by each WERrtc PHB controlled and
> how are these aligned with the QoS resources consumed by other services?

The short Web RTC answer to all of the above =93it doesn=92t do that=94.  T=
he DSCPs are in effect QoS service requests that the various networks invol=
ved may deal with as they wish.  Bleaching is one possibility (in effect an=
 =93I don=92t do that=94 response to the request).  Even in that case, ther=
e may be value in DSCP usage for forwarding by equipment between the browse=
r and the interface to the ISP=92s network (e.g., residential router/NAT/AP=
 box).

In spite of all of the above, I have been told of good experiences with use=
 of CS2 for gaming and CS1 for lower effort (less than best effort), plus E=
F use for gaming was mentioned in the tsvwg meeting yesterday.

This is not the first time I=92ve heard a request for a =93what QoS service=
s are available via which DSCPs?=94 interface/protocol.  I don=92t know of =
one, and I=92m not sure how to incentivize deployment if we designed one.

What is likely to happen is that if certain types of traffic favor specific=
 DSCPs, then operators who want to improve the user experience of those who=
 send/receive such traffic may put in QoS mechanisms for such traffic keyed=
 by those DSCPs at network ingress.  OTOH, a =93tragedy of the commons=94 i=
s possible here, e.g., the story in tsvwg about how a lot of non-voice traf=
fic showing up marked as if it were voice resulted in the network operator =
removing that special treatment of (residential, I presume) voice traffic.

</WG chair hat OFF>

Thanks,
--David

From: tsvwg [mailto:tsvwg-bounces@ietf.org] On Behalf Of Ruediger.Geib@tele=
kom.de<mailto:Ruediger.Geib@telekom.de>
Sent: Friday, July 24, 2015 5:33 AM
To: karen.nielsen@tieto.com<mailto:karen.nielsen@tieto.com>
Cc: Michael.Tuexen@lurchi.franken.de<mailto:Michael.Tuexen@lurchi.franken.d=
e>; rtcweb@ietf.org<mailto:rtcweb@ietf.org>; tsvwg@ietf.org<mailto:tsvwg@ie=
tf.org>; fluffy@iii.ca<mailto:fluffy@iii.ca>
Subject: Re: [tsvwg] diffserv marking for rtcwb data channel

Hi Karen,

QoS  needs to be commercially negotiated at interconnection. If QoS should =
be available at a consumer access, the consumer division of a company must =
be involved in addition as well as the technical staff setting up and maint=
aining technical configurations. 11 PHBs for a single consumer service are =
a challenging requirement.

All DSCPs, which have not been negotiated between access and interconnectio=
n are possibly bleached, especially if a network provider operates QoS with=
in the network.

QoS to me means overprovisioning to ensure transport free of congestion. If=
 I see 11 PHBs for a single flow, I=92m not sure if this assumes more than =
CoS (drop me last). I=92d like to read a little more about how WEBrtc figur=
es out which QoS resources are configured and which of these are available =
for such a flow at an access when required. How and where are the resources=
 consumed by each WERrtc PHB controlled and how are these aligned with the =
QoS resources consumed by other services?

Regards,

Ruediger

Von: Karen Elisabeth Egede Nielsen [mailto:karen.nielsen@tieto.com]
Gesendet: Freitag, 24. Juli 2015 10:28
An: Geib, R=FCdiger
Cc: rtcweb@ietf.org<mailto:rtcweb@ietf.org>; tsvwg@ietf.org<mailto:tsvwg@ie=
tf.org>; Michael Tuexen; fluffy@iii.ca<mailto:fluffy@iii.ca>
Betreff: RE: [tsvwg] diffserv marking for rtcwb data channel

Hi Ruediger,

Thanks a lot.

So I think that you=92re saying that even if one made sure that different d=
iff serv markings were
split on different 5-tuples, this setting likely would be bleached at the n=
etwork.

Is that correctly understood ?

If so, then (it sounds to me =96 not knowing too much about this) that from=
 a coupling congestion control perspective it
would be preferable to  use SCTP scheduling prioritization (within one and =
the same association) for fine granular differentiation in between
multiple data channels in between the same end-points rather than different=
iation by means of diff serv marking.

When/if one get a more consistent and forceful ECN operation in AQMs and SC=
TP end-points, perhaps the coupling congestion
principles achieved by means of sctp stream scheduling get to be less impor=
tant.
On the other hand, again as said before, having the one and the same, or mu=
ltiple, flow control (receiver buffer) contexts may be
a strength (simplicity) or a weakness.

BR, Karen

From: Ruediger.Geib@telekom.de<mailto:Ruediger.Geib@telekom.de> [mailto:Rue=
diger.Geib@telekom.de<mailto:Ruediger.Geib@telekom.de>]
Sent: Friday, July 24, 2015 10:00 AM
To: karen.nielsen@tieto.com<mailto:karen.nielsen@tieto.com>
Cc: rtcweb@ietf.org<mailto:rtcweb@ietf.org>; tsvwg@ietf.org<mailto:tsvwg@ie=
tf.org>
Subject: AW: [tsvwg] diffserv marking for rtcwb data channel

Karen,

      Finally there was a comment at the tsvwg meeting that the same 5-tupl=
e with
       different diffserv markings would not get through the network =96 if=
 that generally
       is correct,  then the solution is very easy: one MUST use different =
associations to
       get port (5-tuple) differentiation.
       I think that if this is correct, one also wants to ask what happens =
if the diffserv is
       changed for the same 5-tuple =96 i.e., is the network robust to this=
 change even if
       it is not robust to concurrent usage of different markings for the s=
ame 5-tuple.

Talking about the policies of the company I work for only:

Possibilities to enforce QoS policies at a provider BRAS or BNG terminating=
 wired
consumer accesses are limited. QoS policies based on full 5 tuple are very
conservative and hard to maintain. I wouldn=92t expect full 5 tuple as basi=
s for a QoS
policy then. Trust in the correct information on which a BRAS/BNG can
base QoS classification is an important issue. In any case, I=92d not expec=
t
more than one DSCP is assigned based on such a policy for the time
being. I=92m not aware of any actual requirement to support e.g. an AFn
class by all three PHBs.

At interconnection points, any unexpected DSCP is bleached. This must
be done to protect backbone QoS production. QoS transport can be
negotiated for a set of defined DSCPs.

A study of the EU found DPI to be more common in wireless communication tha=
n in wired communication. I=92d not expect the QoS policies in that area to=
 be more liberal than for fixed access networks for the time being.

Regards,

Ruediger




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

--_000_E4FD72E43B1C46B9B4E4624BDC4085ECkpncom_
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<div>Sometimes legislation prohibits the QoS model over open internet, all =
traffic must be treated the same.<br>
<br>
A VPN may be a solution, based on a SPs services model like Voice. Setting =
up a VPN on the fly once the connection is established may suit the needs o=
f a developer.</div>
<div>
<div><br>
</div>
<div>Richard</div>
</div>
<div><br>
Op 24 jul. 2015 om 14:29 heeft Black, David &lt;<a href=3D"mailto:david.bla=
ck@emc.com">david.black@emc.com</a>&gt; het volgende geschreven:<br>
<br>
</div>
<blockquote type=3D"cite">
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<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;;color:black">&lt;WG chair hat OFF&gt;<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;;color:black"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;;color:black">&gt;
</span><span style=3D"color:#1F497D">All DSCPs, which have not been negotia=
ted between access and interconnection are possibly bleached, especially if=
 a network<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">&gt; provider operates=
 QoS within the network.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;;color:black"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;;color:black">=93possibly=94 -&gt; =93probably=94 -&gt; =93a=
lmost certainly=94 depending on the network (YMMV).<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;;color:black"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;;color:black">&gt;</span><span style=3D"color:#1F497D"> I=92=
d like to read a little more about how WEBrtc figures out which QoS resourc=
es are configured and which of these are available<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">&gt; for such a flow a=
t an access when required. How and where are the resources consumed by each=
 WERrtc PHB controlled and<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">&gt; how are these ali=
gned with the QoS resources consumed by other services?<o:p></o:p></span></=
p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;;color:black"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;;color:black">The short Web RTC answer to all of the above =
=93it doesn=92t do that=94.&nbsp; The DSCPs are in effect QoS service reque=
sts that the various networks involved may deal with as they
 wish.&nbsp; Bleaching is one possibility (in effect an =93I don=92t do tha=
t=94 response to the request).&nbsp; Even in that case, there may be value =
in DSCP usage for forwarding by equipment between the browser and the inter=
face to the ISP=92s network (e.g., residential router/NAT/AP
 box).<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;;color:black"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;;color:black">In spite of all of the above, I have been told=
 of good experiences with use of CS2 for gaming and CS1 for lower effort (l=
ess than best effort), plus EF use for gaming
 was mentioned in the tsvwg meeting yesterday.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;;color:black"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;;color:black">This is not the first time I=92ve heard a requ=
est for a =93what QoS services are available via which DSCPs?=94 interface/=
protocol.&nbsp; I don=92t know of one, and I=92m not sure how
 to incentivize deployment if we designed one.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;;color:black"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;;color:black">What is likely to happen is that if certain ty=
pes of traffic favor specific DSCPs, then operators who want to improve the=
 user experience of those who send/receive such
 traffic may put in QoS mechanisms for such traffic keyed by those DSCPs at=
 network ingress.&nbsp; OTOH, a =93tragedy of the commons=94 is possible he=
re, e.g., the story in tsvwg about how a lot of non-voice traffic showing u=
p marked as if it were voice resulted in
 the network operator removing that special treatment of (residential, I pr=
esume) voice traffic.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;;color:black"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;;color:black">&lt;/WG chair hat OFF&gt;<o:p></o:p></span></p=
>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;;color:black"><o:p>&nbsp;</o:p></span></p>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;;color:black">Thanks,<br>
--David</span><span style=3D"font-family:&quot;Courier New&quot;;color:blac=
k"><o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;;color:black"><o:p>&nbsp;</o:p></span></p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> tsvwg [<=
a href=3D"mailto:tsvwg-bounces@ietf.org">mailto:tsvwg-bounces@ietf.org</a>]
<b>On Behalf Of </b><a href=3D"mailto:Ruediger.Geib@telekom.de">Ruediger.Ge=
ib@telekom.de</a><br>
<b>Sent:</b> Friday, July 24, 2015 5:33 AM<br>
<b>To:</b> <a href=3D"mailto:karen.nielsen@tieto.com">karen.nielsen@tieto.c=
om</a><br>
<b>Cc:</b> <a href=3D"mailto:Michael.Tuexen@lurchi.franken.de">Michael.Tuex=
en@lurchi.franken.de</a>;
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>; <a href=3D"mailto:t=
svwg@ietf.org">
tsvwg@ietf.org</a>; <a href=3D"mailto:fluffy@iii.ca">fluffy@iii.ca</a><br>
<b>Subject:</b> Re: [tsvwg] diffserv marking for rtcwb data channel<o:p></o=
:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><span lang=3D"DE" style=3D"color:#1F497D">Hi Karen,<=
o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"DE" style=3D"color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">QoS&nbsp; needs to be =
commercially negotiated at interconnection. If QoS should be available at a=
 consumer access, the consumer division of a company must be involved in ad=
dition as well as the technical staff setting
 up and maintaining technical configurations. 11 PHBs for a single consumer=
 service are a challenging requirement.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">All DSCPs, which have =
not been negotiated between access and interconnection are possibly bleache=
d, especially if a network provider operates QoS within the network.<o:p></=
o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">QoS to me means overpr=
ovisioning to ensure transport free of congestion. If I see 11 PHBs for a s=
ingle flow, I=92m not sure if this assumes more than CoS (drop me last). I=
=92d like to read a little more about how
 WEBrtc figures out which QoS resources are configured and which of these a=
re available for such a flow at an access when required. How and where are =
the resources consumed by each WERrtc PHB controlled and how are these alig=
ned with the QoS resources consumed
 by other services?<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Regards,<o:p></o:p></s=
pan></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Ruediger<o:p></o:p></s=
pan></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">Von:</span></b><span style=3D"font-si=
ze:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> Karen Eli=
sabeth Egede Nielsen [<a href=3D"mailto:karen.nielsen@tieto.com">mailto:kar=
en.nielsen@tieto.com</a>]
<br>
<b>Gesendet:</b> Freitag, 24. Juli 2015 10:28<br>
<b>An:</b> Geib, </span><span lang=3D"DE" style=3D"font-size:10.0pt;font-fa=
mily:&quot;Tahoma&quot;,&quot;sans-serif&quot;">R=FCdiger<br>
<b>Cc:</b> <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>; <a href=
=3D"mailto:tsvwg@ietf.org">
tsvwg@ietf.org</a>; Michael Tuexen; <a href=3D"mailto:fluffy@iii.ca">fluffy=
@iii.ca</a><br>
<b>Betreff:</b> RE: [tsvwg] diffserv marking for rtcwb data channel<o:p></o=
:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><span lang=3D"DE"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Hi Ruediger,</span><o:=
p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">&nbsp;</span><o:p></o:=
p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Thanks a lot.</span><o=
:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">&nbsp;</span><o:p></o:=
p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">So I think that you=92=
re saying that even if one made sure that different diff serv markings were=
</span><o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">split on different 5-t=
uples, this setting likely would be bleached at the network.</span><o:p></o=
:p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">&nbsp;</span><o:p></o:=
p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Is that correctly unde=
rstood ?</span><o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">&nbsp;</span><o:p></o:=
p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">If so, then (it sounds=
 to me =96 not knowing too much about this) that from a coupling congestion=
 control perspective it
</span><o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">would be preferable to=
&nbsp; use SCTP scheduling prioritization (within one and the same associat=
ion) for fine granular differentiation in between</span><o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">multiple data channels=
 in between the same end-points rather than differentiation by means of dif=
f serv marking.</span><o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">&nbsp;</span><o:p></o:=
p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">When/if one get a more=
 consistent and forceful ECN operation in AQMs and SCTP end-points, perhaps=
 the coupling congestion</span><o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">principles achieved by=
 means of sctp stream scheduling get to be less important.
</span><o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">On the other hand, aga=
in as said before, having the one and the same, or multiple, flow control (=
receiver buffer) contexts may be</span><o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">a strength (simplicity=
) or a weakness. &nbsp;&nbsp;&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">&nbsp;</span><o:p></o:=
p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">BR, Karen </span><o:p>=
</o:p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">&nbsp;</span><o:p></o:=
p></p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">
<a href=3D"mailto:Ruediger.Geib@telekom.de">Ruediger.Geib@telekom.de</a> [m=
ailto:<a href=3D"mailto:Ruediger.Geib@telekom.de">Ruediger.Geib@telekom.de<=
/a>]
<br>
<b>Sent:</b> Friday, July 24, 2015 10:00 AM<br>
<b>To:</b> <a href=3D"mailto:karen.nielsen@tieto.com">karen.nielsen@tieto.c=
om</a><br>
<b>Cc:</b> <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>; <a href=
=3D"mailto:tsvwg@ietf.org">
tsvwg@ietf.org</a><br>
<b>Subject:</b> AW: [tsvwg] diffserv marking for rtcwb data channel</span><=
o:p></o:p></p>
</div>
</div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Karen,</span><o:p></o:=
p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">&nbsp;</span><o:p></o:=
p></p>
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;
</span></b>Finally there was a comment at the tsvwg meeting that the same 5=
-tuple with
<o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;</span>different diffserv markings would not get<span s=
tyle=3D"color:#1F497D">
</span>through the network =96 if that generally <o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;</span>is correct, &nbsp;then the solution is very easy=
: one MUST use different associations to
<o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;</span>get port (5-tuple) differentiation.<o:p></o:p></=
p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp; </span>I think that if this is correct, one also wants to as=
k what happens if the diffserv is
<o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;</span>changed for the same 5-tuple =96 i.e., is the ne=
twork robust to this change even if
<o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;</span>it is not robust to concurrent usage of differen=
t markings for the same 5-tuple.<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Talking about the poli=
cies of the company I work for only:</span><o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">&nbsp;</span><o:p></o:=
p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Possibilities to enfor=
ce QoS policies at a provider BRAS or</span>&nbsp;<span style=3D"color:#1F4=
97D">BNG terminating wired
</span><o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">consumer accesses are =
limited. QoS policies based on full 5 tuple are very
</span><o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">conservative and hard =
to maintain. I wouldn=92t expect full 5 tuple as basis for a QoS
</span><o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">policy then. Trust in =
the correct information on which a BRAS/BNG can
</span><o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">base QoS classificatio=
n is an important issue. In any case, I=92d not expect
</span><o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">more than one DSCP is =
assigned based on such a policy for the time
</span><o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">being. I=92m not aware=
 of any actual requirement to support e.g. an AFn
</span><o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">class by all three PHB=
s.</span><o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">&nbsp;</span><o:p></o:=
p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">At interconnection poi=
nts, any unexpected DSCP is bleached. This must
</span><o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">be done to protect bac=
kbone QoS production. QoS transport can be
</span><o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">negotiated for a set o=
f defined DSCPs.
</span><o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">&nbsp;</span><o:p></o:=
p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">A study of the EU foun=
d DPI to be more common in wireless communication than in wired communicati=
on. I=92d not expect the QoS policies in that area to be more liberal than =
for fixed access networks for the time
 being.</span><o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">&nbsp;</span><o:p></o:=
p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Regards,</span><o:p></=
o:p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">&nbsp;</span><o:p></o:=
p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Ruediger</span><o:p></=
o:p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">&nbsp;</span><o:p></o:=
p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">&nbsp;</span><o:p></o:=
p></p>
<p class=3D"MsoNormal" style=3D"margin-left:.25in">&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
</div>
</div>
</div>
</blockquote>
<blockquote type=3D"cite">
<div><span>_______________________________________________</span><br>
<span>rtcweb mailing list</span><br>
<span><a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a></span><br>
<span><a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.=
ietf.org/mailman/listinfo/rtcweb</a></span><br>
</div>
</blockquote>
</body>
</html>

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From: <richard.vandet@kpn.com>
To: <goran.ap.eriksson@ericsson.com>
Thread-Topic: [rtcweb] [tsvwg] diffserv marking for rtcweb data channel
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Date: Sat, 25 Jul 2015 17:16:13 +0000
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Cc: karen.nielsen@tieto.com, david.black@emc.com, rtcweb@ietf.org, Ruediger.Geib@telekom.de, tsvwg@ietf.org
Subject: Re: [rtcweb] [tsvwg] diffserv marking for rtcweb data channel
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> Op 25 jul. 2015 om 15:56 heeft G=F6ran Eriksson AP <goran.ap.eriksson@eri=
csson.com> het volgende geschreven:
>=20
>=20
>=20
>> Sometimes legislation prohibits the QoS model over open internet, all
>> traffic must be treated the same.
>>=20
>> A VPN may be a solution, based on a SPs services model like Voice.
>> Setting up a VPN on the fly once the connection is established may suit
>> the needs of a developer.
>=20
> You mean a =93specialised service=94 in the EU net neutrality terminology=
?
>=20
Indeed Net neutrality.=20

> A =93VPN=94 can have many shapes and forms- a closed user group and HTTPS
> would be sufficient, right?
>=20
Something like bandwith reservaties: a temporarily tunnel to the centralize=
d WebRTC server. Especially when using Voice and video.=20

> Or?
>=20
>>=20
>>=20
>> Richard
>>=20
>>=20
>> Op 24 jul. 2015 om 14:29 heeft Black, David <david.black@emc.com> het
>> volgende geschreven:
>>=20
>>=20
>>=20
>> <WG chair hat OFF>
>>=20
>> All DSCPs, which have not been negotiated between access and
>> interconnection are possibly bleached, especially if a network
>>> provider operates QoS within the network.
>>=20
>> =93possibly=94 -> =93probably=94 -> =93almost certainly=94 depending on =
the network
>> (YMMV).
>>=20
>>> I=92d like to read a little more about how WEBrtc figures out which QoS
>>> resources are configured and which of these are available
>>> for such a flow at an access when required. How and where are the
>>> resources consumed by each WERrtc PHB controlled and
>>> how are these aligned with the QoS resources consumed by other services=
?
>>=20
>> The short Web RTC answer to all of the above =93it doesn=92t do that=94.=
  The
>> DSCPs are in effect QoS service requests that the various networks
>> involved may deal with as they
>> wish.  Bleaching is one possibility (in effect an =93I don=92t do that=
=94
>> response to the request).  Even in that case, there may be value in DSCP
>> usage for forwarding by equipment between the browser and the interface
>> to the ISP=92s network (e.g., residential router/NAT/AP
>> box).
>>=20
>> In spite of all of the above, I have been told of good experiences with
>> use of CS2 for gaming and CS1 for lower effort (less than best effort),
>> plus EF use for gaming
>> was mentioned in the tsvwg meeting yesterday.
>>=20
>> This is not the first time I=92ve heard a request for a =93what QoS serv=
ices
>> are available via which DSCPs?=94 interface/protocol.  I don=92t know of=
 one,
>> and I=92m not sure how
>> to incentivize deployment if we designed one.
>>=20
>> What is likely to happen is that if certain types of traffic favor
>> specific DSCPs, then operators who want to improve the user experience o=
f
>> those who send/receive such
>> traffic may put in QoS mechanisms for such traffic keyed by those DSCPs
>> at network ingress.  OTOH, a =93tragedy of the commons=94 is possible he=
re,
>> e.g., the story in tsvwg about how a lot of non-voice traffic showing up
>> marked as if it were voice resulted in
>> the network operator removing that special treatment of (residential, I
>> presume) voice traffic.
>>=20
>> </WG chair hat OFF>
>>=20
>> Thanks,
>> --David
>>=20
>>=20
>>=20
>> From: tsvwg [mailto:tsvwg-bounces@ietf.org]
>> On Behalf Of Ruediger.Geib@telekom.de
>> Sent: Friday, July 24, 2015 5:33 AM
>> To: karen.nielsen@tieto.com
>> Cc: Michael.Tuexen@lurchi.franken.de;
>> rtcweb@ietf.org;=20
>> tsvwg@ietf.org <mailto:tsvwg@ietf.org>; fluffy@iii.ca
>> Subject: Re: [tsvwg] diffserv marking for rtcwb data channel
>>=20
>>=20
>>=20
>> Hi Karen,
>>=20
>> QoS  needs to be commercially negotiated at interconnection. If QoS
>> should be available at a consumer access, the consumer division of a
>> company must be involved in addition as well as the technical staff
>> setting
>> up and maintaining technical configurations. 11 PHBs for a single
>> consumer service are a challenging requirement.
>>=20
>> All DSCPs, which have not been negotiated between access and
>> interconnection are possibly bleached, especially if a network provider
>> operates QoS within the network.
>>=20
>> QoS to me means overprovisioning to ensure transport free of congestion.
>> If I see 11 PHBs for a single flow, I=92m not sure if this assumes more
>> than CoS (drop me last). I=92d like to read a little more about how
>> WEBrtc figures out which QoS resources are configured and which of these
>> are available for such a flow at an access when required. How and where
>> are the resources consumed by each WERrtc PHB controlled and how are
>> these aligned with the QoS resources consumed
>> by other services?
>>=20
>> Regards,
>>=20
>> Ruediger
>>=20
>> Von: Karen Elisabeth Egede Nielsen [mailto:karen.nielsen@tieto.com]
>>=20
>> Gesendet: Freitag, 24. Juli 2015 10:28
>> An: Geib, R=FCdiger
>> Cc: rtcweb@ietf.org;
>> tsvwg@ietf.org <mailto:tsvwg@ietf.org>; Michael Tuexen; fluffy@iii.ca
>> Betreff: RE: [tsvwg] diffserv marking for rtcwb data channel
>>=20
>>=20
>>=20
>> Hi Ruediger,
>>=20
>> Thanks a lot.
>>=20
>> So I think that you=92re saying that even if one made sure that differen=
t
>> diff serv markings were
>> split on different 5-tuples, this setting likely would be bleached at th=
e
>> network.
>>=20
>> Is that correctly understood ?
>>=20
>> If so, then (it sounds to me =96 not knowing too much about this) that f=
rom
>> a coupling congestion control perspective it
>>=20
>> would be preferable to  use SCTP scheduling prioritization (within one
>> and the same association) for fine granular differentiation in between
>> multiple data channels in between the same end-points rather than
>> differentiation by means of diff serv marking.
>>=20
>> When/if one get a more consistent and forceful ECN operation in AQMs and
>> SCTP end-points, perhaps the coupling congestion
>> principles achieved by means of sctp stream scheduling get to be less
>> important.
>>=20
>> On the other hand, again as said before, having the one and the same, or
>> multiple, flow control (receiver buffer) contexts may be
>> a strength (simplicity) or a weakness.
>>=20
>> BR, Karen=20
>>=20
>> From:Ruediger.Geib@telekom.de [mailto:Ruediger.Geib@telekom.de]
>>=20
>> Sent: Friday, July 24, 2015 10:00 AM
>> To: karen.nielsen@tieto.com
>> Cc: rtcweb@ietf.org;
>> tsvwg@ietf.org <mailto:tsvwg@ietf.org>
>> Subject: AW: [tsvwg] diffserv marking for rtcwb data channel
>>=20
>>=20
>>=20
>> Karen,
>>=20
>>=20
>> Finally there was a comment at the tsvwg meeting that the same 5-tuple
>> with
>>=20
>>      different diffserv markings would not getthrough the network =96 if
>> that generally=20
>>      is correct,  then the solution is very easy: one MUST use
>> different associations to
>>=20
>>      get port (5-tuple) differentiation.
>>      I think that if this is correct, one also wants to ask what
>> happens if the diffserv is
>>=20
>>      changed for the same 5-tuple =96 i.e., is the network robust to thi=
s
>> change even if
>>=20
>>      it is not robust to concurrent usage of different markings for the
>> same 5-tuple.
>>=20
>> Talking about the policies of the company I work for only:
>>=20
>> Possibilities to enforce QoS policies at a provider BRAS or BNG
>> terminating wired
>>=20
>> consumer accesses are limited. QoS policies based on full 5 tuple are ve=
ry
>>=20
>> conservative and hard to maintain. I wouldn=92t expect full 5 tuple as
>> basis for a QoS
>>=20
>> policy then. Trust in the correct information on which a BRAS/BNG can
>>=20
>> base QoS classification is an important issue. In any case, I=92d not ex=
pect
>>=20
>> more than one DSCP is assigned based on such a policy for the time
>>=20
>> being. I=92m not aware of any actual requirement to support e.g. an AFn
>>=20
>> class by all three PHBs.
>>=20
>> At interconnection points, any unexpected DSCP is bleached. This must
>>=20
>> be done to protect backbone QoS production. QoS transport can be
>>=20
>> negotiated for a set of defined DSCPs.
>>=20
>>=20
>> A study of the EU found DPI to be more common in wireless communication
>> than in wired communication. I=92d not expect the QoS policies in that a=
rea
>> to be more liberal than for fixed access networks for the time
>> being.
>>=20
>> Regards,
>>=20
>> Ruediger
>>=20
>>=20
>>=20
>>=20
>>=20
>>=20
>>=20
>>=20
>>=20
>>=20
>>=20
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


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All,

The chairs have been in discussions with some IESG members to try to =
resolve a MISREF in the video codec draft [1]; the normative reference =
to the overview draft should have been made an informative reference see =
this thread circa Aug =9213 =
(http://www.ietf.org/mail-archive/web/rtcweb/current/msg08624.html).  =
What we=92ve come up with is a) remove the 1st paragraph of s5 from the =
codec draft, b) copy the definitions from the overview draft (and note =
the definitions are also in the overview drat), and c) move the =
reference to an informative reference.  You can see all of this in an =
RFC editor that Alissa has kind crafted here =
http://datatracker.ietf.org/doc/draft-ietf-rtcweb-video/writeup/.

spt (as the doc shepherd)

[1] http://www.rfc-editor.org/cluster_info.php?cid=3DC238=


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Subject: [rtcweb] Fwd: New WebRTC WG charter officially approved
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FYI

---------- Forwarded message ----------
From: Vivien Lacourba <vivien@w3.org>
Date: Mon, Jul 27, 2015 at 10:25 AM
Subject: New WebRTC WG charter officially approved
To: WebRTC WG <public-webrtc@w3.org>


Hi all,

Great news, the new W3C WebRTC Working Group charter [1] has been
officially approved by the W3C Director [2].

The revised charter adds a deliverable for the next version of WebRTC,
has an updated list of deliverables based on the work started under the
previous charter, clarifies its decision policy, and extends the group
until March 2018.

The charter of this Working Group includes a new deliverable that
require W3C Patent Policy licensing commitments from all Participants.

Consequently, all Participants must join or re-join the group, which
involves agreeing to participate under the terms of the revised charter
and the W3C Patent Policy. Current Participants may continue to attend
meetings (teleconferences and face-to-face meetings) for 45 days after
this announcement, even if they have not yet re-joined the group. After
45 days (ie. September 10, 2015), ongoing participation (including
meeting attendance and voting) is only permitted for those who have
re-joined the group.

Use this form to (re)join:
   https://www.w3.org/2004/01/pp-impl/47318/join

Instructions to join the group are available at:
  http://www.w3.org/2004/01/pp-impl/47318/instructions

Thanks,
Vivien on behalf of the WebRTC WG Chairs and Staff contacts

[1] http://www.w3.org/2015/07/webrtc-charter.html
[2]
https://lists.w3.org/Archives/Member/w3c-ac-members/2015JulSep/0024.html

--
Vivien Lacourba                      World Wide Web Consortium
vivien@w3.org                        http://www.w3.org
http://www.w3.org/People/Vivien      Tel: +33.4.92.38.78.89





--=20
Victor Pascual =C3=81vila

--047d7bf0c59c514c29051bdc1bff
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<div dir=3D"ltr">FYI<div><br><div class=3D"gmail_quote">---------- Forwarde=
d message ----------<br>From: <b class=3D"gmail_sendername">Vivien Lacourba=
</b> <span dir=3D"ltr">&lt;<a href=3D"mailto:vivien@w3.org">vivien@w3.org</=
a>&gt;</span><br>Date: Mon, Jul 27, 2015 at 10:25 AM<br>Subject: New WebRTC=
 WG charter officially approved<br>To: WebRTC WG &lt;<a href=3D"mailto:publ=
ic-webrtc@w3.org">public-webrtc@w3.org</a>&gt;<br><br><br>Hi all,<br>
<br>
Great news, the new W3C WebRTC Working Group charter [1] has been<br>
officially approved by the W3C Director [2].<br>
<br>
The revised charter adds a deliverable for the next version of WebRTC,<br>
has an updated list of deliverables based on the work started under the<br>
previous charter, clarifies its decision policy, and extends the group<br>
until March 2018.<br>
<br>
The charter of this Working Group includes a new deliverable that<br>
require W3C Patent Policy licensing commitments from all Participants.<br>
<br>
Consequently, all Participants must join or re-join the group, which<br>
involves agreeing to participate under the terms of the revised charter<br>
and the W3C Patent Policy. Current Participants may continue to attend<br>
meetings (teleconferences and face-to-face meetings) for 45 days after<br>
this announcement, even if they have not yet re-joined the group. After<br>
45 days (ie. September 10, 2015), ongoing participation (including<br>
meeting attendance and voting) is only permitted for those who have<br>
re-joined the group.<br>
<br>
Use this form to (re)join:<br>
=C2=A0 =C2=A0<a href=3D"https://www.w3.org/2004/01/pp-impl/47318/join" rel=
=3D"noreferrer" target=3D"_blank">https://www.w3.org/2004/01/pp-impl/47318/=
join</a><br>
<br>
Instructions to join the group are available at:<br>
=C2=A0 <a href=3D"http://www.w3.org/2004/01/pp-impl/47318/instructions" rel=
=3D"noreferrer" target=3D"_blank">http://www.w3.org/2004/01/pp-impl/47318/i=
nstructions</a><br>
<br>
Thanks,<br>
Vivien on behalf of the WebRTC WG Chairs and Staff contacts<br>
<br>
[1] <a href=3D"http://www.w3.org/2015/07/webrtc-charter.html" rel=3D"norefe=
rrer" target=3D"_blank">http://www.w3.org/2015/07/webrtc-charter.html</a><b=
r>
[2]<br>
<a href=3D"https://lists.w3.org/Archives/Member/w3c-ac-members/2015JulSep/0=
024.html" rel=3D"noreferrer" target=3D"_blank">https://lists.w3.org/Archive=
s/Member/w3c-ac-members/2015JulSep/0024.html</a><br>
<span class=3D"HOEnZb"><font color=3D"#888888"><br>
--<br>
Vivien Lacourba=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 World Wide Web Consortium<br>
<a href=3D"mailto:vivien@w3.org">vivien@w3.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=A0 <a href=3D"http=
://www.w3.org" rel=3D"noreferrer" target=3D"_blank">http://www.w3.org</a><b=
r>
<a href=3D"http://www.w3.org/People/Vivien" rel=3D"noreferrer" target=3D"_b=
lank">http://www.w3.org/People/Vivien</a>=C2=A0 =C2=A0 =C2=A0 Tel: <a href=
=3D"tel:%2B33.4.92.38.78.89" value=3D"+33492387889">+33.4.92.38.78.89</a><b=
r>
<br>
<br>
</font></span></div><br><br clear=3D"all"><div><br></div>-- <br><div class=
=3D"gmail_signature">Victor Pascual =C3=81vila</div>
</div></div>

--047d7bf0c59c514c29051bdc1bff--


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Date: Sun, 26 Jul 2015 07:57:50 +1200
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Subject: Re: [rtcweb] [tsvwg]   diffserv marking for rtcweb data channel
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On 25/07/2015 18:01, richard.vandet@kpn.com wrote:
> Sometimes legislation prohibits the QoS model over open internet, all t=
raffic must be treated the same.

That is a problem for operators (and their lawyers). Actually I don't thi=
nk that
either the EU regulations or the FCC rules prevent diffserv, as long as i=
t isn't
used to discriminate against individual content providers. But as far as =
the IETF
is concerned, I believe we should ignore this completely. It's our job to=
 define
technical methods to support QoS. It's not our job to decide who uses tho=
se methods.

   Brian

>=20
> A VPN may be a solution, based on a SPs services model like Voice. Sett=
ing up a VPN on the fly once the connection is established may suit the n=
eeds of a developer.
>=20
> Richard
>=20
> Op 24 jul. 2015 om 14:29 heeft Black, David <david.black@emc.com<mailto=
:david.black@emc.com>> het volgende geschreven:
>=20
> <WG chair hat OFF>
>=20
>> All DSCPs, which have not been negotiated between access and interconn=
ection are possibly bleached, especially if a network
>> provider operates QoS within the network.
>=20
> =E2=80=9Cpossibly=E2=80=9D -> =E2=80=9Cprobably=E2=80=9D -> =E2=80=9Cal=
most certainly=E2=80=9D depending on the network (YMMV).
>=20
>> I=E2=80=99d like to read a little more about how WEBrtc figures out wh=
ich QoS resources are configured and which of these are available
>> for such a flow at an access when required. How and where are the reso=
urces consumed by each WERrtc PHB controlled and
>> how are these aligned with the QoS resources consumed by other service=
s?
>=20
> The short Web RTC answer to all of the above =E2=80=9Cit doesn=E2=80=99=
t do that=E2=80=9D.  The DSCPs are in effect QoS service requests that th=
e various networks involved may deal with as they wish.  Bleaching is one=
 possibility (in effect an =E2=80=9CI don=E2=80=99t do that=E2=80=9D resp=
onse to the request).  Even in that case, there may be value in DSCP usag=
e for forwarding by equipment between the browser and the interface to th=
e ISP=E2=80=99s network (e.g., residential router/NAT/AP box).
>=20
> In spite of all of the above, I have been told of good experiences with=
 use of CS2 for gaming and CS1 for lower effort (less than best effort), =
plus EF use for gaming was mentioned in the tsvwg meeting yesterday.
>=20
> This is not the first time I=E2=80=99ve heard a request for a =E2=80=9C=
what QoS services are available via which DSCPs?=E2=80=9D interface/proto=
col.  I don=E2=80=99t know of one, and I=E2=80=99m not sure how to incent=
ivize deployment if we designed one.
>=20
> What is likely to happen is that if certain types of traffic favor spec=
ific DSCPs, then operators who want to improve the user experience of tho=
se who send/receive such traffic may put in QoS mechanisms for such traff=
ic keyed by those DSCPs at network ingress.  OTOH, a =E2=80=9Ctragedy of =
the commons=E2=80=9D is possible here, e.g., the story in tsvwg about how=
 a lot of non-voice traffic showing up marked as if it were voice resulte=
d in the network operator removing that special treatment of (residential=
, I presume) voice traffic.
>=20
> </WG chair hat OFF>
>=20
> Thanks,
> --David
>=20
> From: tsvwg [mailto:tsvwg-bounces@ietf.org] On Behalf Of Ruediger.Geib@=
telekom.de<mailto:Ruediger.Geib@telekom.de>
> Sent: Friday, July 24, 2015 5:33 AM
> To: karen.nielsen@tieto.com<mailto:karen.nielsen@tieto.com>
> Cc: Michael.Tuexen@lurchi.franken.de<mailto:Michael.Tuexen@lurchi.frank=
en.de>; rtcweb@ietf.org<mailto:rtcweb@ietf.org>; tsvwg@ietf.org<mailto:ts=
vwg@ietf.org>; fluffy@iii.ca<mailto:fluffy@iii.ca>
> Subject: Re: [tsvwg] diffserv marking for rtcwb data channel
>=20
> Hi Karen,
>=20
> QoS  needs to be commercially negotiated at interconnection. If QoS sho=
uld be available at a consumer access, the consumer division of a company=
 must be involved in addition as well as the technical staff setting up a=
nd maintaining technical configurations. 11 PHBs for a single consumer se=
rvice are a challenging requirement.
>=20
> All DSCPs, which have not been negotiated between access and interconne=
ction are possibly bleached, especially if a network provider operates Qo=
S within the network.
>=20
> QoS to me means overprovisioning to ensure transport free of congestion=
=2E If I see 11 PHBs for a single flow, I=E2=80=99m not sure if this assu=
mes more than CoS (drop me last). I=E2=80=99d like to read a little more =
about how WEBrtc figures out which QoS resources are configured and which=
 of these are available for such a flow at an access when required. How a=
nd where are the resources consumed by each WERrtc PHB controlled and how=
 are these aligned with the QoS resources consumed by other services?
>=20
> Regards,
>=20
> Ruediger
>=20
> Von: Karen Elisabeth Egede Nielsen [mailto:karen.nielsen@tieto.com]
> Gesendet: Freitag, 24. Juli 2015 10:28
> An: Geib, R=C3=BCdiger
> Cc: rtcweb@ietf.org<mailto:rtcweb@ietf.org>; tsvwg@ietf.org<mailto:tsvw=
g@ietf.org>; Michael Tuexen; fluffy@iii.ca<mailto:fluffy@iii.ca>
> Betreff: RE: [tsvwg] diffserv marking for rtcwb data channel
>=20
> Hi Ruediger,
>=20
> Thanks a lot.
>=20
> So I think that you=E2=80=99re saying that even if one made sure that d=
ifferent diff serv markings were
> split on different 5-tuples, this setting likely would be bleached at t=
he network.
>=20
> Is that correctly understood ?
>=20
> If so, then (it sounds to me =E2=80=93 not knowing too much about this)=
 that from a coupling congestion control perspective it
> would be preferable to  use SCTP scheduling prioritization (within one =
and the same association) for fine granular differentiation in between
> multiple data channels in between the same end-points rather than diffe=
rentiation by means of diff serv marking.
>=20
> When/if one get a more consistent and forceful ECN operation in AQMs an=
d SCTP end-points, perhaps the coupling congestion
> principles achieved by means of sctp stream scheduling get to be less i=
mportant.
> On the other hand, again as said before, having the one and the same, o=
r multiple, flow control (receiver buffer) contexts may be
> a strength (simplicity) or a weakness.
>=20
> BR, Karen
>=20
> From: Ruediger.Geib@telekom.de<mailto:Ruediger.Geib@telekom.de> [mailto=
:Ruediger.Geib@telekom.de<mailto:Ruediger.Geib@telekom.de>]
> Sent: Friday, July 24, 2015 10:00 AM
> To: karen.nielsen@tieto.com<mailto:karen.nielsen@tieto.com>
> Cc: rtcweb@ietf.org<mailto:rtcweb@ietf.org>; tsvwg@ietf.org<mailto:tsvw=
g@ietf.org>
> Subject: AW: [tsvwg] diffserv marking for rtcwb data channel
>=20
> Karen,
>=20
>       Finally there was a comment at the tsvwg meeting that the same 5-=
tuple with
>        different diffserv markings would not get through the network =E2=
=80=93 if that generally
>        is correct,  then the solution is very easy: one MUST use differ=
ent associations to
>        get port (5-tuple) differentiation.
>        I think that if this is correct, one also wants to ask what happ=
ens if the diffserv is
>        changed for the same 5-tuple =E2=80=93 i.e., is the network robu=
st to this change even if
>        it is not robust to concurrent usage of different markings for t=
he same 5-tuple.
>=20
> Talking about the policies of the company I work for only:
>=20
> Possibilities to enforce QoS policies at a provider BRAS or BNG termina=
ting wired
> consumer accesses are limited. QoS policies based on full 5 tuple are v=
ery
> conservative and hard to maintain. I wouldn=E2=80=99t expect full 5 tup=
le as basis for a QoS
> policy then. Trust in the correct information on which a BRAS/BNG can
> base QoS classification is an important issue. In any case, I=E2=80=99d=
 not expect
> more than one DSCP is assigned based on such a policy for the time
> being. I=E2=80=99m not aware of any actual requirement to support e.g. =
an AFn
> class by all three PHBs.
>=20
> At interconnection points, any unexpected DSCP is bleached. This must
> be done to protect backbone QoS production. QoS transport can be
> negotiated for a set of defined DSCPs.
>=20
> A study of the EU found DPI to be more common in wireless communication=
 than in wired communication. I=E2=80=99d not expect the QoS policies in =
that area to be more liberal than for fixed access networks for the time =
being.
>=20
> Regards,
>=20
> Ruediger
>=20
>=20
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org<mailto:rtcweb@ietf.org>
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


From nobody Mon Jul 27 11:54:59 2015
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From: <Ruediger.Geib@telekom.de>
To: <richard.vandet@kpn.com>, <david.black@emc.com>
Date: Mon, 27 Jul 2015 09:46:25 +0200
Thread-Topic: [rtcweb] [tsvwg] diffserv marking for rtcweb data channel
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Subject: Re: [rtcweb] [tsvwg] diffserv marking for rtcweb data channel
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Richard, David,

on the last mile of a consumer access, Deutsche Telekom operates QoS. To pr=
oduce QoS with flows having a high bandwidth as compared to the available r=
esources of a class, admission control is required. That means control plan=
e interaction with a service platform for all streams consuming QoS resourc=
es. This includes awareness of the access resources available per QoS class=
.

QoS interconnection with Deutsche Telekom's Voice platform is offered and o=
perational. A VPN is used. Available PHBs and DSCPs are specified. Also the=
 per flow signaling and payload properties are specified to some extent.

A VPN or an outer tunnel CoS / inner tunnel CoS scheme is useful if

-          no admission control or "live and let live" cooperation respecti=
vely between flows requesting QoS support in the last mile is intended.

-          the outer tunnel QoS may be Best Effort or another CoS like solu=
tion (a don't drop me first, but drop me before any QoS traffic gets droppe=
d approach).

-          Then the inner tunnel can carry any DSCP desired and the outer t=
unnel may consist of one or two markings. The latter may have a chance to b=
e carried end-to-end.

Another option may be to mark admission controlled traffic by DSCP a and pr=
oduce WebRTC with DSCP b and c in the  same queue, however making sure that=
 during congestion in that class all traffic without admission control (Web=
RTC)  gets dropped before traffic with admission control is dropped.

Regards,

Ruediger


Von: richard.vandet@kpn.com [mailto:richard.vandet@kpn.com]
Gesendet: Samstag, 25. Juli 2015 08:01
An: david.black@emc.com
Cc: Geib, R=FCdiger; karen.nielsen@tieto.com; rtcweb@ietf.org; tsvwg@ietf.o=
rg
Betreff: Re: [rtcweb] [tsvwg] diffserv marking for rtcweb data channel

Sometimes legislation prohibits the QoS model over open internet, all traff=
ic must be treated the same.

A VPN may be a solution, based on a SPs services model like Voice. Setting =
up a VPN on the fly once the connection is established may suit the needs o=
f a developer.

Richard

Op 24 jul. 2015 om 14:29 heeft Black, David <david.black@emc.com<mailto:dav=
id.black@emc.com>> het volgende geschreven:
<WG chair hat OFF>

> All DSCPs, which have not been negotiated between access and interconnect=
ion are possibly bleached, especially if a network
> provider operates QoS within the network.

"possibly" -> "probably" -> "almost certainly" depending on the network (YM=
MV).

> I'd like to read a little more about how WEBrtc figures out which QoS res=
ources are configured and which of these are available
> for such a flow at an access when required. How and where are the resourc=
es consumed by each WERrtc PHB controlled and
> how are these aligned with the QoS resources consumed by other services?

The short Web RTC answer to all of the above "it doesn't do that".  The DSC=
Ps are in effect QoS service requests that the various networks involved ma=
y deal with as they wish.  Bleaching is one possibility (in effect an "I do=
n't do that" response to the request).  Even in that case, there may be val=
ue in DSCP usage for forwarding by equipment between the browser and the in=
terface to the ISP's network (e.g., residential router/NAT/AP box).

In spite of all of the above, I have been told of good experiences with use=
 of CS2 for gaming and CS1 for lower effort (less than best effort), plus E=
F use for gaming was mentioned in the tsvwg meeting yesterday.

This is not the first time I've heard a request for a "what QoS services ar=
e available via which DSCPs?" interface/protocol.  I don't know of one, and=
 I'm not sure how to incentivize deployment if we designed one.

What is likely to happen is that if certain types of traffic favor specific=
 DSCPs, then operators who want to improve the user experience of those who=
 send/receive such traffic may put in QoS mechanisms for such traffic keyed=
 by those DSCPs at network ingress.  OTOH, a "tragedy of the commons" is po=
ssible here, e.g., the story in tsvwg about how a lot of non-voice traffic =
showing up marked as if it were voice resulted in the network operator remo=
ving that special treatment of (residential, I presume) voice traffic.

</WG chair hat OFF>

Thanks,
--David

From: tsvwg [mailto:tsvwg-bounces@ietf.org] On Behalf Of Ruediger.Geib@tele=
kom.de<mailto:Ruediger.Geib@telekom.de>
Sent: Friday, July 24, 2015 5:33 AM
To: karen.nielsen@tieto.com<mailto:karen.nielsen@tieto.com>
Cc: Michael.Tuexen@lurchi.franken.de<mailto:Michael.Tuexen@lurchi.franken.d=
e>; rtcweb@ietf.org<mailto:rtcweb@ietf.org>; tsvwg@ietf.org<mailto:tsvwg@ie=
tf.org>; fluffy@iii.ca<mailto:fluffy@iii.ca>
Subject: Re: [tsvwg] diffserv marking for rtcwb data channel

Hi Karen,

QoS  needs to be commercially negotiated at interconnection. If QoS should =
be available at a consumer access, the consumer division of a company must =
be involved in addition as well as the technical staff setting up and maint=
aining technical configurations. 11 PHBs for a single consumer service are =
a challenging requirement.

All DSCPs, which have not been negotiated between access and interconnectio=
n are possibly bleached, especially if a network provider operates QoS with=
in the network.

QoS to me means overprovisioning to ensure transport free of congestion. If=
 I see 11 PHBs for a single flow, I'm not sure if this assumes more than Co=
S (drop me last). I'd like to read a little more about how WEBrtc figures o=
ut which QoS resources are configured and which of these are available for =
such a flow at an access when required. How and where are the resources con=
sumed by each WERrtc PHB controlled and how are these aligned with the QoS =
resources consumed by other services?

Regards,

Ruediger

Von: Karen Elisabeth Egede Nielsen [mailto:karen.nielsen@tieto.com]
Gesendet: Freitag, 24. Juli 2015 10:28
An: Geib, R=FCdiger
Cc: rtcweb@ietf.org<mailto:rtcweb@ietf.org>; tsvwg@ietf.org<mailto:tsvwg@ie=
tf.org>; Michael Tuexen; fluffy@iii.ca<mailto:fluffy@iii.ca>
Betreff: RE: [tsvwg] diffserv marking for rtcwb data channel

Hi Ruediger,

Thanks a lot.

So I think that you're saying that even if one made sure that different dif=
f serv markings were
split on different 5-tuples, this setting likely would be bleached at the n=
etwork.

Is that correctly understood ?

If so, then (it sounds to me - not knowing too much about this) that from a=
 coupling congestion control perspective it
would be preferable to  use SCTP scheduling prioritization (within one and =
the same association) for fine granular differentiation in between
multiple data channels in between the same end-points rather than different=
iation by means of diff serv marking.

When/if one get a more consistent and forceful ECN operation in AQMs and SC=
TP end-points, perhaps the coupling congestion
principles achieved by means of sctp stream scheduling get to be less impor=
tant.
On the other hand, again as said before, having the one and the same, or mu=
ltiple, flow control (receiver buffer) contexts may be
a strength (simplicity) or a weakness.

BR, Karen

From: Ruediger.Geib@telekom.de<mailto:Ruediger.Geib@telekom.de> [mailto:Rue=
diger.Geib@telekom.de<mailto:Ruediger.Geib@telekom.de>]
Sent: Friday, July 24, 2015 10:00 AM
To: karen.nielsen@tieto.com<mailto:karen.nielsen@tieto.com>
Cc: rtcweb@ietf.org<mailto:rtcweb@ietf.org>; tsvwg@ietf.org<mailto:tsvwg@ie=
tf.org>
Subject: AW: [tsvwg] diffserv marking for rtcwb data channel

Karen,

      Finally there was a comment at the tsvwg meeting that the same 5-tupl=
e with
       different diffserv markings would not get through the network - if t=
hat generally
       is correct,  then the solution is very easy: one MUST use different =
associations to
       get port (5-tuple) differentiation.
       I think that if this is correct, one also wants to ask what happens =
if the diffserv is
       changed for the same 5-tuple - i.e., is the network robust to this c=
hange even if
       it is not robust to concurrent usage of different markings for the s=
ame 5-tuple.

Talking about the policies of the company I work for only:

Possibilities to enforce QoS policies at a provider BRAS or BNG terminating=
 wired
consumer accesses are limited. QoS policies based on full 5 tuple are very
conservative and hard to maintain. I wouldn't expect full 5 tuple as basis =
for a QoS
policy then. Trust in the correct information on which a BRAS/BNG can
base QoS classification is an important issue. In any case, I'd not expect
more than one DSCP is assigned based on such a policy for the time
being. I'm not aware of any actual requirement to support e.g. an AFn
class by all three PHBs.

At interconnection points, any unexpected DSCP is bleached. This must
be done to protect backbone QoS production. QoS transport can be
negotiated for a set of defined DSCPs.

A study of the EU found DPI to be more common in wireless communication tha=
n in wired communication. I'd not expect the QoS policies in that area to b=
e more liberal than for fixed access networks for the time being.

Regards,

Ruediger




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

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--></style><!--[if gte mso 9]><xml>
<o:shapedefaults v:ext=3D"edit" spidmax=3D"1026" />
</xml><![endif]--><!--[if gte mso 9]><xml>
<o:shapelayout v:ext=3D"edit">
<o:idmap v:ext=3D"edit" data=3D"1" />
</o:shapelayout></xml><![endif]--></head><body lang=3DDE link=3Dblue vlink=
=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal><span style=3D'col=
or:#1F497D'>Richard, David,<o:p></o:p></span></p><p class=3DMsoNormal><span=
 style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><=
span lang=3DEN-US style=3D'color:#1F497D'>on the last mile of a consumer ac=
cess, Deutsche Telekom operates QoS. To produce QoS with flows having a hig=
h bandwidth as compared to the available resources of a class, admission co=
ntrol is required. That means control plane interaction with a service plat=
form for all streams consuming QoS resources. This includes awareness of th=
e access resources available per QoS class.<o:p></o:p></span></p><p class=
=3DMsoNormal><span lang=3DEN-US style=3D'color:#1F497D'><o:p>&nbsp;</o:p></=
span></p><p class=3DMsoNormal><span lang=3DEN-US style=3D'color:#1F497D'>Qo=
S interconnection with Deutsche Telekom&#8217;s Voice platform is offered a=
nd operational. A VPN is used. Available PHBs and DSCPs are specified. Also=
 the per flow signaling and payload properties are specified to some extent=
. <o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US style=3D'co=
lor:#1F497D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3D=
EN-US style=3D'color:#1F497D'>A VPN or an outer tunnel CoS / inner tunnel C=
oS scheme is useful if<o:p></o:p></span></p><p class=3DMsoListParagraph sty=
le=3D'text-indent:-18.0pt;mso-list:l0 level1 lfo1'><![if !supportLists]><sp=
an lang=3DEN-US style=3D'color:#1F497D'><span style=3D'mso-list:Ignore'>-<s=
pan style=3D'font:7.0pt "Times New Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp; </span></span></span><![endif]><span lang=3DEN-US st=
yle=3D'color:#1F497D'>no admission control or &#8220;live and let live&#822=
1; cooperation respectively between flows requesting QoS support in the las=
t mile is intended.<o:p></o:p></span></p><p class=3DMsoListParagraph style=
=3D'text-indent:-18.0pt;mso-list:l0 level1 lfo1'><![if !supportLists]><span=
 lang=3DEN-US style=3D'color:#1F497D'><span style=3D'mso-list:Ignore'>-<spa=
n style=3D'font:7.0pt "Times New Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp; </span></span></span><![endif]><span lang=3DEN-US styl=
e=3D'color:#1F497D'>the outer tunnel QoS may be Best Effort or another CoS =
like solution (a don&#8217;t drop me first, but drop me before any QoS traf=
fic gets dropped approach).<o:p></o:p></span></p><p class=3DMsoListParagrap=
h style=3D'text-indent:-18.0pt;mso-list:l0 level1 lfo1'><![if !supportLists=
]><span lang=3DEN-US style=3D'color:#1F497D'><span style=3D'mso-list:Ignore=
'>-<span style=3D'font:7.0pt "Times New Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp; </span></span></span><![endif]><span lang=3DEN-=
US style=3D'color:#1F497D'>Then the inner tunnel can carry any DSCP desired=
 and the outer tunnel may consist of one or two markings. The latter may ha=
ve a chance to be carried end-to-end.<o:p></o:p></span></p><p class=3DMsoNo=
rmal><span lang=3DEN-US style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p=
><p class=3DMsoNormal><span lang=3DEN-US style=3D'color:#1F497D'>Another op=
tion may be to mark admission controlled traffic by DSCP a and produce WebR=
TC with DSCP b and c in the=A0 same queue, however making sure that during =
congestion in that class all traffic without admission control (WebRTC) =A0=
gets dropped before traffic with admission control is dropped. <o:p></o:p><=
/span></p><p class=3DMsoNormal><span lang=3DEN-US style=3D'color:#1F497D'><=
o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US style=3D=
'color:#1F497D'>Regards,<o:p></o:p></span></p><p class=3DMsoNormal><span la=
ng=3DEN-US style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p class=3DM=
soNormal><span lang=3DEN-US style=3D'color:#1F497D'>Ruediger<o:p></o:p></sp=
an></p><p class=3DMsoNormal><span lang=3DEN-US style=3D'color:#1F497D'><o:p=
>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US style=3D'co=
lor:#1F497D'><o:p>&nbsp;</o:p></span></p><div><div style=3D'border:none;bor=
der-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm'><p class=3DMsoNormal=
><b><span style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Von:=
</span></b><span style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif=
"'> richard.vandet@kpn.com [mailto:richard.vandet@kpn.com] <br><b>Gesendet:=
</b> Samstag, 25. Juli 2015 08:01<br><b>An:</b> david.black@emc.com<br><b>C=
c:</b> Geib, R=FCdiger; karen.nielsen@tieto.com; rtcweb@ietf.org; tsvwg@iet=
f.org<br><b>Betreff:</b> Re: [rtcweb] [tsvwg] diffserv marking for rtcweb d=
ata channel<o:p></o:p></span></p></div></div><p class=3DMsoNormal><o:p>&nbs=
p;</o:p></p><div><p class=3DMsoNormal>Sometimes legislation prohibits the Q=
oS model over open internet, all traffic must be treated the same.<br><br>A=
 VPN may be a solution, based on a SPs services model like Voice. Setting u=
p a VPN on the fly once the connection is established may suit the needs of=
 a developer.<o:p></o:p></p></div><div><div><p class=3DMsoNormal><o:p>&nbsp=
;</o:p></p></div><div><p class=3DMsoNormal>Richard<o:p></o:p></p></div></di=
v><div><p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><br>Op 24 jul. 2=
015 om 14:29 heeft Black, David &lt;<a href=3D"mailto:david.black@emc.com">=
david.black@emc.com</a>&gt; het volgende geschreven:<o:p></o:p></p></div><b=
lockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><div><p class=3DMs=
oNormal><span style=3D'font-size:10.0pt;font-family:"Courier New";color:bla=
ck'>&lt;WG chair hat OFF&gt;</span><o:p></o:p></p><p class=3DMsoNormal><spa=
n style=3D'font-size:10.0pt;font-family:"Courier New";color:black'>&nbsp;</=
span><o:p></o:p></p><p class=3DMsoNormal><span style=3D'font-size:10.0pt;fo=
nt-family:"Courier New";color:black'>&gt; </span><span style=3D'color:#1F49=
7D'>All DSCPs, which have not been negotiated between access and interconne=
ction are possibly bleached, especially if a network</span><o:p></o:p></p><=
p class=3DMsoNormal><span style=3D'color:#1F497D'>&gt; provider operates Qo=
S within the network.</span><o:p></o:p></p><p class=3DMsoNormal><span style=
=3D'font-size:10.0pt;font-family:"Courier New";color:black'>&nbsp;</span><o=
:p></o:p></p><p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-fami=
ly:"Courier New";color:black'>&#8220;possibly&#8221; -&gt; &#8220;probably&=
#8221; -&gt; &#8220;almost certainly&#8221; depending on the network (YMMV)=
.</span><o:p></o:p></p><p class=3DMsoNormal><span style=3D'font-size:10.0pt=
;font-family:"Courier New";color:black'>&nbsp;</span><o:p></o:p></p><p clas=
s=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier New";col=
or:black'>&gt;</span><span style=3D'color:#1F497D'> I&#8217;d like to read =
a little more about how WEBrtc figures out which QoS resources are configur=
ed and which of these are available</span><o:p></o:p></p><p class=3DMsoNorm=
al><span style=3D'color:#1F497D'>&gt; for such a flow at an access when req=
uired. How and where are the resources consumed by each WERrtc PHB controll=
ed and</span><o:p></o:p></p><p class=3DMsoNormal><span style=3D'color:#1F49=
7D'>&gt; how are these aligned with the QoS resources consumed by other ser=
vices?</span><o:p></o:p></p><p class=3DMsoNormal><span style=3D'font-size:1=
0.0pt;font-family:"Courier New";color:black'>&nbsp;</span><o:p></o:p></p><p=
 class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier New=
";color:black'>The short Web RTC answer to all of the above &#8220;it doesn=
&#8217;t do that&#8221;.&nbsp; The DSCPs are in effect QoS service requests=
 that the various networks involved may deal with as they wish.&nbsp; Bleac=
hing is one possibility (in effect an &#8220;I don&#8217;t do that&#8221; r=
esponse to the request).&nbsp; Even in that case, there may be value in DSC=
P usage for forwarding by equipment between the browser and the interface t=
o the ISP&#8217;s network (e.g., residential router/NAT/AP box).</span><o:p=
></o:p></p><p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family=
:"Courier New";color:black'>&nbsp;</span><o:p></o:p></p><p class=3DMsoNorma=
l><span style=3D'font-size:10.0pt;font-family:"Courier New";color:black'>In=
 spite of all of the above, I have been told of good experiences with use o=
f CS2 for gaming and CS1 for lower effort (less than best effort), plus EF =
use for gaming was mentioned in the tsvwg meeting yesterday.</span><o:p></o=
:p></p><p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Co=
urier New";color:black'>&nbsp;</span><o:p></o:p></p><p class=3DMsoNormal><s=
pan style=3D'font-size:10.0pt;font-family:"Courier New";color:black'>This i=
s not the first time I&#8217;ve heard a request for a &#8220;what QoS servi=
ces are available via which DSCPs?&#8221; interface/protocol.&nbsp; I don&#=
8217;t know of one, and I&#8217;m not sure how to incentivize deployment if=
 we designed one.</span><o:p></o:p></p><p class=3DMsoNormal><span style=3D'=
font-size:10.0pt;font-family:"Courier New";color:black'>&nbsp;</span><o:p><=
/o:p></p><p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"=
Courier New";color:black'>What is likely to happen is that if certain types=
 of traffic favor specific DSCPs, then operators who want to improve the us=
er experience of those who send/receive such traffic may put in QoS mechani=
sms for such traffic keyed by those DSCPs at network ingress.&nbsp; OTOH, a=
 &#8220;tragedy of the commons&#8221; is possible here, e.g., the story in =
tsvwg about how a lot of non-voice traffic showing up marked as if it were =
voice resulted in the network operator removing that special treatment of (=
residential, I presume) voice traffic.</span><o:p></o:p></p><p class=3DMsoN=
ormal><span style=3D'font-size:10.0pt;font-family:"Courier New";color:black=
'>&nbsp;</span><o:p></o:p></p><p class=3DMsoNormal><span style=3D'font-size=
:10.0pt;font-family:"Courier New";color:black'>&lt;/WG chair hat OFF&gt;</s=
pan><o:p></o:p></p><p class=3DMsoNormal><span style=3D'font-size:10.0pt;fon=
t-family:"Courier New";color:black'>&nbsp;</span><o:p></o:p></p><div><div><=
p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier Ne=
w";color:black'>Thanks,<br>--David</span><o:p></o:p></p></div></div><p clas=
s=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier New";col=
or:black'>&nbsp;</span><o:p></o:p></p><div style=3D'border:none;border-left=
:solid blue 1.5pt;padding:0cm 0cm 0cm 4.0pt'><div><div style=3D'border:none=
;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm'><p class=3DMsoNo=
rmal><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"'> tsvwg [<a href=3D"mailto:tsvwg-bounces@ietf.org">mailto:tsvwg-boun=
ces@ietf.org</a>] <b>On Behalf Of </b><a href=3D"mailto:Ruediger.Geib@telek=
om.de">Ruediger.Geib@telekom.de</a><br><b>Sent:</b> Friday, July 24, 2015 5=
:33 AM<br><b>To:</b> <a href=3D"mailto:karen.nielsen@tieto.com">karen.niels=
en@tieto.com</a><br><b>Cc:</b> <a href=3D"mailto:Michael.Tuexen@lurchi.fran=
ken.de">Michael.Tuexen@lurchi.franken.de</a>; <a href=3D"mailto:rtcweb@ietf=
.org">rtcweb@ietf.org</a>; <a href=3D"mailto:tsvwg@ietf.org">tsvwg@ietf.org=
</a>; <a href=3D"mailto:fluffy@iii.ca">fluffy@iii.ca</a><br><b>Subject:</b>=
 Re: [tsvwg] diffserv marking for rtcwb data channel</span><o:p></o:p></p><=
/div></div><p class=3DMsoNormal>&nbsp;<o:p></o:p></p><p class=3DMsoNormal><=
span style=3D'color:#1F497D'>Hi Karen,</span><o:p></o:p></p><p class=3DMsoN=
ormal><span style=3D'color:#1F497D'>&nbsp;</span><o:p></o:p></p><p class=3D=
MsoNormal><span style=3D'color:#1F497D'>QoS&nbsp; needs to be commercially =
negotiated at interconnection. If QoS should be available at a consumer acc=
ess, the consumer division of a company must be involved in addition as wel=
l as the technical staff setting up and maintaining technical configuration=
s. 11 PHBs for a single consumer service are a challenging requirement.</sp=
an><o:p></o:p></p><p class=3DMsoNormal><span style=3D'color:#1F497D'>&nbsp;=
</span><o:p></o:p></p><p class=3DMsoNormal><span style=3D'color:#1F497D'>Al=
l DSCPs, which have not been negotiated between access and interconnection =
are possibly bleached, especially if a network provider operates QoS within=
 the network.</span><o:p></o:p></p><p class=3DMsoNormal><span style=3D'colo=
r:#1F497D'>&nbsp;</span><o:p></o:p></p><p class=3DMsoNormal><span style=3D'=
color:#1F497D'>QoS to me means overprovisioning to ensure transport free of=
 congestion. If I see 11 PHBs for a single flow, I&#8217;m not sure if this=
 assumes more than CoS (drop me last). I&#8217;d like to read a little more=
 about how WEBrtc figures out which QoS resources are configured and which =
of these are available for such a flow at an access when required. How and =
where are the resources consumed by each WERrtc PHB controlled and how are =
these aligned with the QoS resources consumed by other services?</span><o:p=
></o:p></p><p class=3DMsoNormal><span style=3D'color:#1F497D'>&nbsp;</span>=
<o:p></o:p></p><p class=3DMsoNormal><span style=3D'color:#1F497D'>Regards,<=
/span><o:p></o:p></p><p class=3DMsoNormal><span style=3D'color:#1F497D'>&nb=
sp;</span><o:p></o:p></p><p class=3DMsoNormal><span style=3D'color:#1F497D'=
>Ruediger</span><o:p></o:p></p><p class=3DMsoNormal><span style=3D'color:#1=
F497D'>&nbsp;</span><o:p></o:p></p><div><div style=3D'border:none;border-to=
p:solid #B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm'><p class=3DMsoNormal><b><s=
pan style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Von:</span=
></b><span style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Ka=
ren Elisabeth Egede Nielsen [<a href=3D"mailto:karen.nielsen@tieto.com">mai=
lto:karen.nielsen@tieto.com</a>] <br><b>Gesendet:</b> Freitag, 24. Juli 201=
5 10:28<br><b>An:</b> Geib, R=FCdiger<br><b>Cc:</b> <a href=3D"mailto:rtcwe=
b@ietf.org">rtcweb@ietf.org</a>; <a href=3D"mailto:tsvwg@ietf.org">tsvwg@ie=
tf.org</a>; Michael Tuexen; <a href=3D"mailto:fluffy@iii.ca">fluffy@iii.ca<=
/a><br><b>Betreff:</b> RE: [tsvwg] diffserv marking for rtcwb data channel<=
/span><o:p></o:p></p></div></div><p class=3DMsoNormal>&nbsp;<o:p></o:p></p>=
<p class=3DMsoNormal><span style=3D'color:#1F497D'>Hi Ruediger,</span><o:p>=
</o:p></p><p class=3DMsoNormal><span style=3D'color:#1F497D'>&nbsp;</span><=
o:p></o:p></p><p class=3DMsoNormal><span style=3D'color:#1F497D'>Thanks a l=
ot.</span><o:p></o:p></p><p class=3DMsoNormal><span style=3D'color:#1F497D'=
>&nbsp;</span><o:p></o:p></p><p class=3DMsoNormal><span style=3D'color:#1F4=
97D'>So I think that you&#8217;re saying that even if one made sure that di=
fferent diff serv markings were</span><o:p></o:p></p><p class=3DMsoNormal><=
span style=3D'color:#1F497D'>split on different 5-tuples, this setting like=
ly would be bleached at the network.</span><o:p></o:p></p><p class=3DMsoNor=
mal><span style=3D'color:#1F497D'>&nbsp;</span><o:p></o:p></p><p class=3DMs=
oNormal><span style=3D'color:#1F497D'>Is that correctly understood ?</span>=
<o:p></o:p></p><p class=3DMsoNormal><span style=3D'color:#1F497D'>&nbsp;</s=
pan><o:p></o:p></p><p class=3DMsoNormal><span style=3D'color:#1F497D'>If so=
, then (it sounds to me &#8211; not knowing too much about this) that from =
a coupling congestion control perspective it </span><o:p></o:p></p><p class=
=3DMsoNormal><span style=3D'color:#1F497D'>would be preferable to&nbsp; use=
 SCTP scheduling prioritization (within one and the same association) for f=
ine granular differentiation in between</span><o:p></o:p></p><p class=3DMso=
Normal><span style=3D'color:#1F497D'>multiple data channels in between the =
same end-points rather than differentiation by means of diff serv marking.<=
/span><o:p></o:p></p><p class=3DMsoNormal><span style=3D'color:#1F497D'>&nb=
sp;</span><o:p></o:p></p><p class=3DMsoNormal><span style=3D'color:#1F497D'=
>When/if one get a more consistent and forceful ECN operation in AQMs and S=
CTP end-points, perhaps the coupling congestion</span><o:p></o:p></p><p cla=
ss=3DMsoNormal><span style=3D'color:#1F497D'>principles achieved by means o=
f sctp stream scheduling get to be less important. </span><o:p></o:p></p><p=
 class=3DMsoNormal><span style=3D'color:#1F497D'>On the other hand, again a=
s said before, having the one and the same, or multiple, flow control (rece=
iver buffer) contexts may be</span><o:p></o:p></p><p class=3DMsoNormal><spa=
n style=3D'color:#1F497D'>a strength (simplicity) or a weakness. &nbsp;&nbs=
p;&nbsp;</span><o:p></o:p></p><p class=3DMsoNormal><span style=3D'color:#1F=
497D'>&nbsp;</span><o:p></o:p></p><p class=3DMsoNormal><span style=3D'color=
:#1F497D'>BR, Karen </span><o:p></o:p></p><p class=3DMsoNormal><span style=
=3D'color:#1F497D'>&nbsp;</span><o:p></o:p></p><div style=3D'border:none;bo=
rder-left:solid blue 1.5pt;padding:0cm 0cm 0cm 4.0pt'><div><div style=3D'bo=
rder:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm'><p clas=
s=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:"Tahom=
a","sans-serif"'> <a href=3D"mailto:Ruediger.Geib@telekom.de">Ruediger.Geib=
@telekom.de</a> [mailto:<a href=3D"mailto:Ruediger.Geib@telekom.de">Ruedige=
r.Geib@telekom.de</a>] <br><b>Sent:</b> Friday, July 24, 2015 10:00 AM<br><=
b>To:</b> <a href=3D"mailto:karen.nielsen@tieto.com">karen.nielsen@tieto.co=
m</a><br><b>Cc:</b> <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>;=
 <a href=3D"mailto:tsvwg@ietf.org">tsvwg@ietf.org</a><br><b>Subject:</b> AW=
: [tsvwg] diffserv marking for rtcwb data channel</span><o:p></o:p></p></di=
v></div><p class=3DMsoNormal>&nbsp;<o:p></o:p></p><p class=3DMsoNormal><spa=
n style=3D'color:#1F497D'>Karen,</span><o:p></o:p></p><p class=3DMsoNormal>=
<span style=3D'color:#1F497D'>&nbsp;</span><o:p></o:p></p><p class=3DMsoNor=
mal><b><span style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";co=
lor:#1F497D'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span></b>Finally there was a =
comment at the tsvwg meeting that the same 5-tuple with <o:p></o:p></p><p c=
lass=3DMsoNormal><span style=3D'color:#1F497D'>&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;</span>different diffserv markings would not get<span style=
=3D'color:#1F497D'> </span>through the network &#8211; if that generally <o=
:p></o:p></p><p class=3DMsoNormal><span style=3D'color:#1F497D'>&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</span>is correct, &nbsp;then the solution i=
s very easy: one MUST use different associations to <o:p></o:p></p><p class=
=3DMsoNormal><span style=3D'color:#1F497D'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;</span>get port (5-tuple) differentiation.<o:p></o:p></p><p class=
=3DMsoNormal><span style=3D'color:#1F497D'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp; </span>I think that if this is correct, one also wants to ask what hap=
pens if the diffserv is <o:p></o:p></p><p class=3DMsoNormal><span style=3D'=
color:#1F497D'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</span>changed for=
 the same 5-tuple &#8211; i.e., is the network robust to this change even i=
f <o:p></o:p></p><p class=3DMsoNormal><span style=3D'color:#1F497D'>&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</span>it is not robust to concurrent us=
age of different markings for the same 5-tuple.<o:p></o:p></p><p class=3DMs=
oNormal>&nbsp;<o:p></o:p></p><p class=3DMsoNormal><span style=3D'color:#1F4=
97D'>Talking about the policies of the company I work for only:</span><o:p>=
</o:p></p><p class=3DMsoNormal><span style=3D'color:#1F497D'>&nbsp;</span><=
o:p></o:p></p><p class=3DMsoNormal><span style=3D'color:#1F497D'>Possibilit=
ies to enforce QoS policies at a provider BRAS or</span>&nbsp;<span style=
=3D'color:#1F497D'>BNG terminating wired </span><o:p></o:p></p><p class=3DM=
soNormal><span style=3D'color:#1F497D'>consumer accesses are limited. QoS p=
olicies based on full 5 tuple are very </span><o:p></o:p></p><p class=3DMso=
Normal><span style=3D'color:#1F497D'>conservative and hard to maintain. I w=
ouldn&#8217;t expect full 5 tuple as basis for a QoS </span><o:p></o:p></p>=
<p class=3DMsoNormal><span style=3D'color:#1F497D'>policy then. Trust in th=
e correct information on which a BRAS/BNG can </span><o:p></o:p></p><p clas=
s=3DMsoNormal><span style=3D'color:#1F497D'>base QoS classification is an i=
mportant issue. In any case, I&#8217;d not expect </span><o:p></o:p></p><p =
class=3DMsoNormal><span style=3D'color:#1F497D'>more than one DSCP is assig=
ned based on such a policy for the time </span><o:p></o:p></p><p class=3DMs=
oNormal><span style=3D'color:#1F497D'>being. I&#8217;m not aware of any act=
ual requirement to support e.g. an AFn </span><o:p></o:p></p><p class=3DMso=
Normal><span style=3D'color:#1F497D'>class by all three PHBs.</span><o:p></=
o:p></p><p class=3DMsoNormal><span style=3D'color:#1F497D'>&nbsp;</span><o:=
p></o:p></p><p class=3DMsoNormal><span style=3D'color:#1F497D'>At interconn=
ection points, any unexpected DSCP is bleached. This must </span><o:p></o:p=
></p><p class=3DMsoNormal><span style=3D'color:#1F497D'>be done to protect =
backbone QoS production. QoS transport can be </span><o:p></o:p></p><p clas=
s=3DMsoNormal><span style=3D'color:#1F497D'>negotiated for a set of defined=
 DSCPs. </span><o:p></o:p></p><p class=3DMsoNormal><span style=3D'color:#1F=
497D'>&nbsp;</span><o:p></o:p></p><p class=3DMsoNormal><span style=3D'color=
:#1F497D'>A study of the EU found DPI to be more common in wireless communi=
cation than in wired communication. I&#8217;d not expect the QoS policies i=
n that area to be more liberal than for fixed access networks for the time =
being.</span><o:p></o:p></p><p class=3DMsoNormal><span style=3D'color:#1F49=
7D'>&nbsp;</span><o:p></o:p></p><p class=3DMsoNormal><span style=3D'color:#=
1F497D'>Regards,</span><o:p></o:p></p><p class=3DMsoNormal><span style=3D'c=
olor:#1F497D'>&nbsp;</span><o:p></o:p></p><p class=3DMsoNormal><span style=
=3D'color:#1F497D'>Ruediger</span><o:p></o:p></p><p class=3DMsoNormal><span=
 style=3D'color:#1F497D'>&nbsp;</span><o:p></o:p></p><p class=3DMsoNormal><=
span style=3D'color:#1F497D'>&nbsp;</span><o:p></o:p></p><p class=3DMsoNorm=
al style=3D'margin-left:18.0pt'>&nbsp;<o:p></o:p></p><p class=3DMsoNormal>&=
nbsp;<o:p></o:p></p></div></div></div></blockquote><blockquote style=3D'mar=
gin-top:5.0pt;margin-bottom:5.0pt'><div><p class=3DMsoNormal><span style=3D=
'font-size:12.0pt;font-family:"Times New Roman","serif"'>__________________=
_____________________________<br>rtcweb mailing list<br><a href=3D"mailto:r=
tcweb@ietf.org">rtcweb@ietf.org</a><br><a href=3D"https://www.ietf.org/mail=
man/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a><o:p><=
/o:p></span></p></div></blockquote></div></body></html>=

--_000_CA7A7C64CC4ADB458B74477EA99DF6F50529674941HE111643EMEA1_--


From nobody Mon Jul 27 11:55:01 2015
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Date: Mon, 27 Jul 2015 23:48:38 +1200
From: Brian E Carpenter <brian.e.carpenter@gmail.com>
Organization: University of Auckland
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Subject: Re: [rtcweb] [tsvwg]   diffserv marking for rtcweb data channel
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Ruediger brings out an important point.

Diffserv can only work properly if there is admission control
for every traffic class that claims to offer definite guarantees.

That isn't because the people who defined diffserv were trying
to be annoying. It's because of the physics and mathematics of
queuing systems.

Regards
   Brian

On 27/07/2015 19:46, Ruediger.Geib@telekom.de wrote:
> Richard, David,
>=20
> on the last mile of a consumer access, Deutsche Telekom operates QoS. T=
o produce QoS with flows having a high bandwidth as compared to the avail=
able resources of a class, admission control is required. That means cont=
rol plane interaction with a service platform for all streams consuming Q=
oS resources. This includes awareness of the access resources available p=
er QoS class.
>=20
> QoS interconnection with Deutsche Telekom's Voice platform is offered a=
nd operational. A VPN is used. Available PHBs and DSCPs are specified. Al=
so the per flow signaling and payload properties are specified to some ex=
tent.
>=20
> A VPN or an outer tunnel CoS / inner tunnel CoS scheme is useful if
>=20
> -          no admission control or "live and let live" cooperation resp=
ectively between flows requesting QoS support in the last mile is intende=
d.
>=20
> -          the outer tunnel QoS may be Best Effort or another CoS like =
solution (a don't drop me first, but drop me before any QoS traffic gets =
dropped approach).
>=20
> -          Then the inner tunnel can carry any DSCP desired and the out=
er tunnel may consist of one or two markings. The latter may have a chanc=
e to be carried end-to-end.
>=20
> Another option may be to mark admission controlled traffic by DSCP a an=
d produce WebRTC with DSCP b and c in the  same queue, however making sur=
e that during congestion in that class all traffic without admission cont=
rol (WebRTC)  gets dropped before traffic with admission control is dropp=
ed.
>=20
> Regards,
>=20
> Ruediger
>=20
>=20
> Von: richard.vandet@kpn.com [mailto:richard.vandet@kpn.com]
> Gesendet: Samstag, 25. Juli 2015 08:01
> An: david.black@emc.com
> Cc: Geib, R=C3=BCdiger; karen.nielsen@tieto.com; rtcweb@ietf.org; tsvwg=
@ietf.org
> Betreff: Re: [rtcweb] [tsvwg] diffserv marking for rtcweb data channel
>=20
> Sometimes legislation prohibits the QoS model over open internet, all t=
raffic must be treated the same.
>=20
> A VPN may be a solution, based on a SPs services model like Voice. Sett=
ing up a VPN on the fly once the connection is established may suit the n=
eeds of a developer.
>=20
> Richard
>=20
> Op 24 jul. 2015 om 14:29 heeft Black, David <david.black@emc.com<mailto=
:david.black@emc.com>> het volgende geschreven:
> <WG chair hat OFF>
>=20
>> All DSCPs, which have not been negotiated between access and interconn=
ection are possibly bleached, especially if a network
>> provider operates QoS within the network.
>=20
> "possibly" -> "probably" -> "almost certainly" depending on the network=
 (YMMV).
>=20
>> I'd like to read a little more about how WEBrtc figures out which QoS =
resources are configured and which of these are available
>> for such a flow at an access when required. How and where are the reso=
urces consumed by each WERrtc PHB controlled and
>> how are these aligned with the QoS resources consumed by other service=
s?
>=20
> The short Web RTC answer to all of the above "it doesn't do that".  The=
 DSCPs are in effect QoS service requests that the various networks invol=
ved may deal with as they wish.  Bleaching is one possibility (in effect =
an "I don't do that" response to the request).  Even in that case, there =
may be value in DSCP usage for forwarding by equipment between the browse=
r and the interface to the ISP's network (e.g., residential router/NAT/AP=
 box).
>=20
> In spite of all of the above, I have been told of good experiences with=
 use of CS2 for gaming and CS1 for lower effort (less than best effort), =
plus EF use for gaming was mentioned in the tsvwg meeting yesterday.
>=20
> This is not the first time I've heard a request for a "what QoS service=
s are available via which DSCPs?" interface/protocol.  I don't know of on=
e, and I'm not sure how to incentivize deployment if we designed one.
>=20
> What is likely to happen is that if certain types of traffic favor spec=
ific DSCPs, then operators who want to improve the user experience of tho=
se who send/receive such traffic may put in QoS mechanisms for such traff=
ic keyed by those DSCPs at network ingress.  OTOH, a "tragedy of the comm=
ons" is possible here, e.g., the story in tsvwg about how a lot of non-vo=
ice traffic showing up marked as if it were voice resulted in the network=
 operator removing that special treatment of (residential, I presume) voi=
ce traffic.
>=20
> </WG chair hat OFF>
>=20
> Thanks,
> --David
>=20
> From: tsvwg [mailto:tsvwg-bounces@ietf.org] On Behalf Of Ruediger.Geib@=
telekom.de<mailto:Ruediger.Geib@telekom.de>
> Sent: Friday, July 24, 2015 5:33 AM
> To: karen.nielsen@tieto.com<mailto:karen.nielsen@tieto.com>
> Cc: Michael.Tuexen@lurchi.franken.de<mailto:Michael.Tuexen@lurchi.frank=
en.de>; rtcweb@ietf.org<mailto:rtcweb@ietf.org>; tsvwg@ietf.org<mailto:ts=
vwg@ietf.org>; fluffy@iii.ca<mailto:fluffy@iii.ca>
> Subject: Re: [tsvwg] diffserv marking for rtcwb data channel
>=20
> Hi Karen,
>=20
> QoS  needs to be commercially negotiated at interconnection. If QoS sho=
uld be available at a consumer access, the consumer division of a company=
 must be involved in addition as well as the technical staff setting up a=
nd maintaining technical configurations. 11 PHBs for a single consumer se=
rvice are a challenging requirement.
>=20
> All DSCPs, which have not been negotiated between access and interconne=
ction are possibly bleached, especially if a network provider operates Qo=
S within the network.
>=20
> QoS to me means overprovisioning to ensure transport free of congestion=
=2E If I see 11 PHBs for a single flow, I'm not sure if this assumes more=
 than CoS (drop me last). I'd like to read a little more about how WEBrtc=
 figures out which QoS resources are configured and which of these are av=
ailable for such a flow at an access when required. How and where are the=
 resources consumed by each WERrtc PHB controlled and how are these align=
ed with the QoS resources consumed by other services?
>=20
> Regards,
>=20
> Ruediger
>=20
> Von: Karen Elisabeth Egede Nielsen [mailto:karen.nielsen@tieto.com]
> Gesendet: Freitag, 24. Juli 2015 10:28
> An: Geib, R=C3=BCdiger
> Cc: rtcweb@ietf.org<mailto:rtcweb@ietf.org>; tsvwg@ietf.org<mailto:tsvw=
g@ietf.org>; Michael Tuexen; fluffy@iii.ca<mailto:fluffy@iii.ca>
> Betreff: RE: [tsvwg] diffserv marking for rtcwb data channel
>=20
> Hi Ruediger,
>=20
> Thanks a lot.
>=20
> So I think that you're saying that even if one made sure that different=
 diff serv markings were
> split on different 5-tuples, this setting likely would be bleached at t=
he network.
>=20
> Is that correctly understood ?
>=20
> If so, then (it sounds to me - not knowing too much about this) that fr=
om a coupling congestion control perspective it
> would be preferable to  use SCTP scheduling prioritization (within one =
and the same association) for fine granular differentiation in between
> multiple data channels in between the same end-points rather than diffe=
rentiation by means of diff serv marking.
>=20
> When/if one get a more consistent and forceful ECN operation in AQMs an=
d SCTP end-points, perhaps the coupling congestion
> principles achieved by means of sctp stream scheduling get to be less i=
mportant.
> On the other hand, again as said before, having the one and the same, o=
r multiple, flow control (receiver buffer) contexts may be
> a strength (simplicity) or a weakness.
>=20
> BR, Karen
>=20
> From: Ruediger.Geib@telekom.de<mailto:Ruediger.Geib@telekom.de> [mailto=
:Ruediger.Geib@telekom.de<mailto:Ruediger.Geib@telekom.de>]
> Sent: Friday, July 24, 2015 10:00 AM
> To: karen.nielsen@tieto.com<mailto:karen.nielsen@tieto.com>
> Cc: rtcweb@ietf.org<mailto:rtcweb@ietf.org>; tsvwg@ietf.org<mailto:tsvw=
g@ietf.org>
> Subject: AW: [tsvwg] diffserv marking for rtcwb data channel
>=20
> Karen,
>=20
>       Finally there was a comment at the tsvwg meeting that the same 5-=
tuple with
>        different diffserv markings would not get through the network - =
if that generally
>        is correct,  then the solution is very easy: one MUST use differ=
ent associations to
>        get port (5-tuple) differentiation.
>        I think that if this is correct, one also wants to ask what happ=
ens if the diffserv is
>        changed for the same 5-tuple - i.e., is the network robust to th=
is change even if
>        it is not robust to concurrent usage of different markings for t=
he same 5-tuple.
>=20
> Talking about the policies of the company I work for only:
>=20
> Possibilities to enforce QoS policies at a provider BRAS or BNG termina=
ting wired
> consumer accesses are limited. QoS policies based on full 5 tuple are v=
ery
> conservative and hard to maintain. I wouldn't expect full 5 tuple as ba=
sis for a QoS
> policy then. Trust in the correct information on which a BRAS/BNG can
> base QoS classification is an important issue. In any case, I'd not exp=
ect
> more than one DSCP is assigned based on such a policy for the time
> being. I'm not aware of any actual requirement to support e.g. an AFn
> class by all three PHBs.
>=20
> At interconnection points, any unexpected DSCP is bleached. This must
> be done to protect backbone QoS production. QoS transport can be
> negotiated for a set of defined DSCPs.
>=20
> A study of the EU found DPI to be more common in wireless communication=
 than in wired communication. I'd not expect the QoS policies in that are=
a to be more liberal than for fixed access networks for the time being.
>=20
> Regards,
>=20
> Ruediger
>=20
>=20
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org<mailto:rtcweb@ietf.org>
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


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Date: Mon, 27 Jul 2015 12:02:01 -0700
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From: Ted Hardie <ted.ietf@gmail.com>
To: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
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Subject: Re: [rtcweb] Please require user consent for data channels
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On Mon, Jul 20, 2015 at 4:43 AM, I=C3=B1aki Baz Castillo <ibc@aliax.net> wr=
ote:

> 2015-07-20 13:37 GMT+02:00 Tim Panton <thp@westhawk.co.uk>:
>
>> E.g. I=E2=80=99m at home and my chromebook pixel (or firefox tablet) is =
on wifi,
>> but I=E2=80=99ve left LTE enabled.
>> I (or the OS) is configured to prefer wifi wen available - but it happen=
s
>> that for a specific peer LTE completes first.
>> So now my video call goes over LTE without my say-so and with no hint
>> this is happening  - costing me real
>> money. My only option is to completely disable LTE when I get home  (and
>> lose SMS too) ?
>>
>
> Exactly.
>
> Setting the default network route should be enough to avoid that and to
> ensure a global network policy for the whole OS, unless an app (the
> browser) plays to be God.
>
>
=E2=80=8BSo, the work of MIF got mentioned during the IETF WG meeting, but =
I'm not
sure that folks have its coordinates; just in case, the documents are all
here:

http://datatracker.ietf.org/wg/mif/documents/

It's the right IETF venue to discuss the general question of how to manage
multiple interfaces.

regards,

Ted=E2=80=8B

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<div dir=3D"ltr"><div class=3D"gmail_extra">On Mon, Jul 20, 2015 at 4:43 AM=
, I=C3=B1aki Baz Castillo <span dir=3D"ltr">&lt;<a href=3D"mailto:ibc@aliax=
.net" target=3D"_blank">ibc@aliax.net</a>&gt;</span> wrote:<br><div class=
=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8=
ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div class=
=3D"gmail_extra"><span class=3D""><div class=3D"gmail_quote">2015-07-20 13:=
37 GMT+02:00 Tim Panton <span dir=3D"ltr">&lt;<a href=3D"mailto:thp@westhaw=
k.co.uk" target=3D"_blank">thp@westhawk.co.uk</a>&gt;</span>:<br><blockquot=
e class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc sol=
id;padding-left:1ex"><div>E.g. I=E2=80=99m at home and my chromebook pixel =
(or firefox tablet) is on wifi, but I=E2=80=99ve left LTE enabled.</div><di=
v>I (or the OS) is configured to prefer wifi wen available - but it happens=
 that for a specific peer LTE completes first.</div><div>So now my video ca=
ll goes over LTE without my say-so and with no hint this is happening =C2=
=A0- costing me real</div><div>money. My only option is to completely disab=
le LTE when I get home =C2=A0(and lose SMS too) ?</div></blockquote></div><=
br></span>Exactly.</div><div class=3D"gmail_extra"><br></div><div class=3D"=
gmail_extra">Setting the default network route should be enough to avoid th=
at and to ensure a global network policy for the whole OS, unless an app (t=
he browser) plays to be God.<span class=3D"HOEnZb"><font color=3D"#888888">=
<br></font></span><br></div></div></blockquote></div><br><div class=3D"gmai=
l_default" style=3D"font-family:tahoma,sans-serif;font-size:small">=E2=80=
=8BSo, the work of MIF got mentioned during the IETF WG meeting, but I&#39;=
m not sure that folks have its coordinates; just in case, the documents are=
 all here:<br><br><a href=3D"http://datatracker.ietf.org/wg/mif/documents/"=
>http://datatracker.ietf.org/wg/mif/documents/</a><br><br></div><div class=
=3D"gmail_default" style=3D"font-family:tahoma,sans-serif;font-size:small">=
It&#39;s the right IETF venue to discuss the general question of how to man=
age multiple interfaces.<br></div><div class=3D"gmail_default" style=3D"fon=
t-family:tahoma,sans-serif;font-size:small"><br></div><div class=3D"gmail_d=
efault" style=3D"font-family:tahoma,sans-serif;font-size:small">regards,<br=
><br></div><div class=3D"gmail_default" style=3D"font-family:tahoma,sans-se=
rif;font-size:small">Ted=E2=80=8B</div><br></div></div>

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From: Roman Shpount <roman@telurix.com>
To: Ted Hardie <ted.ietf@gmail.com>
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On Mon, Jul 27, 2015 at 3:02 PM, Ted Hardie <ted.ietf@gmail.com> wrote:

> =E2=80=8BSo, the work of MIF got mentioned during the IETF WG meeting, bu=
t I'm not
> sure that folks have its coordinates; just in case, the documents are all
> here:
>
> http://datatracker.ietf.org/wg/mif/documents/
>
> It's the right IETF venue to discuss the general question of how to manag=
e
> multiple interfaces.
>
>
Is this the right venue to discuss how ICE should handle multiple
interfaces as well?
_____________
Roman Shpount

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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On M=
on, Jul 27, 2015 at 3:02 PM, Ted Hardie <span dir=3D"ltr">&lt;<a href=3D"ma=
ilto:ted.ietf@gmail.com" target=3D"_blank">ted.ietf@gmail.com</a>&gt;</span=
> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0=
.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-s=
tyle:solid;padding-left:1ex"><div dir=3D"ltr"><div class=3D"gmail_extra"><d=
iv style=3D"font-family:tahoma,sans-serif;font-size:small">=E2=80=8BSo, the=
 work of MIF got mentioned during the IETF WG meeting, but I&#39;m not sure=
 that folks have its coordinates; just in case, the documents are all here:=
<br><br><a href=3D"http://datatracker.ietf.org/wg/mif/documents/" target=3D=
"_blank">http://datatracker.ietf.org/wg/mif/documents/</a><br><br></div><di=
v style=3D"font-family:tahoma,sans-serif;font-size:small">It&#39;s the righ=
t IETF venue to discuss the general question of how to manage multiple inte=
rfaces.<br></div><div style=3D"font-family:tahoma,sans-serif;font-size:smal=
l"><br></div></div></div></blockquote><div><br></div><div><div class=3D"gma=
il_signature">Is this the right venue to discuss how ICE should handle mult=
iple interfaces as well?</div><div class=3D"gmail_signature">_____________<=
br>Roman Shpount</div></div><div>=C2=A0</div></div></div></div>

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On Mon, Jul 27, 2015 at 12:09 PM, Roman Shpount <roman@telurix.com> wrote:

> On Mon, Jul 27, 2015 at 3:02 PM, Ted Hardie <ted.ietf@gmail.com> wrote:
>
>> =E2=80=8BSo, the work of MIF got mentioned during the IETF WG meeting, b=
ut I'm
>> not sure that folks have its coordinates; just in case, the documents ar=
e
>> all here:
>>
>> http://datatracker.ietf.org/wg/mif/documents/
>>
>> It's the right IETF venue to discuss the general question of how to
>> manage multiple interfaces.
>>
>>
> Is this the right venue to discuss how ICE should handle multiple
> interfaces as well?
>
=E2=80=8B
I don't think that question has enough detail to answer.  For ICE, the
current theory is questions on what ICE does with the interface information
it has go to MMUSIC, though that may get updated to a dedicated group.  MIF
would be for discussing what interface states get surfaced and when.


(Other views of this are possible, of course, and your mileage may vary
when posting).

Ted

--047d7bb03a92d2e0fa051be0568d
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<div dir=3D"ltr"><div class=3D"gmail_extra">On Mon, Jul 27, 2015 at 12:09 P=
M, Roman Shpount <span dir=3D"ltr">&lt;<a href=3D"mailto:roman@telurix.com"=
 target=3D"_blank">roman@telurix.com</a>&gt;</span> wrote:<br><div class=3D=
"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;=
border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div class=3D=
"gmail_extra"><div class=3D"gmail_quote"><span class=3D"">On Mon, Jul 27, 2=
015 at 3:02 PM, Ted Hardie <span dir=3D"ltr">&lt;<a href=3D"mailto:ted.ietf=
@gmail.com" target=3D"_blank">ted.ietf@gmail.com</a>&gt;</span> wrote:<br><=
blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-l=
eft-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;pa=
dding-left:1ex"><div dir=3D"ltr"><div class=3D"gmail_extra"><div style=3D"f=
ont-family:tahoma,sans-serif;font-size:small">=E2=80=8BSo, the work of MIF =
got mentioned during the IETF WG meeting, but I&#39;m not sure that folks h=
ave its coordinates; just in case, the documents are all here:<br><br><a hr=
ef=3D"http://datatracker.ietf.org/wg/mif/documents/" target=3D"_blank">http=
://datatracker.ietf.org/wg/mif/documents/</a><br><br></div><div style=3D"fo=
nt-family:tahoma,sans-serif;font-size:small">It&#39;s the right IETF venue =
to discuss the general question of how to manage multiple interfaces.<br></=
div><div style=3D"font-family:tahoma,sans-serif;font-size:small"><br></div>=
</div></div></blockquote><div><br></div></span><div><div>Is this the right =
venue to discuss how ICE should handle multiple interfaces as well?</div></=
div></div></div></div></blockquote><div><div class=3D"gmail_default" style=
=3D"font-family:tahoma,sans-serif;font-size:small">=E2=80=8B<br></div><div =
class=3D"gmail_default" style=3D"font-family:tahoma,sans-serif;font-size:sm=
all">I don&#39;t think that question has enough detail to answer.=C2=A0 For=
 ICE, the current theory is questions on what ICE does with the interface i=
nformation it has go to MMUSIC, though that may get updated to a dedicated =
group.=C2=A0 MIF would be for discussing what interface states get surfaced=
 and when.<br><br><br></div><div class=3D"gmail_default" style=3D"font-fami=
ly:tahoma,sans-serif;font-size:small">(Other views of this are possible, of=
 course, and your mileage may vary when posting).<br></div><div class=3D"gm=
ail_default" style=3D"font-family:tahoma,sans-serif;font-size:small"><br></=
div><div class=3D"gmail_default" style=3D"font-family:tahoma,sans-serif;fon=
t-size:small">Ted<br></div>=C2=A0</div></div></div></div>

--047d7bb03a92d2e0fa051be0568d--


From nobody Tue Jul 28 05:35:19 2015
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Subject: [rtcweb] Barry Leiba's No Objection on draft-ietf-rtcweb-stun-consent-freshness-15: (with COMMENT)
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Barry Leiba has entered the following ballot position for
draft-ietf-rtcweb-stun-consent-freshness-15: No Objection

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----------------------------------------------------------------------
COMMENT:
----------------------------------------------------------------------

A couple of very minor comments only:

FWIW, I think rtcweb-security-arch need only be an informative reference;
it seems only explanatory.  I also think that about RFC 6263.

-- Section 5.1 --

   A full ICE implementation obtains consent to send using ICE.  After
   ICE concludes on a particular candidate pair and whenever the
   endpoint sends application data on that pair consent MUST be
   maintained following the procedure described in this document.

I don't understand the "MUST" here, given that Section 4 says this is "an
optional extension".  Why "MUST", then, rather than "can be"?



From nobody Tue Jul 28 06:04:27 2015
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On 28 July 2015 at 14:35, Barry Leiba <barryleiba@computer.org> wrote:
> FWIW, I think rtcweb-security-arch need only be an informative reference;
> it seems only explanatory.  I also think that about RFC 6263.
>
> -- Section 5.1 --
>
>    A full ICE implementation obtains consent to send using ICE.  After
>    ICE concludes on a particular candidate pair and whenever the
>    endpoint sends application data on that pair consent MUST be
>    maintained following the procedure described in this document.
>
> I don't understand the "MUST" here, given that Section 4 says this is "an
> optional extension".  Why "MUST", then, rather than "can be"?

The extension might be optional, but we need to be clear about what is
mandatory and not within the mechanism itself.

Note that this *won't* be optional for WebRTC implementations
(browsers, certainly, maybe other classes of implementation too).


From nobody Tue Jul 28 06:07:49 2015
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Subject: Re: [rtcweb] Barry Leiba's No Objection on draft-ietf-rtcweb-stun-consent-freshness-15: (with COMMENT)
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>>    A full ICE implementation obtains consent to send using ICE.  After
>>    ICE concludes on a particular candidate pair and whenever the
>>    endpoint sends application data on that pair consent MUST be
>>    maintained following the procedure described in this document.
>>
>> I don't understand the "MUST" here, given that Section 4 says this is "an
>> optional extension".  Why "MUST", then, rather than "can be"?
>
> The extension might be optional, but we need to be clear about what is
> mandatory and not within the mechanism itself.

That's why I think the *other* MUSTs and SHOULDs are fine.  But *this*
one says "MUST use the procedure in this document", which doesn't seem
right.  (But, as I said, a minor comment.)

> Note that this *won't* be optional for WebRTC implementations
> (browsers, certainly, maybe other classes of implementation too).

Sure, but then those should say "MUST use <this>".

Barry


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On 28 July 2015 at 15:07, Barry Leiba <barryleiba@computer.org> wrote:
> That's why I think the *other* MUSTs and SHOULDs are fine.  But *this*
> one says "MUST use the procedure in this document", which doesn't seem
> right.  (But, as I said, a minor comment.)

I don't think that we lose anything by changing the "MUST be" to "is":

>    A full ICE implementation obtains consent to send using ICE.  After
>    ICE concludes on a particular candidate pair and whenever the
>    endpoint sends application data on that pair consent is
>    maintained following the procedure described in this document.

Axiomatic wins against prescriptive generally for me.


From nobody Wed Jul 29 23:33:03 2015
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Subject: Re: [rtcweb] Please require user consent for data channels
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Note: responding to an older message. Also, Mozilla is planning to land 
a number of optional controls for ICE candidate generation and TURN 
server use.  We also plan to add hooks that extensions can use to prompt 
(depending on circumstance) or monitor 'background' use of 
PeerConnections (without user-granted permission).

On 7/11/2015 12:42 PM, Daniel Roesler wrote:
> One of the items in the new proposal was "WebRTC already requires
> permission to access getUserMedia. Why not use that permission to
> control interface enumeration?" That item didn't really get discussed
> much in the thread, but I think it's one of the most important issues.

In addition to Kaufman's answer, I'll add that this would effectively 
make WebRTC unusable for users - and/or would destroy the utility of 
camera access requests (and WebRTC), due to user request-fatigue.  (If 
you ask a user for permission too many times, especially for something 
they asked to do and the question is about something they don't really 
understand, they either a) always hit yes regardless of the question, or 
b) stop using the feature entirely, because it's too annoying.)

Even camera-access requests are arguably a problem - but the alternative 
(granting permanent permission by default) is itself problematic.

> Why? There is now a documented case where a third party on nytimes.com
> is using a fake webRTC datachannel to silently gather user local (and
> potentially "real" ISP) IP addresses.

Sure, and gathering those for normal users is already trivially possible 
*without* webrtc - external IP's are trivial: whatismyipaddress.com, and 
for LAN addresses you can infer the LAN address with high probability by 
trying to load images from likely gateway addresses 
(192.168.0.1/192.168.1.1/10.0.0.1/etc) to find the likely subnet, and 
then scan the subnet using the same trick, recording the time-to-fail 
for each.  That will map the entire list of devices on the LAN, and you 
may be able to infer from it which address is yours (probably the 
fastest failure).  Note the LAN map itself is often a larger fingerprint 
reveal than the LAN IP address alone.  And yes, I've seen real examples 
of these tricks.

The case this doesn't directly cover fully is the "partial VPN" case, 
where someone has a VPN, but *also* still allows routing out directly 
without going through the VPN.  This is already hugely risky if you're 
trying to maintain anonymity - it's a false sense of security.  However 
it is at least a real concern, as is the "hiding at a shelter" case 
discussed a long time ago, where the issue is exposing external IP's to 
a remote caller (not to the service provider or STUN/TURN server).  I 
will note that there are many ways someone can find your external IP 
address unless you're very paranoid and careful (never load images in 
email, etc).

> So I'd like to voice my support for adding a consent requirement that
> would prevent this type of silent behavior. A user that is
> purposefully visiting a site that has a legitimate reason for using a
> webRTC data channel will have the necessary context to give consent,
> just like they have to get consent on getUserMedia.

You could disable DataChannels and have the same problem, note - this 
would have to apply to all PeerConnections (think of a call with 
offerToReceiveAudio - no gUM request would be made).

> I'm really unclear on why it's so important to have the silent webRTC
> data channel behavior. P2P data connections should be held to the same
> consent requirements just like P2P video and voice connections.

They are.

> In addition to the silent third party local IP tracking, a silent P2P
> data channel opens up users to silently becoming nodes in a web-based
> file sharing network. For example, webtorrent[2] can be silently
> embedded on a website so that all of the users to that website start
> seeding content they don't know they are seeding.

Note we don't require permissions to open WebSockets either.  While 
those don't allow P2P access, downloaded JS can use your CPU for bitcoin 
mining (though not efficiently yet), SETI searches, can store data and 
serve it back to the same (or other) servers, etc.

> 1. Require user consent before calling the callback on createOffer().
> 2. If the user has already given consent via getUserMedia, the user
> consent requirement is satisfied.
> 3. If there's no previous gUM consent, then ask the user for consent.

This could be done technically, and would cover the 
offerToReceiveAudio/Video case.  And there are still ways this could be 
abused, though not as easily.

It would cause people using WebRTC for the advantages DataChannels has 
over WebSockets to be annoyed (such as games).  Anything that benefits 
from UDP-like semantics has cause to use WebRTC/DataChannels, even if 
it's just talking directly to the server, and this would put a huge 
roadblock in the face of such utility.   Web-based games are starting to 
make use of DataChannels, and this would be a real problem for them.

Working around this for access back to the server would require some 
variant of CORS (to bypass the request if it's back to the same origin 
you loaded the page from - and that means you'd make it trivial again 
for the site to find your IPs, which was the reason this was brought 
up).  So that's not really useful.

One thing that *could* be done is to create an option to force such 
requests; we can debate what the visibility of such an option should be, 
and what the defaults should be - but a designed-in option opens the 
door for trivial extensions to change the or to provide UI to expose it 
(either in prefs or in main browser chrome or in a modal popup).  We 
could also flip the pref by default in private browsing mode, though 
that might require in-browser UI since random sites may trigger it.


Speaking to the non-fingerprinting aspects of this which had previously 
been discussed:

I understand your concerns, however the IP address exposure and LAN 
exposure is a red herring (and even more so in IPV6 world), except for 
the "I'm hiding" cases (poorly-configured VPN hiding or hiding from 
another user of the service).  We do surface the ability to not only 
know what PeerConnections are in use, but also that have been in use, so 
you can vet if sites are actually making connections. Right now it 
doesn't list unconnected PeerConnections (used for IP address 
harvesting), but it could.

One could also provide some sort of semi-persistent indicator of such 
background usage without requiring the user to interact with it.  (See 
the hooks I mention at the top of this post.)

-- 
Randell Jesup -- rjesup a t mozilla d o t com
Please please please don't emailrandell-ietf@jesup.org!  Way too much spam


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From: Ted Hardie <ted.ietf@gmail.com>
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The chairs have discussed how to make progress between now and Yokohama,
and it seems like another focused 2-3 hour meeting would be useful.  That
might be a completely virtual interim or one associated with the WebRTC
interim, but with a remote participation component.

If you have thoughts on that, please share them between now and August 7th,
2015.

thanks,

Ted, Cullen, Sean

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<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:tahoma,s=
ans-serif;font-size:small">The chairs have discussed how to make progress b=
etween now and Yokohama, and it seems like another focused 2-3 hour meeting=
 would be useful.=C2=A0 That might be a completely virtual interim or one a=
ssociated with the WebRTC interim, but with a remote participation componen=
t.=C2=A0 <br><br></div><div class=3D"gmail_default" style=3D"font-family:ta=
homa,sans-serif;font-size:small">If you have thoughts on that, please share=
 them between now and August 7th, 2015.<br><br></div><div class=3D"gmail_de=
fault" style=3D"font-family:tahoma,sans-serif;font-size:small">thanks,<br><=
br></div><div class=3D"gmail_default" style=3D"font-family:tahoma,sans-seri=
f;font-size:small">Ted, Cullen, Sean<br></div></div>

--f46d043c825a2502c0051c1b3c7d--


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From: Peter Thatcher <pthatcher@google.com>
Date: Thu, 30 Jul 2015 11:53:43 -0700
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Making it a part of the W3C interim sounds like a good idea to me.  We've
done both IETF and W3C at interims in the past, and they have been
productive (perhaps more productive than the non-interim meetings).

On Thu, Jul 30, 2015 at 10:43 AM, Ted Hardie <ted.ietf@gmail.com> wrote:

> The chairs have discussed how to make progress between now and Yokohama,
> and it seems like another focused 2-3 hour meeting would be useful.  That
> might be a completely virtual interim or one associated with the WebRTC
> interim, but with a remote participation component.
>
> If you have thoughts on that, please share them between now and August
> 7th, 2015.
>
> thanks,
>
> Ted, Cullen, Sean
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:arial,he=
lvetica,sans-serif">Making it a part of the W3C interim sounds like a good =
idea to me.=C2=A0 We&#39;ve done both IETF and W3C at interims in the past,=
 and they have been productive (perhaps more productive than the non-interi=
m meetings).</div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote"=
>On Thu, Jul 30, 2015 at 10:43 AM, Ted Hardie <span dir=3D"ltr">&lt;<a href=
=3D"mailto:ted.ietf@gmail.com" target=3D"_blank">ted.ietf@gmail.com</a>&gt;=
</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .=
8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div clas=
s=3D"gmail_default" style=3D"font-family:tahoma,sans-serif;font-size:small"=
>The chairs have discussed how to make progress between now and Yokohama, a=
nd it seems like another focused 2-3 hour meeting would be useful.=C2=A0 Th=
at might be a completely virtual interim or one associated with the WebRTC =
interim, but with a remote participation component.=C2=A0 <br><br></div><di=
v class=3D"gmail_default" style=3D"font-family:tahoma,sans-serif;font-size:=
small">If you have thoughts on that, please share them between now and Augu=
st 7th, 2015.<br><br></div><div class=3D"gmail_default" style=3D"font-famil=
y:tahoma,sans-serif;font-size:small">thanks,<br><br></div><div class=3D"gma=
il_default" style=3D"font-family:tahoma,sans-serif;font-size:small">Ted, Cu=
llen, Sean<br></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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div></div>

--e89a8fb1fa764a901e051c1c3b30--


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Cc: Cullen Jennings <fluffy@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Interim discussion?
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--001a1140c18a7f973e051c1c6255
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+1

On Thu, Jul 30, 2015 at 11:53 AM, Peter Thatcher <pthatcher@google.com>
wrote:

> Making it a part of the W3C interim sounds like a good idea to me.  We've
> done both IETF and W3C at interims in the past, and they have been
> productive (perhaps more productive than the non-interim meetings).
>
> On Thu, Jul 30, 2015 at 10:43 AM, Ted Hardie <ted.ietf@gmail.com> wrote:
>
>> The chairs have discussed how to make progress between now and Yokohama,
>> and it seems like another focused 2-3 hour meeting would be useful.  That
>> might be a completely virtual interim or one associated with the WebRTC
>> interim, but with a remote participation component.
>>
>> If you have thoughts on that, please share them between now and August
>> 7th, 2015.
>>
>> thanks,
>>
>> Ted, Cullen, Sean
>>
>> _______________________________________________
>> 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
>
>

--001a1140c18a7f973e051c1c6255
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<div dir=3D"ltr">+1</div><div class=3D"gmail_extra"><br><div class=3D"gmail=
_quote">On Thu, Jul 30, 2015 at 11:53 AM, Peter Thatcher <span dir=3D"ltr">=
&lt;<a href=3D"mailto:pthatcher@google.com" target=3D"_blank">pthatcher@goo=
gle.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 dir=3D=
"ltr"><div class=3D"gmail_default" style=3D"font-family:arial,helvetica,san=
s-serif">Making it a part of the W3C interim sounds like a good idea to me.=
=C2=A0 We&#39;ve done both IETF and W3C at interims in the past, and they h=
ave been productive (perhaps more productive than the non-interim meetings)=
.</div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote"><div><div =
class=3D"h5">On Thu, Jul 30, 2015 at 10:43 AM, Ted Hardie <span dir=3D"ltr"=
>&lt;<a href=3D"mailto:ted.ietf@gmail.com" target=3D"_blank">ted.ietf@gmail=
.com</a>&gt;</span> wrote:<br></div></div><blockquote class=3D"gmail_quote"=
 style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><d=
iv><div class=3D"h5"><div dir=3D"ltr"><div class=3D"gmail_default" style=3D=
"font-family:tahoma,sans-serif;font-size:small">The chairs have discussed h=
ow to make progress between now and Yokohama, and it seems like another foc=
used 2-3 hour meeting would be useful.=C2=A0 That might be a completely vir=
tual interim or one associated with the WebRTC interim, but with a remote p=
articipation component.=C2=A0 <br><br></div><div class=3D"gmail_default" st=
yle=3D"font-family:tahoma,sans-serif;font-size:small">If you have thoughts =
on that, please share them between now and August 7th, 2015.<br><br></div><=
div class=3D"gmail_default" style=3D"font-family:tahoma,sans-serif;font-siz=
e:small">thanks,<br><br></div><div class=3D"gmail_default" style=3D"font-fa=
mily:tahoma,sans-serif;font-size:small">Ted, Cullen, Sean<br></div></div>
<br></div></div>_______________________________________________<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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>

--001a1140c18a7f973e051c1c6255--


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Subject: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it would be Christmas in July!)
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--001a1134fd4033feb8051c1c90bd
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At IETF 93 last week,  Cullen pointed out that he looked into ICE endpoints
and found that all of them support rtcp-mux.  Inspired by that, I went
around and asked about a dozen people "would you be OK if we removed
support for non-muxed RTCP?"  The range of responses went from "I'm OK with
it if Cullen is" to "yeah, that would be great".  No one said no.  Cullen
said yes.

For me, removing support for non-muxed RTCP would be like Christmas in
July.  The amount of gross code in our implementation just to support this
case is very large, especially considering the very small benefit it gives
us.  Talking to the developers at Mozilla, it sounds like they have a
similar cost in code complexity.  I'm guessing every ICE stack does.  We
could clean up a lot of code complexity in a lot of code bases by requiring
rtcp-mux.


So here's my proposal:  make RtcpMuxPolicy == "required" be not only the
default, but the only RtcpMuxPolicy available.   All rtcweb endpoints must
support rtcp-mux.   An rtcweb endpoint would be free to fail if the remote
endpoint does not support rtcp-mux.



The whole reason we didn't do this before was because we wanted to work
with legacy equipment.  But Cullen's claim is that there is no legacy
equipment that can speak ICE (and DTLS) but not rtcp-mux.

So is there any remaining reason that we cannot require rtcp-mux?

--001a1134fd4033feb8051c1c90bd
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<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:arial,he=
lvetica,sans-serif">At IETF 93 last week, =C2=A0Cullen pointed out that he =
looked into ICE endpoints and found that all of them support rtcp-mux.=C2=
=A0 Inspired by that, I went around and asked about a dozen people &quot;wo=
uld you be OK if we removed support for non-muxed RTCP?&quot; =C2=A0The ran=
ge of responses went from &quot;I&#39;m OK with it if Cullen is&quot; to &q=
uot;yeah, that would be great&quot;.=C2=A0 No one said no.=C2=A0 Cullen sai=
d yes. =C2=A0</div><div class=3D"gmail_default" style=3D"font-family:arial,=
helvetica,sans-serif"><br></div><div class=3D"gmail_default" style=3D"font-=
family:arial,helvetica,sans-serif">For me, removing support for non-muxed R=
TCP would be like Christmas in July.=C2=A0 The amount of gross code in our =
implementation just to support this case is very large, especially consider=
ing the very small benefit it gives us.=C2=A0 Talking to the developers at =
Mozilla, it sounds like they have a similar cost in code complexity.=C2=A0 =
I&#39;m guessing every ICE stack does.=C2=A0 We could clean up a lot of cod=
e complexity in a lot of code bases by requiring rtcp-mux. =C2=A0</div><div=
 class=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-serif"><=
br></div><div class=3D"gmail_default" style=3D"font-family:arial,helvetica,=
sans-serif"><br></div><div class=3D"gmail_default" style=3D"font-family:ari=
al,helvetica,sans-serif">So here&#39;s my proposal: =C2=A0make RtcpMuxPolic=
y =3D=3D &quot;required&quot; be not only the default, but the only RtcpMux=
Policy available. =C2=A0 All rtcweb endpoints must support rtcp-mux. =C2=A0=
 An rtcweb endpoint would be free to fail if the remote endpoint does not s=
upport rtcp-mux.</div><div class=3D"gmail_default" style=3D"font-family:ari=
al,helvetica,sans-serif"><br></div><div class=3D"gmail_default" style=3D"fo=
nt-family:arial,helvetica,sans-serif"><br></div><div class=3D"gmail_default=
" style=3D"font-family:arial,helvetica,sans-serif"><br></div><div class=3D"=
gmail_default" style=3D"font-family:arial,helvetica,sans-serif">The whole r=
eason we didn&#39;t do this before was because we wanted to work with legac=
y equipment.=C2=A0 But Cullen&#39;s claim is that there is no legacy equipm=
ent that can speak ICE (and DTLS) but not rtcp-mux.=C2=A0</div><div class=
=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-serif"><br></d=
iv><div class=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-s=
erif">So is there any remaining reason that we cannot require rtcp-mux?</di=
v><div class=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-se=
rif"><br></div><div class=3D"gmail_default" style=3D"font-family:arial,helv=
etica,sans-serif"><br></div><div class=3D"gmail_default" style=3D"font-fami=
ly:arial,helvetica,sans-serif"><br></div></div>

--001a1134fd4033feb8051c1c90bd--


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From: Roman Shpount <roman@telurix.com>
To: Randell Jesup <randell-ietf@jesup.org>
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Subject: Re: [rtcweb] Please require user consent for data channels
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Hi Randell,

I do agree with most of your argument regarding fingerprinting and gUM. I
also agree that this is not just the data channel issue and similar
exploits can be created using audio or video connections. Furthermore,
these usage patterns are very difficult to distinguish from legitimate
applications. On the other hand I would argue that all of this exploits do
not matter, since you can get the same information using non webrtc related
HTML functionality.

One thing that I do not agree with is that PeerConnection introduced a
significant new behavior -- routing data over the non-default network
interface. This is something that was not possible before. There is no way
to overwrite default routing with HTTP request or anything similar
generated by the browser. Overwriting default routing would allow creation
of a whole new class of exploits, from determining user other IP addresses
to forcing traffic over unexpected networks and generating expenses for the
 user. My suggestion is that until system wide preferences are exposed via
some other mechanism defined by MIF, web browser should follow the local
policy and do not overwrite local routing metrics by sending ICE requests
over specific interfaces. At the very least, this should only be enabled
via some sort of configuration option which should be disabled by default.
I understand that using only default routes can decrease the chances of
connection, but from my experience* we are talking about less then 0.1% of
all the users.

Also, please note that a lot of software VPNs do allow routing over
secondary gateways. There is nothing fake or unusual about them. For
instance Microsoft default PPTP VPN simply creates a virtual adapter for
the VPN connection and creates a route with lower metric. This virtual
adapter, on the other hand is using the previous default route for PPTP
connection.

* We have deployed a VoIP system with 500K users that used ICE in
combination with TURNs over default route with HTTP connect option. Total
number of users who could not place a call even though they could connect
to the web server was less then 0.1%.
_____________
Roman Shpount

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<div dir=3D"ltr"><div class=3D"gmail_extra">Hi Randell,</div><div class=3D"=
gmail_extra"><br></div><div class=3D"gmail_extra">I do agree with most of y=
our argument regarding fingerprinting and gUM. I also agree that this is no=
t just the data channel issue and similar exploits can be created using aud=
io or video connections. Furthermore, these usage patterns are very difficu=
lt to distinguish from legitimate applications. On the other hand I would a=
rgue that all of this exploits do not matter, since you can get the same in=
formation using non webrtc related HTML functionality.</div><div class=3D"g=
mail_extra"><br></div><div class=3D"gmail_extra">One thing that I do not ag=
ree with is that PeerConnection introduced a significant new behavior -- ro=
uting data over the non-default network interface. This is something that w=
as not possible before. There is no way to overwrite default routing with H=
TTP request or anything similar generated by the browser. Overwriting defau=
lt routing would allow creation of a whole new class of exploits, from dete=
rmining user other IP addresses to forcing traffic over unexpected networks=
 and generating expenses for the =C2=A0user. My suggestion is that until sy=
stem wide preferences are exposed via some other mechanism defined by MIF, =
web browser should follow the local policy and do not overwrite local routi=
ng metrics by sending ICE requests over specific interfaces. At the very le=
ast, this should only be enabled via some sort of configuration option whic=
h should be disabled by default. I understand that using only default route=
s can decrease the chances of connection, but from my experience* we are ta=
lking about less then 0.1% of all the users.</div><div class=3D"gmail_extra=
"><br></div><div class=3D"gmail_extra">Also, please note that a lot of soft=
ware VPNs do allow routing over secondary gateways. There is nothing fake o=
r unusual about them. For instance Microsoft default PPTP VPN simply create=
s a virtual adapter for the VPN connection and creates a route with lower m=
etric. This virtual adapter, on the other hand is using the previous defaul=
t route for PPTP connection.</div><div class=3D"gmail_extra"><br></div><div=
 class=3D"gmail_extra">*=C2=A0We have deployed a VoIP system with 500K user=
s that used ICE in combination with TURNs over default route with HTTP conn=
ect option. Total number of users who could not place a call even though th=
ey could connect to the web server was less then 0.1%.<br clear=3D"all"><di=
v><div class=3D"gmail_signature">_____________<br>Roman Shpount</div></div>
<br></div></div>

--001a114043c607b567051c1cafcb--


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From: Bernard Aboba <bernard.aboba@gmail.com>
Date: Thu, 30 Jul 2015 12:28:04 -0700
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To: Peter Thatcher <pthatcher@google.com>
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Subject: Re: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it would be Christmas in July!)
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--001a11c37d22efb86b051c1cb473
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Peter said:

"So is there any remaining reason that we cannot require rtcp-mux?"

[BA] To be clear, you are talking about making RTP/RTCP
multiplexing mandatory-to-use?

The RTP-Usage document already mandates support for RFC 5761.  From Section
4.5:

   implementations are REQUIRED to support multiplexing RTP data packets
   and RTCP control packets on a single transport-layer flow [RFC5761
<https://tools.ietf.org/html/rfc5761>].

On Thu, Jul 30, 2015 at 12:17 PM, Peter Thatcher <pthatcher@google.com>
wrote:

> At IETF 93 last week,  Cullen pointed out that he looked into ICE
> endpoints and found that all of them support rtcp-mux.  Inspired by that, I
> went around and asked about a dozen people "would you be OK if we removed
> support for non-muxed RTCP?"  The range of responses went from "I'm OK with
> it if Cullen is" to "yeah, that would be great".  No one said no.  Cullen
> said yes.
>
> For me, removing support for non-muxed RTCP would be like Christmas in
> July.  The amount of gross code in our implementation just to support this
> case is very large, especially considering the very small benefit it gives
> us.  Talking to the developers at Mozilla, it sounds like they have a
> similar cost in code complexity.  I'm guessing every ICE stack does.  We
> could clean up a lot of code complexity in a lot of code bases by requiring
> rtcp-mux.
>
>
> So here's my proposal:  make RtcpMuxPolicy == "required" be not only the
> default, but the only RtcpMuxPolicy available.   All rtcweb endpoints must
> support rtcp-mux.   An rtcweb endpoint would be free to fail if the remote
> endpoint does not support rtcp-mux.
>
>
>
> The whole reason we didn't do this before was because we wanted to work
> with legacy equipment.  But Cullen's claim is that there is no legacy
> equipment that can speak ICE (and DTLS) but not rtcp-mux.
>
> So is there any remaining reason that we cannot require rtcp-mux?
>
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--001a11c37d22efb86b051c1cb473
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<div dir=3D"ltr"><div>Peter said: </div><div><br></div><div>&quot;So is the=
re any remaining reason that we cannot require rtcp-mux?&quot;</div><div><b=
r></div><div>[BA] To be clear, you are talking about making RTP/RTCP multip=
lexing=C2=A0mandatory-to-use? </div><div><br></div><div>The RTP-Usage docum=
ent already=C2=A0mandates support for RFC 5761.=C2=A0 From Section 4.5: </d=
iv><div><br></div><div>=C2=A0=C2=A0 implementations are REQUIRED to support=
 multiplexing RTP data packets<br>=C2=A0=C2=A0 and RTCP control packets on =
a single transport-layer flow [<a title=3D"&quot;Multiplexing RTP Data and =
Control Packets on a Single Port&quot;" href=3D"https://tools.ietf.org/html=
/rfc5761"><font color=3D"#0066cc">RFC5761</font></a>].</div></div><div clas=
s=3D"gmail_extra"><br><div class=3D"gmail_quote">On Thu, Jul 30, 2015 at 12=
:17 PM, Peter Thatcher <span dir=3D"ltr">&lt;<a href=3D"mailto:pthatcher@go=
ogle.com" target=3D"_blank">pthatcher@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"><div dir=3D"ltr"><div class=3D"gmail_default"=
 style=3D"font-family:arial,helvetica,sans-serif">At IETF 93 last week, =C2=
=A0Cullen pointed out that he looked into ICE endpoints and found that all =
of them support rtcp-mux.=C2=A0 Inspired by that, I went around and asked a=
bout a dozen people &quot;would you be OK if we removed support for non-mux=
ed RTCP?&quot; =C2=A0The range of responses went from &quot;I&#39;m OK with=
 it if Cullen is&quot; to &quot;yeah, that would be great&quot;.=C2=A0 No o=
ne said no.=C2=A0 Cullen said yes. =C2=A0</div><div class=3D"gmail_default"=
 style=3D"font-family:arial,helvetica,sans-serif"><br></div><div class=3D"g=
mail_default" style=3D"font-family:arial,helvetica,sans-serif">For me, remo=
ving support for non-muxed RTCP would be like Christmas in July.=C2=A0 The =
amount of gross code in our implementation just to support this case is ver=
y large, especially considering the very small benefit it gives us.=C2=A0 T=
alking to the developers at Mozilla, it sounds like they have a similar cos=
t in code complexity.=C2=A0 I&#39;m guessing every ICE stack does.=C2=A0 We=
 could clean up a lot of code complexity in a lot of code bases by requirin=
g rtcp-mux. =C2=A0</div><div class=3D"gmail_default" style=3D"font-family:a=
rial,helvetica,sans-serif"><br></div><div class=3D"gmail_default" style=3D"=
font-family:arial,helvetica,sans-serif"><br></div><div class=3D"gmail_defau=
lt" style=3D"font-family:arial,helvetica,sans-serif">So here&#39;s my propo=
sal: =C2=A0make RtcpMuxPolicy =3D=3D &quot;required&quot; be not only the d=
efault, but the only RtcpMuxPolicy available. =C2=A0 All rtcweb endpoints m=
ust support rtcp-mux. =C2=A0 An rtcweb endpoint would be free to fail if th=
e remote endpoint does not support rtcp-mux.</div><div class=3D"gmail_defau=
lt" style=3D"font-family:arial,helvetica,sans-serif"><br></div><div class=
=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-serif"><br></d=
iv><div class=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-s=
erif"><br></div><div class=3D"gmail_default" style=3D"font-family:arial,hel=
vetica,sans-serif">The whole reason we didn&#39;t do this before was becaus=
e we wanted to work with legacy equipment.=C2=A0 But Cullen&#39;s claim is =
that there is no legacy equipment that can speak ICE (and DTLS) but not rtc=
p-mux.=C2=A0</div><div class=3D"gmail_default" style=3D"font-family:arial,h=
elvetica,sans-serif"><br></div><div class=3D"gmail_default" style=3D"font-f=
amily:arial,helvetica,sans-serif">So is there any remaining reason that we =
cannot require rtcp-mux?</div><div class=3D"gmail_default" style=3D"font-fa=
mily:arial,helvetica,sans-serif"><br></div><div class=3D"gmail_default" sty=
le=3D"font-family:arial,helvetica,sans-serif"><br></div><div class=3D"gmail=
_default" style=3D"font-family:arial,helvetica,sans-serif"><br></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" =
rel=3D"noreferrer">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>

--001a11c37d22efb86b051c1cb473--


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Subject: Re: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it would be Christmas in July!)
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Please, please, please make rtcp-mux mandatory to use! For me this would be
Christmas, Hanukkah and Kwanzaa rolled into one.

_____________
Roman Shpount

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<div dir=3D"ltr">Please, please, please make rtcp-mux mandatory to use! For=
 me this would be Christmas, Hanukkah and Kwanzaa rolled into one.=C2=A0<br=
><div class=3D"gmail_extra"><br clear=3D"all"><div><div class=3D"gmail_sign=
ature">_____________<br>Roman Shpount</div></div>
<br></div></div>

--001a11403fd4c76d9e051c1cb961--


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From: Peter Thatcher <pthatcher@google.com>
Date: Thu, 30 Jul 2015 12:35:17 -0700
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Subject: Re: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it would be Christmas in July!)
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I didn't quite mean mandatory-to-use.  If an rtcweb endpoint wants to use
non-muxed RTCP, it could go ahead and do that, so it's not quite
mandatory-to-use-rtcp-mux, but instead free-to-not-support-non-muxed-rtcp.

Perhaps my proposal should instead be:  rtcweb endpoints MAY make RtcpMuxPolicy
== "required" the default and MAY remove support for RtcpMuxPolicy ==
"negotiate".

That would leave the door open to endpoints that want to support non-muxed
RTCP, which mandatory-to-use would not.  But it would also leave the door
open to endpoints not having to support non-muxed RTCP.



On Thu, Jul 30, 2015 at 12:28 PM, Bernard Aboba <bernard.aboba@gmail.com>
wrote:

> Peter said:
>
> "So is there any remaining reason that we cannot require rtcp-mux?"
>
> [BA] To be clear, you are talking about making RTP/RTCP
> multiplexing mandatory-to-use?
>
> The RTP-Usage document already mandates support for RFC 5761.  From
> Section 4.5:
>
>    implementations are REQUIRED to support multiplexing RTP data packets
>    and RTCP control packets on a single transport-layer flow [RFC5761
> <https://tools.ietf.org/html/rfc5761>].
>
> On Thu, Jul 30, 2015 at 12:17 PM, Peter Thatcher <pthatcher@google.com>
> wrote:
>
>> At IETF 93 last week,  Cullen pointed out that he looked into ICE
>> endpoints and found that all of them support rtcp-mux.  Inspired by that, I
>> went around and asked about a dozen people "would you be OK if we removed
>> support for non-muxed RTCP?"  The range of responses went from "I'm OK with
>> it if Cullen is" to "yeah, that would be great".  No one said no.  Cullen
>> said yes.
>>
>> For me, removing support for non-muxed RTCP would be like Christmas in
>> July.  The amount of gross code in our implementation just to support this
>> case is very large, especially considering the very small benefit it gives
>> us.  Talking to the developers at Mozilla, it sounds like they have a
>> similar cost in code complexity.  I'm guessing every ICE stack does.  We
>> could clean up a lot of code complexity in a lot of code bases by requiring
>> rtcp-mux.
>>
>>
>> So here's my proposal:  make RtcpMuxPolicy == "required" be not only the
>> default, but the only RtcpMuxPolicy available.   All rtcweb endpoints must
>> support rtcp-mux.   An rtcweb endpoint would be free to fail if the remote
>> endpoint does not support rtcp-mux.
>>
>>
>>
>> The whole reason we didn't do this before was because we wanted to work
>> with legacy equipment.  But Cullen's claim is that there is no legacy
>> equipment that can speak ICE (and DTLS) but not rtcp-mux.
>>
>> So is there any remaining reason that we cannot require rtcp-mux?
>>
>>
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>>
>

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<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:arial,he=
lvetica,sans-serif">I didn&#39;t quite mean mandatory-to-use.=C2=A0 If an r=
tcweb endpoint wants to use non-muxed RTCP, it could go ahead and do that, =
so it&#39;s not quite mandatory-to-use-rtcp-mux, but instead free-to-not-su=
pport-non-muxed-rtcp.</div><div class=3D"gmail_default" style=3D"font-famil=
y:arial,helvetica,sans-serif"><br></div><div class=3D"gmail_default" style=
=3D"font-family:arial,helvetica,sans-serif">Perhaps my proposal should inst=
ead be: =C2=A0rtcweb endpoints MAY make<span style=3D"font-size:12.8px">=C2=
=A0RtcpMuxPolicy =3D=3D &quot;required&quot; the default and MAY remove sup=
port for RtcpMuxPolicy =3D=3D &quot;negotiate&quot;.</span></div><div class=
=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-serif"><span s=
tyle=3D"font-size:12.8px"><br></span></div><div class=3D"gmail_default" sty=
le=3D"font-family:arial,helvetica,sans-serif"><span style=3D"font-size:12.8=
px">That would leave the door open to endpoints that want to support non-mu=
xed RTCP, which mandatory-to-use would not.=C2=A0 But it would also leave t=
he door open to endpoints not having to support non-muxed RTCP.</span></div=
><div class=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-ser=
if"><br></div><div class=3D"gmail_default" style=3D"font-family:arial,helve=
tica,sans-serif"><br></div><div class=3D"gmail_extra"><br><div class=3D"gma=
il_quote">On Thu, Jul 30, 2015 at 12:28 PM, Bernard Aboba <span dir=3D"ltr"=
>&lt;<a href=3D"mailto:bernard.aboba@gmail.com" target=3D"_blank">bernard.a=
boba@gmail.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" s=
tyle=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div=
 dir=3D"ltr"><span><div>Peter said: </div><div><br></div><div>&quot;So is t=
here any remaining reason that we cannot require rtcp-mux?&quot;</div><div>=
<br></div></span><div>[BA] To be clear, you are talking about making RTP/RT=
CP multiplexing=C2=A0mandatory-to-use? </div><div><br></div><div>The RTP-Us=
age document already=C2=A0mandates support for RFC 5761.=C2=A0 From Section=
 4.5: </div><div><br></div><div>=C2=A0=C2=A0 implementations are REQUIRED t=
o support multiplexing RTP data packets<br>=C2=A0=C2=A0 and RTCP control pa=
ckets on a single transport-layer flow [<a title=3D"&quot;Multiplexing RTP =
Data and Control Packets on a Single Port&quot;" href=3D"https://tools.ietf=
.org/html/rfc5761" target=3D"_blank"><font color=3D"#0066cc">RFC5761</font>=
</a>].</div></div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote"=
><div><div>On Thu, Jul 30, 2015 at 12:17 PM, Peter Thatcher <span dir=3D"lt=
r">&lt;<a href=3D"mailto:pthatcher@google.com" target=3D"_blank">pthatcher@=
google.com</a>&gt;</span> wrote:<br></div></div><blockquote class=3D"gmail_=
quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1=
ex"><div><div><div dir=3D"ltr"><div style=3D"font-family:arial,helvetica,sa=
ns-serif">At IETF 93 last week, =C2=A0Cullen pointed out that he looked int=
o ICE endpoints and found that all of them support rtcp-mux.=C2=A0 Inspired=
 by that, I went around and asked about a dozen people &quot;would you be O=
K if we removed support for non-muxed RTCP?&quot; =C2=A0The range of respon=
ses went from &quot;I&#39;m OK with it if Cullen is&quot; to &quot;yeah, th=
at would be great&quot;.=C2=A0 No one said no.=C2=A0 Cullen said yes. =C2=
=A0</div><div style=3D"font-family:arial,helvetica,sans-serif"><br></div><d=
iv style=3D"font-family:arial,helvetica,sans-serif">For me, removing suppor=
t for non-muxed RTCP would be like Christmas in July.=C2=A0 The amount of g=
ross code in our implementation just to support this case is very large, es=
pecially considering the very small benefit it gives us.=C2=A0 Talking to t=
he developers at Mozilla, it sounds like they have a similar cost in code c=
omplexity.=C2=A0 I&#39;m guessing every ICE stack does.=C2=A0 We could clea=
n up a lot of code complexity in a lot of code bases by requiring rtcp-mux.=
 =C2=A0</div><div style=3D"font-family:arial,helvetica,sans-serif"><br></di=
v><div style=3D"font-family:arial,helvetica,sans-serif"><br></div><div styl=
e=3D"font-family:arial,helvetica,sans-serif">So here&#39;s my proposal: =C2=
=A0make RtcpMuxPolicy =3D=3D &quot;required&quot; be not only the default, =
but the only RtcpMuxPolicy available. =C2=A0 All rtcweb endpoints must supp=
ort rtcp-mux. =C2=A0 An rtcweb endpoint would be free to fail if the remote=
 endpoint does not support rtcp-mux.</div><div style=3D"font-family:arial,h=
elvetica,sans-serif"><br></div><div style=3D"font-family:arial,helvetica,sa=
ns-serif"><br></div><div style=3D"font-family:arial,helvetica,sans-serif"><=
br></div><div style=3D"font-family:arial,helvetica,sans-serif">The whole re=
ason we didn&#39;t do this before was because we wanted to work with legacy=
 equipment.=C2=A0 But Cullen&#39;s claim is that there is no legacy equipme=
nt that can speak ICE (and DTLS) but not rtcp-mux.=C2=A0</div><div style=3D=
"font-family:arial,helvetica,sans-serif"><br></div><div style=3D"font-famil=
y:arial,helvetica,sans-serif">So is there any remaining reason that we cann=
ot require rtcp-mux?</div><div style=3D"font-family:arial,helvetica,sans-se=
rif"><br></div><div style=3D"font-family:arial,helvetica,sans-serif"><br></=
div><div style=3D"font-family:arial,helvetica,sans-serif"><br></div></div>
<br></div></div>_______________________________________________<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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>
</blockquote></div><br></div></div>

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Subject: Re: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it would be Christmas in July!)
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Le 2015-07-30 15:29, Roman Shpount a écrit :
> Please, please, please make rtcp-mux mandatory to use! For me this would
> be Christmas, Hanukkah and Kwanzaa rolled into one.

+1

I prefer mandatory-to-use to free-to-not-support-non-muxed-rtcp.

Simon


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From: Peter Thatcher <pthatcher@google.com>
Date: Thu, 30 Jul 2015 12:38:30 -0700
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Subject: Re: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it would be Christmas in July!)
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I guess I should have said "Winter Break in July".

On Thu, Jul 30, 2015 at 12:29 PM, Roman Shpount <roman@telurix.com> wrote:

> Please, please, please make rtcp-mux mandatory to use! For me this would
> be Christmas, Hanukkah and Kwanzaa rolled into one.
>
> _____________
> Roman Shpount
>
>

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<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:arial,he=
lvetica,sans-serif">I guess I should have said &quot;Winter Break in July&q=
uot;. =C2=A0</div></div><div class=3D"gmail_extra"><br><div class=3D"gmail_=
quote">On Thu, Jul 30, 2015 at 12:29 PM, Roman Shpount <span dir=3D"ltr">&l=
t;<a href=3D"mailto:roman@telurix.com" target=3D"_blank">roman@telurix.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 dir=3D"ltr">Pl=
ease, please, please make rtcp-mux mandatory to use! For me this would be C=
hristmas, Hanukkah and Kwanzaa rolled into one.=C2=A0<br><div class=3D"gmai=
l_extra"><br clear=3D"all"><div><div>_____________<span class=3D"HOEnZb"><f=
ont color=3D"#888888"><br>Roman Shpount</font></span></div></div>
<br></div></div>
</blockquote></div><br></div>

--001a113cd41a704424051c1cdb43--


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Subject: Re: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it would be Christmas in July!)
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"free-to-not-support-non-muxed-rtcp."

[BA] OK.   This would involve changing "are also REQUIRED to" to "MAY" in
the following sentence within RTP-Usage Section 4.5:

   For backwards compatibility, implementations are also REQUIRED to support
   RTP and RTCP sent on separate transport-layer flows.


On Thu, Jul 30, 2015 at 12:35 PM, Peter Thatcher <pthatcher@google.com>
wrote:

> I didn't quite mean mandatory-to-use.  If an rtcweb endpoint wants to use
> non-muxed RTCP, it could go ahead and do that, so it's not quite
> mandatory-to-use-rtcp-mux, but instead free-to-not-support-non-muxed-rtcp.
>
> Perhaps my proposal should instead be:  rtcweb endpoints MAY make RtcpMuxPolicy
> == "required" the default and MAY remove support for RtcpMuxPolicy ==
> "negotiate".
>
> That would leave the door open to endpoints that want to support non-muxed
> RTCP, which mandatory-to-use would not.  But it would also leave the door
> open to endpoints not having to support non-muxed RTCP.
>
>
>
> On Thu, Jul 30, 2015 at 12:28 PM, Bernard Aboba <bernard.aboba@gmail.com>
> wrote:
>
>> Peter said:
>>
>> "So is there any remaining reason that we cannot require rtcp-mux?"
>>
>> [BA] To be clear, you are talking about making RTP/RTCP
>> multiplexing mandatory-to-use?
>>
>> The RTP-Usage document already mandates support for RFC 5761.  From
>> Section 4.5:
>>
>>    implementations are REQUIRED to support multiplexing RTP data packets
>>    and RTCP control packets on a single transport-layer flow [RFC5761
>> <https://tools.ietf.org/html/rfc5761>].
>>
>> On Thu, Jul 30, 2015 at 12:17 PM, Peter Thatcher <pthatcher@google.com>
>> wrote:
>>
>>> At IETF 93 last week,  Cullen pointed out that he looked into ICE
>>> endpoints and found that all of them support rtcp-mux.  Inspired by that, I
>>> went around and asked about a dozen people "would you be OK if we removed
>>> support for non-muxed RTCP?"  The range of responses went from "I'm OK with
>>> it if Cullen is" to "yeah, that would be great".  No one said no.  Cullen
>>> said yes.
>>>
>>> For me, removing support for non-muxed RTCP would be like Christmas in
>>> July.  The amount of gross code in our implementation just to support this
>>> case is very large, especially considering the very small benefit it gives
>>> us.  Talking to the developers at Mozilla, it sounds like they have a
>>> similar cost in code complexity.  I'm guessing every ICE stack does.  We
>>> could clean up a lot of code complexity in a lot of code bases by requiring
>>> rtcp-mux.
>>>
>>>
>>> So here's my proposal:  make RtcpMuxPolicy == "required" be not only the
>>> default, but the only RtcpMuxPolicy available.   All rtcweb endpoints must
>>> support rtcp-mux.   An rtcweb endpoint would be free to fail if the remote
>>> endpoint does not support rtcp-mux.
>>>
>>>
>>>
>>> The whole reason we didn't do this before was because we wanted to work
>>> with legacy equipment.  But Cullen's claim is that there is no legacy
>>> equipment that can speak ICE (and DTLS) but not rtcp-mux.
>>>
>>> So is there any remaining reason that we cannot require rtcp-mux?
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>
>>>
>>
>

--047d7bf19850c11f07051c1cf42b
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<div dir=3D"ltr">&quot;<span style=3D"font-family:arial,helvetica,sans-seri=
f;font-size:12.8px">free-to-not-support-non-muxed-</span><span style=3D"fon=
t-family:arial,helvetica,sans-serif;font-size:12.8px">rtcp.</span>&quot;<di=
v><br></div><div>[BA] OK. =C2=A0 This would involve changing &quot;are also=
 REQUIRED to&quot; to &quot;MAY&quot; in the following sentence within RTP-=
Usage Section 4.5:=C2=A0</div><div><br></div><div><pre class=3D"newpage">  =
 For backwards compatibility, implementations are also REQUIRED to support
   RTP and RTCP sent on separate transport-layer flows.
</pre></div></div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote"=
>On Thu, Jul 30, 2015 at 12:35 PM, Peter Thatcher <span dir=3D"ltr">&lt;<a =
href=3D"mailto:pthatcher@google.com" target=3D"_blank">pthatcher@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"><div dir=3D"ltr"><=
div class=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-serif=
">I didn&#39;t quite mean mandatory-to-use.=C2=A0 If an rtcweb endpoint wan=
ts to use non-muxed RTCP, it could go ahead and do that, so it&#39;s not qu=
ite mandatory-to-use-rtcp-mux, but instead free-to-not-support-non-muxed-rt=
cp.</div><div class=3D"gmail_default" style=3D"font-family:arial,helvetica,=
sans-serif"><br></div><div class=3D"gmail_default" style=3D"font-family:ari=
al,helvetica,sans-serif">Perhaps my proposal should instead be: =C2=A0rtcwe=
b endpoints MAY make<span style=3D"font-size:12.8px">=C2=A0RtcpMuxPolicy =
=3D=3D &quot;required&quot; the default and MAY remove support for RtcpMuxP=
olicy =3D=3D &quot;negotiate&quot;.</span></div><div class=3D"gmail_default=
" style=3D"font-family:arial,helvetica,sans-serif"><span style=3D"font-size=
:12.8px"><br></span></div><div class=3D"gmail_default" style=3D"font-family=
:arial,helvetica,sans-serif"><span style=3D"font-size:12.8px">That would le=
ave the door open to endpoints that want to support non-muxed RTCP, which m=
andatory-to-use would not.=C2=A0 But it would also leave the door open to e=
ndpoints not having to support non-muxed RTCP.</span></div><div><div class=
=3D"h5"><div class=3D"gmail_default" style=3D"font-family:arial,helvetica,s=
ans-serif"><br></div><div class=3D"gmail_default" style=3D"font-family:aria=
l,helvetica,sans-serif"><br></div><div class=3D"gmail_extra"><br><div class=
=3D"gmail_quote">On Thu, Jul 30, 2015 at 12:28 PM, Bernard Aboba <span dir=
=3D"ltr">&lt;<a href=3D"mailto:bernard.aboba@gmail.com" target=3D"_blank">b=
ernard.aboba@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 dir=3D"ltr"><span><div>Peter said: </div><div><br></div><div>&quot=
;So is there any remaining reason that we cannot require rtcp-mux?&quot;</d=
iv><div><br></div></span><div>[BA] To be clear, you are talking about makin=
g RTP/RTCP multiplexing=C2=A0mandatory-to-use? </div><div><br></div><div>Th=
e RTP-Usage document already=C2=A0mandates support for RFC 5761.=C2=A0 From=
 Section 4.5: </div><div><br></div><div>=C2=A0=C2=A0 implementations are RE=
QUIRED to support multiplexing RTP data packets<br>=C2=A0=C2=A0 and RTCP co=
ntrol packets on a single transport-layer flow [<a title=3D"&quot;Multiplex=
ing RTP Data and Control Packets on a Single Port&quot;" href=3D"https://to=
ols.ietf.org/html/rfc5761" target=3D"_blank"><font color=3D"#0066cc">RFC576=
1</font></a>].</div></div><div class=3D"gmail_extra"><br><div class=3D"gmai=
l_quote"><div><div>On Thu, Jul 30, 2015 at 12:17 PM, Peter Thatcher <span d=
ir=3D"ltr">&lt;<a href=3D"mailto:pthatcher@google.com" target=3D"_blank">pt=
hatcher@google.com</a>&gt;</span> wrote:<br></div></div><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex"><div><div><div dir=3D"ltr"><div style=3D"font-family:arial,he=
lvetica,sans-serif">At IETF 93 last week, =C2=A0Cullen pointed out that he =
looked into ICE endpoints and found that all of them support rtcp-mux.=C2=
=A0 Inspired by that, I went around and asked about a dozen people &quot;wo=
uld you be OK if we removed support for non-muxed RTCP?&quot; =C2=A0The ran=
ge of responses went from &quot;I&#39;m OK with it if Cullen is&quot; to &q=
uot;yeah, that would be great&quot;.=C2=A0 No one said no.=C2=A0 Cullen sai=
d yes. =C2=A0</div><div style=3D"font-family:arial,helvetica,sans-serif"><b=
r></div><div style=3D"font-family:arial,helvetica,sans-serif">For me, remov=
ing support for non-muxed RTCP would be like Christmas in July.=C2=A0 The a=
mount of gross code in our implementation just to support this case is very=
 large, especially considering the very small benefit it gives us.=C2=A0 Ta=
lking to the developers at Mozilla, it sounds like they have a similar cost=
 in code complexity.=C2=A0 I&#39;m guessing every ICE stack does.=C2=A0 We =
could clean up a lot of code complexity in a lot of code bases by requiring=
 rtcp-mux. =C2=A0</div><div style=3D"font-family:arial,helvetica,sans-serif=
"><br></div><div style=3D"font-family:arial,helvetica,sans-serif"><br></div=
><div style=3D"font-family:arial,helvetica,sans-serif">So here&#39;s my pro=
posal: =C2=A0make RtcpMuxPolicy =3D=3D &quot;required&quot; be not only the=
 default, but the only RtcpMuxPolicy available. =C2=A0 All rtcweb endpoints=
 must support rtcp-mux. =C2=A0 An rtcweb endpoint would be free to fail if =
the remote endpoint does not support rtcp-mux.</div><div style=3D"font-fami=
ly:arial,helvetica,sans-serif"><br></div><div style=3D"font-family:arial,he=
lvetica,sans-serif"><br></div><div style=3D"font-family:arial,helvetica,san=
s-serif"><br></div><div style=3D"font-family:arial,helvetica,sans-serif">Th=
e whole reason we didn&#39;t do this before was because we wanted to work w=
ith legacy equipment.=C2=A0 But Cullen&#39;s claim is that there is no lega=
cy equipment that can speak ICE (and DTLS) but not rtcp-mux.=C2=A0</div><di=
v style=3D"font-family:arial,helvetica,sans-serif"><br></div><div style=3D"=
font-family:arial,helvetica,sans-serif">So is there any remaining reason th=
at we cannot require rtcp-mux?</div><div style=3D"font-family:arial,helveti=
ca,sans-serif"><br></div><div style=3D"font-family:arial,helvetica,sans-ser=
if"><br></div><div style=3D"font-family:arial,helvetica,sans-serif"><br></d=
iv></div>
<br></div></div>_______________________________________________<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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>
</blockquote></div><br></div></div></div></div>
</blockquote></div><br></div>

--047d7bf19850c11f07051c1cf42b--


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Date: Thu, 30 Jul 2015 15:56:09 -0400
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Cc: Colin Perkins <csp@csperkins.org>
Subject: Re: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it would be Christmas in July!)
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--------------080002010802050801070102
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On 7/30/2015 3:17 PM, Peter Thatcher wrote:
> So here's my proposal:  make RtcpMuxPolicy == "required" be not only 
> the default, but the only RtcpMuxPolicy available.   All rtcweb 
> endpoints must support rtcp-mux.   An rtcweb endpoint would be free to 
> fail if the remote endpoint does not support rtcp-mux.
>
>
>
> The whole reason we didn't do this before was because we wanted to 
> work with legacy equipment.  But Cullen's claim is that there is no 
> legacy equipment that can speak ICE (and DTLS) but not rtcp-mux.
>
> So is there any remaining reason that we cannot require rtcp-mux?

I agree it would clean up code considerably, and simplify candidate 
gathering, etc.

This means any gateways from WebRTC/ICE/etc to legacy RTP devices (PSTN 
gateways, whatever) will need to be able to do rtcp-mux on one side, and 
non-muxed on the other, and deal with possible payload rewriting to 
avoid conflicts.  Or punt.  I just want to make that clear; I am ok with 
this restriction (to require use of (not just support of) RTCP-mux in 
WebRTC).

Cc: Colin

-- 
Randell Jesup -- rjesup a t mozilla d o t com
Please please please don't email randell-ietf@jesup.org!  Way too much spam


--------------080002010802050801070102
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<html>
  <head>
    <meta content=3D"text/html; charset=3Dwindows-1252"
      http-equiv=3D"Content-Type">
  </head>
  <body bgcolor=3D"#FFFFFF" text=3D"#000000">
    <div class=3D"moz-cite-prefix">On 7/30/2015 3:17 PM, Peter Thatcher
      wrote:<br>
    </div>
    <blockquote
cite=3D"mid:CAJrXDUGs_+AKNHhaJjB5udr9TZv7oFpJ7h3cJ_ToXUfUC5TEpQ@mail.gmai=
l.com"
      type=3D"cite">
      <div dir=3D"ltr">So here's my proposal: =A0make RtcpMuxPolicy =3D=3D
        "required" be not only the default, but the only RtcpMuxPolicy
        available. =A0 All rtcweb endpoints must support rtcp-mux. =A0 An
        rtcweb endpoint would be free to fail if the remote endpoint
        does not support rtcp-mux.
        <div class=3D"gmail_default"
          style=3D"font-family:arial,helvetica,sans-serif"><br>
        </div>
        <div class=3D"gmail_default"
          style=3D"font-family:arial,helvetica,sans-serif"><br>
        </div>
        <div class=3D"gmail_default"
          style=3D"font-family:arial,helvetica,sans-serif"><br>
        </div>
        <div class=3D"gmail_default"
          style=3D"font-family:arial,helvetica,sans-serif">The whole
          reason we didn't do this before was because we wanted to work
          with legacy equipment.=A0 But Cullen's claim is that there is n=
o
          legacy equipment that can speak ICE (and DTLS) but not
          rtcp-mux.=A0</div>
        <div class=3D"gmail_default"
          style=3D"font-family:arial,helvetica,sans-serif"><br>
        </div>
        <div class=3D"gmail_default"
          style=3D"font-family:arial,helvetica,sans-serif">So is there an=
y
          remaining reason that we cannot require rtcp-mux?</div>
      </div>
    </blockquote>
    <br>
    I agree it would clean up code considerably, and simplify candidate
    gathering, etc.=A0 <br>
    <br>
    This means any gateways from WebRTC/ICE/etc to legacy RTP devices
    (PSTN gateways, whatever) will need to be able to do rtcp-mux on one
    side, and non-muxed on the other, and deal with possible payload
    rewriting to avoid conflicts.=A0 Or punt.=A0 I just want to make that
    clear; I am ok with this restriction (to require use of (not just
    support of) RTCP-mux in WebRTC).<br>
    <br>
    Cc: Colin<br>
    <pre class=3D"moz-signature" cols=3D"72">--=20
Randell Jesup -- rjesup a t mozilla d o t com
Please please please don't email <a class=3D"moz-txt-link-abbreviated" hr=
ef=3D"mailto:randell-ietf@jesup.org">randell-ietf@jesup.org</a>!  Way too=
 much spam
</pre>
  </body>
</html>

--------------080002010802050801070102--


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--f46d0442806e1a6153051c1d28b5
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Peter said:

"Making it a part of the W3C interim sounds like a good idea to me.  We've
done both IETF and W3C at interims in the past, and they have been
productive (perhaps more productive than the non-interim meetings)."

[BA] That could be made to work, though to piggyback on the WEBRTC WG
interim in September, it would be necessary to quickly understand the
schedule/attendance impact on the existing plans (e.g. impact on the
estimated attendance, whether the existing dates would stay the same or
would need extending, etc.).

On Thu, Jul 30, 2015 at 11:53 AM, Peter Thatcher <pthatcher@google.com>
wrote:

> Making it a part of the W3C interim sounds like a good idea to me.  We've
> done both IETF and W3C at interims in the past, and they have been
> productive (perhaps more productive than the non-interim meetings).
>
> On Thu, Jul 30, 2015 at 10:43 AM, Ted Hardie <ted.ietf@gmail.com> wrote:
>
>> The chairs have discussed how to make progress between now and Yokohama,
>> and it seems like another focused 2-3 hour meeting would be useful.  That
>> might be a completely virtual interim or one associated with the WebRTC
>> interim, but with a remote participation component.
>>
>> If you have thoughts on that, please share them between now and August
>> 7th, 2015.
>>
>> thanks,
>>
>> Ted, Cullen, Sean
>>
>> _______________________________________________
>> 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
>
>

--f46d0442806e1a6153051c1d28b5
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<div dir=3D"ltr">Peter said:=C2=A0<div><br></div><div>&quot;<span style=3D"=
font-family:arial,helvetica,sans-serif;font-size:12.8px">Making it a part o=
f the W3C interim sounds like a good idea to me.=C2=A0 We&#39;ve done both =
IETF and W3C at interims in the past, and they have been productive (perhap=
s more productive than the non-interim meetings).</span>&quot;</div><div><b=
r></div><div>[BA] That could be made to work, though to piggyback on the WE=
BRTC WG interim in September, it would be necessary to quickly understand t=
he schedule/attendance impact on the existing plans (e.g. impact on the est=
imated attendance, whether the existing dates would stay the same or would =
need extending, etc.).=C2=A0</div></div><div class=3D"gmail_extra"><br><div=
 class=3D"gmail_quote">On Thu, Jul 30, 2015 at 11:53 AM, Peter Thatcher <sp=
an dir=3D"ltr">&lt;<a href=3D"mailto:pthatcher@google.com" target=3D"_blank=
">pthatcher@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:1=
ex"><div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:aria=
l,helvetica,sans-serif">Making it a part of the W3C interim sounds like a g=
ood idea to me.=C2=A0 We&#39;ve done both IETF and W3C at interims in the p=
ast, and they have been productive (perhaps more productive than the non-in=
terim meetings).</div><div class=3D"gmail_extra"><br><div class=3D"gmail_qu=
ote"><div><div class=3D"h5">On Thu, Jul 30, 2015 at 10:43 AM, Ted Hardie <s=
pan dir=3D"ltr">&lt;<a href=3D"mailto:ted.ietf@gmail.com" target=3D"_blank"=
>ted.ietf@gmail.com</a>&gt;</span> wrote:<br></div></div><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex"><div><div class=3D"h5"><div dir=3D"ltr"><div class=3D"gmail_d=
efault" style=3D"font-family:tahoma,sans-serif;font-size:small">The chairs =
have discussed how to make progress between now and Yokohama, and it seems =
like another focused 2-3 hour meeting would be useful.=C2=A0 That might be =
a completely virtual interim or one associated with the WebRTC interim, but=
 with a remote participation component.=C2=A0 <br><br></div><div class=3D"g=
mail_default" style=3D"font-family:tahoma,sans-serif;font-size:small">If yo=
u have thoughts on that, please share them between now and August 7th, 2015=
.<br><br></div><div class=3D"gmail_default" style=3D"font-family:tahoma,san=
s-serif;font-size:small">thanks,<br><br></div><div class=3D"gmail_default" =
style=3D"font-family:tahoma,sans-serif;font-size:small">Ted, Cullen, Sean<b=
r></div></div>
<br></div></div>_______________________________________________<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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>

--f46d0442806e1a6153051c1d28b5--


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From: Ted Hardie <ted.ietf@gmail.com>
To: Bernard Aboba <bernard.aboba@gmail.com>
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Cc: Cullen Jennings <fluffy@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Interim discussion?
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On Thu, Jul 30, 2015 at 1:00 PM, Bernard Aboba <bernard.aboba@gmail.com>
wrote

>
> [BA] That could be made to work, though to piggyback on the WEBRTC WG
> interim in September, it would be necessary to quickly understand the
> schedule/attendance impact on the existing plans (e.g. impact on the
> estimated attendance, whether the existing dates would stay the same or
> would need extending, etc.).
>
>
=E2=80=8BOne option here would be to make it a virtual interim within the u=
sual
terms (conference-based call), but understand that lots of attendees might
happen to be in the same place.  That would be particularly useful if we
could do the virtual one at some mid-point of the W3C f2f interim, since we
seem to run into issue passing a fair amount during these meetings.  We
could add a physical interim on one side or the other (maybe nearby, if
Microsoft could no host both), but it is close timing to do so.

Ted=E2=80=8B

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<div dir=3D"ltr"><div class=3D"gmail_extra">On Thu, Jul 30, 2015 at 1:00 PM=
, Bernard Aboba <span dir=3D"ltr">&lt;<a href=3D"mailto:bernard.aboba@gmail=
.com" target=3D"_blank">bernard.aboba@gmail.com</a>&gt;</span> wrote<div cl=
ass=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0=
 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><span c=
lass=3D""><div><br></div></span><div>[BA] That could be made to work, thoug=
h to piggyback on the WEBRTC WG interim in September, it would be necessary=
 to quickly understand the schedule/attendance impact on the existing plans=
 (e.g. impact on the estimated attendance, whether the existing dates would=
 stay the same or would need extending, etc.).=C2=A0</div></div><br></block=
quote></div><br><div class=3D"gmail_default" style=3D"font-family:tahoma,sa=
ns-serif;font-size:small">=E2=80=8BOne option here would be to make it a vi=
rtual interim within the usual terms (conference-based call), but understan=
d that lots of attendees might happen to be in the same place.=C2=A0 That w=
ould be particularly useful if we could do the virtual one at some mid-poin=
t of the W3C f2f interim, since we seem to run into issue passing a fair am=
ount during these meetings.=C2=A0 We could add a physical interim on one si=
de or the other (maybe nearby, if Microsoft could no host both), but it is =
close timing to do so.<br><br></div><div class=3D"gmail_default" style=3D"f=
ont-family:tahoma,sans-serif;font-size:small">Ted=E2=80=8B</div><br></div><=
/div>

--f46d043c825a6b6ab6051c1d79a1--


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From: Roman Shpount <roman@telurix.com>
To: Peter Thatcher <pthatcher@google.com>
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Cc: "<rtcweb@ietf.org>" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it would be Christmas in July!)
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As long as webrtc compliant end point MUST offer rtcp-mux and MUST always
accepted if offered by the remote end point I am ok with this.

Compliant end points MAY offer additional candidates for RTCP and accept
offers and answers without rtcp-mux.
_____________
Roman Shpount

On Thu, Jul 30, 2015 at 3:35 PM, Peter Thatcher <pthatcher@google.com>
wrote:

> I didn't quite mean mandatory-to-use.  If an rtcweb endpoint wants to use
> non-muxed RTCP, it could go ahead and do that, so it's not quite
> mandatory-to-use-rtcp-mux, but instead free-to-not-support-non-muxed-rtcp.
>
> Perhaps my proposal should instead be:  rtcweb endpoints MAY make RtcpMuxPolicy
> == "required" the default and MAY remove support for RtcpMuxPolicy ==
> "negotiate".
>
> That would leave the door open to endpoints that want to support non-muxed
> RTCP, which mandatory-to-use would not.  But it would also leave the door
> open to endpoints not having to support non-muxed RTCP.
>
>

--001a11403fd43a5fbc051c1e2404
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<div dir=3D"ltr">As long as webrtc compliant end point MUST offer rtcp-mux =
and MUST always accepted if offered by the remote end point I am ok with th=
is.<div><br></div><div>Compliant end points MAY offer additional candidates=
 for RTCP and accept offers and answers without rtcp-mux.<div class=3D"gmai=
l_extra"><div><div class=3D"gmail_signature">_____________<br>Roman Shpount=
</div></div>
<br><div class=3D"gmail_quote">On Thu, Jul 30, 2015 at 3:35 PM, Peter Thatc=
her <span dir=3D"ltr">&lt;<a href=3D"mailto:pthatcher@google.com" target=3D=
"_blank">pthatcher@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"><div dir=3D"ltr"><div style=3D"font-family:arial,helvetica,sans-=
serif">I didn&#39;t quite mean mandatory-to-use.=C2=A0 If an rtcweb endpoin=
t wants to use non-muxed RTCP, it could go ahead and do that, so it&#39;s n=
ot quite mandatory-to-use-rtcp-mux, but instead free-to-not-support-non-mux=
ed-rtcp.</div><div style=3D"font-family:arial,helvetica,sans-serif"><br></d=
iv><div style=3D"font-family:arial,helvetica,sans-serif">Perhaps my proposa=
l should instead be: =C2=A0rtcweb endpoints MAY make<span style=3D"font-siz=
e:12.8px">=C2=A0RtcpMuxPolicy =3D=3D &quot;required&quot; the default and M=
AY remove support for RtcpMuxPolicy =3D=3D &quot;negotiate&quot;.</span></d=
iv><div style=3D"font-family:arial,helvetica,sans-serif"><span style=3D"fon=
t-size:12.8px"><br></span></div><div style=3D"font-family:arial,helvetica,s=
ans-serif"><span style=3D"font-size:12.8px">That would leave the door open =
to endpoints that want to support non-muxed RTCP, which mandatory-to-use wo=
uld not.=C2=A0 But it would also leave the door open to endpoints not havin=
g to support non-muxed RTCP.</span></div><div><div class=3D"h5"><div style=
=3D"font-family:arial,helvetica,sans-serif"><br></div></div></div></div></b=
lockquote></div></div></div></div>

--001a11403fd43a5fbc051c1e2404--


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From: Eric Rescorla <ekr@rtfm.com>
Date: Fri, 31 Jul 2015 05:04:18 +0200
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Subject: Re: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it would be Christmas in July!)
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On Thu, Jul 30, 2015 at 9:17 PM, Peter Thatcher <pthatcher@google.com>
wrote:

> At IETF 93 last week,  Cullen pointed out that he looked into ICE
> endpoints and found that all of them support rtcp-mux.  Inspired by that, I
> went around and asked about a dozen people "would you be OK if we removed
> support for non-muxed RTCP?"  The range of responses went from "I'm OK with
> it if Cullen is" to "yeah, that would be great".  No one said no.  Cullen
> said yes.
>
> For me, removing support for non-muxed RTCP would be like Christmas in
> July.
>

Realistically, August at this point.


So here's my proposal:  make RtcpMuxPolicy == "required" be not only the
> default, but the only RtcpMuxPolicy available.   All rtcweb endpoints must
> support rtcp-mux.
>

This seems reasonable.



> An rtcweb endpoint would be free to fail if the remote endpoint does not
> support rtcp-mux.
>

Given the current specification, I believe this would need to be strong.
I.e., an rtcweb
endpoint *must* fail if the remote endpoint does not support rtcp-mux. As
should
be obvious, this only applies when the rtcweb endpoint is an answerer since
with
the above setting, you will not provide an offer that the other side can
accept, but
the spec says:

"When acting as answerer, the browser will reject any m= section that does
not provide an "a=rtcp-mux" attribute."

The reason for that is to avoid asymmetrical situations in which endpoint NM
(non-mux) makes an offer to endpoint M (mux) which is accepted, but then
re-offers from NM to M fail.

So I believe the best version of this proposal is simply requiring rtcp-mux
entirely.

-Ekr



> The whole reason we didn't do this before was because we wanted to work
> with legacy equipment.  But Cullen's claim is that there is no legacy
> equipment that can speak ICE (and DTLS) but not rtcp-mux.
>
> So is there any remaining reason that we cannot require rtcp-mux?
>
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Thu, Jul 30, 2015 at 9:17 PM, Peter Thatcher <span dir=3D"ltr">&lt;<=
a href=3D"mailto:pthatcher@google.com" target=3D"_blank">pthatcher@google.c=
om</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"marg=
in:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"=
><div style=3D"font-family:arial,helvetica,sans-serif">At IETF 93 last week=
, =C2=A0Cullen pointed out that he looked into ICE endpoints and found that=
 all of them support rtcp-mux.=C2=A0 Inspired by that, I went around and as=
ked about a dozen people &quot;would you be OK if we removed support for no=
n-muxed RTCP?&quot; =C2=A0The range of responses went from &quot;I&#39;m OK=
 with it if Cullen is&quot; to &quot;yeah, that would be great&quot;.=C2=A0=
 No one said no.=C2=A0 Cullen said yes. =C2=A0</div><div style=3D"font-fami=
ly:arial,helvetica,sans-serif"><br></div><div style=3D"font-family:arial,he=
lvetica,sans-serif">For me, removing support for non-muxed RTCP would be li=
ke Christmas in July.</div></div></blockquote><div><br></div><div>Realistic=
ally, August at this point.</div><div><br></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 dir=3D"ltr"><div style=3D"font-family:arial,helveti=
ca,sans-serif">So here&#39;s my proposal: =C2=A0make RtcpMuxPolicy =3D=3D &=
quot;required&quot; be not only the default, but the only RtcpMuxPolicy ava=
ilable. =C2=A0 All rtcweb endpoints must support rtcp-mux. =C2=A0</div></di=
v></blockquote><div><br></div><div>This seems reasonable.</div><div><br></d=
iv><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0=
 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div st=
yle=3D"font-family:arial,helvetica,sans-serif"> An rtcweb endpoint would be=
 free to fail if the remote endpoint does not support rtcp-mux.</div></div>=
</blockquote><div><br></div><div>Given the current specification, I believe=
 this would need to be strong. I.e., an rtcweb</div><div>endpoint *must* fa=
il if the remote endpoint does not support rtcp-mux. As should=C2=A0</div><=
div>be obvious, this only applies when the rtcweb endpoint is an answerer s=
ince with</div><div>the above setting, you will not provide an offer that t=
he other side can accept, but</div><div>the spec says:</div><div><br></div>=
<div>&quot;When acting as answerer, the browser will reject any m=3D sectio=
n that does not provide an &quot;a=3Drtcp-mux&quot; attribute.&quot;</div><=
div><br></div><div>The reason for that is to avoid asymmetrical situations =
in which endpoint NM</div><div>(non-mux) makes an offer to endpoint M (mux)=
 which is accepted, but then</div><div>re-offers from NM to M fail.</div><d=
iv><br></div><div>So I believe the best version of this proposal is simply =
requiring rtcp-mux</div><div>entirely.</div><div><br></div><div>-Ekr</div><=
div><br></div><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"m=
argin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"l=
tr"><div style=3D"font-family:arial,helvetica,sans-serif"></div><div style=
=3D"font-family:arial,helvetica,sans-serif">The whole reason we didn&#39;t =
do this before was because we wanted to work with legacy equipment.=C2=A0 B=
ut Cullen&#39;s claim is that there is no legacy equipment that can speak I=
CE (and DTLS) but not rtcp-mux.=C2=A0</div><div style=3D"font-family:arial,=
helvetica,sans-serif"><br></div><div style=3D"font-family:arial,helvetica,s=
ans-serif">So is there any remaining reason that we cannot require rtcp-mux=
?</div><div style=3D"font-family:arial,helvetica,sans-serif"><br></div><div=
 style=3D"font-family:arial,helvetica,sans-serif"><br></div><div style=3D"f=
ont-family:arial,helvetica,sans-serif"><br></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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div></div>

--047d7b10c903c6d911051c231529--


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From: Peter Thatcher <pthatcher@google.com>
Date: Thu, 30 Jul 2015 21:01:15 -0700
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To: Eric Rescorla <ekr@rtfm.com>
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Cc: "<rtcweb@ietf.org>" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it would be Christmas in July!)
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--047d7b33d4aa653b37051c23e117
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On Thu, Jul 30, 2015 at 8:04 PM, Eric Rescorla <ekr@rtfm.com> wrote:

>
>
> On Thu, Jul 30, 2015 at 9:17 PM, Peter Thatcher <pthatcher@google.com>
> wrote:
>
>> At IETF 93 last week,  Cullen pointed out that he looked into ICE
>> endpoints and found that all of them support rtcp-mux.  Inspired by that=
, I
>> went around and asked about a dozen people "would you be OK if we remove=
d
>> support for non-muxed RTCP?"  The range of responses went from "I'm OK w=
ith
>> it if Cullen is" to "yeah, that would be great".  No one said no.  Culle=
n
>> said yes.
>>
>> For me, removing support for non-muxed RTCP would be like Christmas in
>> July.
>>
>
> Realistically, August at this point.
>

=E2=80=8BClose enough :)
=E2=80=8B


>
>
> So here's my proposal:  make RtcpMuxPolicy =3D=3D "required" be not only =
the
>> default, but the only RtcpMuxPolicy available.   All rtcweb endpoints mu=
st
>> support rtcp-mux.
>>
>
> This seems reasonable.
>
>
>
>> An rtcweb endpoint would be free to fail if the remote endpoint does not
>> support rtcp-mux.
>>
>
> Given the current specification, I believe this would need to be strong.
> I.e., an rtcweb
> endpoint *must* fail if the remote endpoint does not support rtcp-mux. As
> should
> be obvious, this only applies when the rtcweb endpoint is an answerer
> since with
> the above setting, you will not provide an offer that the other side can
> accept, but
> the spec says:
>
> "When acting as answerer, the browser will reject any m=3D section that d=
oes
> not provide an "a=3Drtcp-mux" attribute."
>
> The reason for that is to avoid asymmetrical situations in which endpoint
> NM
> (non-mux) makes an offer to endpoint M (mux) which is accepted, but then
> re-offers from NM to M fail.
>
> So I believe the best version of this proposal is simply requiring rtcp-m=
ux
> entirely.
>

=E2=80=8BThat makes sense, and I would be happy with "reject any m=3D secti=
on without
a=3Drtcp-mux".=E2=80=8B



>
> -Ekr
>
>
>
>> The whole reason we didn't do this before was because we wanted to work
>> with legacy equipment.  But Cullen's claim is that there is no legacy
>> equipment that can speak ICE (and DTLS) but not rtcp-mux.
>>
>> So is there any remaining reason that we cannot require rtcp-mux?
>>
>>
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>>
>

--047d7b33d4aa653b37051c23e117
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<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:arial,he=
lvetica,sans-serif"><br></div><div class=3D"gmail_extra"><br><div class=3D"=
gmail_quote">On Thu, Jul 30, 2015 at 8:04 PM, Eric Rescorla <span dir=3D"lt=
r">&lt;<a href=3D"mailto:ekr@rtfm.com" target=3D"_blank">ekr@rtfm.com</a>&g=
t;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0=
 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><br><di=
v class=3D"gmail_extra"><br><div class=3D"gmail_quote"><span class=3D"">On =
Thu, Jul 30, 2015 at 9:17 PM, Peter Thatcher <span dir=3D"ltr">&lt;<a href=
=3D"mailto:pthatcher@google.com" target=3D"_blank">pthatcher@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"><div dir=3D"ltr"><div =
style=3D"font-family:arial,helvetica,sans-serif">At IETF 93 last week, =C2=
=A0Cullen pointed out that he looked into ICE endpoints and found that all =
of them support rtcp-mux.=C2=A0 Inspired by that, I went around and asked a=
bout a dozen people &quot;would you be OK if we removed support for non-mux=
ed RTCP?&quot; =C2=A0The range of responses went from &quot;I&#39;m OK with=
 it if Cullen is&quot; to &quot;yeah, that would be great&quot;.=C2=A0 No o=
ne said no.=C2=A0 Cullen said yes. =C2=A0</div><div style=3D"font-family:ar=
ial,helvetica,sans-serif"><br></div><div style=3D"font-family:arial,helveti=
ca,sans-serif">For me, removing support for non-muxed RTCP would be like Ch=
ristmas in July.</div></div></blockquote><div><br></div></span><div>Realist=
ically, August at this point.</div></div></div></div></blockquote><div><br>=
</div><div><div class=3D"gmail_default" style=3D"font-family:arial,helvetic=
a,sans-serif;display:inline">=E2=80=8BClose enough :)</div></div><div><div =
class=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-serif;dis=
play:inline">=E2=80=8B</div>=C2=A0</div><blockquote class=3D"gmail_quote" s=
tyle=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div=
 dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote"><span cl=
ass=3D""><div><br></div><div><br></div><blockquote class=3D"gmail_quote" st=
yle=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div =
dir=3D"ltr"><div style=3D"font-family:arial,helvetica,sans-serif">So here&#=
39;s my proposal: =C2=A0make RtcpMuxPolicy =3D=3D &quot;required&quot; be n=
ot only the default, but the only RtcpMuxPolicy available. =C2=A0 All rtcwe=
b endpoints must support rtcp-mux. =C2=A0</div></div></blockquote><div><br>=
</div></span><div>This seems reasonable.</div><span class=3D""><div><br></d=
iv><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0=
 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div st=
yle=3D"font-family:arial,helvetica,sans-serif"> An rtcweb endpoint would be=
 free to fail if the remote endpoint does not support rtcp-mux.</div></div>=
</blockquote><div><br></div></span><div>Given the current specification, I =
believe this would need to be strong. I.e., an rtcweb</div><div>endpoint *m=
ust* fail if the remote endpoint does not support rtcp-mux. As should=C2=A0=
</div><div>be obvious, this only applies when the rtcweb endpoint is an ans=
werer since with</div><div>the above setting, you will not provide an offer=
 that the other side can accept, but</div><div>the spec says:</div><div><br=
></div><div>&quot;When acting as answerer, the browser will reject any m=3D=
 section that does not provide an &quot;a=3Drtcp-mux&quot; attribute.&quot;=
</div><div><br></div><div>The reason for that is to avoid asymmetrical situ=
ations in which endpoint NM</div><div>(non-mux) makes an offer to endpoint =
M (mux) which is accepted, but then</div><div>re-offers from NM to M fail.<=
/div><div><br></div><div>So I believe the best version of this proposal is =
simply requiring rtcp-mux</div><div>entirely.</div></div></div></div></bloc=
kquote><div><br></div><div><div class=3D"gmail_default" style=3D"font-famil=
y:arial,helvetica,sans-serif">=E2=80=8BThat makes sense, and I would be hap=
py with &quot;reject any m=3D section without a=3Drtcp-mux&quot;.=E2=80=8B<=
/div><br></div><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"=
margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"=
ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote"><div><br></div><=
div>-Ekr</div><div><br></div><div>=C2=A0</div><blockquote class=3D"gmail_qu=
ote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex=
"><span class=3D""><div dir=3D"ltr"><div style=3D"font-family:arial,helveti=
ca,sans-serif"></div><div style=3D"font-family:arial,helvetica,sans-serif">=
The whole reason we didn&#39;t do this before was because we wanted to work=
 with legacy equipment.=C2=A0 But Cullen&#39;s claim is that there is no le=
gacy equipment that can speak ICE (and DTLS) but not rtcp-mux.=C2=A0</div><=
div style=3D"font-family:arial,helvetica,sans-serif"><br></div><div style=
=3D"font-family:arial,helvetica,sans-serif">So is there any remaining reaso=
n that we cannot require rtcp-mux?</div><div style=3D"font-family:arial,hel=
vetica,sans-serif"><br></div><div style=3D"font-family:arial,helvetica,sans=
-serif"><br></div><div style=3D"font-family:arial,helvetica,sans-serif"><br=
></div></div>
<br></span><span class=3D"">_______________________________________________=
<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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></span></blockquote></div><br></div></div>
</blockquote></div><br></div></div>

--047d7b33d4aa653b37051c23e117--


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From: "Asveren, Tolga" <tasveren@sonusnet.com>
To: Eric Rescorla <ekr@rtfm.com>, Peter Thatcher <pthatcher@google.com>
Thread-Topic: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it would be Christmas in July!)
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Subject: Re: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it would be Christmas in July!)
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"The whole reason we didn't do this before was because we wanted to work wi=
th legacy equipment.  But Cullen's claim is that there is no legacy equipme=
nt that can speak ICE (and DTLS) but not rtcp-mux."


I am not sure whether this is entirely true, especially on GW front. It cou=
ld be a better idea to be lenient for enhanced interoperability and making =
RTCWeb more deployable in practice. Can't the decision of being firm on usi=
ng rtcp-mux left to the WebRTC endpoint with signaling aspects supporting n=
egotiation of the capability?


Thanks,

Tolga


________________________________
From: rtcweb <rtcweb-bounces@ietf.org> on behalf of Eric Rescorla <ekr@rtfm=
.com>
Sent: Thursday, July 30, 2015 11:04 PM
To: Peter Thatcher
Cc: <rtcweb@ietf.org>
Subject: Re: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it=
 would be Christmas in July!)



On Thu, Jul 30, 2015 at 9:17 PM, Peter Thatcher <pthatcher@google.com<mailt=
o:pthatcher@google.com>> wrote:
At IETF 93 last week,  Cullen pointed out that he looked into ICE endpoints=
 and found that all of them support rtcp-mux.  Inspired by that, I went aro=
und and asked about a dozen people "would you be OK if we removed support f=
or non-muxed RTCP?"  The range of responses went from "I'm OK with it if Cu=
llen is" to "yeah, that would be great".  No one said no.  Cullen said yes.

For me, removing support for non-muxed RTCP would be like Christmas in July=
.

Realistically, August at this point.


So here's my proposal:  make RtcpMuxPolicy =3D=3D "required" be not only th=
e default, but the only RtcpMuxPolicy available.   All rtcweb endpoints mus=
t support rtcp-mux.

This seems reasonable.


An rtcweb endpoint would be free to fail if the remote endpoint does not su=
pport rtcp-mux.

Given the current specification, I believe this would need to be strong. I.=
e., an rtcweb
endpoint *must* fail if the remote endpoint does not support rtcp-mux. As s=
hould
be obvious, this only applies when the rtcweb endpoint is an answerer since=
 with
the above setting, you will not provide an offer that the other side can ac=
cept, but
the spec says:

"When acting as answerer, the browser will reject any m=3D section that doe=
s not provide an "a=3Drtcp-mux" attribute."

The reason for that is to avoid asymmetrical situations in which endpoint N=
M
(non-mux) makes an offer to endpoint M (mux) which is accepted, but then
re-offers from NM to M fail.

So I believe the best version of this proposal is simply requiring rtcp-mux
entirely.

-Ekr


The whole reason we didn't do this before was because we wanted to work wit=
h legacy equipment.  But Cullen's claim is that there is no legacy equipmen=
t that can speak ICE (and DTLS) but not rtcp-mux.

So is there any remaining reason that we cannot require rtcp-mux?




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



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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Diso-8859-=
1">
<style type=3D"text/css" style=3D"display:none;"><!-- P {margin-top:0;margi=
n-bottom:0;} --></style>
</head>
<body dir=3D"ltr">
<div id=3D"divtagdefaultwrapper" style=3D"font-size:12pt;color:#000000;back=
ground-color:#FFFFFF;font-family:Calibri,Arial,Helvetica,sans-serif;">
<p><span style=3D"color: rgb(33, 33, 33); font-family: arial, helvetica, sa=
ns-serif; font-size: 15px;">&quot;The whole reason we didn't do this before=
 was because we wanted to work with legacy equipment.&nbsp; But Cullen's cl=
aim is that there is no legacy equipment that
 can speak ICE (and DTLS) but not rtcp-mux.&quot;</span><br>
</p>
<p><span style=3D"color: rgb(33, 33, 33); font-family: arial, helvetica, sa=
ns-serif; font-size: 15px;"><br>
</span></p>
<p><font color=3D"#212121" face=3D"arial, helvetica, sans-serif"><span styl=
e=3D"font-size: 15px;">I am not sure whether this is entirely true, especia=
lly on GW front. It could be a better idea to be lenient for enhanced inter=
operability and making RTCWeb more&nbsp;deployable
 in practice. Can't the decision of being firm on using rtcp-mux left to th=
e WebRTC endpoint with signaling aspects supporting negotiation of the capa=
bility?</span></font></p>
<p><span style=3D"color: rgb(33, 33, 33); font-family: arial, helvetica, sa=
ns-serif; font-size: 15px;"><br>
</span></p>
<p><span style=3D"color: rgb(33, 33, 33); font-family: arial, helvetica, sa=
ns-serif; font-size: 15px;">Thanks,</span></p>
<p><span style=3D"color: rgb(33, 33, 33); font-family: arial, helvetica, sa=
ns-serif; font-size: 15px;">Tolga</span></p>
<br>
<br>
<div style=3D"color: rgb(0, 0, 0);">
<hr tabindex=3D"-1" style=3D"display:inline-block; width:98%">
<div id=3D"divRplyFwdMsg" dir=3D"ltr"><font face=3D"Calibri, sans-serif" co=
lor=3D"#000000" style=3D"font-size:11pt"><b>From:</b> rtcweb &lt;rtcweb-bou=
nces@ietf.org&gt; on behalf of Eric Rescorla &lt;ekr@rtfm.com&gt;<br>
<b>Sent:</b> Thursday, July 30, 2015 11:04 PM<br>
<b>To:</b> Peter Thatcher<br>
<b>Cc:</b> &lt;rtcweb@ietf.org&gt;<br>
<b>Subject:</b> Re: [rtcweb] Proposal: require rtcp-mux (remove non-muxed R=
TCP; it would be Christmas in July!)</font>
<div>&nbsp;</div>
</div>
<div>
<div dir=3D"ltr"><br>
<div class=3D"gmail_extra"><br>
<div class=3D"gmail_quote">On Thu, Jul 30, 2015 at 9:17 PM, Peter Thatcher =
<span dir=3D"ltr">
&lt;<a href=3D"mailto:pthatcher@google.com" target=3D"_blank">pthatcher@goo=
gle.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex; border-left:1=
px #ccc solid; padding-left:1ex">
<div dir=3D"ltr">
<div style=3D"font-family:arial,helvetica,sans-serif">At IETF 93 last week,=
 &nbsp;Cullen pointed out that he looked into ICE endpoints and found that =
all of them support rtcp-mux.&nbsp; Inspired by that, I went around and ask=
ed about a dozen people &quot;would you be OK if
 we removed support for non-muxed RTCP?&quot; &nbsp;The range of responses =
went from &quot;I'm OK with it if Cullen is&quot; to &quot;yeah, that would=
 be great&quot;.&nbsp; No one said no.&nbsp; Cullen said yes. &nbsp;</div>
<div style=3D"font-family:arial,helvetica,sans-serif"><br>
</div>
<div style=3D"font-family:arial,helvetica,sans-serif">For me, removing supp=
ort for non-muxed RTCP would be like Christmas in July.</div>
</div>
</blockquote>
<div><br>
</div>
<div>Realistically, August at this point.</div>
<div><br>
</div>
<div><br>
</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex; border-left:1=
px #ccc solid; padding-left:1ex">
<div dir=3D"ltr">
<div style=3D"font-family:arial,helvetica,sans-serif">So here's my proposal=
: &nbsp;make RtcpMuxPolicy =3D=3D &quot;required&quot; be not only the defa=
ult, but the only RtcpMuxPolicy available. &nbsp; All rtcweb endpoints must=
 support rtcp-mux. &nbsp;</div>
</div>
</blockquote>
<div><br>
</div>
<div>This seems reasonable.</div>
<div><br>
</div>
<div>&nbsp;</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex; border-left:1=
px #ccc solid; padding-left:1ex">
<div dir=3D"ltr">
<div style=3D"font-family:arial,helvetica,sans-serif">An rtcweb endpoint wo=
uld be free to fail if the remote endpoint does not support rtcp-mux.</div>
</div>
</blockquote>
<div><br>
</div>
<div>Given the current specification, I believe this would need to be stron=
g. I.e., an rtcweb</div>
<div>endpoint *must* fail if the remote endpoint does not support rtcp-mux.=
 As should&nbsp;</div>
<div>be obvious, this only applies when the rtcweb endpoint is an answerer =
since with</div>
<div>the above setting, you will not provide an offer that the other side c=
an accept, but</div>
<div>the spec says:</div>
<div><br>
</div>
<div>&quot;When acting as answerer, the browser will reject any m=3D sectio=
n that does not provide an &quot;a=3Drtcp-mux&quot; attribute.&quot;</div>
<div><br>
</div>
<div>The reason for that is to avoid asymmetrical situations in which endpo=
int NM</div>
<div>(non-mux) makes an offer to endpoint M (mux) which is accepted, but th=
en</div>
<div>re-offers from NM to M fail.</div>
<div><br>
</div>
<div>So I believe the best version of this proposal is simply requiring rtc=
p-mux</div>
<div>entirely.</div>
<div><br>
</div>
<div>-Ekr</div>
<div><br>
</div>
<div>&nbsp;</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex; border-left:1=
px #ccc solid; padding-left:1ex">
<div dir=3D"ltr">
<div style=3D"font-family:arial,helvetica,sans-serif"></div>
<div style=3D"font-family:arial,helvetica,sans-serif">The whole reason we d=
idn't do this before was because we wanted to work with legacy equipment.&n=
bsp; But Cullen's claim is that there is no legacy equipment that can speak=
 ICE (and DTLS) but not rtcp-mux.&nbsp;</div>
<div style=3D"font-family:arial,helvetica,sans-serif"><br>
</div>
<div style=3D"font-family:arial,helvetica,sans-serif">So is there any remai=
ning reason that we cannot require rtcp-mux?</div>
<div style=3D"font-family:arial,helvetica,sans-serif"><br>
</div>
<div style=3D"font-family:arial,helvetica,sans-serif"><br>
</div>
<div style=3D"font-family:arial,helvetica,sans-serif"><br>
</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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br>
</blockquote>
</div>
<br>
</div>
</div>
</div>
</div>
</div>
</body>
</html>

--_000_SN1PR0301MB1551E28A489A90A93FA2F8BDB28A0SN1PR0301MB1551_--


From nobody Fri Jul 31 03:47:15 2015
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Date: Fri, 31 Jul 2015 12:47:12 +0200
From: carlo von lynX <lynX@i.know.you.are.psyced.org>
To: rtcweb@ietf.org
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References: <55352067.6070007@nostrum.com> <20150421154657.GA4071@lo.psyced.org> <20150715123803.GA24976@lo.psyced.org> <2BCEF545-D322-4589-AC85-AA80E7F934D9@phonefromhere.com> <20150715164129.GA27105@lo.psyced.org> <AF6485D4-1579-45AF-937C-1682AC1AA645@phonefromhere.com> <20150716084225.GA29652@lo.psyced.org> <0AF88216-374E-4DBA-9DB5-3E55B358E6AA@phonefromhere.com>
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Subject: Re: [rtcweb] Prompt for mic, camera *and* network access
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Alright, thanks Tim for digging deeper and confirming that
we have a serious problem here. Unfortunately everybody
else is talking of other things and giving the impression
we are having just a side discussion.. and the problem 
remains: WebRTC threatens lives of dissidents. If we can't
publish news that rtcweb has fixed the problem then the
logical consequence is to publish that n months have passed 
and rtcweb is still not willing to fix the issue.... which
can be interpreted as a political news all on its own - how
industry does not care about collateral damages. Certainly
we can't just keep quiet and let this become a working 
strategy to ignore issues.

So, dear folks, this is your chance to keep the next
shitstorm from coming your way. Do something!


On Mon, Jul 20, 2015 at 01:23:08PM +0100, Tim Panton wrote:
> 
> > On 16 Jul 2015, at 09:42, carlo von lynX <lynX@i.know.you.are.psyced.org <mailto:lynX@i.know.you.are.psyced.org>> wrote:
> > 
> > Yes, I dare to speak out of my 25 years of experience on the
> > Internet even if I didn't have to use much VPN ever. I can
> > imagine plenty of constellations where the VPN is NOT supposed
> > to take over all communications on the system but rather offer
> > a side route into a company's servers or suchlike. Therefore
> > I worry if the software provider can always come up with a
> > configuration that still does what the customers expect and
> > at the same time deals with the new backdoor introduced by WebRTC.
> > Of course anyone with better understanding can help clear up
> > these worries.
> > 
> >> That isnâ€™t what I meant. I expect secure software providers to keep their
> >> software abreast of new developments. 
> 
> It turns out that I was completely wrong. The browsers are currently so 
> thorough in their discovery of available interfaces and routes that no
> amount of changes by the sys admin or VPN supplier will fix this problem.
> 
> Sorry about that. 
> 
> > 
> > Uh, not sure if that is an established pattern in the industry.
> > Usually somebody does whatever fits into their business model
> > best, then there is a public outcry and possibly some bloodspill,
> > then all the players look at how they can make a good impression
> > by reducing the collateral damages.. but if that doesn't work out
> > cheap and easy.. tant pis. Let's continue with business.
> > 
> >> I happen to have the (no doubt under informed) opinion that
> >> this isnâ€™t as simple as it has been made out to be.
> 
> At least I was right here. - It is even more complex than I believed.
> 
> > 
> > Well, in case of doubt it is important to pick the conservative
> > option until the issue is properly analyzed rather than hoping
> > for the general public to forget about the problem.
> > 
> >>> Also the "solution" you suggest AFAIU does not work for people
> >>> who use Tor in a non-NATted environment, for example on a
> >>> fixed system in a university library with a static IP address.
> >> 
> >> That is an interesting case. In that instance hiding the local IP would be 
> >> valuable. I thought however those users would normally run a TOR os off 
> >> a usb stick in a VM. That would probably mask the local IP.
> > 
> > Now that this is more clear, have another look at my original mail
> > that describes how such a dissident would use any Windows OS system
> > standing around. I doubt anything advanced like TAILS sticks are
> > out there as merely possessing one could be life-threatening.
> 
> 
> Here too I find that I was wrong and the current situation not what I assumed
> and is unsatisfactory.
> 
> Most SOCKS proxies don't have support for ICE/STUN nor arbitrary UDP.
> The current browser implementations (backed by the ICE spec) take the view that
> if SOCKS doesnâ€™t do UDP, then they should go directly and send UDP anyhow.
> 
> Iâ€™m not clear that this is what a user might expect. 
> An alternative might be to detect a proxy config _only_ offer TURN over TLS 
> candidates (and route the traffic over the proxy) - This would result in
> poor audio/video but would be more in keeping with the spirit of the proxy config.
> 
> (a variant might be to inspect the proxy in detail and use the address whitelists
> to filter allowed traffic somehow?) 
> 
> 
> >> 
> >> I totally agree. My solution is imho a better fix, since it
> >> also protects against many other apps that might get started from
> >> a browser page (email, dns, ipv6 tunnels, flash, mp3 players etc).
> 
> 
> Turns out I was wrong - sigh - as I said the browsers are more aggressive
> in finding routes than I assumed (or indeed than my ICE implementations are).
> 
> > Breaking the proxy principle is neither a Tor nor a VPN problem.
> > It's just browser vendors breaking a promise given to users that
> > configured proxies would be respected.
> > 
> >> Iâ€™m SOCKs ignorant.
> > 
> > So we all have our knowledge deficiencies...
> > good to be honest about them.
> > 
> 
> And now that Iâ€™ve educated myself somewhat, Iâ€™m agreeing with you.
> 
> 
> >>> And if local IP information is so super harmless, then why is it
> >>> an obvious reason for TBB devs to disable the entire thing?
> >>> I presume for the reasons I stated before.
> >> 
> >> I have no idea, but I fear it is because no one has split these issues out.
> > 
> > So let's dissect some more until we know what we are doing.
> 
> Which turns out to be the only useful contribution my <rant/> made.
> 
> 
> > 
> >> Iâ€™m of the view that there are _useful_ instances of P2P crypto
> >> in-browser - we will clearly have to disagree about that.
> > 
> > I think we could improve the user's ability to ensure end-to-end
> > authentication beyond server-based negotiation. Something like
> > rendering persistent public keys into QR codes and having people
> > exchange those QR codes at cocktail parties and hackathons.
> > 
> > Servers would still be able to provide the signaling code, but users
> > would be granted a guarantee that the E2E connection is authentic.
> > 
> 
> https://yopet.us <https://yopet.us/> :-) 
> 
> Tim.

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From nobody Fri Jul 31 03:52:47 2015
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From: "Wyss, Felix" <Felix.Wyss@inin.com>
To: "Asveren, Tolga" <tasveren@sonusnet.com>, Eric Rescorla <ekr@rtfm.com>, Peter Thatcher <pthatcher@google.com>
Thread-Topic: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it would be Christmas in July!)
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Date: Fri, 31 Jul 2015 10:52:38 +0000
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Cc: "<rtcweb@ietf.org>" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it would be Christmas in July!)
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As somebody who has to interoperate between WebRTC and legacy VoIP, I whole=
heartedly support making RTP/RTCP muxing mandatory.  Having just one "pipe"=
 for everything is IMHO an incredibly compelling feature of WebRTC.  Furthe=
rmore, the fewer RECOMMENDED vs. MUST we have in the specification, the mor=
e likely the WebRTC endpoints from different vendors will be interoperable =
without hacks or workarounds.

--Felix

________________________________
From: rtcweb <rtcweb-bounces@ietf.org> on behalf of Asveren, Tolga <tasvere=
n@sonusnet.com>
Sent: Friday, July 31, 2015 06:18
To: Eric Rescorla; Peter Thatcher
Cc: <rtcweb@ietf.org>
Subject: Re: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it=
 would be Christmas in July!)


"The whole reason we didn't do this before was because we wanted to work wi=
th legacy equipment.  But Cullen's claim is that there is no legacy equipme=
nt that can speak ICE (and DTLS) but not rtcp-mux."


I am not sure whether this is entirely true, especially on GW front. It cou=
ld be a better idea to be lenient for enhanced interoperability and making =
RTCWeb more deployable in practice. Can't the decision of being firm on usi=
ng rtcp-mux left to the WebRTC endpoint with signaling aspects supporting n=
egotiation of the capability?


Thanks,

Tolga


________________________________
From: rtcweb <rtcweb-bounces@ietf.org> on behalf of Eric Rescorla <ekr@rtfm=
.com>
Sent: Thursday, July 30, 2015 11:04 PM
To: Peter Thatcher
Cc: <rtcweb@ietf.org>
Subject: Re: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it=
 would be Christmas in July!)



On Thu, Jul 30, 2015 at 9:17 PM, Peter Thatcher <pthatcher@google.com<mailt=
o:pthatcher@google.com>> wrote:
At IETF 93 last week,  Cullen pointed out that he looked into ICE endpoints=
 and found that all of them support rtcp-mux.  Inspired by that, I went aro=
und and asked about a dozen people "would you be OK if we removed support f=
or non-muxed RTCP?"  The range of responses went from "I'm OK with it if Cu=
llen is" to "yeah, that would be great".  No one said no.  Cullen said yes.

For me, removing support for non-muxed RTCP would be like Christmas in July=
.

Realistically, August at this point.


So here's my proposal:  make RtcpMuxPolicy =3D=3D "required" be not only th=
e default, but the only RtcpMuxPolicy available.   All rtcweb endpoints mus=
t support rtcp-mux.

This seems reasonable.


An rtcweb endpoint would be free to fail if the remote endpoint does not su=
pport rtcp-mux.

Given the current specification, I believe this would need to be strong. I.=
e., an rtcweb
endpoint *must* fail if the remote endpoint does not support rtcp-mux. As s=
hould
be obvious, this only applies when the rtcweb endpoint is an answerer since=
 with
the above setting, you will not provide an offer that the other side can ac=
cept, but
the spec says:

"When acting as answerer, the browser will reject any m=3D section that doe=
s not provide an "a=3Drtcp-mux" attribute."

The reason for that is to avoid asymmetrical situations in which endpoint N=
M
(non-mux) makes an offer to endpoint M (mux) which is accepted, but then
re-offers from NM to M fail.

So I believe the best version of this proposal is simply requiring rtcp-mux
entirely.

-Ekr


The whole reason we didn't do this before was because we wanted to work wit=
h legacy equipment.  But Cullen's claim is that there is no legacy equipmen=
t that can speak ICE (and DTLS) but not rtcp-mux.

So is there any remaining reason that we cannot require rtcp-mux?




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



--_000_CY1PR0501MB1579BC38B75CBC58744A970FEB8A0CY1PR0501MB1579_
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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Diso-8859-=
1">
<style type=3D"text/css" style=3D"display:none;"><!-- P {margin-top:0;margi=
n-bottom:0;} --></style>
</head>
<body dir=3D"ltr">
<div id=3D"divtagdefaultwrapper" style=3D"font-size:12pt;color:#000000;back=
ground-color:#FFFFFF;font-family:Calibri,Arial,Helvetica,sans-serif;">
<p>As somebody who has to interoperate between WebRTC and legacy VoIP, I wh=
oleheartedly support making RTP/RTCP muxing mandatory.&nbsp; Having just on=
e &quot;pipe&quot; for everything is IMHO an incredibly compelling feature =
of WebRTC.&nbsp; Furthermore, the fewer RECOMMENDED
 vs. MUST we have in the specification, the more likely the WebRTC endpoint=
s from different vendors will be interoperable without hacks or workarounds=
.&nbsp;&nbsp;
<br>
</p>
<br>
--Felix<br>
<br>
<div style=3D"color: rgb(0, 0, 0);">
<hr tabindex=3D"-1" style=3D"display:inline-block; width:98%">
<div id=3D"divRplyFwdMsg" dir=3D"ltr"><font style=3D"font-size:11pt" face=
=3D"Calibri, sans-serif" color=3D"#000000"><b>From:</b> rtcweb &lt;rtcweb-b=
ounces@ietf.org&gt; on behalf of Asveren, Tolga &lt;tasveren@sonusnet.com&g=
t;<br>
<b>Sent:</b> Friday, July 31, 2015 06:18<br>
<b>To:</b> Eric Rescorla; Peter Thatcher<br>
<b>Cc:</b> &lt;rtcweb@ietf.org&gt;<br>
<b>Subject:</b> Re: [rtcweb] Proposal: require rtcp-mux (remove non-muxed R=
TCP; it would be Christmas in July!)</font>
<div>&nbsp;</div>
</div>
<div>
<div id=3D"divtagdefaultwrapper" style=3D"font-size:12pt; color:#000000; ba=
ckground-color:#FFFFFF; font-family:Calibri,Arial,Helvetica,sans-serif">
<p><span style=3D"color:rgb(33,33,33); font-family:arial,helvetica,sans-ser=
if; font-size:15px">&quot;The whole reason we didn't do this before was bec=
ause we wanted to work with legacy equipment.&nbsp; But Cullen's claim is t=
hat there is no legacy equipment that can speak
 ICE (and DTLS) but not rtcp-mux.&quot;</span><br>
</p>
<p><span style=3D"color:rgb(33,33,33); font-family:arial,helvetica,sans-ser=
if; font-size:15px"><br>
</span></p>
<p><font face=3D"arial, helvetica, sans-serif" color=3D"#212121"><span styl=
e=3D"font-size:15px">I am not sure whether this is entirely true, especiall=
y on GW front. It could be a better idea to be lenient for enhanced interop=
erability and making RTCWeb more&nbsp;deployable
 in practice. Can't the decision of being firm on using rtcp-mux left to th=
e WebRTC endpoint with signaling aspects supporting negotiation of the capa=
bility?</span></font></p>
<p><span style=3D"color:rgb(33,33,33); font-family:arial,helvetica,sans-ser=
if; font-size:15px"><br>
</span></p>
<p><span style=3D"color:rgb(33,33,33); font-family:arial,helvetica,sans-ser=
if; font-size:15px">Thanks,</span></p>
<p><span style=3D"color:rgb(33,33,33); font-family:arial,helvetica,sans-ser=
if; font-size:15px">Tolga</span></p>
<br>
<br>
<div style=3D"color:rgb(0,0,0)">
<hr tabindex=3D"-1" style=3D"display:inline-block; width:98%">
<div id=3D"divRplyFwdMsg" dir=3D"ltr"><font style=3D"font-size:11pt" face=
=3D"Calibri, sans-serif" color=3D"#000000"><b>From:</b> rtcweb &lt;rtcweb-b=
ounces@ietf.org&gt; on behalf of Eric Rescorla &lt;ekr@rtfm.com&gt;<br>
<b>Sent:</b> Thursday, July 30, 2015 11:04 PM<br>
<b>To:</b> Peter Thatcher<br>
<b>Cc:</b> &lt;rtcweb@ietf.org&gt;<br>
<b>Subject:</b> Re: [rtcweb] Proposal: require rtcp-mux (remove non-muxed R=
TCP; it would be Christmas in July!)</font>
<div>&nbsp;</div>
</div>
<div>
<div dir=3D"ltr"><br>
<div class=3D"gmail_extra"><br>
<div class=3D"gmail_quote">On Thu, Jul 30, 2015 at 9:17 PM, Peter Thatcher =
<span dir=3D"ltr">
&lt;<a href=3D"mailto:pthatcher@google.com" target=3D"_blank">pthatcher@goo=
gle.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex; border-left:1=
px #ccc solid; padding-left:1ex">
<div dir=3D"ltr">
<div style=3D"font-family:arial,helvetica,sans-serif">At IETF 93 last week,=
 &nbsp;Cullen pointed out that he looked into ICE endpoints and found that =
all of them support rtcp-mux.&nbsp; Inspired by that, I went around and ask=
ed about a dozen people &quot;would you be OK if
 we removed support for non-muxed RTCP?&quot; &nbsp;The range of responses =
went from &quot;I'm OK with it if Cullen is&quot; to &quot;yeah, that would=
 be great&quot;.&nbsp; No one said no.&nbsp; Cullen said yes. &nbsp;</div>
<div style=3D"font-family:arial,helvetica,sans-serif"><br>
</div>
<div style=3D"font-family:arial,helvetica,sans-serif">For me, removing supp=
ort for non-muxed RTCP would be like Christmas in July.</div>
</div>
</blockquote>
<div><br>
</div>
<div>Realistically, August at this point.</div>
<div><br>
</div>
<div><br>
</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex; border-left:1=
px #ccc solid; padding-left:1ex">
<div dir=3D"ltr">
<div style=3D"font-family:arial,helvetica,sans-serif">So here's my proposal=
: &nbsp;make RtcpMuxPolicy =3D=3D &quot;required&quot; be not only the defa=
ult, but the only RtcpMuxPolicy available. &nbsp; All rtcweb endpoints must=
 support rtcp-mux. &nbsp;</div>
</div>
</blockquote>
<div><br>
</div>
<div>This seems reasonable.</div>
<div><br>
</div>
<div>&nbsp;</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex; border-left:1=
px #ccc solid; padding-left:1ex">
<div dir=3D"ltr">
<div style=3D"font-family:arial,helvetica,sans-serif">An rtcweb endpoint wo=
uld be free to fail if the remote endpoint does not support rtcp-mux.</div>
</div>
</blockquote>
<div><br>
</div>
<div>Given the current specification, I believe this would need to be stron=
g. I.e., an rtcweb</div>
<div>endpoint *must* fail if the remote endpoint does not support rtcp-mux.=
 As should&nbsp;</div>
<div>be obvious, this only applies when the rtcweb endpoint is an answerer =
since with</div>
<div>the above setting, you will not provide an offer that the other side c=
an accept, but</div>
<div>the spec says:</div>
<div><br>
</div>
<div>&quot;When acting as answerer, the browser will reject any m=3D sectio=
n that does not provide an &quot;a=3Drtcp-mux&quot; attribute.&quot;</div>
<div><br>
</div>
<div>The reason for that is to avoid asymmetrical situations in which endpo=
int NM</div>
<div>(non-mux) makes an offer to endpoint M (mux) which is accepted, but th=
en</div>
<div>re-offers from NM to M fail.</div>
<div><br>
</div>
<div>So I believe the best version of this proposal is simply requiring rtc=
p-mux</div>
<div>entirely.</div>
<div><br>
</div>
<div>-Ekr</div>
<div><br>
</div>
<div>&nbsp;</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex; border-left:1=
px #ccc solid; padding-left:1ex">
<div dir=3D"ltr">
<div style=3D"font-family:arial,helvetica,sans-serif"></div>
<div style=3D"font-family:arial,helvetica,sans-serif">The whole reason we d=
idn't do this before was because we wanted to work with legacy equipment.&n=
bsp; But Cullen's claim is that there is no legacy equipment that can speak=
 ICE (and DTLS) but not rtcp-mux.&nbsp;</div>
<div style=3D"font-family:arial,helvetica,sans-serif"><br>
</div>
<div style=3D"font-family:arial,helvetica,sans-serif">So is there any remai=
ning reason that we cannot require rtcp-mux?</div>
<div style=3D"font-family:arial,helvetica,sans-serif"><br>
</div>
<div style=3D"font-family:arial,helvetica,sans-serif"><br>
</div>
<div style=3D"font-family:arial,helvetica,sans-serif"><br>
</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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br>
</blockquote>
</div>
<br>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</body>
</html>

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From: "Asveren, Tolga" <tasveren@sonusnet.com>
To: "Wyss, Felix" <Felix.Wyss@inin.com>, Eric Rescorla <ekr@rtfm.com>, "Peter Thatcher" <pthatcher@google.com>
Thread-Topic: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it would be Christmas in July!)
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Cc: "<rtcweb@ietf.org>" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it would be Christmas in July!)
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It is not really about "what is technically better" (and event there, the b=
eauty is in the eye of the beholder depending on the deployment scenario) b=
ut more wider interoperability. As long as there is a negotiation mechanism=
/freedom to reject, I don't see any harm of allowing not to mandate rtcp-mu=
x. You still can mandate its use in a specific implementation.

Thanks,
Tolga

From: Wyss, Felix [mailto:Felix.Wyss@inin.com]
Sent: Friday, July 31, 2015 6:53 AM
To: Asveren, Tolga <tasveren@sonusnet.com>; Eric Rescorla <ekr@rtfm.com>; P=
eter Thatcher <pthatcher@google.com>
Cc: <rtcweb@ietf.org> <rtcweb@ietf.org>
Subject: Re: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it=
 would be Christmas in July!)


As somebody who has to interoperate between WebRTC and legacy VoIP, I whole=
heartedly support making RTP/RTCP muxing mandatory.  Having just one "pipe"=
 for everything is IMHO an incredibly compelling feature of WebRTC.  Furthe=
rmore, the fewer RECOMMENDED vs. MUST we have in the specification, the mor=
e likely the WebRTC endpoints from different vendors will be interoperable =
without hacks or workarounds.

--Felix
________________________________
From: rtcweb <rtcweb-bounces@ietf.org<mailto:rtcweb-bounces@ietf.org>> on b=
ehalf of Asveren, Tolga <tasveren@sonusnet.com<mailto:tasveren@sonusnet.com=
>>
Sent: Friday, July 31, 2015 06:18
To: Eric Rescorla; Peter Thatcher
Cc: <rtcweb@ietf.org<mailto:rtcweb@ietf.org>>
Subject: Re: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it=
 would be Christmas in July!)


"The whole reason we didn't do this before was because we wanted to work wi=
th legacy equipment.  But Cullen's claim is that there is no legacy equipme=
nt that can speak ICE (and DTLS) but not rtcp-mux."



I am not sure whether this is entirely true, especially on GW front. It cou=
ld be a better idea to be lenient for enhanced interoperability and making =
RTCWeb more deployable in practice. Can't the decision of being firm on usi=
ng rtcp-mux left to the WebRTC endpoint with signaling aspects supporting n=
egotiation of the capability?



Thanks,

Tolga

________________________________
From: rtcweb <rtcweb-bounces@ietf.org<mailto:rtcweb-bounces@ietf.org>> on b=
ehalf of Eric Rescorla <ekr@rtfm.com<mailto:ekr@rtfm.com>>
Sent: Thursday, July 30, 2015 11:04 PM
To: Peter Thatcher
Cc: <rtcweb@ietf.org<mailto:rtcweb@ietf.org>>
Subject: Re: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it=
 would be Christmas in July!)



On Thu, Jul 30, 2015 at 9:17 PM, Peter Thatcher <pthatcher@google.com<mailt=
o:pthatcher@google.com>> wrote:
At IETF 93 last week,  Cullen pointed out that he looked into ICE endpoints=
 and found that all of them support rtcp-mux.  Inspired by that, I went aro=
und and asked about a dozen people "would you be OK if we removed support f=
or non-muxed RTCP?"  The range of responses went from "I'm OK with it if Cu=
llen is" to "yeah, that would be great".  No one said no.  Cullen said yes.

For me, removing support for non-muxed RTCP would be like Christmas in July=
.

Realistically, August at this point.


So here's my proposal:  make RtcpMuxPolicy =3D=3D "required" be not only th=
e default, but the only RtcpMuxPolicy available.   All rtcweb endpoints mus=
t support rtcp-mux.

This seems reasonable.


An rtcweb endpoint would be free to fail if the remote endpoint does not su=
pport rtcp-mux.

Given the current specification, I believe this would need to be strong. I.=
e., an rtcweb
endpoint *must* fail if the remote endpoint does not support rtcp-mux. As s=
hould
be obvious, this only applies when the rtcweb endpoint is an answerer since=
 with
the above setting, you will not provide an offer that the other side can ac=
cept, but
the spec says:

"When acting as answerer, the browser will reject any m=3D section that doe=
s not provide an "a=3Drtcp-mux" attribute."

The reason for that is to avoid asymmetrical situations in which endpoint N=
M
(non-mux) makes an offer to endpoint M (mux) which is accepted, but then
re-offers from NM to M fail.

So I believe the best version of this proposal is simply requiring rtcp-mux
entirely.

-Ekr


The whole reason we didn't do this before was because we wanted to work wit=
h legacy equipment.  But Cullen's claim is that there is no legacy equipmen=
t that can speak ICE (and DTLS) but not rtcp-mux.

So is there any remaining reason that we cannot require rtcp-mux?




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


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</head>
<body lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1F497D">It is not really about &#8220;what is=
 technically better&#8221; (and event there, the beauty is in the eye of th=
e beholder depending on the deployment scenario) but more
 wider interoperability. As long as there is a negotiation mechanism/freedo=
m to reject, I don&#8217;t see any harm of allowing not to mandate rtcp-mux=
. You still can mandate its use in a specific implementation.<o:p></o:p></s=
pan></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1F497D"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1F497D">Thanks,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1F497D">Tolga<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1F497D"><o:p>&nbsp;</o:p></span></p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #E1E1E1 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:11.0pt;font-family:&quot=
;Calibri&quot;,sans-serif">From:</span></b><span style=3D"font-size:11.0pt;=
font-family:&quot;Calibri&quot;,sans-serif"> Wyss, Felix [mailto:Felix.Wyss=
@inin.com]
<br>
<b>Sent:</b> Friday, July 31, 2015 6:53 AM<br>
<b>To:</b> Asveren, Tolga &lt;tasveren@sonusnet.com&gt;; Eric Rescorla &lt;=
ekr@rtfm.com&gt;; Peter Thatcher &lt;pthatcher@google.com&gt;<br>
<b>Cc:</b> &lt;rtcweb@ietf.org&gt; &lt;rtcweb@ietf.org&gt;<br>
<b>Subject:</b> Re: [rtcweb] Proposal: require rtcp-mux (remove non-muxed R=
TCP; it would be Christmas in July!)<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div id=3D"divtagdefaultwrapper">
<p style=3D"background:white"><span style=3D"font-family:&quot;Calibri&quot=
;,sans-serif;color:black">As somebody who has to interoperate between WebRT=
C and legacy VoIP, I wholeheartedly support making RTP/RTCP muxing mandator=
y.&nbsp; Having just one &quot;pipe&quot; for everything is IMHO
 an incredibly compelling feature of WebRTC.&nbsp; Furthermore, the fewer R=
ECOMMENDED vs. MUST we have in the specification, the more likely the WebRT=
C endpoints from different vendors will be interoperable without hacks or w=
orkarounds.&nbsp;&nbsp;
<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt;background:white"><spa=
n style=3D"font-family:&quot;Calibri&quot;,sans-serif;color:black"><br>
--Felix<o:p></o:p></span></p>
<div>
<div class=3D"MsoNormal" align=3D"center" style=3D"text-align:center;backgr=
ound:white">
<span style=3D"font-family:&quot;Calibri&quot;,sans-serif;color:black">
<hr size=3D"2" width=3D"98%" align=3D"center">
</span></div>
<div id=3D"divRplyFwdMsg">
<p class=3D"MsoNormal" style=3D"background:white"><b><span style=3D"font-si=
ze:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:black">From:</sp=
an></b><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,sans=
-serif;color:black"> rtcweb &lt;<a href=3D"mailto:rtcweb-bounces@ietf.org">=
rtcweb-bounces@ietf.org</a>&gt;
 on behalf of Asveren, Tolga &lt;<a href=3D"mailto:tasveren@sonusnet.com">t=
asveren@sonusnet.com</a>&gt;<br>
<b>Sent:</b> Friday, July 31, 2015 06:18<br>
<b>To:</b> Eric Rescorla; Peter Thatcher<br>
<b>Cc:</b> &lt;<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>&gt;<b=
r>
<b>Subject:</b> Re: [rtcweb] Proposal: require rtcp-mux (remove non-muxed R=
TCP; it would be Christmas in July!)</span><span style=3D"font-family:&quot=
;Calibri&quot;,sans-serif;color:black">
<o:p></o:p></span></p>
<div>
<p class=3D"MsoNormal" style=3D"background:white"><span style=3D"font-famil=
y:&quot;Calibri&quot;,sans-serif;color:black">&nbsp;<o:p></o:p></span></p>
</div>
</div>
<div>
<div id=3D"divtagdefaultwrapper">
<p style=3D"background:white"><span style=3D"font-size:11.5pt;font-family:&=
quot;Arial&quot;,sans-serif;color:#212121">&quot;The whole reason we didn't=
 do this before was because we wanted to work with legacy equipment.&nbsp; =
But Cullen's claim is that there is no legacy equipment
 that can speak ICE (and DTLS) but not rtcp-mux.&quot;</span><span style=3D=
"font-family:&quot;Calibri&quot;,sans-serif;color:black"><o:p></o:p></span>=
</p>
<p style=3D"background:white"><span style=3D"font-family:&quot;Calibri&quot=
;,sans-serif;color:black"><o:p>&nbsp;</o:p></span></p>
<p style=3D"background:white"><span style=3D"font-size:11.5pt;font-family:&=
quot;Arial&quot;,sans-serif;color:#212121">I am not sure whether this is en=
tirely true, especially on GW front. It could be a better idea to be lenien=
t for enhanced interoperability and making RTCWeb
 more&nbsp;deployable in practice. Can't the decision of being firm on usin=
g rtcp-mux left to the WebRTC endpoint with signaling aspects supporting ne=
gotiation of the capability?</span><span style=3D"font-family:&quot;Calibri=
&quot;,sans-serif;color:black"><o:p></o:p></span></p>
<p style=3D"background:white"><span style=3D"font-family:&quot;Calibri&quot=
;,sans-serif;color:black"><o:p>&nbsp;</o:p></span></p>
<p style=3D"background:white"><span style=3D"font-size:11.5pt;font-family:&=
quot;Arial&quot;,sans-serif;color:#212121">Thanks,</span><span style=3D"fon=
t-family:&quot;Calibri&quot;,sans-serif;color:black"><o:p></o:p></span></p>
<p style=3D"background:white"><span style=3D"font-size:11.5pt;font-family:&=
quot;Arial&quot;,sans-serif;color:#212121">Tolga</span><span style=3D"font-=
family:&quot;Calibri&quot;,sans-serif;color:black"><o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt;background:white"><spa=
n style=3D"font-family:&quot;Calibri&quot;,sans-serif;color:black"><o:p>&nb=
sp;</o:p></span></p>
<div>
<div class=3D"MsoNormal" align=3D"center" style=3D"text-align:center;backgr=
ound:white">
<span style=3D"font-family:&quot;Calibri&quot;,sans-serif;color:black">
<hr size=3D"2" width=3D"98%" align=3D"center">
</span></div>
<div id=3D"divRplyFwdMsg">
<p class=3D"MsoNormal" style=3D"background:white"><b><span style=3D"font-si=
ze:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:black">From:</sp=
an></b><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,sans=
-serif;color:black"> rtcweb &lt;<a href=3D"mailto:rtcweb-bounces@ietf.org">=
rtcweb-bounces@ietf.org</a>&gt;
 on behalf of Eric Rescorla &lt;<a href=3D"mailto:ekr@rtfm.com">ekr@rtfm.co=
m</a>&gt;<br>
<b>Sent:</b> Thursday, July 30, 2015 11:04 PM<br>
<b>To:</b> Peter Thatcher<br>
<b>Cc:</b> &lt;<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>&gt;<b=
r>
<b>Subject:</b> Re: [rtcweb] Proposal: require rtcp-mux (remove non-muxed R=
TCP; it would be Christmas in July!)</span><span style=3D"font-family:&quot=
;Calibri&quot;,sans-serif;color:black">
<o:p></o:p></span></p>
<div>
<p class=3D"MsoNormal" style=3D"background:white"><span style=3D"font-famil=
y:&quot;Calibri&quot;,sans-serif;color:black">&nbsp;<o:p></o:p></span></p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal" style=3D"background:white"><span style=3D"font-famil=
y:&quot;Calibri&quot;,sans-serif;color:black"><o:p>&nbsp;</o:p></span></p>
<div>
<p class=3D"MsoNormal" style=3D"background:white"><span style=3D"font-famil=
y:&quot;Calibri&quot;,sans-serif;color:black"><o:p>&nbsp;</o:p></span></p>
<div>
<p class=3D"MsoNormal" style=3D"background:white"><span style=3D"font-famil=
y:&quot;Calibri&quot;,sans-serif;color:black">On Thu, Jul 30, 2015 at 9:17 =
PM, Peter Thatcher &lt;<a href=3D"mailto:pthatcher@google.com" target=3D"_b=
lank">pthatcher@google.com</a>&gt; wrote:<o:p></o:p></span></p>
<blockquote style=3D"border:none;border-left:solid #CCCCCC 1.0pt;padding:0i=
n 0in 0in 6.0pt;margin-left:4.8pt;margin-right:0in">
<div>
<div>
<p class=3D"MsoNormal" style=3D"background:white"><span style=3D"font-famil=
y:&quot;Arial&quot;,sans-serif;color:black">At IETF 93 last week, &nbsp;Cul=
len pointed out that he looked into ICE endpoints and found that all of the=
m support rtcp-mux.&nbsp; Inspired by that, I went around
 and asked about a dozen people &quot;would you be OK if we removed support=
 for non-muxed RTCP?&quot; &nbsp;The range of responses went from &quot;I'm=
 OK with it if Cullen is&quot; to &quot;yeah, that would be great&quot;.&nb=
sp; No one said no.&nbsp; Cullen said yes. &nbsp;<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"background:white"><span style=3D"font-famil=
y:&quot;Arial&quot;,sans-serif;color:black"><o:p>&nbsp;</o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"background:white"><span style=3D"font-famil=
y:&quot;Arial&quot;,sans-serif;color:black">For me, removing support for no=
n-muxed RTCP would be like Christmas in July.<o:p></o:p></span></p>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal" style=3D"background:white"><span style=3D"font-famil=
y:&quot;Calibri&quot;,sans-serif;color:black"><o:p>&nbsp;</o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"background:white"><span style=3D"font-famil=
y:&quot;Calibri&quot;,sans-serif;color:black">Realistically, August at this=
 point.<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"background:white"><span style=3D"font-famil=
y:&quot;Calibri&quot;,sans-serif;color:black"><o:p>&nbsp;</o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"background:white"><span style=3D"font-famil=
y:&quot;Calibri&quot;,sans-serif;color:black"><o:p>&nbsp;</o:p></span></p>
</div>
<blockquote style=3D"border:none;border-left:solid #CCCCCC 1.0pt;padding:0i=
n 0in 0in 6.0pt;margin-left:4.8pt;margin-right:0in">
<div>
<div>
<p class=3D"MsoNormal" style=3D"background:white"><span style=3D"font-famil=
y:&quot;Arial&quot;,sans-serif;color:black">So here's my proposal: &nbsp;ma=
ke RtcpMuxPolicy =3D=3D &quot;required&quot; be not only the default, but t=
he only RtcpMuxPolicy available. &nbsp; All rtcweb endpoints must support
 rtcp-mux. &nbsp;<o:p></o:p></span></p>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal" style=3D"background:white"><span style=3D"font-famil=
y:&quot;Calibri&quot;,sans-serif;color:black"><o:p>&nbsp;</o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"background:white"><span style=3D"font-famil=
y:&quot;Calibri&quot;,sans-serif;color:black">This seems reasonable.<o:p></=
o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"background:white"><span style=3D"font-famil=
y:&quot;Calibri&quot;,sans-serif;color:black"><o:p>&nbsp;</o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"background:white"><span style=3D"font-famil=
y:&quot;Calibri&quot;,sans-serif;color:black">&nbsp;<o:p></o:p></span></p>
</div>
<blockquote style=3D"border:none;border-left:solid #CCCCCC 1.0pt;padding:0i=
n 0in 0in 6.0pt;margin-left:4.8pt;margin-right:0in">
<div>
<div>
<p class=3D"MsoNormal" style=3D"background:white"><span style=3D"font-famil=
y:&quot;Arial&quot;,sans-serif;color:black">An rtcweb endpoint would be fre=
e to fail if the remote endpoint does not support rtcp-mux.<o:p></o:p></spa=
n></p>
</div>
</div>
</blockquote>
<div>
<p class=3D"MsoNormal" style=3D"background:white"><span style=3D"font-famil=
y:&quot;Calibri&quot;,sans-serif;color:black"><o:p>&nbsp;</o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"background:white"><span style=3D"font-famil=
y:&quot;Calibri&quot;,sans-serif;color:black">Given the current specificati=
on, I believe this would need to be strong. I.e., an rtcweb<o:p></o:p></spa=
n></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"background:white"><span style=3D"font-famil=
y:&quot;Calibri&quot;,sans-serif;color:black">endpoint *must* fail if the r=
emote endpoint does not support rtcp-mux. As should&nbsp;<o:p></o:p></span>=
</p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"background:white"><span style=3D"font-famil=
y:&quot;Calibri&quot;,sans-serif;color:black">be obvious, this only applies=
 when the rtcweb endpoint is an answerer since with<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"background:white"><span style=3D"font-famil=
y:&quot;Calibri&quot;,sans-serif;color:black">the above setting, you will n=
ot provide an offer that the other side can accept, but<o:p></o:p></span></=
p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"background:white"><span style=3D"font-famil=
y:&quot;Calibri&quot;,sans-serif;color:black">the spec says:<o:p></o:p></sp=
an></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"background:white"><span style=3D"font-famil=
y:&quot;Calibri&quot;,sans-serif;color:black"><o:p>&nbsp;</o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"background:white"><span style=3D"font-famil=
y:&quot;Calibri&quot;,sans-serif;color:black">&quot;When acting as answerer=
, the browser will reject any m=3D section that does not provide an &quot;a=
=3Drtcp-mux&quot; attribute.&quot;<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"background:white"><span style=3D"font-famil=
y:&quot;Calibri&quot;,sans-serif;color:black"><o:p>&nbsp;</o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"background:white"><span style=3D"font-famil=
y:&quot;Calibri&quot;,sans-serif;color:black">The reason for that is to avo=
id asymmetrical situations in which endpoint NM<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"background:white"><span style=3D"font-famil=
y:&quot;Calibri&quot;,sans-serif;color:black">(non-mux) makes an offer to e=
ndpoint M (mux) which is accepted, but then<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"background:white"><span style=3D"font-famil=
y:&quot;Calibri&quot;,sans-serif;color:black">re-offers from NM to M fail.<=
o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"background:white"><span style=3D"font-famil=
y:&quot;Calibri&quot;,sans-serif;color:black"><o:p>&nbsp;</o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"background:white"><span style=3D"font-famil=
y:&quot;Calibri&quot;,sans-serif;color:black">So I believe the best version=
 of this proposal is simply requiring rtcp-mux<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"background:white"><span style=3D"font-famil=
y:&quot;Calibri&quot;,sans-serif;color:black">entirely.<o:p></o:p></span></=
p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"background:white"><span style=3D"font-famil=
y:&quot;Calibri&quot;,sans-serif;color:black"><o:p>&nbsp;</o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"background:white"><span style=3D"font-famil=
y:&quot;Calibri&quot;,sans-serif;color:black">-Ekr<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"background:white"><span style=3D"font-famil=
y:&quot;Calibri&quot;,sans-serif;color:black"><o:p>&nbsp;</o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"background:white"><span style=3D"font-famil=
y:&quot;Calibri&quot;,sans-serif;color:black">&nbsp;<o:p></o:p></span></p>
</div>
<blockquote style=3D"border:none;border-left:solid #CCCCCC 1.0pt;padding:0i=
n 0in 0in 6.0pt;margin-left:4.8pt;margin-right:0in">
<div>
<div>
<p class=3D"MsoNormal" style=3D"background:white"><span style=3D"font-famil=
y:&quot;Arial&quot;,sans-serif;color:black">The whole reason we didn't do t=
his before was because we wanted to work with legacy equipment.&nbsp; But C=
ullen's claim is that there is no legacy equipment that
 can speak ICE (and DTLS) but not rtcp-mux.&nbsp;<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"background:white"><span style=3D"font-famil=
y:&quot;Arial&quot;,sans-serif;color:black"><o:p>&nbsp;</o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"background:white"><span style=3D"font-famil=
y:&quot;Arial&quot;,sans-serif;color:black">So is there any remaining reaso=
n that we cannot require rtcp-mux?<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"background:white"><span style=3D"font-famil=
y:&quot;Arial&quot;,sans-serif;color:black"><o:p>&nbsp;</o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"background:white"><span style=3D"font-famil=
y:&quot;Arial&quot;,sans-serif;color:black"><o:p>&nbsp;</o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"background:white"><span style=3D"font-famil=
y:&quot;Arial&quot;,sans-serif;color:black"><o:p>&nbsp;</o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt;background:white"><spa=
n style=3D"font-family:&quot;Calibri&quot;,sans-serif;color:black"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><o:p></o:p></span></p>
</blockquote>
</div>
<p class=3D"MsoNormal" style=3D"background:white"><span style=3D"font-famil=
y:&quot;Calibri&quot;,sans-serif;color:black"><o:p>&nbsp;</o:p></span></p>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</body>
</html>

--_000_SN1PR0301MB15518198D375F21B973C091DB28A0SN1PR0301MB1551_--


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Date: Fri, 31 Jul 2015 09:45:22 -0400
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From: Roman Shpount <roman@telurix.com>
To: "Asveren, Tolga" <tasveren@sonusnet.com>
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Cc: "<rtcweb@ietf.org>" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it would be Christmas in July!)
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On Fri, Jul 31, 2015 at 7:58 AM, Asveren, Tolga <tasveren@sonusnet.com>
wrote:

> It is not really about =E2=80=9Cwhat is technically better=E2=80=9D (and =
event there, the
> beauty is in the eye of the beholder depending on the deployment scenario=
)
> but more wider interoperability. As long as there is a negotiation
> mechanism/freedom to reject, I don=E2=80=99t see any harm of allowing not=
 to
> mandate rtcp-mux. You still can mandate its use in a specific
> implementation.
>
>
>
Is there a WebRTC gateway or some other device that you have in mind that
supports DTLS-SRTP and does not support rtcp-mux?

The whole point of this discussion is to remove the negotiation for
rtcp-mux. If there are no devices that WebRTC endpoint can communicate with
(which now means devices supporting required DTLS-SRTP) which do not
support rtcp-mux, it would be better to reduce the code size by removing
rtcp-mux negotiation and all the code related to collecting second set of
candidates for RTCP connection, supporting second component per media
stream, negotiating second DTLS-SRTP for RTCP connection, etc. This is a
non-trivial amount of code, which is not being actively tested right now.
Most of the open source webrtc gateways I looked at have it broken.
Maintaining this code without a compelling use case is too much of the
burden this is why I strongly suggest rtcp-mux must be supported.

Regards,
_____________
Roman Shpount

--089e0111c0a40c72c0051c2c0851
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div><div class=3D"gmail_signat=
ure"><br></div></div><div class=3D"gmail_quote">On Fri, Jul 31, 2015 at 7:5=
8 AM, Asveren, Tolga <span dir=3D"ltr">&lt;<a href=3D"mailto:tasveren@sonus=
net.com" target=3D"_blank">tasveren@sonusnet.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 lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1f497d">It is not really about =E2=80=9Cwhat =
is technically better=E2=80=9D (and event there, the beauty is in the eye o=
f the beholder depending on the deployment scenario) but more
 wider interoperability. As long as there is a negotiation mechanism/freedo=
m to reject, I don=E2=80=99t see any harm of allowing not to mandate rtcp-m=
ux. You still can mandate its use in a specific implementation.<u></u><u></=
u></span></p>
<p class=3D"MsoNormal"><br></p></div></div></blockquote><div><br></div><div=
>Is there a WebRTC gateway or some other device that you have in mind that =
supports DTLS-SRTP and does not support rtcp-mux?</div><div><br></div><div>=
The whole point of this discussion is to remove the negotiation for rtcp-mu=
x. If there are no devices that WebRTC endpoint can communicate with (which=
 now means devices supporting required DTLS-SRTP) which do not support rtcp=
-mux, it would be better to reduce the code size by removing rtcp-mux negot=
iation and all the code related to collecting second set of candidates for =
RTCP connection, supporting second component per media stream, negotiating =
second DTLS-SRTP for RTCP connection, etc. This is a non-trivial amount of =
code, which is not being actively tested right now. Most of the open source=
 webrtc gateways I looked at have it broken. Maintaining this code without =
a compelling use case is too much of the burden this is why I strongly sugg=
est rtcp-mux must be supported.</div><div><br></div><div>Regards,</div><div=
><div class=3D"gmail_signature">_____________<br>Roman Shpount</div></div><=
div>=C2=A0</div></div></div></div>

--089e0111c0a40c72c0051c2c0851--


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References: <55352067.6070007@nostrum.com> <20150421154657.GA4071@lo.psyced.org> <20150715123803.GA24976@lo.psyced.org> <2BCEF545-D322-4589-AC85-AA80E7F934D9@phonefromhere.com> <20150715164129.GA27105@lo.psyced.org> <AF6485D4-1579-45AF-937C-1682AC1AA645@phonefromhere.com> <20150716084225.GA29652@lo.psyced.org> <0AF88216-374E-4DBA-9DB5-3E55B358E6AA@phonefromhere.com> <20150731104712.GA5228@lo.psyced.org>
Date: Fri, 31 Jul 2015 09:53:09 -0400
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From: Roman Shpount <roman@telurix.com>
To: carlo von lynX <lynX@i.know.you.are.psyced.org>
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Subject: Re: [rtcweb] Prompt for mic, camera *and* network access
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On Fri, Jul 31, 2015 at 6:47 AM, carlo von lynX <
lynX@i.know.you.are.psyced.org> wrote:

> Alright, thanks Tim for digging deeper and confirming that
> we have a serious problem here. Unfortunately everybody
> else is talking of other things and giving the impression
> we are having just a side discussion.. and the problem
> remains: WebRTC threatens lives of dissidents. If we can't
> publish news that rtcweb has fixed the problem then the
> logical consequence is to publish that n months have passed
> and rtcweb is still not willing to fix the issue.... which
> can be interpreted as a political news all on its own - how
> industry does not care about collateral damages. Certainly
> we can't just keep quiet and let this become a working
> strategy to ignore issues.
>
> So, dear folks, this is your chance to keep the next
> shitstorm from coming your way. Do something!
>
>
In IETF we typically discuss technical issues and not political PR. So, the
other discussions that you so blatantly ignore are actually about the
actual technical implications of webrtc on user privacy and possible ways
to improve it. If you have a particular technical solution that you want
propose then please do so. Screaming that industry needs to do something or
else the puppy will die is not helpful.

Regards,
_____________
Roman Shpount

--089e0122a5b8dc0843051c2c23ab
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Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On F=
ri, Jul 31, 2015 at 6:47 AM, carlo von lynX <span dir=3D"ltr">&lt;<a href=
=3D"mailto:lynX@i.know.you.are.psyced.org" target=3D"_blank">lynX@i.know.yo=
u.are.psyced.org</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">Al=
right, thanks Tim for digging deeper and confirming that<br>
we have a serious problem here. Unfortunately everybody<br>
else is talking of other things and giving the impression<br>
we are having just a side discussion.. and the problem<br>
remains: WebRTC threatens lives of dissidents. If we can&#39;t<br>
publish news that rtcweb has fixed the problem then the<br>
logical consequence is to publish that n months have passed<br>
and rtcweb is still not willing to fix the issue.... which<br>
can be interpreted as a political news all on its own - how<br>
industry does not care about collateral damages. Certainly<br>
we can&#39;t just keep quiet and let this become a working<br>
strategy to ignore issues.<br>
<br>
So, dear folks, this is your chance to keep the next<br>
shitstorm from coming your way. Do something!<br>
<br></blockquote><div><br></div>In IETF we typically discuss technical issu=
es and not political PR. So, the other discussions that you so blatantly ig=
nore are actually about the actual technical implications of webrtc on user=
 privacy and possible ways to improve it. If you have a particular technica=
l solution that you want propose then please do so. Screaming that industry=
 needs to do something or else the puppy will die is not helpful.<br><div c=
lass=3D"gmail_extra"><br></div><div class=3D"gmail_extra">Regards,<br clear=
=3D"all"><div><div class=3D"gmail_signature">_____________<br>Roman Shpount=
</div></div></div><div>=C2=A0</div></div></div></div>

--089e0122a5b8dc0843051c2c23ab--


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From: <richard.vandet@kpn.com>
To: <Felix.Wyss@inin.com>
Thread-Topic: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it would be Christmas in July!)
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Date: Fri, 31 Jul 2015 13:56:52 +0000
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+1

M.vr.gr<http://M.vr.gr>.,

Richard van Det
06 2054 7291

Op 31 jul. 2015 om 12:52 heeft Wyss, Felix <Felix.Wyss@inin.com<mailto:Feli=
x.Wyss@inin.com>> het volgende geschreven:


As somebody who has to interoperate between WebRTC and legacy VoIP, I whole=
heartedly support making RTP/RTCP muxing mandatory.  Having just one "pipe"=
 for everything is IMHO an incredibly compelling feature of WebRTC.  Furthe=
rmore, the fewer RECOMMENDED vs. MUST we have in the specification, the mor=
e likely the WebRTC endpoints from different vendors will be interoperable =
without hacks or workarounds.

--Felix

________________________________
From: rtcweb <rtcweb-bounces@ietf.org<mailto:rtcweb-bounces@ietf.org>> on b=
ehalf of Asveren, Tolga <tasveren@sonusnet.com<mailto:tasveren@sonusnet.com=
>>
Sent: Friday, July 31, 2015 06:18
To: Eric Rescorla; Peter Thatcher
Cc: <rtcweb@ietf.org<mailto:rtcweb@ietf.org>>
Subject: Re: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it=
 would be Christmas in July!)


"The whole reason we didn't do this before was because we wanted to work wi=
th legacy equipment.  But Cullen's claim is that there is no legacy equipme=
nt that can speak ICE (and DTLS) but not rtcp-mux."


I am not sure whether this is entirely true, especially on GW front. It cou=
ld be a better idea to be lenient for enhanced interoperability and making =
RTCWeb more deployable in practice. Can't the decision of being firm on usi=
ng rtcp-mux left to the WebRTC endpoint with signaling aspects supporting n=
egotiation of the capability?


Thanks,

Tolga


________________________________
From: rtcweb <rtcweb-bounces@ietf.org<mailto:rtcweb-bounces@ietf.org>> on b=
ehalf of Eric Rescorla <ekr@rtfm.com<mailto:ekr@rtfm.com>>
Sent: Thursday, July 30, 2015 11:04 PM
To: Peter Thatcher
Cc: <rtcweb@ietf.org<mailto:rtcweb@ietf.org>>
Subject: Re: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it=
 would be Christmas in July!)



On Thu, Jul 30, 2015 at 9:17 PM, Peter Thatcher <pthatcher@google.com<mailt=
o:pthatcher@google.com>> wrote:
At IETF 93 last week,  Cullen pointed out that he looked into ICE endpoints=
 and found that all of them support rtcp-mux.  Inspired by that, I went aro=
und and asked about a dozen people "would you be OK if we removed support f=
or non-muxed RTCP?"  The range of responses went from "I'm OK with it if Cu=
llen is" to "yeah, that would be great".  No one said no.  Cullen said yes.

For me, removing support for non-muxed RTCP would be like Christmas in July=
.

Realistically, August at this point.


So here's my proposal:  make RtcpMuxPolicy =3D=3D "required" be not only th=
e default, but the only RtcpMuxPolicy available.   All rtcweb endpoints mus=
t support rtcp-mux.

This seems reasonable.


An rtcweb endpoint would be free to fail if the remote endpoint does not su=
pport rtcp-mux.

Given the current specification, I believe this would need to be strong. I.=
e., an rtcweb
endpoint *must* fail if the remote endpoint does not support rtcp-mux. As s=
hould
be obvious, this only applies when the rtcweb endpoint is an answerer since=
 with
the above setting, you will not provide an offer that the other side can ac=
cept, but
the spec says:

"When acting as answerer, the browser will reject any m=3D section that doe=
s not provide an "a=3Drtcp-mux" attribute."

The reason for that is to avoid asymmetrical situations in which endpoint N=
M
(non-mux) makes an offer to endpoint M (mux) which is accepted, but then
re-offers from NM to M fail.

So I believe the best version of this proposal is simply requiring rtcp-mux
entirely.

-Ekr


The whole reason we didn't do this before was because we wanted to work wit=
h legacy equipment.  But Cullen's claim is that there is no legacy equipmen=
t that can speak ICE (and DTLS) but not rtcp-mux.

So is there any remaining reason that we cannot require rtcp-mux?




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


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

--_000_4DE3D9E88B364EC3A2D95A7C9BD01922kpncom_
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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
</head>
<body dir=3D"auto">
<div>&#43;1<br>
<br>
<a href=3D"http://M.vr.gr">M.vr.gr</a>.,
<div><br>
</div>
<div>Richard van Det</div>
<div>06 2054 7291</div>
</div>
<div><br>
Op 31 jul. 2015 om 12:52 heeft Wyss, Felix &lt;<a href=3D"mailto:Felix.Wyss=
@inin.com">Felix.Wyss@inin.com</a>&gt; het volgende geschreven:<br>
<br>
</div>
<blockquote type=3D"cite">
<div>
<div id=3D"divtagdefaultwrapper" style=3D"font-size:12pt;color:#000000;back=
ground-color:#FFFFFF;font-family:Calibri,Arial,Helvetica,sans-serif;">
<p>As somebody who has to interoperate between WebRTC and legacy VoIP, I wh=
oleheartedly support making RTP/RTCP muxing mandatory.&nbsp; Having just on=
e &quot;pipe&quot; for everything is IMHO an incredibly compelling feature =
of WebRTC.&nbsp; Furthermore, the fewer RECOMMENDED
 vs. MUST we have in the specification, the more likely the WebRTC endpoint=
s from different vendors will be interoperable without hacks or workarounds=
.&nbsp;&nbsp;
<br>
</p>
<br>
--Felix<br>
<br>
<div style=3D"color: rgb(0, 0, 0);">
<hr tabindex=3D"-1" style=3D"display:inline-block; width:98%">
<div id=3D"divRplyFwdMsg" dir=3D"ltr"><font style=3D"font-size:11pt" face=
=3D"Calibri, sans-serif" color=3D"#000000"><b>From:</b> rtcweb &lt;<a href=
=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a>&gt; on beha=
lf of Asveren, Tolga &lt;<a href=3D"mailto:tasveren@sonusnet.com">tasveren@=
sonusnet.com</a>&gt;<br>
<b>Sent:</b> Friday, July 31, 2015 06:18<br>
<b>To:</b> Eric Rescorla; Peter Thatcher<br>
<b>Cc:</b> &lt;<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>&gt;<b=
r>
<b>Subject:</b> Re: [rtcweb] Proposal: require rtcp-mux (remove non-muxed R=
TCP; it would be Christmas in July!)</font>
<div>&nbsp;</div>
</div>
<div>
<div id=3D"divtagdefaultwrapper" style=3D"font-size:12pt; color:#000000; ba=
ckground-color:#FFFFFF; font-family:Calibri,Arial,Helvetica,sans-serif">
<p><span style=3D"color:rgb(33,33,33); font-family:arial,helvetica,sans-ser=
if; font-size:15px">&quot;The whole reason we didn't do this before was bec=
ause we wanted to work with legacy equipment.&nbsp; But Cullen's claim is t=
hat there is no legacy equipment that can speak
 ICE (and DTLS) but not rtcp-mux.&quot;</span><br>
</p>
<p><span style=3D"color:rgb(33,33,33); font-family:arial,helvetica,sans-ser=
if; font-size:15px"><br>
</span></p>
<p><font face=3D"arial, helvetica, sans-serif" color=3D"#212121"><span styl=
e=3D"font-size:15px">I am not sure whether this is entirely true, especiall=
y on GW front. It could be a better idea to be lenient for enhanced interop=
erability and making RTCWeb more&nbsp;deployable
 in practice. Can't the decision of being firm on using rtcp-mux left to th=
e WebRTC endpoint with signaling aspects supporting negotiation of the capa=
bility?</span></font></p>
<p><span style=3D"color:rgb(33,33,33); font-family:arial,helvetica,sans-ser=
if; font-size:15px"><br>
</span></p>
<p><span style=3D"color:rgb(33,33,33); font-family:arial,helvetica,sans-ser=
if; font-size:15px">Thanks,</span></p>
<p><span style=3D"color:rgb(33,33,33); font-family:arial,helvetica,sans-ser=
if; font-size:15px">Tolga</span></p>
<br>
<br>
<div style=3D"color:rgb(0,0,0)">
<hr tabindex=3D"-1" style=3D"display:inline-block; width:98%">
<div id=3D"divRplyFwdMsg" dir=3D"ltr"><font style=3D"font-size:11pt" face=
=3D"Calibri, sans-serif" color=3D"#000000"><b>From:</b> rtcweb &lt;<a href=
=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a>&gt; on beha=
lf of Eric Rescorla &lt;<a href=3D"mailto:ekr@rtfm.com">ekr@rtfm.com</a>&gt=
;<br>
<b>Sent:</b> Thursday, July 30, 2015 11:04 PM<br>
<b>To:</b> Peter Thatcher<br>
<b>Cc:</b> &lt;<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>&gt;<b=
r>
<b>Subject:</b> Re: [rtcweb] Proposal: require rtcp-mux (remove non-muxed R=
TCP; it would be Christmas in July!)</font>
<div>&nbsp;</div>
</div>
<div>
<div dir=3D"ltr"><br>
<div class=3D"gmail_extra"><br>
<div class=3D"gmail_quote">On Thu, Jul 30, 2015 at 9:17 PM, Peter Thatcher =
<span dir=3D"ltr">
&lt;<a href=3D"mailto:pthatcher@google.com" target=3D"_blank">pthatcher@goo=
gle.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex; border-left:1=
px #ccc solid; padding-left:1ex">
<div dir=3D"ltr">
<div style=3D"font-family:arial,helvetica,sans-serif">At IETF 93 last week,=
 &nbsp;Cullen pointed out that he looked into ICE endpoints and found that =
all of them support rtcp-mux.&nbsp; Inspired by that, I went around and ask=
ed about a dozen people &quot;would you be OK if
 we removed support for non-muxed RTCP?&quot; &nbsp;The range of responses =
went from &quot;I'm OK with it if Cullen is&quot; to &quot;yeah, that would=
 be great&quot;.&nbsp; No one said no.&nbsp; Cullen said yes. &nbsp;</div>
<div style=3D"font-family:arial,helvetica,sans-serif"><br>
</div>
<div style=3D"font-family:arial,helvetica,sans-serif">For me, removing supp=
ort for non-muxed RTCP would be like Christmas in July.</div>
</div>
</blockquote>
<div><br>
</div>
<div>Realistically, August at this point.</div>
<div><br>
</div>
<div><br>
</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex; border-left:1=
px #ccc solid; padding-left:1ex">
<div dir=3D"ltr">
<div style=3D"font-family:arial,helvetica,sans-serif">So here's my proposal=
: &nbsp;make RtcpMuxPolicy =3D=3D &quot;required&quot; be not only the defa=
ult, but the only RtcpMuxPolicy available. &nbsp; All rtcweb endpoints must=
 support rtcp-mux. &nbsp;</div>
</div>
</blockquote>
<div><br>
</div>
<div>This seems reasonable.</div>
<div><br>
</div>
<div>&nbsp;</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex; border-left:1=
px #ccc solid; padding-left:1ex">
<div dir=3D"ltr">
<div style=3D"font-family:arial,helvetica,sans-serif">An rtcweb endpoint wo=
uld be free to fail if the remote endpoint does not support rtcp-mux.</div>
</div>
</blockquote>
<div><br>
</div>
<div>Given the current specification, I believe this would need to be stron=
g. I.e., an rtcweb</div>
<div>endpoint *must* fail if the remote endpoint does not support rtcp-mux.=
 As should&nbsp;</div>
<div>be obvious, this only applies when the rtcweb endpoint is an answerer =
since with</div>
<div>the above setting, you will not provide an offer that the other side c=
an accept, but</div>
<div>the spec says:</div>
<div><br>
</div>
<div>&quot;When acting as answerer, the browser will reject any m=3D sectio=
n that does not provide an &quot;a=3Drtcp-mux&quot; attribute.&quot;</div>
<div><br>
</div>
<div>The reason for that is to avoid asymmetrical situations in which endpo=
int NM</div>
<div>(non-mux) makes an offer to endpoint M (mux) which is accepted, but th=
en</div>
<div>re-offers from NM to M fail.</div>
<div><br>
</div>
<div>So I believe the best version of this proposal is simply requiring rtc=
p-mux</div>
<div>entirely.</div>
<div><br>
</div>
<div>-Ekr</div>
<div><br>
</div>
<div>&nbsp;</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex; border-left:1=
px #ccc solid; padding-left:1ex">
<div dir=3D"ltr">
<div style=3D"font-family:arial,helvetica,sans-serif"></div>
<div style=3D"font-family:arial,helvetica,sans-serif">The whole reason we d=
idn't do this before was because we wanted to work with legacy equipment.&n=
bsp; But Cullen's claim is that there is no legacy equipment that can speak=
 ICE (and DTLS) but not rtcp-mux.&nbsp;</div>
<div style=3D"font-family:arial,helvetica,sans-serif"><br>
</div>
<div style=3D"font-family:arial,helvetica,sans-serif">So is there any remai=
ning reason that we cannot require rtcp-mux?</div>
<div style=3D"font-family:arial,helvetica,sans-serif"><br>
</div>
<div style=3D"font-family:arial,helvetica,sans-serif"><br>
</div>
<div style=3D"font-family:arial,helvetica,sans-serif"><br>
</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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br>
</blockquote>
</div>
<br>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</blockquote>
<blockquote type=3D"cite">
<div><span>_______________________________________________</span><br>
<span>rtcweb mailing list</span><br>
<span><a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a></span><br>
<span><a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.=
ietf.org/mailman/listinfo/rtcweb</a></span><br>
</div>
</blockquote>
</body>
</html>

--_000_4DE3D9E88B364EC3A2D95A7C9BD01922kpncom_--


From nobody Fri Jul 31 07:12:34 2015
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From: "Asveren, Tolga" <tasveren@sonusnet.com>
To: Roman Shpount <roman@telurix.com>
Thread-Topic: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it would be Christmas in July!)
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Date: Fri, 31 Jul 2015 14:12:13 +0000
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Archived-At: <http://mailarchive.ietf.org/arch/msg/rtcweb/blBCWyFcEaVxZwHRzURu5M5BukQ>
Cc: "<rtcweb@ietf.org>" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it would be Christmas in July!)
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From: Bernard Aboba <bernard.aboba@gmail.com>
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To: Roman Shpount <roman@telurix.com>
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Cc: "<rtcweb@ietf.org>" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it would be Christmas in July!)
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On Jul 31, 2015, at 06:45, Roman Shpount <roman@telurix.com> wrote:
>=20
> The whole point of this discussion is to remove the negotiation for rtcp-m=
ux.

[BA] I support removing the mandate for RTP/RTCP non-mux but do not support r=
emoving the ability to negotiate it.  If an implementation supports it,  why=
 shouldn't it be possible to negotiate its use?

> It would be better to reduce the code size by removing rtcp-mux negotiatio=
n and all the code related to collecting second set of candidates for RTCP c=
onnection, supporting second component per media stream, negotiating second D=
TLS-SRTP for RTCP connection, etc.This is a non-trivial amount of code, whic=
h is not being actively tested right now. Most of the open source webrtc gat=
eways I looked at have it broken.

[BA] I agree that removing this code is attractive, particularly given the t=
ricky issues that arise in supporting non-mux properly.=20=

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<html><head><meta http-equiv=3D"content-type" content=3D"text/html; charset=3D=
utf-8"></head><body dir=3D"auto"><div>On Jul 31, 2015, at 06:45, Roman Shpou=
nt &lt;<a href=3D"mailto:roman@telurix.com">roman@telurix.com</a>&gt; wrote:=
</div><blockquote type=3D"cite"><div dir=3D"ltr"><div class=3D"gmail_extra">=
<div class=3D"gmail_quote"><div><br></div><div>The whole point of this discu=
ssion is to remove the negotiation for rtcp-mux.</div></div></div></div></bl=
ockquote><div><br></div><div>[BA] I support removing the mandate for RTP/RTC=
P non-mux but do not support removing the ability to negotiate it. &nbsp;If a=
n implementation supports it, &nbsp;why shouldn't it be possible to negotiat=
e its use?</div><div><br></div><blockquote type=3D"cite"><div dir=3D"ltr"><d=
iv class=3D"gmail_extra"><div class=3D"gmail_quote"><div>It would be better t=
o reduce the code size by removing rtcp-mux negotiation and all the code rel=
ated to collecting second set of candidates for RTCP connection, supporting s=
econd component per media stream, negotiating second DTLS-SRTP for RTCP conn=
ection, etc.<font color=3D"#000000"><span style=3D"background-color: rgba(25=
5, 255, 255, 0);">This is a non-trivial amount of code, which is not being a=
ctively tested right now. Most of the open source webrtc gateways I looked a=
t have it broken.</span></font></div></div></div></div></blockquote><div><br=
></div><div>[BA] I agree that removing this code is attractive, particularly=
 given the tricky issues that arise in supporting non-mux properly.&nbsp;</d=
iv></body></html>=

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From: Justin Uberti <juberti@google.com>
Date: Fri, 31 Jul 2015 09:44:56 -0700
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To: Roman Shpount <roman@telurix.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Prompt for mic, camera *and* network access
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On the Chrome side, we have taken the concrete step of allowing this
functionality to be controlled via a Chrome extension:
https://chrome.google.com/webstore/detail/webrtc-network-limiter/npeicpdbkakmehahjeeohfdhnlpdklia/reviews?hl=en

We are still sorting out some issues, which is why it is currently opt-in
(i.e., an extension). However, we expect to have a more comprehensive
solution in the near future.

On Fri, Jul 31, 2015 at 6:53 AM, Roman Shpount <roman@telurix.com> wrote:

> On Fri, Jul 31, 2015 at 6:47 AM, carlo von lynX <
> lynX@i.know.you.are.psyced.org> wrote:
>
>> Alright, thanks Tim for digging deeper and confirming that
>> we have a serious problem here. Unfortunately everybody
>> else is talking of other things and giving the impression
>> we are having just a side discussion.. and the problem
>> remains: WebRTC threatens lives of dissidents. If we can't
>> publish news that rtcweb has fixed the problem then the
>> logical consequence is to publish that n months have passed
>> and rtcweb is still not willing to fix the issue.... which
>> can be interpreted as a political news all on its own - how
>> industry does not care about collateral damages. Certainly
>> we can't just keep quiet and let this become a working
>> strategy to ignore issues.
>>
>> So, dear folks, this is your chance to keep the next
>> shitstorm from coming your way. Do something!
>>
>>
> In IETF we typically discuss technical issues and not political PR. So,
> the other discussions that you so blatantly ignore are actually about the
> actual technical implications of webrtc on user privacy and possible ways
> to improve it. If you have a particular technical solution that you want
> propose then please do so. Screaming that industry needs to do something or
> else the puppy will die is not helpful.
>
> Regards,
> _____________
> Roman Shpount
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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<div dir=3D"ltr">On the Chrome side, we have taken the concrete step of all=
owing this functionality to be controlled via a Chrome extension:<div><a hr=
ef=3D"https://chrome.google.com/webstore/detail/webrtc-network-limiter/npei=
cpdbkakmehahjeeohfdhnlpdklia/reviews?hl=3Den">https://chrome.google.com/web=
store/detail/webrtc-network-limiter/npeicpdbkakmehahjeeohfdhnlpdklia/review=
s?hl=3Den</a><br></div><div><br></div><div>We are still sorting out some is=
sues, which is why it is currently opt-in (i.e., an extension). However, we=
 expect to have a more comprehensive solution in the near future.</div></di=
v><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Fri, Jul 31,=
 2015 at 6:53 AM, Roman Shpount <span dir=3D"ltr">&lt;<a href=3D"mailto:rom=
an@telurix.com" target=3D"_blank">roman@telurix.com</a>&gt;</span> wrote:<b=
r><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:=
1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div class=3D"gmail_extra=
"><div class=3D"gmail_quote"><span class=3D"">On Fri, Jul 31, 2015 at 6:47 =
AM, carlo von lynX <span dir=3D"ltr">&lt;<a href=3D"mailto:lynX@i.know.you.=
are.psyced.org" target=3D"_blank">lynX@i.know.you.are.psyced.org</a>&gt;</s=
pan> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex=
;border-left:1px #ccc solid;padding-left:1ex">Alright, thanks Tim for diggi=
ng deeper and confirming that<br>
we have a serious problem here. Unfortunately everybody<br>
else is talking of other things and giving the impression<br>
we are having just a side discussion.. and the problem<br>
remains: WebRTC threatens lives of dissidents. If we can&#39;t<br>
publish news that rtcweb has fixed the problem then the<br>
logical consequence is to publish that n months have passed<br>
and rtcweb is still not willing to fix the issue.... which<br>
can be interpreted as a political news all on its own - how<br>
industry does not care about collateral damages. Certainly<br>
we can&#39;t just keep quiet and let this become a working<br>
strategy to ignore issues.<br>
<br>
So, dear folks, this is your chance to keep the next<br>
shitstorm from coming your way. Do something!<br>
<br></blockquote><div><br></div></span>In IETF we typically discuss technic=
al issues and not political PR. So, the other discussions that you so blata=
ntly ignore are actually about the actual technical implications of webrtc =
on user privacy and possible ways to improve it. If you have a particular t=
echnical solution that you want propose then please do so. Screaming that i=
ndustry needs to do something or else the puppy will die is not helpful.<br=
><div class=3D"gmail_extra"><br></div><div class=3D"gmail_extra">Regards,<b=
r clear=3D"all"><div><div>_____________<span class=3D"HOEnZb"><font color=
=3D"#888888"><br>Roman Shpount</font></span></div></div></div><div>=C2=A0</=
div></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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>

--bcaec54864ba665d5e051c2e8bd4--


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From: Justin Uberti <juberti@google.com>
Date: Fri, 31 Jul 2015 09:51:16 -0700
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To: Bernard Aboba <bernard.aboba@gmail.com>
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Subject: Re: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it would be Christmas in July!)
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On Fri, Jul 31, 2015 at 9:20 AM, Bernard Aboba <bernard.aboba@gmail.com>
wrote:

> On Jul 31, 2015, at 06:45, Roman Shpount <roman@telurix.com> wrote:
>
>
> The whole point of this discussion is to remove the negotiation for
> rtcp-mux.
>
>
> [BA] I support removing the mandate for RTP/RTCP non-mux but do not
> support removing the ability to negotiate it.  If an implementation
> supports it,  why shouldn't it be possible to negotiate its use?
>

Are you saying that compliant RTCWEB endpoints MUST support muxed RTCP, but
MAY support non-muxed RTCP (i.e. it is not explicitly prohibited)? That
would work for us.

>
> It would be better to reduce the code size by removing rtcp-mux
> negotiation and all the code related to collecting second set of candidates
> for RTCP connection, supporting second component per media stream,
> negotiating second DTLS-SRTP for RTCP connection, etc.This is a
> non-trivial amount of code, which is not being actively tested right now.
> Most of the open source webrtc gateways I looked at have it broken.
>
>
> [BA] I agree that removing this code is attractive, particularly given the
> tricky issues that arise in supporting non-mux properly.
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Fri, Jul 31, 2015 at 9:20 AM, Bernard Aboba <span dir=3D"ltr">&lt;<a=
 href=3D"mailto:bernard.aboba@gmail.com" target=3D"_blank">bernard.aboba@gm=
ail.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 dir=3D=
"auto"><span class=3D""><div>On Jul 31, 2015, at 06:45, Roman Shpount &lt;<=
a href=3D"mailto:roman@telurix.com" target=3D"_blank">roman@telurix.com</a>=
&gt; wrote:</div><blockquote type=3D"cite"><div dir=3D"ltr"><div class=3D"g=
mail_extra"><div class=3D"gmail_quote"><div><br></div><div>The whole point =
of this discussion is to remove the negotiation for rtcp-mux.</div></div></=
div></div></blockquote><div><br></div></span><div>[BA] I support removing t=
he mandate for RTP/RTCP non-mux but do not support removing the ability to =
negotiate it.=C2=A0 If an implementation supports it, =C2=A0why shouldn&#39=
;t it be possible to negotiate its use?</div></div></blockquote><div><br></=
div><div>Are you saying that compliant RTCWEB endpoints MUST support muxed =
RTCP, but MAY support non-muxed RTCP (i.e. it is not explicitly prohibited)=
? That would work for us.</div><blockquote class=3D"gmail_quote" style=3D"m=
argin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"a=
uto"><div><br></div><blockquote type=3D"cite"><div dir=3D"ltr"><div class=
=3D"gmail_extra"><div class=3D"gmail_quote"><div>It would be better to redu=
ce the code size by removing rtcp-mux negotiation and all the code related =
to collecting second set of candidates for RTCP connection, supporting seco=
nd component per media stream, negotiating second DTLS-SRTP for RTCP connec=
tion, etc.<font color=3D"#000000"><span style=3D"background-color:rgba(255,=
255,255,0)">This is a non-trivial amount of code, which is not being active=
ly tested right now. Most of the open source webrtc gateways I looked at ha=
ve it broken.</span></font></div></div></div></div></blockquote><div><br></=
div><div>[BA] I agree that removing this code is attractive, particularly g=
iven the tricky issues that arise in supporting non-mux properly.=C2=A0</di=
v></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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div></div>

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From: Eric Rescorla <ekr@rtfm.com>
Date: Fri, 31 Jul 2015 19:03:18 +0200
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To: Justin Uberti <juberti@google.com>
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Subject: Re: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it would be Christmas in July!)
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On Fri, Jul 31, 2015 at 6:51 PM, Justin Uberti <juberti@google.com> wrote:

>
>
> On Fri, Jul 31, 2015 at 9:20 AM, Bernard Aboba <bernard.aboba@gmail.com>
> wrote:
>
>> On Jul 31, 2015, at 06:45, Roman Shpount <roman@telurix.com> wrote:
>>
>>
>> The whole point of this discussion is to remove the negotiation for
>> rtcp-mux.
>>
>>
>> [BA] I support removing the mandate for RTP/RTCP non-mux but do not
>> support removing the ability to negotiate it.  If an implementation
>> supports it,  why shouldn't it be possible to negotiate its use?
>>
>
> Are you saying that compliant RTCWEB endpoints MUST support muxed RTCP,
> but MAY support non-muxed RTCP (i.e. it is not explicitly prohibited)? That
> would work for us.
>

In this case, would browsers be able to reject attempts to set the policy to
anything other than always-mux? Because I'm pretty sure that's what both
we and Chrome want to do.

-Ekr

It would be better to reduce the code size by removing rtcp-mux negotiation
>> and all the code related to collecting second set of candidates for RTCP
>> connection, supporting second component per media stream, negotiating
>> second DTLS-SRTP for RTCP connection, etc.This is a non-trivial amount
>> of code, which is not being actively tested right now. Most of the open
>> source webrtc gateways I looked at have it broken.
>>
>>
>> [BA] I agree that removing this code is attractive, particularly given
>> the tricky issues that arise in supporting non-mux properly.
>>
>> _______________________________________________
>> 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
>
>

--001a11c37eb8454333051c2ece3d
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Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Fri, Jul 31, 2015 at 6:51 PM, Justin Uberti <span dir=3D"ltr">&lt;<a=
 href=3D"mailto:juberti@google.com" target=3D"_blank">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"><div dir=3D"ltr"><br>=
<div class=3D"gmail_extra"><br><div class=3D"gmail_quote"><span class=3D"">=
On Fri, Jul 31, 2015 at 9:20 AM, Bernard Aboba <span dir=3D"ltr">&lt;<a hre=
f=3D"mailto:bernard.aboba@gmail.com" target=3D"_blank">bernard.aboba@gmail.=
com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"mar=
gin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"aut=
o"><span><div>On Jul 31, 2015, at 06:45, Roman Shpount &lt;<a href=3D"mailt=
o:roman@telurix.com" target=3D"_blank">roman@telurix.com</a>&gt; wrote:</di=
v><blockquote type=3D"cite"><div dir=3D"ltr"><div class=3D"gmail_extra"><di=
v class=3D"gmail_quote"><div><br></div><div>The whole point of this discuss=
ion is to remove the negotiation for rtcp-mux.</div></div></div></div></blo=
ckquote><div><br></div></span><div>[BA] I support removing the mandate for =
RTP/RTCP non-mux but do not support removing the ability to negotiate it.=
=C2=A0 If an implementation supports it, =C2=A0why shouldn&#39;t it be poss=
ible to negotiate its use?</div></div></blockquote><div><br></div></span><d=
iv>Are you saying that compliant RTCWEB endpoints MUST support muxed RTCP, =
but MAY support non-muxed RTCP (i.e. it is not explicitly prohibited)? That=
 would work for us.</div></div></div></div></blockquote><div><br></div><div=
>In this case, would browsers be able to reject attempts to set the policy =
to</div><div>anything other than always-mux? Because I&#39;m pretty sure th=
at&#39;s what both</div><div>we and Chrome want to do.</div><div><br></div>=
<div>-Ekr</div><div><br></div><blockquote class=3D"gmail_quote" style=3D"ma=
rgin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"lt=
r"><div class=3D"gmail_extra"><div class=3D"gmail_quote"><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex"><span class=3D""><div dir=3D"auto"><blockquote type=3D"cite">=
<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote"><div=
>It would be better to reduce the code size by removing rtcp-mux negotiatio=
n and all the code related to collecting second set of candidates for RTCP =
connection, supporting second component per media stream, negotiating secon=
d DTLS-SRTP for RTCP connection, etc.<font color=3D"#000000"><span style=3D=
"background-color:rgba(255,255,255,0)">This is a non-trivial amount of code=
, which is not being actively tested right now. Most of the open source web=
rtc gateways I looked at have it broken.</span></font></div></div></div></d=
iv></blockquote><div><br></div><div>[BA] I agree that removing this code is=
 attractive, particularly given the tricky issues that arise in supporting =
non-mux properly.=C2=A0</div></div><br></span><span class=3D"">____________=
___________________________________<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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></span></blockquote></div><br></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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div></div>

--001a11c37eb8454333051c2ece3d--


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Bernard Aboba <bernard.aboba@gmail.com>, Roman Shpount <roman@telurix.com>
Thread-Topic: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it would be Christmas in July!)
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References: <CAJrXDUGs_+AKNHhaJjB5udr9TZv7oFpJ7h3cJ_ToXUfUC5TEpQ@mail.gmail.com> <CABcZeBOH6zCEfuH6xh8-sPSmW3y4Vt2cDudfAAtw5MXq90oyWQ@mail.gmail.com> <SN1PR0301MB1551E28A489A90A93FA2F8BDB28A0@SN1PR0301MB1551.namprd03.prod.outlook.com> <CY1PR0501MB1579BC38B75CBC58744A970FEB8A0@CY1PR0501MB1579.namprd05.prod.outlook.com> <SN1PR0301MB15518198D375F21B973C091DB28A0@SN1PR0301MB1551.namprd03.prod.outlook.com> <CAD5OKxvUUSEyCfJcrxTJj68qKZuw6Ddze1Aw1QxXvTnGXT-qfA@mail.gmail.com>, <AF9C5DDA-451E-4ED4-A0AA-9853A3983D59@gmail.com>
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Cc: "<rtcweb@ietf.org>" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it would be Christmas in July!)
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Hi,

In my opinion, even if we mandate the usage of rtcp-mux, the negotiation (r=
ead: include the SDP rtcp-mux attribute in offers and answers) should not b=
e removed. If there are ways to signal features we should do it.

Regards,

Christer

Sent from my Windows Phone
________________________________
From: Bernard Aboba<mailto:bernard.aboba@gmail.com>
Sent: =FD31/=FD07/=FD2015 19:20
To: Roman Shpount<mailto:roman@telurix.com>
Cc: <rtcweb@ietf.org><mailto:rtcweb@ietf.org>
Subject: Re: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it=
 would be Christmas in July!)

On Jul 31, 2015, at 06:45, Roman Shpount <roman@telurix.com<mailto:roman@te=
lurix.com>> wrote:

The whole point of this discussion is to remove the negotiation for rtcp-mu=
x.

[BA] I support removing the mandate for RTP/RTCP non-mux but do not support=
 removing the ability to negotiate it.  If an implementation supports it,  =
why shouldn't it be possible to negotiate its use?

It would be better to reduce the code size by removing rtcp-mux negotiation=
 and all the code related to collecting second set of candidates for RTCP c=
onnection, supporting second component per media stream, negotiating second=
 DTLS-SRTP for RTCP connection, etc.This is a non-trivial amount of code, w=
hich is not being actively tested right now. Most of the open source webrtc=
 gateways I looked at have it broken.

[BA] I agree that removing this code is attractive, particularly given the =
tricky issues that arise in supporting non-mux properly.

--_000_7594FB04B1934943A5C02806D1A2204B348E29DAESESSMB209erics_
Content-Type: text/html; charset="windows-1256"
Content-Transfer-Encoding: quoted-printable

<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dwindows-1=
256">
</head>
<body dir=3D"auto">
<div>
<div style=3D"font-family:Calibri,sans-serif; font-size:11pt">Hi,<br>
<br>
In my opinion, even if we mandate the usage of rtcp-mux, the negotiation (r=
ead: include the SDP rtcp-mux attribute in offers and answers) should not b=
e removed. If there are ways to signal features we should do it.<br>
<br>
Regards,<br>
<br>
Christer<br>
<br>
Sent from my Windows Phone</div>
</div>
<div dir=3D"ltr">
<hr>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">From:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt"><a hr=
ef=3D"mailto:bernard.aboba@gmail.com">Bernard Aboba</a></span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">Sent:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt">=FD31=
/=FD07/=FD2015 19:20</span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">To:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt"><a hr=
ef=3D"mailto:roman@telurix.com">Roman Shpount</a></span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">Cc:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt"><a hr=
ef=3D"mailto:rtcweb@ietf.org">&lt;rtcweb@ietf.org&gt;</a></span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">Subject:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt">Re: [=
rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it would be Chri=
stmas in July!)</span><br>
<br>
</div>
<div>
<div>On Jul 31, 2015, at 06:45, Roman Shpount &lt;<a href=3D"mailto:roman@t=
elurix.com">roman@telurix.com</a>&gt; wrote:</div>
<blockquote type=3D"cite">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><br>
</div>
<div>The whole point of this discussion is to remove the negotiation for rt=
cp-mux.</div>
</div>
</div>
</div>
</blockquote>
<div><br>
</div>
<div>[BA] I support removing the mandate for RTP/RTCP non-mux but do not su=
pport removing the ability to negotiate it. &nbsp;If an implementation supp=
orts it, &nbsp;why shouldn't it be possible to negotiate its use?</div>
<div><br>
</div>
<blockquote type=3D"cite">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div>It would be better to reduce the code size by removing rtcp-mux negoti=
ation and all the code related to collecting second set of candidates for R=
TCP connection, supporting second component per media stream, negotiating s=
econd DTLS-SRTP for RTCP connection,
 etc.<font color=3D"#000000"><span style=3D"">This is a non-trivial amount =
of code, which is not being actively tested right now. Most of the open sou=
rce webrtc gateways I looked at have it broken.</span></font></div>
</div>
</div>
</div>
</blockquote>
<div><br>
</div>
<div>[BA] I agree that removing this code is attractive, particularly given=
 the tricky issues that arise in supporting non-mux properly.&nbsp;</div>
</div>
</body>
</html>

--_000_7594FB04B1934943A5C02806D1A2204B348E29DAESESSMB209erics_--


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From: Justin Uberti <juberti@google.com>
Date: Fri, 31 Jul 2015 10:11:37 -0700
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: "<rtcweb@ietf.org>" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it would be Christmas in July!)
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Mandating usage rtcp-mux means that an implementation is free to not
negotiate it (or fail if the remote side does not support it). Right?

On Fri, Jul 31, 2015 at 10:04 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> Hi,
>
> In my opinion, even if we mandate the usage of rtcp-mux, the negotiation
> (read: include the SDP rtcp-mux attribute in offers and answers) should n=
ot
> be removed. If there are ways to signal features we should do it.
>
> Regards,
>
> Christer
>
> Sent from my Windows Phone
> ------------------------------
> From: Bernard Aboba <bernard.aboba@gmail.com>
> Sent: =E2=80=8E31/=E2=80=8E07/=E2=80=8E2015 19:20
> To: Roman Shpount <roman@telurix.com>
> Cc: <rtcweb@ietf.org> <rtcweb@ietf.org>
> Subject: Re: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP;
> it would be Christmas in July!)
>
> On Jul 31, 2015, at 06:45, Roman Shpount <roman@telurix.com> wrote:
>
>
> The whole point of this discussion is to remove the negotiation for
> rtcp-mux.
>
>
> [BA] I support removing the mandate for RTP/RTCP non-mux but do not
> support removing the ability to negotiate it.  If an implementation
> supports it,  why shouldn't it be possible to negotiate its use?
>
> It would be better to reduce the code size by removing rtcp-mux
> negotiation and all the code related to collecting second set of candidat=
es
> for RTCP connection, supporting second component per media stream,
> negotiating second DTLS-SRTP for RTCP connection, etc.This is a
> non-trivial amount of code, which is not being actively tested right now.
> Most of the open source webrtc gateways I looked at have it broken.
>
>
> [BA] I agree that removing this code is attractive, particularly given th=
e
> tricky issues that arise in supporting non-mux properly.
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--bcaec517a990d0befc051c2eea3a
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<div dir=3D"ltr">Mandating usage rtcp-mux means that an implementation is f=
ree to not negotiate it (or fail if the remote side does not support it). R=
ight?<div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Fri, Jul =
31, 2015 at 10:04 AM, Christer Holmberg <span dir=3D"ltr">&lt;<a href=3D"ma=
ilto:christer.holmberg@ericsson.com" target=3D"_blank">christer.holmberg@er=
icsson.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 dir=3D"auto">
<div>
<div style=3D"font-family:Calibri,sans-serif;font-size:11pt">Hi,<br>
<br>
In my opinion, even if we mandate the usage of rtcp-mux, the negotiation (r=
ead: include the SDP rtcp-mux attribute in offers and answers) should not b=
e removed. If there are ways to signal features we should do it.<br>
<br>
Regards,<br>
<br>
Christer<br>
<br>
Sent from my Windows Phone</div>
</div>
<div dir=3D"ltr">
<hr>
<span style=3D"font-family:Calibri,sans-serif;font-size:11pt;font-weight:bo=
ld">From:
</span><span style=3D"font-family:Calibri,sans-serif;font-size:11pt"><a hre=
f=3D"mailto:bernard.aboba@gmail.com" target=3D"_blank">Bernard Aboba</a></s=
pan><br>
<span style=3D"font-family:Calibri,sans-serif;font-size:11pt;font-weight:bo=
ld">Sent:
</span><span style=3D"font-family:Calibri,sans-serif;font-size:11pt">=E2=80=
=8E31/=E2=80=8E07/=E2=80=8E2015 19:20</span><br>
<span style=3D"font-family:Calibri,sans-serif;font-size:11pt;font-weight:bo=
ld">To:
</span><span style=3D"font-family:Calibri,sans-serif;font-size:11pt"><a hre=
f=3D"mailto:roman@telurix.com" target=3D"_blank">Roman Shpount</a></span><s=
pan><br>
<span style=3D"font-family:Calibri,sans-serif;font-size:11pt;font-weight:bo=
ld">Cc:
</span><span style=3D"font-family:Calibri,sans-serif;font-size:11pt"><a hre=
f=3D"mailto:rtcweb@ietf.org" target=3D"_blank">&lt;rtcweb@ietf.org&gt;</a><=
/span><br>
<span style=3D"font-family:Calibri,sans-serif;font-size:11pt;font-weight:bo=
ld">Subject:
</span><span style=3D"font-family:Calibri,sans-serif;font-size:11pt">Re: [r=
tcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it would be Chris=
tmas in July!)</span><br>
<br>
</span></div><div><div>
<div>
<div>On Jul 31, 2015, at 06:45, Roman Shpount &lt;<a href=3D"mailto:roman@t=
elurix.com" target=3D"_blank">roman@telurix.com</a>&gt; wrote:</div>
<blockquote type=3D"cite">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div><br>
</div>
<div>The whole point of this discussion is to remove the negotiation for rt=
cp-mux.</div>
</div>
</div>
</div>
</blockquote>
<div><br>
</div>
<div>[BA] I support removing the mandate for RTP/RTCP non-mux but do not su=
pport removing the ability to negotiate it.=C2=A0 If an implementation supp=
orts it, =C2=A0why shouldn&#39;t it be possible to negotiate its use?</div>
<div><br>
</div>
<blockquote type=3D"cite">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div>It would be better to reduce the code size by removing rtcp-mux negoti=
ation and all the code related to collecting second set of candidates for R=
TCP connection, supporting second component per media stream, negotiating s=
econd DTLS-SRTP for RTCP connection,
 etc.<font color=3D"#000000"><span>This is a non-trivial amount of code, wh=
ich is not being actively tested right now. Most of the open source webrtc =
gateways I looked at have it broken.</span></font></div>
</div>
</div>
</div>
</blockquote>
<div><br>
</div>
<div>[BA] I agree that removing this code is attractive, particularly given=
 the tricky issues that arise in supporting non-mux properly.=C2=A0</div>
</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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div></div>

--bcaec517a990d0befc051c2eea3a--


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On Fri, Jul 31, 2015 at 1:11 PM, Justin Uberti <juberti@google.com> wrote:

> Mandating usage rtcp-mux means that an implementation is free to not
> negotiate it (or fail if the remote side does not support it). Right?
>
>
What I want to see precisely is the following:

1. WebRTC end point MUST include rtcp-mux for every SDP m-line in any offer
it generates
2. WebRTC end point MAY include RTCP ICE component, RTCP ICE candidates,
and "rtcp" SDP attribute with distinct RTCP address/port in the offer if it
wants to support non mux RTCP traffic or it MAY ommit all of those
attributes.
3. WebRTC end point MUST include rtcp-mux in response to any offer which
contains rtcp-mux. It should not include RTCP ICE components or RTCP ICE
candidates in the answer. "rtcp" SDP attribute MUST contain only port which
should be the same as port associated with m-line (or address from c-line
and port from m-line).
4. WebRTC end point MAY accept offers without rtcp-mux or MAY reject them.

So, basically, rtcp-mux must be offered and must be accepted if offered.
End point can continue to create offers which will work without rtcp-mux or
accept offers without rtcp-mux if it needs to interop with legacy.
_____________
Roman Shpount

--089e0122a5b84f999a051c2f2192
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div><div class=3D"gmail_signat=
ure">On Fri, Jul 31, 2015 at 1:11 PM, Justin Uberti <span dir=3D"ltr">&lt;<=
a href=3D"mailto:juberti@google.com" target=3D"_blank">juberti@google.com</=
a>&gt;</span> wrote:<br></div></div><div class=3D"gmail_quote"><blockquote =
class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid=
;padding-left:1ex"><div dir=3D"ltr">Mandating usage rtcp-mux means that an =
implementation is free to not negotiate it (or fail if the remote side does=
 not support it). Right?<div class=3D"gmail_extra"><div class=3D"gmail_quot=
e"><div><div class=3D"h5"><br></div></div></div></div></div></blockquote><d=
iv><br></div><div>What I want to see precisely is the following:</div><div>=
<br></div><div>1. WebRTC end point MUST include rtcp-mux for every=C2=A0SDP=
=C2=A0m-line in any offer it generates</div><div>2. WebRTC end point MAY in=
clude RTCP ICE component, RTCP ICE candidates, and &quot;rtcp&quot; SDP att=
ribute with distinct RTCP address/port in the offer if it wants to support =
non mux RTCP traffic or it MAY ommit all of those attributes.</div><div>3. =
WebRTC end point MUST include rtcp-mux in response to any offer which conta=
ins rtcp-mux. It should not include RTCP ICE components or RTCP ICE candida=
tes in the answer. &quot;rtcp&quot; SDP attribute MUST contain only port wh=
ich should be the same as port associated with m-line (or address from c-li=
ne and port from m-line).</div><div>4. WebRTC end point MAY accept offers w=
ithout rtcp-mux or MAY reject them.</div><div><br></div><div>So, basically,=
 rtcp-mux must be offered and must be accepted if offered. End point can co=
ntinue to create offers which will work without rtcp-mux or accept offers w=
ithout rtcp-mux if it needs to interop with legacy.</div><div><div class=3D=
"gmail_signature">_____________<br>Roman Shpount</div></div><div>=C2=A0</di=
v></div></div></div>

--089e0122a5b84f999a051c2f2192--


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Date: Fri, 31 Jul 2015 19:31:57 +0200
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From: Victor Pascual Avila <victor.pascual.avila@gmail.com>
To: Roman Shpount <roman@telurix.com>
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Subject: Re: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it would be Christmas in July!)
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--047d7bf0c59c574a89051c2f323f
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On Fri, Jul 31, 2015 at 7:27 PM, Roman Shpount <roman@telurix.com> wrote:

> On Fri, Jul 31, 2015 at 1:11 PM, Justin Uberti <juberti@google.com> wrote:
>
>> Mandating usage rtcp-mux means that an implementation is free to not
>> negotiate it (or fail if the remote side does not support it). Right?
>>
>>
> What I want to see precisely is the following:
>
> 1. WebRTC end point MUST include rtcp-mux for every SDP m-line in any
> offer it generates
> 2. WebRTC end point MAY include RTCP ICE component, RTCP ICE candidates,
> and "rtcp" SDP attribute with distinct RTCP address/port in the offer if it
> wants to support non mux RTCP traffic or it MAY ommit all of those
> attributes.
> 3. WebRTC end point MUST include rtcp-mux in response to any offer which
> contains rtcp-mux. It should not include RTCP ICE components or RTCP ICE
> candidates in the answer. "rtcp" SDP attribute MUST contain only port which
> should be the same as port associated with m-line (or address from c-line
> and port from m-line).
> 4. WebRTC end point MAY accept offers without rtcp-mux or MAY reject them.
>
> So, basically, rtcp-mux must be offered and must be accepted if offered.
> End point can continue to create offers which will work without rtcp-mux or
> accept offers without rtcp-mux if it needs to interop with legacy.
>

+1

Cheers,
-Victor

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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On F=
ri, Jul 31, 2015 at 7:27 PM, Roman Shpount <span dir=3D"ltr">&lt;<a href=3D=
"mailto:roman@telurix.com" target=3D"_blank">roman@telurix.com</a>&gt;</spa=
n> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;b=
order-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div class=3D"=
gmail_extra"><span class=3D""><div><div>On Fri, Jul 31, 2015 at 1:11 PM, Ju=
stin Uberti <span dir=3D"ltr">&lt;<a href=3D"mailto:juberti@google.com" tar=
get=3D"_blank">juberti@google.com</a>&gt;</span> wrote:<br></div></div></sp=
an><div class=3D"gmail_quote"><span class=3D""><blockquote class=3D"gmail_q=
uote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1e=
x"><div dir=3D"ltr">Mandating usage rtcp-mux means that an implementation i=
s free to not negotiate it (or fail if the remote side does not support it)=
. Right?<div class=3D"gmail_extra"><div class=3D"gmail_quote"><div><div><br=
></div></div></div></div></div></blockquote><div><br></div></span><div>What=
 I want to see precisely is the following:</div><div><br></div><div>1. WebR=
TC end point MUST include rtcp-mux for every=C2=A0SDP=C2=A0m-line in any of=
fer it generates</div><div>2. WebRTC end point MAY include RTCP ICE compone=
nt, RTCP ICE candidates, and &quot;rtcp&quot; SDP attribute with distinct R=
TCP address/port in the offer if it wants to support non mux RTCP traffic o=
r it MAY ommit all of those attributes.</div><div>3. WebRTC end point MUST =
include rtcp-mux in response to any offer which contains rtcp-mux. It shoul=
d not include RTCP ICE components or RTCP ICE candidates in the answer. &qu=
ot;rtcp&quot; SDP attribute MUST contain only port which should be the same=
 as port associated with m-line (or address from c-line and port from m-lin=
e).</div><div>4. WebRTC end point MAY accept offers without rtcp-mux or MAY=
 reject them.</div><div><br></div><div>So, basically, rtcp-mux must be offe=
red and must be accepted if offered. End point can continue to create offer=
s which will work without rtcp-mux or accept offers without rtcp-mux if it =
needs to interop with legacy.</div></div></div></div></blockquote><div><br>=
</div><div>+1</div><div><br></div><div>Cheers,</div><div>-Victor<br></div><=
/div>
</div></div>

--047d7bf0c59c574a89051c2f323f--


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To: Eric Rescorla <ekr@rtfm.com>
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Cc: "<rtcweb@ietf.org>" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it would be Christmas in July!)
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On Jul 31, 2015, at 10:03, Eric Rescorla <ekr@rtfm.com> wrote:
>>=20
>> Are you saying that compliant RTCWEB endpoints MUST support muxed RTCP, b=
ut MAY support non-muxed RTCP (i.e. it is not explicitly prohibited)? That w=
ould work for us.

[BA] Yes.  Since RTP-Usage already requires mux support, we would only need t=
o change the sentence requiring non-mux to make it optional.=20

> In this case, would browsers be able to reject attempts to set the policy t=
o
> anything other than always-mux? Because I'm pretty sure that's what both
> we and Chrome want to do.

[BA] Yes, that should be possible. If non-mux is optional, then an implement=
ation that doesn't support it would only allow setting the policy to always m=
ux.=20=

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<html><head><meta http-equiv=3D"content-type" content=3D"text/html; charset=3D=
utf-8"></head><body dir=3D"auto"><div>On Jul 31, 2015, at 10:03, Eric Rescor=
la &lt;<a href=3D"mailto:ekr@rtfm.com">ekr@rtfm.com</a>&gt; wrote:</div><blo=
ckquote type=3D"cite"><div dir=3D"ltr"><div class=3D"gmail_extra"><div class=
=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8e=
x;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div class=3D=
"gmail_extra"><div class=3D"gmail_quote"><span class=3D""><div><br></div></s=
pan><div>Are you saying that compliant RTCWEB endpoints MUST support muxed R=
TCP, but MAY support non-muxed RTCP (i.e. it is not explicitly prohibited)? T=
hat would work for us.</div></div></div></div></blockquote></div></div></div=
></blockquote><div><br></div><div>[BA] Yes. &nbsp;Since RTP-Usage already re=
quires mux support, we would only need to change the sentence requiring non-=
mux to make it optional.&nbsp;</div><div><br></div><blockquote type=3D"cite"=
><div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote"><div=
>In this case, would browsers be able to reject attempts to set the policy t=
o</div><div>anything other than always-mux? Because I'm pretty sure that's w=
hat both</div><div>we and Chrome want to do.</div></div></div></div>
</blockquote><br><div>[BA] Yes, that should be possible. If non-mux is optio=
nal, then an implementation that doesn't support it would only allow setting=
 the policy to always mux.&nbsp;</div></body></html>=

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From: Eric Rescorla <ekr@rtfm.com>
Date: Fri, 31 Jul 2015 19:42:00 +0200
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Subject: Re: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it would be Christmas in July!)
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On Fri, Jul 31, 2015 at 7:39 PM, Bernard Aboba <bernard.aboba@gmail.com>
wrote:

> On Jul 31, 2015, at 10:03, Eric Rescorla <ekr@rtfm.com> wrote:
>
>
>> Are you saying that compliant RTCWEB endpoints MUST support muxed RTCP,
>> but MAY support non-muxed RTCP (i.e. it is not explicitly prohibited)? That
>> would work for us.
>>
>
> [BA] Yes.  Since RTP-Usage already requires mux support, we would only
> need to change the sentence requiring non-mux to make it optional.
>
> In this case, would browsers be able to reject attempts to set the policy
> to
> anything other than always-mux? Because I'm pretty sure that's what both
> we and Chrome want to do.
>
>
> [BA] Yes, that should be possible. If non-mux is optional, then an
> implementation that doesn't support it would only allow setting the policy
> to always mux.
>

That seems satisfactory to me, since it means we can remove the code for
non-mux.

-Ekr

--001a11c37eb8a2f1ba051c2f5805
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Fri, Jul 31, 2015 at 7:39 PM, Bernard Aboba <span dir=3D"ltr">&lt;<a=
 href=3D"mailto:bernard.aboba@gmail.com" target=3D"_blank">bernard.aboba@gm=
ail.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 dir=3D=
"auto"><span class=3D""><div>On Jul 31, 2015, at 10:03, Eric Rescorla &lt;<=
a href=3D"mailto:ekr@rtfm.com" target=3D"_blank">ekr@rtfm.com</a>&gt; wrote=
:</div><blockquote type=3D"cite"><div dir=3D"ltr"><div class=3D"gmail_extra=
"><div class=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"mar=
gin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr=
"><div class=3D"gmail_extra"><div class=3D"gmail_quote"><span><div><br></di=
v></span><div>Are you saying that compliant RTCWEB endpoints MUST support m=
uxed RTCP, but MAY support non-muxed RTCP (i.e. it is not explicitly prohib=
ited)? That would work for us.</div></div></div></div></blockquote></div></=
div></div></blockquote><div><br></div></span><div>[BA] Yes.=C2=A0 Since RTP=
-Usage already requires mux support, we would only need to change the sente=
nce requiring non-mux to make it optional.=C2=A0</div><span class=3D""><div=
><br></div><blockquote type=3D"cite"><div dir=3D"ltr"><div class=3D"gmail_e=
xtra"><div class=3D"gmail_quote"><div>In this case, would browsers be able =
to reject attempts to set the policy to</div><div>anything other than alway=
s-mux? Because I&#39;m pretty sure that&#39;s what both</div><div>we and Ch=
rome want to do.</div></div></div></div>
</blockquote><br></span><div>[BA] Yes, that should be possible. If non-mux =
is optional, then an implementation that doesn&#39;t support it would only =
allow setting the policy to always mux.=C2=A0</div></div></blockquote></div=
><br></div><div class=3D"gmail_extra">That seems satisfactory to me, since =
it means we can remove the code for</div><div class=3D"gmail_extra">non-mux=
.</div><div class=3D"gmail_extra"><br></div><div class=3D"gmail_extra">-Ekr=
</div><div class=3D"gmail_extra"><br></div></div>

--001a11c37eb8a2f1ba051c2f5805--


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To: Roman Shpount <roman@telurix.com>
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Cc: "<rtcweb@ietf.org>" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it would be Christmas in July!)
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>> On Fri, Jul 31, 2015 at 7:27 PM, Roman Shpount <roman@telurix.com> wrote:=

>>=20
>> So, basically, rtcp-mux must be offered and must be accepted if offered. E=
nd point can continue to create offers which will work without rtcp-mux or a=
ccept offers without rtcp-mux if it needs to interop with legacy.

[BA] This is a bit weird because if a browser supporting non-mux negotiates w=
ith a gateway that supports non-mux, the implication is that if the browser i=
s the offerer, mux is negotiated, but if the gateway is the offerer, non-mux=
 can be negotiated. That would seem to imply that the gateway would need to r=
eject the browser's offer, then send its own offer in order to negotiate non=
-mux.=

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<html><head><meta http-equiv=3D"content-type" content=3D"text/html; charset=3D=
utf-8"></head><body dir=3D"auto"><blockquote type=3D"cite"><div dir=3D"ltr">=
<div class=3D"gmail_extra"><div class=3D"gmail_quote">On Fri, Jul 31, 2015 a=
t 7:27 PM, Roman Shpount <span dir=3D"ltr">&lt;<a href=3D"mailto:roman@telur=
ix.com" target=3D"_blank">roman@telurix.com</a>&gt;</span> wrote:<blockquote=
 class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid=
;padding-left:1ex"><div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D=
"gmail_quote"><div><br></div><div>So, basically, rtcp-mux must be offered an=
d must be accepted if offered. End point can continue to create offers which=
 will work without rtcp-mux or accept offers without rtcp-mux if it needs to=
 interop with legacy.</div></div></div></div></blockquote></div></div></div>=
</blockquote><br><div>[BA] This is a bit weird because if a browser supporti=
ng non-mux negotiates with a gateway that supports non-mux, the implication i=
s that if the browser is the offerer, mux is negotiated, but if the gateway i=
s the offerer, non-mux can be negotiated. That would seem to imply that the g=
ateway would need to reject the browser's offer, then send its own offer in o=
rder to negotiate non-mux.</div></body></html>=

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From: Roman Shpount <roman@telurix.com>
To: Bernard Aboba <bernard.aboba@gmail.com>
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Subject: Re: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it would be Christmas in July!)
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--089e0122a5b83685ea051c2f8f02
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On Fri, Jul 31, 2015 at 1:46 PM, Bernard Aboba <bernard.aboba@gmail.com>
wrote:

> On Fri, Jul 31, 2015 at 7:27 PM, Roman Shpount <roman@telurix.com> wrote:
>>
>>
>> So, basically, rtcp-mux must be offered and must be accepted if offered.
>> End point can continue to create offers which will work without rtcp-mux or
>> accept offers without rtcp-mux if it needs to interop with legacy.
>>
>
> [BA] This is a bit weird because if a browser supporting non-mux
> negotiates with a gateway that supports non-mux, the implication is that if
> the browser is the offerer, mux is negotiated, but if the gateway is the
> offerer, non-mux can be negotiated. That would seem to imply that the
> gateway would need to reject the browser's offer, then send its own offer
> in order to negotiate non-mux.
>

Why would the gateway need to negotiate non-mix if rtcp-mux is supported? I
have not seen anything that would normally accept an offer with rtcp-mux,
but will send an offer without rtcp-mux. In any case, if the gateway
behaves in such asymmetric manner, it will produce asymmetric results.
_____________
Roman Shpount

--089e0122a5b83685ea051c2f8f02
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div><div class=3D"gmail_signat=
ure">On Fri, Jul 31, 2015 at 1:46 PM, Bernard Aboba <span dir=3D"ltr">&lt;<=
a href=3D"mailto:bernard.aboba@gmail.com" target=3D"_blank">bernard.aboba@g=
mail.com</a>&gt;</span> wrote:<br></div></div><div class=3D"gmail_quote"><b=
lockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px =
#ccc solid;padding-left:1ex"><div dir=3D"auto"><blockquote type=3D"cite"><d=
iv dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote"><span =
class=3D"">On Fri, Jul 31, 2015 at 7:27 PM, Roman Shpount <span dir=3D"ltr"=
>&lt;<a href=3D"mailto:roman@telurix.com" target=3D"_blank">roman@telurix.c=
om</a>&gt;</span> wrote:</span><span class=3D""><blockquote class=3D"gmail_=
quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1=
ex"><div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">=
<div><br></div><div>So, basically, rtcp-mux must be offered and must be acc=
epted if offered. End point can continue to create offers which will work w=
ithout rtcp-mux or accept offers without rtcp-mux if it needs to interop wi=
th legacy.</div></div></div></div></blockquote></span></div></div></div></b=
lockquote><br><div>[BA] This is a bit weird because if a browser supporting=
 non-mux negotiates with a gateway that supports non-mux, the implication i=
s that if the browser is the offerer, mux is negotiated, but if the gateway=
 is the offerer, non-mux can be negotiated. That would seem to imply that t=
he gateway would need to reject the browser&#39;s offer, then send its own =
offer in order to negotiate non-mux.</div></div></blockquote></div><br></di=
v><div class=3D"gmail_extra">Why would the gateway need to negotiate non-mi=
x if rtcp-mux is supported? I have not seen anything that would normally ac=
cept an offer with rtcp-mux, but will send an offer without rtcp-mux. In an=
y case, if the gateway behaves in such asymmetric manner, it will produce a=
symmetric results.</div><div class=3D"gmail_extra"><div><div class=3D"gmail=
_signature">_____________<br>Roman Shpount</div></div><br></div></div>

--089e0122a5b83685ea051c2f8f02--


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To: Roman Shpount <roman@telurix.com>
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Subject: Re: [rtcweb] What the gateway draft should say about mux/non-mux
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On Jul 31, 2015, at 10:57, Roman Shpount <roman@telurix.com> wrote:
>=20
> Why would the gateway need to negotiate non-mux if rtcp-mux is supported?

[BA] IMHO, Gateways should be required to support mux like any other WEBRTC e=
ndpoint, but this is not what it says in Section 2 of https://tools.ietf.org=
/html/draft-ietf-rtcweb-gateways :

"If a gateway serves as a media relay into another RTP domain, it MAY choose=
 to support only features available in that network. This means that it MAY c=
hoose to not support Bundle and any of the RTP/ RTCP extensions related to i=
t, RTCP-Mux, or Trickle Ice. However, the gateway MUST support DTLS-SRTP, si=
nce this is required for interworking with WebRTC endpoints."

Assuming that browsers remove or do not implement non-mux, it seems prudent t=
o require gateways to support mux so as to avoid negotiation failures. If we=
 make that change then gateways would always negotiate RTCP-mux with browser=
s.=

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<html><head><meta http-equiv=3D"content-type" content=3D"text/html; charset=3D=
utf-8"></head><body dir=3D"auto"><div>On Jul 31, 2015, at 10:57, Roman Shpou=
nt &lt;<a href=3D"mailto:roman@telurix.com">roman@telurix.com</a>&gt; wrote:=
</div><blockquote type=3D"cite"><div dir=3D"ltr"><div class=3D"gmail_extra">=
<br></div><div class=3D"gmail_extra">Why would the gateway need to negotiate=
 non-mux if rtcp-mux is supported? </div></div></blockquote><div><br></div><=
div>[BA] IMHO, Gateways should be required to support mux like any other WEB=
RTC endpoint, but this is not what it says in Section 2 of <a href=3D"https:=
//tools.ietf.org/html/draft-ietf-rtcweb-gateways">https://tools.ietf.org/htm=
l/draft-ietf-rtcweb-gateways</a> :</div><div><br></div><div><pre class=3D"ne=
wpage" style=3D"margin-top: 0px; margin-bottom: 0px; page-break-before: alwa=
ys;"><font face=3D"UICTFontTextStyleBody"><span style=3D"white-space: normal=
; background-color: rgba(255, 255, 255, 0);">"If a gateway serves as a media=
 relay into another RTP domain, it MAY
   choose to support only features available in that network.  This
   means that it MAY choose to not support Bundle and any of the RTP/
   RTCP extensions related to it, RTCP-Mux, or Trickle Ice. However, the
   gateway MUST support DTLS-SRTP, since this is required for
   interworking with WebRTC endpoints."</span></font></pre><pre class=3D"new=
page" style=3D"margin-top: 0px; margin-bottom: 0px; page-break-before: alway=
s;"><font face=3D"UICTFontTextStyleBody"><span style=3D"white-space: normal;=
 background-color: rgba(255, 255, 255, 0);"><br></span></font></pre><pre cla=
ss=3D"newpage" style=3D"margin-top: 0px; margin-bottom: 0px; page-break-befo=
re: always;"><font face=3D"UICTFontTextStyleBody"><span style=3D"white-space=
: normal;">Assuming that browsers remove or do not implement non-mux, it see=
ms prudent to require gateways to support mux so as to avoid negotiation fai=
lures. If we make that change then gateways would always negotiate RTCP-mux w=
ith browsers.</span></font></pre></div></body></html>=

--Apple-Mail-B817CC60-2FEA-4BA5-B3F0-C7BA5CC5C66F--


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From: Justin Uberti <juberti@google.com>
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To: Eric Rescorla <ekr@rtfm.com>
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Subject: Re: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it would be Christmas in July!)
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likewise. That is the salient point here, after all.

On Fri, Jul 31, 2015 at 10:42 AM, Eric Rescorla <ekr@rtfm.com> wrote:

>
>
> On Fri, Jul 31, 2015 at 7:39 PM, Bernard Aboba <bernard.aboba@gmail.com>
> wrote:
>
>> On Jul 31, 2015, at 10:03, Eric Rescorla <ekr@rtfm.com> wrote:
>>
>>
>>> Are you saying that compliant RTCWEB endpoints MUST support muxed RTCP,
>>> but MAY support non-muxed RTCP (i.e. it is not explicitly prohibited)? That
>>> would work for us.
>>>
>>
>> [BA] Yes.  Since RTP-Usage already requires mux support, we would only
>> need to change the sentence requiring non-mux to make it optional.
>>
>> In this case, would browsers be able to reject attempts to set the policy
>> to
>> anything other than always-mux? Because I'm pretty sure that's what both
>> we and Chrome want to do.
>>
>>
>> [BA] Yes, that should be possible. If non-mux is optional, then an
>> implementation that doesn't support it would only allow setting the policy
>> to always mux.
>>
>
> That seems satisfactory to me, since it means we can remove the code for
> non-mux.
>
> -Ekr
>
>

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<div dir=3D"ltr">likewise. That is the salient point here, after all.</div>=
<div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Fri, Jul 31, 2=
015 at 10:42 AM, Eric Rescorla <span dir=3D"ltr">&lt;<a href=3D"mailto:ekr@=
rtfm.com" target=3D"_blank">ekr@rtfm.com</a>&gt;</span> wrote:<br><blockquo=
te class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc so=
lid;padding-left:1ex"><div dir=3D"ltr"><div><div class=3D"h5"><br><div clas=
s=3D"gmail_extra"><br><div class=3D"gmail_quote">On Fri, Jul 31, 2015 at 7:=
39 PM, Bernard Aboba <span dir=3D"ltr">&lt;<a href=3D"mailto:bernard.aboba@=
gmail.com" target=3D"_blank">bernard.aboba@gmail.com</a>&gt;</span> wrote:<=
br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left=
:1px #ccc solid;padding-left:1ex"><div dir=3D"auto"><span><div>On Jul 31, 2=
015, at 10:03, Eric Rescorla &lt;<a href=3D"mailto:ekr@rtfm.com" target=3D"=
_blank">ekr@rtfm.com</a>&gt; wrote:</div><blockquote type=3D"cite"><div dir=
=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote"><blockquote =
class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid=
;padding-left:1ex"><div dir=3D"ltr"><div class=3D"gmail_extra"><div class=
=3D"gmail_quote"><span><div><br></div></span><div>Are you saying that compl=
iant RTCWEB endpoints MUST support muxed RTCP, but MAY support non-muxed RT=
CP (i.e. it is not explicitly prohibited)? That would work for us.</div></d=
iv></div></div></blockquote></div></div></div></blockquote><div><br></div><=
/span><div>[BA] Yes.=C2=A0 Since RTP-Usage already requires mux support, we=
 would only need to change the sentence requiring non-mux to make it option=
al.=C2=A0</div><span><div><br></div><blockquote type=3D"cite"><div dir=3D"l=
tr"><div class=3D"gmail_extra"><div class=3D"gmail_quote"><div>In this case=
, would browsers be able to reject attempts to set the policy to</div><div>=
anything other than always-mux? Because I&#39;m pretty sure that&#39;s what=
 both</div><div>we and Chrome want to do.</div></div></div></div>
</blockquote><br></span><div>[BA] Yes, that should be possible. If non-mux =
is optional, then an implementation that doesn&#39;t support it would only =
allow setting the policy to always mux.=C2=A0</div></div></blockquote></div=
><br></div></div></div><div class=3D"gmail_extra">That seems satisfactory t=
o me, since it means we can remove the code for</div><div class=3D"gmail_ex=
tra">non-mux.</div><div class=3D"gmail_extra"><br></div><div class=3D"gmail=
_extra">-Ekr</div><div class=3D"gmail_extra"><br></div></div>
</blockquote></div><br></div>

--bcaec548aa59f81434051c300e0a--


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Justin Uberti <juberti@google.com>
Thread-Topic: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it would be Christmas in July!)
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Date: Fri, 31 Jul 2015 18:39:47 +0000
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Subject: Re: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it would be Christmas in July!)
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Roman Shpount <roman@telurix.com>, Justin Uberti <juberti@google.com>
Thread-Topic: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it would be Christmas in July!)
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Subject: Re: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it would be Christmas in July!)
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Bernard Aboba <bernard.aboba@gmail.com>, Roman Shpount <roman@telurix.com>
Thread-Topic: [rtcweb] What the gateway draft should say about mux/non-mux
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Date: Fri, 31 Jul 2015 18:43:42 +0000
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Subject: Re: [rtcweb] What the gateway draft should say about mux/non-mux
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From: "Rauschenbach, Uwe (Nokia - DE/Munich)" <uwe.rauschenbach@nokia.com>
To: ext Christer Holmberg <christer.holmberg@ericsson.com>, Roman Shpount <roman@telurix.com>, Justin Uberti <juberti@google.com>
Thread-Topic: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it would be Christmas in July!)
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Subject: Re: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it would be Christmas in July!)
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From nobody Fri Jul 31 11:45:35 2015
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From: "Rauschenbach, Uwe (Nokia - DE/Munich)" <uwe.rauschenbach@nokia.com>
To: ext Bernard Aboba <bernard.aboba@gmail.com>, Roman Shpount <roman@telurix.com>
Thread-Topic: [rtcweb] What the gateway draft should say about mux/non-mux
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Date: Fri, 31 Jul 2015 18:45:10 +0000
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Subject: Re: [rtcweb] What the gateway draft should say about mux/non-mux
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Subject: Re: [rtcweb] Prompt for mic, camera *and* network access
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On 7/31/15 11:44, Justin Uberti wrote:
> On the Chrome side, we have taken the concrete step of allowing this 
> functionality to be controlled via a Chrome extension:
> https://chrome.google.com/webstore/detail/webrtc-network-limiter/npeicpdbkakmehahjeeohfdhnlpdklia/reviews?hl=en
>
> We are still sorting out some issues, which is why it is currently 
> opt-in (i.e., an extension). However, we expect to have a more 
> comprehensive solution in the near future.

Mozilla is taking the same approach for Firefox; for details, see:

https://groups.google.com/forum/#!topic/mozilla.dev.media/L6Rx9ubSjMc

/a


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From: "Asveren, Tolga" <tasveren@sonusnet.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, Bernard Aboba <bernard.aboba@gmail.com>, Roman Shpount <roman@telurix.com>
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From: Roman Shpount <roman@telurix.com>
To: "Rauschenbach, Uwe (Nokia - DE/Munich)" <uwe.rauschenbach@nokia.com>
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Subject: Re: [rtcweb] What the gateway draft should say about mux/non-mux
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On Fri, Jul 31, 2015 at 2:45 PM, Rauschenbach, Uwe (Nokia - DE/Munich) <
uwe.rauschenbach@nokia.com> wrote:

> I would say that to interwork with legacy networks, browsers need to
> implement non-mux. Whether or not it is actually negotiated is a different
> story.
>
>
>
Can you provide an example of a legacy network that implements DTLS-SRTP
and does not implement rtcp-mux?

>From my point of view this is no different that requirement to support ICE.
This is supposed to be negotiated as well, but failure to negotiate ICE
with WebRTC end point will cause connection to fail. I think suggestion
here is that rtcp-mux should be treated the same.
_____________
Roman Shpount

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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On F=
ri, Jul 31, 2015 at 2:45 PM, Rauschenbach, Uwe (Nokia - DE/Munich) <span di=
r=3D"ltr">&lt;<a href=3D"mailto:uwe.rauschenbach@nokia.com" target=3D"_blan=
k">uwe.rauschenbach@nokia.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 lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;">I would say that to interwork with legacy=
 networks, browsers need to implement non-mux. Whether or not it is actuall=
y negotiated is a different story.</span><span style=3D"font-family:Arial,s=
ans-serif;font-size:10pt">=C2=A0</span></p>
<p class=3D"MsoNormal"><br></p></div></div></blockquote><div><br></div><div=
>Can you provide an example of a legacy network that implements DTLS-SRTP a=
nd does not implement rtcp-mux?</div><div><br></div><div>From my point of v=
iew this is no different that requirement to support ICE. This is supposed =
to be negotiated as well, but failure to negotiate ICE with WebRTC end poin=
t will cause connection to fail. I think suggestion here is that rtcp-mux s=
hould be treated the same.</div><div><div class=3D"gmail_signature">_______=
______<br>Roman Shpount</div></div><div>=C2=A0</div></div></div></div>

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From: "Rauschenbach, Uwe (Nokia - DE/Munich)" <uwe.rauschenbach@nokia.com>
To: "ext Asveren, Tolga" <tasveren@sonusnet.com>, Christer Holmberg <christer.holmberg@ericsson.com>, Bernard Aboba <bernard.aboba@gmail.com>, Roman Shpount <roman@telurix.com>
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From: Daniel Roesler <diafygi@gmail.com>
Date: Fri, 31 Jul 2015 11:59:01 -0700
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Subject: Re: [rtcweb] Prompt for mic, camera *and* network access
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I very much sympathize with carlo's frustration with this working
group. So far the response here to the introduced privacy exploits has
been pretty appalling. As you mentioned in the other consent thread,
this standard is the first time a browser can send requests over
non-default route network interfaces.

<strong>_This is a life threatening issue!_</strong>

Many, many, many Chinese/Arab/Persian users buy VPNs to get around
censorship and participate in free speech. The default configurations
for these VPNs set the default route, but often still allow requests
to be made over the real network interface (due to the way Windows
configures VPNs). Thus, when ICE sends STUN requests through that real
network interface, they expose the real IP of the VPN user. So any
website can silently identify VPN users, which can lead to a knock a
the door. If you don't believe me, just search for "ipleak" on
twitter[1] and see how many foreign language references there are to
the exploit test page.

Very real implication: Loading a website (or even an embedded ad) is
now much more life-threatening than it was before.

Thanks everyone! Great work!

I've heard many times on threads here that this is the fault of the
VPN user for not configuring their VPN better. I couldn't disagree
more. Up until WebRTC, the browser wasn't able to make non-default
route requests, which means that up until WebRTC, you couldn't expose
your real IP just by visiting a webpage. So WebRTC is the thing that
triggered exposure of the user, not their VPN configuration. Also,
blaming the user when they are using the default configuration (and
don't know any better) and you introduced something that exposes them
is offensive.

So far there have been two proposed solutions, both of which help
solve the issue.

1. Restrict STUN requests to default route. This obviously has
performance implications, so I understand why people here are hesitant
to require this.

2. Prompt the user for consent before sending STUN requests. This
would prevent silent IP leak exploits like the one we saw in the wild
embedded on the NYTimes.

How about a combined solution?

a) By default, restrict STUN requests to the default route.

b) Add an option to PeerConnection to use non-default routes, too.

c) When createOffer() or createAnswer() is called, if the non-default
routes flag is present, ask the user for consent. If there are video
or audio streams already attached to the peerconnection, assume that
consent has already been obtained (since getUserMedia asks for
consent).

This solution is backwards compatible with existing webapps, and
minimizes the introduction of any new prompts. The only change in user
experience is when a webapp explicitly wants to use non-default routes
for their connection. If a webapp is okay with the default route, then
they can silently add a data channel (no prompts).

I very much do not agree with Justin/Chrome's proposal to add a
configuration extension that requires a user to opt-in to restricting
non-default route STUN requests. That's like having the setting to
show the warning for self-signed https certificates turned off by
default and asking users to turn it on if they really want to be safe.
The default behavior should be safe.

Daniel

[1]: https://twitter.com/search?q=ipleak

On Fri, Jul 31, 2015 at 6:53 AM, Roman Shpount <roman@telurix.com> wrote:
> On Fri, Jul 31, 2015 at 6:47 AM, carlo von lynX
> <lynX@i.know.you.are.psyced.org> wrote:
>>
>> Alright, thanks Tim for digging deeper and confirming that
>> we have a serious problem here. Unfortunately everybody
>> else is talking of other things and giving the impression
>> we are having just a side discussion.. and the problem
>> remains: WebRTC threatens lives of dissidents. If we can't
>> publish news that rtcweb has fixed the problem then the
>> logical consequence is to publish that n months have passed
>> and rtcweb is still not willing to fix the issue.... which
>> can be interpreted as a political news all on its own - how
>> industry does not care about collateral damages. Certainly
>> we can't just keep quiet and let this become a working
>> strategy to ignore issues.
>>
>> So, dear folks, this is your chance to keep the next
>> shitstorm from coming your way. Do something!
>>
>
> In IETF we typically discuss technical issues and not political PR. So, the
> other discussions that you so blatantly ignore are actually about the actual
> technical implications of webrtc on user privacy and possible ways to
> improve it. If you have a particular technical solution that you want
> propose then please do so. Screaming that industry needs to do something or
> else the puppy will die is not helpful.
>
> Regards,
> _____________
> Roman Shpount
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


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From: Roman Shpount <roman@telurix.com>
To: "Asveren, Tolga" <tasveren@sonusnet.com>
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On Fri, Jul 31, 2015 at 2:49 PM, Asveren, Tolga <tasveren@sonusnet.com>
wrote:

> Yes, you **could** mandate the whole universe to support it, that
> shouldn=E2=80=99t be too difficult to describe on paper ;-)
>
>
>
We are trying to mandate that WebRTC endpoints (browsers and gateways) must
support. The rest of the universe can continue to negotiate rtcp-mux and/or
not support it.
_____________
Roman Shpount

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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On F=
ri, Jul 31, 2015 at 2:49 PM, Asveren, Tolga <span dir=3D"ltr">&lt;<a href=
=3D"mailto:tasveren@sonusnet.com" target=3D"_blank">tasveren@sonusnet.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 lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1f497d">Yes, you *<b>could</b>* mandate the w=
hole universe to support it, that shouldn=E2=80=99t be too difficult to des=
cribe on paper ;-)<u></u><u></u></span></p>
<p class=3D"MsoNormal"><br></p></div></div></blockquote><div><br></div><div=
>We are trying to mandate that WebRTC endpoints (browsers and gateways) mus=
t support. The rest of the universe can continue to negotiate rtcp-mux and/=
or not support it.</div><div><div class=3D"gmail_signature">_____________<b=
r>Roman Shpount</div></div><div>=C2=A0</div></div></div></div>

--001a113fd5e09541b5051c306b5d--


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From: Roman Shpount <roman@telurix.com>
To: "Rauschenbach, Uwe (Nokia - DE/Munich)" <uwe.rauschenbach@nokia.com>
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On Fri, Jul 31, 2015 at 2:58 PM, Rauschenbach, Uwe (Nokia - DE/Munich) <
uwe.rauschenbach@nokia.com> wrote:

> I think this is the core of the discussion that we are having =E2=80=93 d=
o we want
> to retire the original way RTP+RTCP were working w.r.t. port usage?
>
> I have my doubts (for the time being).
>

I agree this is the core of this discussion, so can you please clarify why
you are having doubts? What is you argument or use case on why original
RTP+RTCP on separate ports is needed in context of WebRTC?

Regards,
_____________
Roman Shpount

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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On F=
ri, Jul 31, 2015 at 2:58 PM, Rauschenbach, Uwe (Nokia - DE/Munich) <span di=
r=3D"ltr">&lt;<a href=3D"mailto:uwe.rauschenbach@nokia.com" target=3D"_blan=
k">uwe.rauschenbach@nokia.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 lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal"><span style=3D"font-family:Arial,sans-serif;font-siz=
e:10pt">I think this is the core of the discussion that we are having =E2=
=80=93 do we want to retire the original way RTP+RTCP were working w.r.t. p=
ort usage?</span><br></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;">I have my doubts (for the time being).</s=
pan></p></div></div></blockquote><div><br></div><div>I agree this is the co=
re of this discussion, so can you please clarify why you are having doubts?=
 What is you argument or use case on why original RTP+RTCP on separate port=
s is needed in context of WebRTC?</div><div><br></div><div>Regards,</div><d=
iv><div class=3D"gmail_signature">_____________<br>Roman Shpount</div></div=
><div>=C2=A0</div></div></div></div>

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From: Roman Shpount <roman@telurix.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [rtcweb] What the gateway draft should say about mux/non-mux
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On Fri, Jul 31, 2015 at 2:43 PM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> If we choose to mandate usage of rtcp-mux, we could also mandate gateways
> to support it.
>
>
+1
_____________
Roman Shpount

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<div dir=3D"ltr"><div class=3D"gmail_extra"><div><div class=3D"gmail_signat=
ure"><br></div></div><div class=3D"gmail_quote">On Fri, Jul 31, 2015 at 2:4=
3 PM, Christer Holmberg <span dir=3D"ltr">&lt;<a href=3D"mailto:christer.ho=
lmberg@ericsson.com" target=3D"_blank">christer.holmberg@ericsson.com</a>&g=
t;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0=
 .8ex;border-left:1px #ccc solid;padding-left:1ex">





<div lang=3D"EN-GB" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal"><span style=3D"color:rgb(31,73,125);font-family:Cali=
bri,sans-serif;font-size:11pt">If we choose to mandate usage of rtcp-mux, w=
e could also mandate gateways to support it.</span><br></p>
<p class=3D"MsoNormal"><a name=3D"14ee56d592baedad__MailEndCompose"><span s=
tyle=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#=
1f497d"></span></a></p></div></div></blockquote><div><br></div><div>+1</div=
><div><div class=3D"gmail_signature">_____________<br>Roman Shpount</div></=
div><div>=C2=A0</div></div></div></div>

--001a11403fd420f7e8051c3075b4--


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From: "Asveren, Tolga" <tasveren@sonusnet.com>
To: "Rauschenbach, Uwe (Nokia - DE/Munich)" <uwe.rauschenbach@nokia.com>, Christer Holmberg <christer.holmberg@ericsson.com>, Bernard Aboba <bernard.aboba@gmail.com>, Roman Shpount <roman@telurix.com>
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From: "Rauschenbach, Uwe (Nokia - DE/Munich)" <uwe.rauschenbach@nokia.com>
To: "ext Asveren, Tolga" <tasveren@sonusnet.com>, Christer Holmberg <christer.holmberg@ericsson.com>, Bernard Aboba <bernard.aboba@gmail.com>, Roman Shpount <roman@telurix.com>
Thread-Topic: [rtcweb] What the gateway draft should say about mux/non-mux
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From: Justin Uberti <juberti@google.com>
Date: Fri, 31 Jul 2015 12:27:20 -0700
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To: Daniel Roesler <diafygi@gmail.com>
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Subject: Re: [rtcweb] Prompt for mic, camera *and* network access
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On Fri, Jul 31, 2015 at 11:59 AM, Daniel Roesler <diafygi@gmail.com> wrote:

> I very much sympathize with carlo's frustration with this working
> group. So far the response here to the introduced privacy exploits has
> been pretty appalling. As you mentioned in the other consent thread,
> this standard is the first time a browser can send requests over
> non-default route network interfaces.
>
> <strong>_This is a life threatening issue!_</strong>
>
> Many, many, many Chinese/Arab/Persian users buy VPNs to get around
> censorship and participate in free speech. The default configurations
> for these VPNs set the default route, but often still allow requests
> to be made over the real network interface (due to the way Windows
> configures VPNs). Thus, when ICE sends STUN requests through that real
> network interface, they expose the real IP of the VPN user. So any
> website can silently identify VPN users, which can lead to a knock a
> the door. If you don't believe me, just search for "ipleak" on
> twitter[1] and see how many foreign language references there are to
> the exploit test page.




> Very real implication: Loading a website (or even an embedded ad) is
> now much more life-threatening than it was before.
>
> Thanks everyone! Great work!
>
> I've heard many times on threads here that this is the fault of the
> VPN user for not configuring their VPN better. I couldn't disagree
> more. Up until WebRTC, the browser wasn't able to make non-default
> route requests, which means that up until WebRTC, you couldn't expose
> your real IP just by visiting a webpage. So WebRTC is the thing that
> triggered exposure of the user, not their VPN configuration. Also,
> blaming the user when they are using the default configuration (and
> don't know any better) and you introduced something that exposes them
> is offensive.


I don't really want to debate nits, but this is not true. Flash has had
this 'feature', since the beginning of the modern browser era.
Regardless, we think we can do better.


> So far there have been two proposed solutions, both of which help
> solve the issue.
>
> 1. Restrict STUN requests to default route. This obviously has
> performance implications, so I understand why people here are hesitant
> to require this.
>
> 2. Prompt the user for consent before sending STUN requests. This
> would prevent silent IP leak exploits like the one we saw in the wild
> embedded on the NYTimes.
>
> How about a combined solution?
>
> a) By default, restrict STUN requests to the default route.
>
> b) Add an option to PeerConnection to use non-default routes, too.
>
> c) When createOffer() or createAnswer() is called, if the non-default
> routes flag is present, ask the user for consent. If there are video
> or audio streams already attached to the peerconnection, assume that
> consent has already been obtained (since getUserMedia asks for
> consent).
>
> This solution is backwards compatible with existing webapps, and
> minimizes the introduction of any new prompts. The only change in user
> experience is when a webapp explicitly wants to use non-default routes
> for their connection. If a webapp is okay with the default route, then
> they can silently add a data channel (no prompts).


> I very much do not agree with Justin/Chrome's proposal to add a
> configuration extension that requires a user to opt-in to restricting
> non-default route STUN requests. That's like having the setting to
> show the warning for self-signed https certificates turned off by
> default and asking users to turn it on if they really want to be safe.
> The default behavior should be safe.
>
>
I concur. As I said, what we have done in Chrome so far is just the first
step.

Opt-in allows us to ensure the solution a) works and b) doesn't break
anything, in a more contained population. Once we are confident with this,
we can move forward with a more comprehensive solution.

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Fri, Jul 31, 2015 at 11:59 AM, Daniel Roesler <span dir=3D"ltr">&lt;=
<a href=3D"mailto:diafygi@gmail.com" target=3D"_blank">diafygi@gmail.com</a=
>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 =
0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">I very much sympathiz=
e with carlo&#39;s frustration with this working<br>
group. So far the response here to the introduced privacy exploits has<br>
been pretty appalling. As you mentioned in the other consent thread,<br>
this standard is the first time a browser can send requests over<br>
non-default route network interfaces.<br>
<br>
&lt;strong&gt;_This is a life threatening issue!_&lt;/strong&gt;<br>
<br>
Many, many, many Chinese/Arab/Persian users buy VPNs to get around<br>
censorship and participate in free speech. The default configurations<br>
for these VPNs set the default route, but often still allow requests<br>
to be made over the real network interface (due to the way Windows<br>
configures VPNs). Thus, when ICE sends STUN requests through that real<br>
network interface, they expose the real IP of the VPN user. So any<br>
website can silently identify VPN users, which can lead to a knock a<br>
the door. If you don&#39;t believe me, just search for &quot;ipleak&quot; o=
n<br>
twitter[1] and see how many foreign language references there are to<br>
the exploit test page.=C2=A0</blockquote><blockquote class=3D"gmail_quote" =
style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">=C2=
=A0</blockquote><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8e=
x;border-left:1px #ccc solid;padding-left:1ex">
<br>
Very real implication: Loading a website (or even an embedded ad) is<br>
now much more life-threatening than it was before.<br>
<br>
Thanks everyone! Great work!<br>
<br>
I&#39;ve heard many times on threads here that this is the fault of the<br>
VPN user for not configuring their VPN better. I couldn&#39;t disagree<br>
more. Up until WebRTC, the browser wasn&#39;t able to make non-default<br>
route requests, which means that up until WebRTC, you couldn&#39;t expose<b=
r>
your real IP just by visiting a webpage. So WebRTC is the thing that<br>
triggered exposure of the user, not their VPN configuration. Also,<br>
blaming the user when they are using the default configuration (and<br>
don&#39;t know any better) and you introduced something that exposes them<b=
r>
is offensive.=C2=A0</blockquote><div><br></div><div>I don&#39;t really want=
 to debate nits, but this is not true. Flash has had this &#39;feature&#39;=
, since the beginning of the modern browser era.</div><div>Regardless, we t=
hink we can do better.</div><div>=C2=A0</div><blockquote class=3D"gmail_quo=
te" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"=
>
So far there have been two proposed solutions, both of which help<br>
solve the issue.<br>
<br>
1. Restrict STUN requests to default route. This obviously has<br>
performance implications, so I understand why people here are hesitant<br>
to require this.<br>
<br>
2. Prompt the user for consent before sending STUN requests. This<br>
would prevent silent IP leak exploits like the one we saw in the wild<br>
embedded on the NYTimes.<br>
<br>
How about a combined solution?<br>
<br>
a) By default, restrict STUN requests to the default route.<br>
<br>
b) Add an option to PeerConnection to use non-default routes, too.<br>
<br>
c) When createOffer() or createAnswer() is called, if the non-default<br>
routes flag is present, ask the user for consent. If there are video<br>
or audio streams already attached to the peerconnection, assume that<br>
consent has already been obtained (since getUserMedia asks for<br>
consent).<br>
<br>
This solution is backwards compatible with existing webapps, and<br>
minimizes the introduction of any new prompts. The only change in user<br>
experience is when a webapp explicitly wants to use non-default routes<br>
for their connection. If a webapp is okay with the default route, then<br>
they can silently add a data channel (no prompts).</blockquote><blockquote =
class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid=
;padding-left:1ex">
<br>
I very much do not agree with Justin/Chrome&#39;s proposal to add a<br>
configuration extension that requires a user to opt-in to restricting<br>
non-default route STUN requests. That&#39;s like having the setting to<br>
show the warning for self-signed https certificates turned off by<br>
default and asking users to turn it on if they really want to be safe.<br>
The default behavior should be safe.<br>
<br></blockquote><div><br></div><div>I concur. As I said, what we have done=
 in Chrome so far is just the first step.=C2=A0</div><div><br></div><div>Op=
t-in allows us to ensure the solution a) works and b) doesn&#39;t break any=
thing, in a more contained population. Once we are confident with this, we =
can move forward with a more comprehensive solution.</div></div></div></div=
>

--bcaec54864ba2a68c0051c30d0f9--


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References: <55352067.6070007@nostrum.com> <20150421154657.GA4071@lo.psyced.org> <20150715123803.GA24976@lo.psyced.org> <2BCEF545-D322-4589-AC85-AA80E7F934D9@phonefromhere.com> <20150715164129.GA27105@lo.psyced.org> <AF6485D4-1579-45AF-937C-1682AC1AA645@phonefromhere.com> <20150716084225.GA29652@lo.psyced.org> <0AF88216-374E-4DBA-9DB5-3E55B358E6AA@phonefromhere.com> <20150731104712.GA5228@lo.psyced.org> <CAD5OKxvYcBYLDXFYMh8DijZrYcEbkxf++WqWzrZF8UxjJxfrTQ@mail.gmail.com> <CA+65OsrXcMY92bR+AafvkBBvEcq_rbgeUpRv9C7OCzNBNXJYyA@mail.gmail.com>
Date: Fri, 31 Jul 2015 15:28:05 -0400
Message-ID: <CAD5OKxvvJ7D6_jVj9Z8CaMyet4BaB8nZFjJ2YWGwMSYDC7mvGA@mail.gmail.com>
From: Roman Shpount <roman@telurix.com>
To: Daniel Roesler <diafygi@gmail.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Prompt for mic, camera *and* network access
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--089e01295380b2e477051c30d157
Content-Type: text/plain; charset=UTF-8

On Fri, Jul 31, 2015 at 2:59 PM, Daniel Roesler <diafygi@gmail.com> wrote:

> I very much sympathize with carlo's frustration with this working
> group. So far the response here to the introduced privacy exploits has
> been pretty appalling. As you mentioned in the other consent thread,
> this standard is the first time a browser can send requests over
> non-default route network interfaces.
>
> <strong>_This is a life threatening issue!_</strong>
>

I actually appreciate the seriousness of the issue. I think part of the
reason this group did not respond in more timely manner is that not
everyone realized the behavior change (use of non-default routes). Once it
was clarified, the issue started to be handled. I understand that this and
any other security issue can be extremely frustrating to the end users, but
people, including people who build web browsers or design web browser
standards do make mistakes.


> Many, many, many Chinese/Arab/Persian users buy VPNs to get around
> censorship and participate in free speech. The default configurations
> for these VPNs set the default route, but often still allow requests
> to be made over the real network interface (due to the way Windows
> configures VPNs). Thus, when ICE sends STUN requests through that real
> network interface, they expose the real IP of the VPN user. So any
> website can silently identify VPN users, which can lead to a knock a
> the door. If you don't believe me, just search for "ipleak" on
> twitter[1] and see how many foreign language references there are to
> the exploit test page.
>
> Very real implication: Loading a website (or even an embedded ad) is
> now much more life-threatening than it was before.
>
> Thanks everyone! Great work!
>

Once again, it took a while to understand the real nature of the problem.
At least for me it took a while to identify what exactly changed and why it
is different from previous browser behavior.

I've heard many times on threads here that this is the fault of the
> VPN user for not configuring their VPN better. I couldn't disagree
> more. Up until WebRTC, the browser wasn't able to make non-default
> route requests, which means that up until WebRTC, you couldn't expose
> your real IP just by visiting a webpage. So WebRTC is the thing that
> triggered exposure of the user, not their VPN configuration. Also,
> blaming the user when they are using the default configuration (and
> don't know any better) and you introduced something that exposes them
> is offensive.
>

I actually agree with you here. I've spent considerable amount of effort to
explain the problem with VPN since I did not think it was completely
understood.


> So far there have been two proposed solutions, both of which help
> solve the issue.
>
> 1. Restrict STUN requests to default route. This obviously has
> performance implications, so I understand why people here are hesitant
> to require this.
>
> 2. Prompt the user for consent before sending STUN requests. This
> would prevent silent IP leak exploits like the one we saw in the wild
> embedded on the NYTimes.
>
> How about a combined solution?
>
> a) By default, restrict STUN requests to the default route.
>
> b) Add an option to PeerConnection to use non-default routes, too.
>
> c) When createOffer() or createAnswer() is called, if the non-default
> routes flag is present, ask the user for consent. If there are video
> or audio streams already attached to the peerconnection, assume that
> consent has already been obtained (since getUserMedia asks for
> consent).
>
> This solution is backwards compatible with existing webapps, and
> minimizes the introduction of any new prompts. The only change in user
> experience is when a webapp explicitly wants to use non-default routes
> for their connection. If a webapp is okay with the default route, then
> they can silently add a data channel (no prompts).
>

Justin is probably the best person to evaluate this solution. It looks
workable to me.

I very much do not agree with Justin/Chrome's proposal to add a
> configuration extension that requires a user to opt-in to restricting
> non-default route STUN requests. That's like having the setting to
> show the warning for self-signed https certificates turned off by
> default and asking users to turn it on if they really want to be safe.
> The default behavior should be safe.
>
>
I think you do not correctly characterize what Justin/Chrome is doing here.
It is not a final solution. It is a way to test the final solution before
it gets widely deployed. Since this is a major change for the browsers I
appreciate the need to deploy this carefully and gradually.
_____________
Roman Shpount

--089e01295380b2e477051c30d157
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On F=
ri, Jul 31, 2015 at 2:59 PM, Daniel Roesler <span dir=3D"ltr">&lt;<a href=
=3D"mailto:diafygi@gmail.com" target=3D"_blank">diafygi@gmail.com</a>&gt;</=
span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8e=
x;border-left:1px #ccc solid;padding-left:1ex">I very much sympathize with =
carlo&#39;s frustration with this working<br>
group. So far the response here to the introduced privacy exploits has<br>
been pretty appalling. As you mentioned in the other consent thread,<br>
this standard is the first time a browser can send requests over<br>
non-default route network interfaces.<br>
<br>
&lt;strong&gt;_This is a life threatening issue!_&lt;/strong&gt;<br></block=
quote><div><br></div><div>I actually appreciate the seriousness of the issu=
e. I think part of the reason this group did not respond in more timely man=
ner is that not everyone realized the behavior change (use of non-default r=
outes). Once it was clarified, the issue started to be handled. I understan=
d that this and any other security issue can be extremely frustrating to th=
e end users, but people, including people who build web browsers or design =
web browser standards do make mistakes.</div><div>=C2=A0</div><blockquote c=
lass=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;=
padding-left:1ex">Many, many, many Chinese/Arab/Persian users buy VPNs to g=
et around<br>
censorship and participate in free speech. The default configurations<br>
for these VPNs set the default route, but often still allow requests<br>
to be made over the real network interface (due to the way Windows<br>
configures VPNs). Thus, when ICE sends STUN requests through that real<br>
network interface, they expose the real IP of the VPN user. So any<br>
website can silently identify VPN users, which can lead to a knock a<br>
the door. If you don&#39;t believe me, just search for &quot;ipleak&quot; o=
n<br>
twitter[1] and see how many foreign language references there are to<br>
the exploit test page.<br>
<br>
Very real implication: Loading a website (or even an embedded ad) is<br>
now much more life-threatening than it was before.<br>
<br>
Thanks everyone! Great work!<br></blockquote><div><br></div><div>Once again=
, it took a while to understand the real nature of the problem. At least fo=
r me it took a while to identify what exactly changed and why it is differe=
nt from previous browser behavior.</div><div><br></div><blockquote class=3D=
"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding=
-left:1ex">I&#39;ve heard many times on threads here that this is the fault=
 of the<br>
VPN user for not configuring their VPN better. I couldn&#39;t disagree<br>
more. Up until WebRTC, the browser wasn&#39;t able to make non-default<br>
route requests, which means that up until WebRTC, you couldn&#39;t expose<b=
r>
your real IP just by visiting a webpage. So WebRTC is the thing that<br>
triggered exposure of the user, not their VPN configuration. Also,<br>
blaming the user when they are using the default configuration (and<br>
don&#39;t know any better) and you introduced something that exposes them<b=
r>
is offensive.<br></blockquote><div><br></div><div>I actually agree with you=
 here. I&#39;ve spent considerable amount of effort to explain the problem =
with VPN since I did not think it was completely understood.</div><div>=C2=
=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;borde=
r-left:1px #ccc solid;padding-left:1ex">So far there have been two proposed=
 solutions, both of which help<br>
solve the issue.<br>
<br>
1. Restrict STUN requests to default route. This obviously has<br>
performance implications, so I understand why people here are hesitant<br>
to require this.<br>
<br>
2. Prompt the user for consent before sending STUN requests. This<br>
would prevent silent IP leak exploits like the one we saw in the wild<br>
embedded on the NYTimes.<br>
<br>
How about a combined solution?<br>
<br>
a) By default, restrict STUN requests to the default route.<br>
<br>
b) Add an option to PeerConnection to use non-default routes, too.<br>
<br>
c) When createOffer() or createAnswer() is called, if the non-default<br>
routes flag is present, ask the user for consent. If there are video<br>
or audio streams already attached to the peerconnection, assume that<br>
consent has already been obtained (since getUserMedia asks for<br>
consent).<br>
<br>
This solution is backwards compatible with existing webapps, and<br>
minimizes the introduction of any new prompts. The only change in user<br>
experience is when a webapp explicitly wants to use non-default routes<br>
for their connection. If a webapp is okay with the default route, then<br>
they can silently add a data channel (no prompts).<br></blockquote><div><br=
></div><div>Justin is probably the best person to evaluate this solution. I=
t looks workable to me.</div><div><br></div><blockquote class=3D"gmail_quot=
e" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">=
I very much do not agree with Justin/Chrome&#39;s proposal to add a<br>
configuration extension that requires a user to opt-in to restricting<br>
non-default route STUN requests. That&#39;s like having the setting to<br>
show the warning for self-signed https certificates turned off by<br>
default and asking users to turn it on if they really want to be safe.<br>
The default behavior should be safe.<br>
<br></blockquote><div><br></div><div>I think you do not correctly character=
ize what Justin/Chrome is doing here. It is not a final solution. It is a w=
ay to test the final solution before it gets widely deployed. Since this is=
 a major change for the browsers I appreciate the need to deploy this caref=
ully and gradually.</div><div><div class=3D"gmail_signature">_____________<=
br>Roman Shpount</div></div><div>=C2=A0</div></div></div></div>

--089e01295380b2e477051c30d157--


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From: Roman Shpount <roman@telurix.com>
To: "Rauschenbach, Uwe (Nokia - DE/Munich)" <uwe.rauschenbach@nokia.com>
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Subject: Re: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it would be Christmas in July!)
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On Fri, Jul 31, 2015 at 2:44 PM, Rauschenbach, Uwe (Nokia - DE/Munich) <
uwe.rauschenbach@nokia.com> wrote:

> I can accept =E2=80=9Cmust be offered=E2=80=9D. But I do not understand =
=E2=80=9Cmust be accepted
> if offered=E2=80=9D. This is against the spirit of negotiation.
>
> If the other endpoint does not support rtcp-mux it rejects this. Am I
> missing something?
>
>
>
I am trying to differentiate here end points that can be compliant with RFC
5761 by never offering rtcp-mux and rejecting all offers with rtcp-mux.
What we are proposing here is that rtcp-mux is actually used for all the
RTP/RTCP streams, not just correctly negotiated away.
_____________
Roman Shpount

--001a113fd5e0deb3b8051c30ec73
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On F=
ri, Jul 31, 2015 at 2:44 PM, Rauschenbach, Uwe (Nokia - DE/Munich) <span di=
r=3D"ltr">&lt;<a href=3D"mailto:uwe.rauschenbach@nokia.com" target=3D"_blan=
k">uwe.rauschenbach@nokia.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 lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal"><span style=3D"font-family:Arial,sans-serif;font-siz=
e:10pt">I can accept =E2=80=9Cmust be offered=E2=80=9D. But I do not unders=
tand =E2=80=9Cmust be accepted if offered=E2=80=9D. This is against the spi=
rit of negotiation.</span><br></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ar=
ial&quot;,&quot;sans-serif&quot;">If the other endpoint does not support rt=
cp-mux it rejects this. Am I missing something?</span><span style=3D"font-f=
amily:Arial,sans-serif;font-size:10pt">=C2=A0</span></p>
<p class=3D"MsoNormal"><br></p></div></div></blockquote><div><br></div><div=
>I am trying to differentiate here end points that can be compliant with RF=
C 5761 by never offering rtcp-mux and rejecting all offers with rtcp-mux. W=
hat we are proposing here is that rtcp-mux is actually used for all the RTP=
/RTCP streams, not just correctly negotiated away.</div><div><div class=3D"=
gmail_signature">_____________<br>Roman Shpount</div></div><div>=C2=A0</div=
></div></div></div>

--001a113fd5e0deb3b8051c30ec73--


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From: <richard.vandet@kpn.com>
To: <ekr@rtfm.com>
Thread-Topic: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it would be Christmas in July!)
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When we choose this path, can we still support webrtc that is applied in de=
vices / no-browser's?

Richard

Op 31 jul. 2015 om 19:42 heeft Eric Rescorla <ekr@rtfm.com<mailto:ekr@rtfm.=
com>> het volgende geschreven:



On Fri, Jul 31, 2015 at 7:39 PM, Bernard Aboba <bernard.aboba@gmail.com<mai=
lto:bernard.aboba@gmail.com>> wrote:
On Jul 31, 2015, at 10:03, Eric Rescorla <ekr@rtfm.com<mailto:ekr@rtfm.com>=
> wrote:

Are you saying that compliant RTCWEB endpoints MUST support muxed RTCP, but=
 MAY support non-muxed RTCP (i.e. it is not explicitly prohibited)? That wo=
uld work for us.

[BA] Yes.  Since RTP-Usage already requires mux support, we would only need=
 to change the sentence requiring non-mux to make it optional.

In this case, would browsers be able to reject attempts to set the policy t=
o
anything other than always-mux? Because I'm pretty sure that's what both
we and Chrome want to do.

[BA] Yes, that should be possible. If non-mux is optional, then an implemen=
tation that doesn't support it would only allow setting the policy to alway=
s mux.

That seems satisfactory to me, since it means we can remove the code for
non-mux.

-Ekr

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

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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
</head>
<body dir=3D"auto">
<div>When we choose this path, can we still support webrtc that is applied =
in devices / no-browser's?<br>
<br>
Richard&nbsp;</div>
<div><br>
Op 31 jul. 2015 om 19:42 heeft Eric Rescorla &lt;<a href=3D"mailto:ekr@rtfm=
.com">ekr@rtfm.com</a>&gt; het volgende geschreven:<br>
<br>
</div>
<blockquote type=3D"cite">
<div>
<div dir=3D"ltr"><br>
<div class=3D"gmail_extra"><br>
<div class=3D"gmail_quote">On Fri, Jul 31, 2015 at 7:39 PM, Bernard Aboba <=
span dir=3D"ltr">
&lt;<a href=3D"mailto:bernard.aboba@gmail.com" target=3D"_blank">bernard.ab=
oba@gmail.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 dir=3D"auto"><span class=3D"">
<div>On Jul 31, 2015, at 10:03, Eric Rescorla &lt;<a href=3D"mailto:ekr@rtf=
m.com" target=3D"_blank">ekr@rtfm.com</a>&gt; wrote:</div>
<blockquote type=3D"cite">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote"><span>
<div><br>
</div>
</span>
<div>Are you saying that compliant RTCWEB endpoints MUST support muxed RTCP=
, but MAY support non-muxed RTCP (i.e. it is not explicitly prohibited)? Th=
at would work for us.</div>
</div>
</div>
</div>
</blockquote>
</div>
</div>
</div>
</blockquote>
<div><br>
</div>
</span>
<div>[BA] Yes.&nbsp; Since RTP-Usage already requires mux support, we would=
 only need to change the sentence requiring non-mux to make it optional.&nb=
sp;</div>
<span class=3D"">
<div><br>
</div>
<blockquote type=3D"cite">
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div class=3D"gmail_quote">
<div>In this case, would browsers be able to reject attempts to set the pol=
icy to</div>
<div>anything other than always-mux? Because I'm pretty sure that's what bo=
th</div>
<div>we and Chrome want to do.</div>
</div>
</div>
</div>
</blockquote>
<br>
</span>
<div>[BA] Yes, that should be possible. If non-mux is optional, then an imp=
lementation that doesn't support it would only allow setting the policy to =
always mux.&nbsp;</div>
</div>
</blockquote>
</div>
<br>
</div>
<div class=3D"gmail_extra">That seems satisfactory to me, since it means we=
 can remove the code for</div>
<div class=3D"gmail_extra">non-mux.</div>
<div class=3D"gmail_extra"><br>
</div>
<div class=3D"gmail_extra">-Ekr</div>
<div class=3D"gmail_extra"><br>
</div>
</div>
</div>
</blockquote>
<blockquote type=3D"cite">
<div><span>_______________________________________________</span><br>
<span>rtcweb mailing list</span><br>
<span><a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a></span><br>
<span><a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.=
ietf.org/mailman/listinfo/rtcweb</a></span><br>
</div>
</blockquote>
</body>
</html>

--_000_A083DF1FE6514C848A1D4A43D6DD5CEDkpncom_--


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Subject: Re: [rtcweb] Prompt for mic, camera *and* network access
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On 31/07/15 20:28, Roman Shpount wrote:
> I actually appreciate the seriousness of the issue.

I'm wondering if there's anything we could (get someone to)
do so that issues like this get handled better in future?

For example I'm wondering if a description of the various
ways in which current VPN services operate and some of the
consequences when things go wrong might be useful input to
WGs faced with similar issues.

I've no idea if someone would be motivated to write that
up, nor for how long such text would be useful but I suspect
there's a document there that'd be good to have. (Or if
there's a non-RFC that has a good survey/description then
a pointer to that would be good.)

Anyway, I'd be happy to help someone who was willing and
able to try that with any process stuff.

Cheers,
S.


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References: <55352067.6070007@nostrum.com> <20150421154657.GA4071@lo.psyced.org> <20150715123803.GA24976@lo.psyced.org> <2BCEF545-D322-4589-AC85-AA80E7F934D9@phonefromhere.com> <20150715164129.GA27105@lo.psyced.org> <AF6485D4-1579-45AF-937C-1682AC1AA645@phonefromhere.com> <20150716084225.GA29652@lo.psyced.org> <0AF88216-374E-4DBA-9DB5-3E55B358E6AA@phonefromhere.com> <20150731104712.GA5228@lo.psyced.org> <CAD5OKxvYcBYLDXFYMh8DijZrYcEbkxf++WqWzrZF8UxjJxfrTQ@mail.gmail.com> <CA+65OsrXcMY92bR+AafvkBBvEcq_rbgeUpRv9C7OCzNBNXJYyA@mail.gmail.com> <CAOJ7v-3aU9gD7aywdi0zToNC2uN1z=8ubgnnJu-wuV0+xdr38w@mail.gmail.com>
From: Eric Rescorla <ekr@rtfm.com>
Date: Fri, 31 Jul 2015 22:30:05 +0200
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To: Justin Uberti <juberti@google.com>
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--f46d0438eb9dbe7aba051c31b1e6
Content-Type: text/plain; charset=UTF-8

On Fri, Jul 31, 2015 at 9:27 PM, Justin Uberti <juberti@google.com> wrote:

>
>
> On Fri, Jul 31, 2015 at 11:59 AM, Daniel Roesler <diafygi@gmail.com>
> wrote:
>
>> I very much sympathize with carlo's frustration with this working
>> group. So far the response here to the introduced privacy exploits has
>> been pretty appalling. As you mentioned in the other consent thread,
>> this standard is the first time a browser can send requests over
>> non-default route network interfaces.
>>
>> <strong>_This is a life threatening issue!_</strong>
>>
>> Many, many, many Chinese/Arab/Persian users buy VPNs to get around
>> censorship and participate in free speech. The default configurations
>> for these VPNs set the default route, but often still allow requests
>> to be made over the real network interface (due to the way Windows
>> configures VPNs). Thus, when ICE sends STUN requests through that real
>> network interface, they expose the real IP of the VPN user. So any
>> website can silently identify VPN users, which can lead to a knock a
>> the door. If you don't believe me, just search for "ipleak" on
>> twitter[1] and see how many foreign language references there are to
>> the exploit test page.
>
>
>
>
>> Very real implication: Loading a website (or even an embedded ad) is
>> now much more life-threatening than it was before.
>>
>> Thanks everyone! Great work!
>>
>> I've heard many times on threads here that this is the fault of the
>> VPN user for not configuring their VPN better. I couldn't disagree
>> more. Up until WebRTC, the browser wasn't able to make non-default
>> route requests, which means that up until WebRTC, you couldn't expose
>> your real IP just by visiting a webpage. So WebRTC is the thing that
>> triggered exposure of the user, not their VPN configuration. Also,
>> blaming the user when they are using the default configuration (and
>> don't know any better) and you introduced something that exposes them
>> is offensive.
>
>
> I don't really want to debate nits, but this is not true. Flash has had
> this 'feature', since the beginning of the modern browser era.
> Regardless, we think we can do better.
>
>

See also:
https://www.usenix.org/system/files/conference/foci12/foci12-final8.pdf
and
http://www.eecs.qmul.ac.uk/~hamed/papers/PETS2015VPN.pdf

-Ekr

--f46d0438eb9dbe7aba051c31b1e6
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Fri, Jul 31, 2015 at 9:27 PM, Justin Uberti <span dir=3D"ltr">&lt;<a=
 href=3D"mailto:juberti@google.com" target=3D"_blank">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"><div dir=3D"ltr"><br>=
<div class=3D"gmail_extra"><br><div class=3D"gmail_quote"><div><div class=
=3D"h5">On Fri, Jul 31, 2015 at 11:59 AM, Daniel Roesler <span dir=3D"ltr">=
&lt;<a href=3D"mailto:diafygi@gmail.com" target=3D"_blank">diafygi@gmail.co=
m</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margi=
n:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">I very much sympa=
thize with carlo&#39;s frustration with this working<br>
group. So far the response here to the introduced privacy exploits has<br>
been pretty appalling. As you mentioned in the other consent thread,<br>
this standard is the first time a browser can send requests over<br>
non-default route network interfaces.<br>
<br>
&lt;strong&gt;_This is a life threatening issue!_&lt;/strong&gt;<br>
<br>
Many, many, many Chinese/Arab/Persian users buy VPNs to get around<br>
censorship and participate in free speech. The default configurations<br>
for these VPNs set the default route, but often still allow requests<br>
to be made over the real network interface (due to the way Windows<br>
configures VPNs). Thus, when ICE sends STUN requests through that real<br>
network interface, they expose the real IP of the VPN user. So any<br>
website can silently identify VPN users, which can lead to a knock a<br>
the door. If you don&#39;t believe me, just search for &quot;ipleak&quot; o=
n<br>
twitter[1] and see how many foreign language references there are to<br>
the exploit test page.=C2=A0</blockquote><blockquote class=3D"gmail_quote" =
style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">=C2=
=A0</blockquote><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8e=
x;border-left:1px #ccc solid;padding-left:1ex">
<br>
Very real implication: Loading a website (or even an embedded ad) is<br>
now much more life-threatening than it was before.<br>
<br>
Thanks everyone! Great work!<br>
<br>
I&#39;ve heard many times on threads here that this is the fault of the<br>
VPN user for not configuring their VPN better. I couldn&#39;t disagree<br>
more. Up until WebRTC, the browser wasn&#39;t able to make non-default<br>
route requests, which means that up until WebRTC, you couldn&#39;t expose<b=
r>
your real IP just by visiting a webpage. So WebRTC is the thing that<br>
triggered exposure of the user, not their VPN configuration. Also,<br>
blaming the user when they are using the default configuration (and<br>
don&#39;t know any better) and you introduced something that exposes them<b=
r>
is offensive.=C2=A0</blockquote><div><br></div></div></div><div>I don&#39;t=
 really want to debate nits, but this is not true. Flash has had this &#39;=
feature&#39;, since the beginning of the modern browser era.</div><div>Rega=
rdless, we think we can do better.</div><div><div class=3D"h5"><div>=C2=A0<=
/div></div></div></div></div></div></blockquote><div><br></div><div>See als=
o:</div><div><a href=3D"https://www.usenix.org/system/files/conference/foci=
12/foci12-final8.pdf">https://www.usenix.org/system/files/conference/foci12=
/foci12-final8.pdf</a></div><div>and</div><div><a href=3D"http://www.eecs.q=
mul.ac.uk/~hamed/papers/PETS2015VPN.pdf">http://www.eecs.qmul.ac.uk/~hamed/=
papers/PETS2015VPN.pdf</a></div><div><br></div><div>-Ekr</div></div></div><=
/div>

--f46d0438eb9dbe7aba051c31b1e6--


From nobody Fri Jul 31 13:57:06 2015
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From: Bernard Aboba <bernard.aboba@gmail.com>
Date: Fri, 31 Jul 2015 13:56:47 -0700
To: "<richard.vandet@kpn.com>" <richard.vandet@kpn.com>
Archived-At: <http://mailarchive.ietf.org/arch/msg/rtcweb/_mst5y2cD7oZNiS3zQdTJ9ZL5ZQ>
Cc: "<rtcweb@ietf.org>" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Proposal: require rtcp-mux (remove non-muxed RTCP; it would be Christmas in July!)
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New WebRTC devices typically use mux since it results in sending fewer packe=
ts, using fewer ports, better energy consumption and reliability, etc.=20



> On Jul 31, 2015, at 12:37 PM, <richard.vandet@kpn.com> <richard.vandet@kpn=
.com> wrote:
>=20
> When we choose this path, can we still support webrtc that is applied in d=
evices / no-browser's?
>=20
> Richard=20
>=20
> Op 31 jul. 2015 om 19:42 heeft Eric Rescorla <ekr@rtfm.com> het volgende g=
eschreven:
>=20
>>=20
>>=20
>>> On Fri, Jul 31, 2015 at 7:39 PM, Bernard Aboba <bernard.aboba@gmail.com>=
 wrote:
>>>> On Jul 31, 2015, at 10:03, Eric Rescorla <ekr@rtfm.com> wrote:
>>>>>=20
>>>>> Are you saying that compliant RTCWEB endpoints MUST support muxed RTCP=
, but MAY support non-muxed RTCP (i.e. it is not explicitly prohibited)? Tha=
t would work for us.
>>>=20
>>> [BA] Yes.  Since RTP-Usage already requires mux support, we would only n=
eed to change the sentence requiring non-mux to make it optional.=20
>>>=20
>>>> In this case, would browsers be able to reject attempts to set the poli=
cy to
>>>> anything other than always-mux? Because I'm pretty sure that's what bot=
h
>>>> we and Chrome want to do.
>>>=20
>>> [BA] Yes, that should be possible. If non-mux is optional, then an imple=
mentation that doesn't support it would only allow setting the policy to alw=
ays mux.=20
>>=20
>> That seems satisfactory to me, since it means we can remove the code for
>> non-mux.
>>=20
>> -Ekr
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb

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<html><head><meta http-equiv="content-type" content="text/html; charset=utf-8"></head><body dir="auto"><div>New WebRTC devices typically use mux since it results in sending fewer packets, using fewer ports, better energy consumption and reliability, etc.&nbsp;</div><div><br><br></div><div><br>On Jul 31, 2015, at 12:37 PM, &lt;<a href="mailto:richard.vandet@kpn.com">richard.vandet@kpn.com</a>&gt; &lt;<a href="mailto:richard.vandet@kpn.com">richard.vandet@kpn.com</a>&gt; wrote:<br><br></div><blockquote type="cite"><div>

<meta http-equiv="Content-Type" content="text/html; charset=us-ascii">


<div>When we choose this path, can we still support webrtc that is applied in devices / no-browser's?<br>
<br>
Richard&nbsp;</div>
<div><br>
Op 31 jul. 2015 om 19:42 heeft Eric Rescorla &lt;<a href="mailto:ekr@rtfm.com">ekr@rtfm.com</a>&gt; het volgende geschreven:<br>
<br>
</div>
<blockquote type="cite">
<div>
<div dir="ltr"><br>
<div class="gmail_extra"><br>
<div class="gmail_quote">On Fri, Jul 31, 2015 at 7:39 PM, Bernard Aboba <span dir="ltr">
&lt;<a href="mailto:bernard.aboba@gmail.com" target="_blank">bernard.aboba@gmail.com</a>&gt;</span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="auto"><span class="">
<div>On Jul 31, 2015, at 10:03, Eric Rescorla &lt;<a href="mailto:ekr@rtfm.com" target="_blank">ekr@rtfm.com</a>&gt; wrote:</div>
<blockquote type="cite">
<div dir="ltr">
<div class="gmail_extra">
<div class="gmail_quote">
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr">
<div class="gmail_extra">
<div class="gmail_quote"><span>
<div><br>
</div>
</span>
<div>Are you saying that compliant RTCWEB endpoints MUST support muxed RTCP, but MAY support non-muxed RTCP (i.e. it is not explicitly prohibited)? That would work for us.</div>
</div>
</div>
</div>
</blockquote>
</div>
</div>
</div>
</blockquote>
<div><br>
</div>
</span>
<div>[BA] Yes.&nbsp; Since RTP-Usage already requires mux support, we would only need to change the sentence requiring non-mux to make it optional.&nbsp;</div>
<span class="">
<div><br>
</div>
<blockquote type="cite">
<div dir="ltr">
<div class="gmail_extra">
<div class="gmail_quote">
<div>In this case, would browsers be able to reject attempts to set the policy to</div>
<div>anything other than always-mux? Because I'm pretty sure that's what both</div>
<div>we and Chrome want to do.</div>
</div>
</div>
</div>
</blockquote>
<br>
</span>
<div>[BA] Yes, that should be possible. If non-mux is optional, then an implementation that doesn't support it would only allow setting the policy to always mux.&nbsp;</div>
</div>
</blockquote>
</div>
<br>
</div>
<div class="gmail_extra">That seems satisfactory to me, since it means we can remove the code for</div>
<div class="gmail_extra">non-mux.</div>
<div class="gmail_extra"><br>
</div>
<div class="gmail_extra">-Ekr</div>
<div class="gmail_extra"><br>
</div>
</div>
</div>
</blockquote>
<blockquote type="cite">
<div><span>_______________________________________________</span><br>
<span>rtcweb mailing list</span><br>
<span><a href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a></span><br>
<span><a href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a></span><br>
</div>
</blockquote>


</div></blockquote></body></html>
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Subject: Re: [rtcweb] Prompt for mic, camera *and* network access
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On Fri, Jul 31, 2015 at 11:59 AM, Daniel Roesler <diafygi@gmail.com> wrote:
> >> more. Up until WebRTC, the browser wasn't able to make non-default
> >> route requests, which means that up until WebRTC, you couldn't expose
> >> your real IP just by visiting a webpage. So WebRTC is the thing that
> >> triggered exposure of the user, not their VPN configuration. Also,
> >> blaming the user when they are using the default configuration (and
> >> don't know any better) and you introduced something that exposes them
> >> is offensive.

On Fri, Jul 31, 2015 at 9:27 PM, Justin Uberti <juberti@google.com> wrote:
> > I don't really want to debate nits, but this is not true. Flash has had
> > this 'feature', since the beginning of the modern browser era.
> > Regardless, we think we can do better.

Wait wait.. I was developing Flash media apps some time ago in the
pre-Snowden era and I never bumped into any such functionality. My 
business partner of those days just dropped me this link:

http://help.adobe.com/en_US/FlashPlatform/reference/actionscript/3/flash/net/Socket.html#localAddress

Looks like the feature exists for installed AIR apps, not Flash plugin.
Then he pointed me to Mr Kaufman's leak that RTMFP, which he introduced
into Flash in 2009 AFAIR, can be abused to obtain such IP numbers.

Well, if I who actually wrote push-to-talk apps with RTMP didn't know
that, I doubt any HackingTeam company figured out this method for
de-anonymizing VPN users. Hardly anyone looked at RTMFP all that much
since the only existing server implementation was running at Adobe.

Quite different from the WebRTC case where the API quite clearly 
explains how to get at local IP information.

So let's indeed not excuse this larger problem with a previous problem
that rather wasn't known until this moment where it may seem practical
to leak such info for the sake of argument.

Flash has NOT been offering such a feature since the beginning of
time AFAICT and the sneaky way it got introduced with RTMFP and
apparently wasn't actually secure as such is a too late scandal
in itself.

On Fri, Jul 31, 2015 at 10:30:05PM +0200, Eric Rescorla wrote:
> See also:
> https://www.usenix.org/system/files/conference/foci12/foci12-final8.pdf
> and
> http://www.eecs.qmul.ac.uk/~hamed/papers/PETS2015VPN.pdf

Interesting papers, but do they describe a method for Javascript code
in websites to break out of the VPN cage? I didn't see anything like that,
maybe I skipped the content too fast.

Also let's not talk about IPv6.
It is not relevant for the dissident scenario of today and by the time
industry stops preferring NAT over IPv6 we'd better already be deploying
public-key based routing whereby worldwide IP addressing is no longer
a worthwhile goal. Thus it is my assumption that IPv6 will hardly ever
play a relevant role in Internet history.


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Subject: Re: [rtcweb] Prompt for mic, camera *and* network access
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On Fri, Jul 31, 2015 at 03:28:05PM -0400, Roman Shpount wrote:
> I actually appreciate the seriousness of the issue. I think part of the
> reason this group did not respond in more timely manner is that not
> everyone realized the behavior change (use of non-default routes). Once it
> was clarified, the issue started to be handled. I understand that this and
> any other security issue can be extremely frustrating to the end users, but
> people, including people who build web browsers or design web browser
> standards do make mistakes.

Woah, everyone reading those blogs understood the seriousness of the
problem except for the developers that culled themselves into the idea
of everyone out there having gotten everything wrong? Whoops.

On Fri, Jul 31, 2015 at 2:59 PM, Daniel Roesler <diafygi@gmail.com> wrote:
> > How about a combined solution?

Sounds like going in the right direction. I still haven't understood
why there can't be one single prompt that allows a site to do webrtc
things.. allowing mic, cam *AND* STUN access in a single gesture.
Anything else would be intransparent anyway.

> > I very much do not agree with Justin/Chrome's proposal to add a
> > configuration extension that requires a user to opt-in to restricting
> > non-default route STUN requests. That's like having the setting to
> > show the warning for self-signed https certificates turned off by
> > default and asking users to turn it on if they really want to be safe.
> > The default behavior should be safe.

Yes, the proposal completely fails the scenario I described. You
can't expect non-tech people that are simply using a web browser to
understand why they suddenly need to flick some switches or install
some odd addons in order to not get deanonymized. WebRTC needs a
general opt-in.

On Fri, Jul 31, 2015 at 03:28:05PM -0400, Roman Shpount wrote:
> I think you do not correctly characterize what Justin/Chrome is doing here.
> It is not a final solution. It is a way to test the final solution before
> it gets widely deployed. Since this is a major change for the browsers I
> appreciate the need to deploy this carefully and gradually.

Features need to be introduced carefully, but this isn't a feature.
This is a top notch security bug risking human lives and I can't
believe it takes months of talking to get anything going here.
This needs to be shut down first, then experiment alternate routes.
Period.

-- 
  E-mail is public! Talk to me in private using encryption:
         http://loupsycedyglgamf.onion/LynX/
          irc://loupsycedyglgamf.onion:67/lynX
         https://psyced.org:34443/LynX/


From nobody Fri Jul 31 17:18:14 2015
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Subject: Re: [rtcweb] Prompt for mic, camera *and* network access
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On 7/31/15 3:27 PM, carlo von lynX wrote:
> On Fri, Jul 31, 2015 at 03:28:05PM -0400, Roman Shpount wrote:
>> I actually appreciate the seriousness of the issue. I think part of the
>> reason this group did not respond in more timely manner is that not
>> everyone realized the behavior change (use of non-default routes). Once it
>> was clarified, the issue started to be handled. I understand that this and
>> any other security issue can be extremely frustrating to the end users, but
>> people, including people who build web browsers or design web browser
>> standards do make mistakes.
> Woah, everyone reading those blogs understood the seriousness of the
> problem except for the developers that culled themselves into the idea
> of everyone out there having gotten everything wrong? Whoops.

I understood the magnitude of the issue, and I still think you're 
overreacting.

>
> On Fri, Jul 31, 2015 at 2:59 PM, Daniel Roesler <diafygi@gmail.com> wrote:
>>> How about a combined solution?
> Sounds like going in the right direction. I still haven't understood
> why there can't be one single prompt that allows a site to do webrtc
> things.. allowing mic, cam *AND* STUN access in a single gesture.
> Anything else would be intransparent anyway.
>
>>> I very much do not agree with Justin/Chrome's proposal to add a
>>> configuration extension that requires a user to opt-in to restricting
>>> non-default route STUN requests. That's like having the setting to
>>> show the warning for self-signed https certificates turned off by
>>> default and asking users to turn it on if they really want to be safe.
>>> The default behavior should be safe.
> Yes, the proposal completely fails the scenario I described. You
> can't expect non-tech people that are simply using a web browser to
> understand why they suddenly need to flick some switches or install
> some odd addons in order to not get deanonymized. WebRTC needs a
> general opt-in.

Disagree.

>
> On Fri, Jul 31, 2015 at 03:28:05PM -0400, Roman Shpount wrote:
>> I think you do not correctly characterize what Justin/Chrome is doing here.
>> It is not a final solution. It is a way to test the final solution before
>> it gets widely deployed. Since this is a major change for the browsers I
>> appreciate the need to deploy this carefully and gradually.
> Features need to be introduced carefully, but this isn't a feature.
> This is a top notch security bug risking human lives and I can't
> believe it takes months of talking to get anything going here.
> This needs to be shut down first, then experiment alternate routes.
> Period.
>
If you're worried about "top notch security bugs risking human lives" 
there's at least a dozen I'd tackle before thinking about this one. It 
allows browsers to do little more than people have been accepting for 
years, in their browsers.

Matthew Kaufman


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> Well, if I who actually wrote push-to-talk apps with RTMP didn't know
> that, I doubt any HackingTeam company figured out this method for
> de-anonymizing VPN users.

At least hackingteam didn't figure out the webrtc method themselves either 
according to https://wikileaks.org/hackingteam/emails/emailid/119217

Now if you assume a smarter opponent than that, why is that opponent not
smart enough to have figured out the RTMFP method as well?

Granted, RTMFP is a pretty tough nut to reverse engineer but it had been 
done back in late 2010.

And it is P2P. It has to exchange some ip addresses somehow.
Even despite the encryption scheme, you can use wireshark to see ice-like 
packets. Their content will be unreadable, but the IP addresses tell you 
what is going on.

I do recall LOL'ing when finding out that Adobe leaked 10.x server 
addresses in the early days of cirrus.

> Hardly anyone looked at RTMFP all that much since the only
> existing server implementation was running at Adobe.

I disagree with your "hardly anyone" FWIW.

> Quite different from the WebRTC case where the API quite clearly
> explains how to get at local IP information.

I do not understand your argument. RTMFP did not expose the local IP 
information at the API level. However, that did not mean it was not 
discoverable as explained above.

Assuming that your point is that WebRTC shouldn't expose this and use a 
signaling channel that isn't under control of the browser, how would 
that increase security?

