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To: "mmusic (E-mail)" <mmusic@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: [rtcweb] AVTCORE discussion of extmap and BUNDLE
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MMUSIC and RTCWeb,

I would like to make you aware of an ongoing discussion around how the 
SDP attribute for RTP header extensions (a=extmap) should work with 
BUNDLE. This is going on in the context of the update of RFC5285:

https://datatracker.ietf.org/doc/draft-ietf-avtcore-rfc5285-bis/

Thread start:
https://www.ietf.org/mail-archive/web/avt/current/msg17531.html

This discussion started as result of the SDP expert reviewer who noticed 
that we failed to take muxing into account for the SDP attributes. And 
extmap is non-trivial and classified as special. So far we agreed that 
clarificatio is needed, but we have not yet agreed to what 
clarifications and their impact. So please review the discussion and 
contribute on the AVTCORE mailing list.

Cheers

Magnus Westerlund

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


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--001a1143847cfc15ae054f1f15c0
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General points:

a. Please expand acronyms on first use (e.g. HTTP, VPN, ICE, etc.).



Details

1 <https://tools.ietf.org/html/draft-ietf-rtcweb-ip-handling-03#section-1>.
Introduction


   As a technology that supports peer-to-peer connections, WebRTC may
   send data over different network paths than the path used for HTTP
   traffic.  This may allow a web application to learn additional
   information about the user, which may be problematic in certain
   cases.  This document summarizes the concerns, and makes
   recommendations on how best to handle the tradeoff between privacy
   and media performance.


[BA] There are a number of distinct privacy issues that WebRTC

needs to address. I believe this document is focussed on IP

address info that can be gleaned by Web applications (the

second sentence).  That is somewhat distinct from what information

a web server might glean, or what a passive observer might be

able to obtain looking at traffic generated by a WebRTC client.

The first sentence therefore is a bit confusing to me.


A suggestion:


"As a technology that attempts to optimize peer-to-peer

connections by testing connectivity between available

addresses, WebRTC may allow a web application to learn

additional information about the user compared to an

application which only utilizes HTTP.  This may be

problematic in certain cases."


2 <https://tools.ietf.org/html/draft-ietf-rtcweb-ip-handling-03#section-2>.
Problem Statement


   WebRTC enables real-time peer-to-peer communications by enumerating
   network interfaces and discovering the best route through the ICE
   [RFC5245 <https://tools.ietf.org/html/rfc5245>]protocol.  During
the ICE process, the peers involved in a
   session gather and exchange all the IP addresses they can discover,
   so that the connectivity of each IP pair can be checked, and the best
   path chosen.  The addresses that are gathered usually consist of an
   endpoint's private physical/virtual addresses, and its public
   Internet addresses.


[BA] The terms "route" and "path" used here do not refer to IP

routing and so is a bit confusing.  How about this?


   "In order to enable peer-to-peer communications, WebRTC

   peers utilize ICE [RFC5245] which gathers and exchanges

   all the IP addresses it can discover, so that the

   connectivity of each IP pair can be checked, and the best
   pair can be chosen.  The addresses that are gathered

   usually consist of an endpoint's private physical/virtual

   addresses, and its public Internet addresses."


   2.  If the client tries to hide its physical location through a VPN,
       and the VPN and local OS support routing over multiple interfaces
       (i.e., a "split-tunnel" VPN), WebRTC will discover the public
       address for the VPN as well as the ISP public address that the
       VPN runs over.


[BA] Even if the local OS doesn't support split-tunnel,

isn't it still possible for an ICE implementation to

discover other addresses?


3 <https://tools.ietf.org/html/draft-ietf-rtcweb-ip-handling-03#section-3>.
Goals

   Being peer-to-peer, WebRTC represents a privacy-enabling technology,
   and therefore we want to avoid solutions that disable WebRTC or make
   it harder to use.  This means that WebRTC should be configured by
   default to only reveal the minimum amount of information needed to
   establish a performant WebRTC session, while providing options to
   reveal additional information upon user consent, or further limit
   this information if the user has specifically requested this.
   Specifically, WebRTC should:


   o  Provide a privacy-friendly default behavior which strikes the
      right balance between privacy and media performance for most users
      and use cases.


   o  For users who care more about one versus the other, provide a
      means to customize the experience.


[BA] This section might state whether it is a

goal to enable WebRTC to be used with

The Onion Routing network (Tor). There

are a number of reasons why enabling Javascript

for  use with Tor may represent a problem so that

use of WebRTC might not be advisable:

https://security.stackexchange.com/questions/95046/why-disable-javascript-in-tor

 4 <https://tools.ietf.org/html/draft-ietf-rtcweb-ip-handling-03#section-4>.
Detailed Design


   1.  By default, WebRTC should follow normal IP routing rules, to the
       extent that this is easy to determine (i.e., not considering
       proxies).  This can be accomplished by binding local sockets to
       the wildcard addresses (0.0.0.0 for IPv4, :: for IPv6), which
       allows the OS to route WebRTC traffic the same way as it would
       HTTP traffic, and allows only the 'typical' public addresses to
       be discovered.


[BA] This paragraph discusses routing behavior and
address binding together and leads me to wonder whether
some assumptions aren't being made (e.g. weak versus
strong host model). My suggestion is to simplify the
the paragraph and leave details to later on. For example:

"1. By default WebRTC should only allow applications
to discover the same address information made available
by an HTTP application: the 'typical' public address."

--001a1143847cfc15ae054f1f15c0
Content-Type: text/html; charset=UTF-8
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<div dir=3D"ltr"><br><div><br></div><div>General points:</div><div><br></di=
v><div>a. Please expand acronyms on first use (e.g. HTTP, VPN, ICE, etc.).=
=C2=A0</div><div><br></div><div><br></div><div><br></div><div>Details</div>=
<div><br></div><div><pre class=3D"gmail-newpage" style=3D"font-size:13.3333=
px;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><span class=3D"gmail-=
h2" style=3D"line-height:0pt;display:inline;font-size:1em;font-weight:bold"=
><h2 style=3D"line-height:0pt;display:inline;font-size:1em"><a class=3D"gma=
il-selflink" name=3D"section-1" href=3D"https://tools.ietf.org/html/draft-i=
etf-rtcweb-ip-handling-03#section-1" style=3D"color:black;text-decoration-l=
ine:none">1</a>.  Introduction</h2></span></pre></div><div><br></div><div><=
pre class=3D"gmail-newpage" style=3D"font-size:13.3333px;margin-top:0px;mar=
gin-bottom:0px;color:rgb(0,0,0)">   As a technology that supports peer-to-p=
eer connections, WebRTC may
   send data over different network paths than the path used for HTTP
   traffic.  This may allow a web application to learn additional
   information about the user, which may be problematic in certain
   cases.  This document summarizes the concerns, and makes
   recommendations on how best to handle the tradeoff between privacy
   and media performance.</pre><pre class=3D"gmail-newpage" style=3D"font-s=
ize:13.3333px;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><br></pre>=
<pre class=3D"gmail-newpage" style=3D"font-size:13.3333px;margin-top:0px;ma=
rgin-bottom:0px;color:rgb(0,0,0)">[BA] There are a number of distinct priva=
cy issues that WebRTC</pre><pre class=3D"gmail-newpage" style=3D"font-size:=
13.3333px;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)">needs to addre=
ss. I believe this document is focussed on IP</pre><pre class=3D"gmail-newp=
age" style=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px;color:rg=
b(0,0,0)">address info that can be gleaned by Web applications (the</pre><p=
re class=3D"gmail-newpage" style=3D"font-size:13.3333px;margin-top:0px;marg=
in-bottom:0px;color:rgb(0,0,0)">second sentence).  That is somewhat distinc=
t from what information</pre><pre class=3D"gmail-newpage" style=3D"font-siz=
e:13.3333px;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)">a web server=
 might glean, or what a passive observer might be</pre><pre class=3D"gmail-=
newpage" style=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px;colo=
r:rgb(0,0,0)">able to obtain looking at traffic generated by a WebRTC clien=
t.</pre><pre class=3D"gmail-newpage" style=3D"font-size:13.3333px;margin-to=
p:0px;margin-bottom:0px;color:rgb(0,0,0)">The first sentence therefore is a=
 bit confusing to me.</pre><pre class=3D"gmail-newpage" style=3D"font-size:=
13.3333px;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><br></pre><pre=
 class=3D"gmail-newpage" style=3D"font-size:13.3333px;margin-top:0px;margin=
-bottom:0px;color:rgb(0,0,0)">A suggestion:</pre><pre class=3D"gmail-newpag=
e" style=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px;color:rgb(=
0,0,0)"><br></pre><pre class=3D"gmail-newpage" style=3D"font-size:13.3333px=
;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)">&quot;As a technology t=
hat attempts to optimize peer-to-peer</pre><pre class=3D"gmail-newpage" sty=
le=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)=
">connections by testing connectivity between available</pre><pre class=3D"=
gmail-newpage" style=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0p=
x;color:rgb(0,0,0)">addresses, WebRTC may allow a web application to learn<=
/pre><pre class=3D"gmail-newpage" style=3D"font-size:13.3333px;margin-top:0=
px;margin-bottom:0px;color:rgb(0,0,0)">additional information about the use=
r compared to an</pre><pre class=3D"gmail-newpage" style=3D"font-size:13.33=
33px;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)">application which o=
nly utilizes HTTP.  This may be</pre><pre class=3D"gmail-newpage" style=3D"=
font-size:13.3333px;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)">prob=
lematic in certain cases.&quot;</pre><pre class=3D"gmail-newpage" style=3D"=
font-size:13.3333px;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><br>=
</pre><pre class=3D"gmail-newpage" style=3D"font-size:13.3333px;margin-top:=
0px;margin-bottom:0px;color:rgb(0,0,0)"><pre class=3D"gmail-newpage" style=
=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px"><span class=3D"gm=
ail-h2" style=3D"line-height:0pt;display:inline;font-size:1em;font-weight:b=
old"><h2 style=3D"line-height:0pt;display:inline;font-size:1em"><a class=3D=
"gmail-selflink" name=3D"section-2" href=3D"https://tools.ietf.org/html/dra=
ft-ietf-rtcweb-ip-handling-03#section-2" style=3D"color:black;text-decorati=
on-line:none">2</a>.  Problem Statement</h2></span></pre></pre><pre class=
=3D"gmail-newpage" style=3D"font-size:13.3333px;margin-top:0px;margin-botto=
m:0px;color:rgb(0,0,0)"><br></pre><pre class=3D"gmail-newpage" style=3D"fon=
t-size:13.3333px;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><pre cl=
ass=3D"gmail-newpage" style=3D"font-size:13.3333px;margin-top:0px;margin-bo=
ttom:0px">   WebRTC enables real-time peer-to-peer communications by enumer=
ating
   network interfaces and discovering the best route through the ICE
   [<a href=3D"https://tools.ietf.org/html/rfc5245" title=3D"&quot;Interact=
ive Connectivity Establishment (ICE): A Protocol for Network Address Transl=
ator (NAT) Traversal for Offer/Answer Protocols&quot;">RFC5245</a>]protocol=
.  During the ICE process, the peers involved in a
   session gather and exchange all the IP addresses they can discover,
   so that the connectivity of each IP pair can be checked, and the best
   path chosen.  The addresses that are gathered usually consist of an
   endpoint&#39;s private physical/virtual addresses, and its public
   Internet addresses.</pre><pre class=3D"gmail-newpage" style=3D"font-size=
:13.3333px;margin-top:0px;margin-bottom:0px"><br></pre><pre class=3D"gmail-=
newpage" style=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px">[BA=
] The terms &quot;route&quot; and &quot;path&quot; used here do not refer t=
o IP</pre><pre class=3D"gmail-newpage" style=3D"font-size:13.3333px;margin-=
top:0px;margin-bottom:0px">routing and so is a bit confusing.  How about th=
is? </pre><pre class=3D"gmail-newpage" style=3D"font-size:13.3333px;margin-=
top:0px;margin-bottom:0px"><br></pre><pre class=3D"gmail-newpage" style=3D"=
font-size:13.3333px;margin-top:0px;margin-bottom:0px"><pre class=3D"gmail-n=
ewpage" style=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px">   &=
quot;In order to enable peer-to-peer communications, WebRTC</pre><pre class=
=3D"gmail-newpage" style=3D"font-size:13.3333px;margin-top:0px;margin-botto=
m:0px">   peers utilize ICE [RFC5245] which gathers and exchanges</pre><pre=
 class=3D"gmail-newpage" style=3D"font-size:13.3333px;margin-top:0px;margin=
-bottom:0px">   all the IP addresses it can discover, so that the </pre><pr=
e class=3D"gmail-newpage" style=3D"font-size:13.3333px;margin-top:0px;margi=
n-bottom:0px">   connectivity of each IP pair can be checked, and the best
   pair can be chosen.  The addresses that are gathered</pre><pre class=3D"=
gmail-newpage" style=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0p=
x">   usually consist of an endpoint&#39;s private physical/virtual</pre><p=
re class=3D"gmail-newpage" style=3D"font-size:13.3333px;margin-top:0px;marg=
in-bottom:0px">   addresses, and its public Internet addresses.&quot;</pre>=
<pre class=3D"gmail-newpage" style=3D"font-size:13.3333px;margin-top:0px;ma=
rgin-bottom:0px"><br></pre><pre class=3D"gmail-newpage" style=3D"font-size:=
13.3333px;margin-top:0px;margin-bottom:0px"><pre class=3D"gmail-newpage" st=
yle=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px">   2.  If the =
client tries to hide its physical location through a VPN,
       and the VPN and local OS support routing over multiple interfaces
       (i.e., a &quot;split-tunnel&quot; VPN), WebRTC will discover the pub=
lic
       address for the VPN as well as the ISP public address that the
       VPN runs over.</pre><pre class=3D"gmail-newpage" style=3D"font-size:=
13.3333px;margin-top:0px;margin-bottom:0px"><br></pre><pre class=3D"gmail-n=
ewpage" style=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px">[BA]=
 Even if the local OS doesn&#39;t support split-tunnel,</pre><pre class=3D"=
gmail-newpage" style=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0p=
x">isn&#39;t it still possible for an ICE implementation to</pre><pre class=
=3D"gmail-newpage" style=3D"font-size:13.3333px;margin-top:0px;margin-botto=
m:0px">discover other addresses? </pre><pre class=3D"gmail-newpage" style=
=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px"><br></pre><pre cl=
ass=3D"gmail-newpage" style=3D"font-size:13.3333px;margin-top:0px;margin-bo=
ttom:0px"><pre class=3D"gmail-newpage" style=3D"font-size:13.3333px;margin-=
top:0px;margin-bottom:0px"><span class=3D"gmail-h2" style=3D"line-height:0p=
t;display:inline;font-size:1em;font-weight:bold"><h2 style=3D"line-height:0=
pt;display:inline;font-size:1em"><a class=3D"gmail-selflink" name=3D"sectio=
n-3" href=3D"https://tools.ietf.org/html/draft-ietf-rtcweb-ip-handling-03#s=
ection-3" style=3D"color:black;text-decoration-line:none">3</a>.  Goals</h2=
></span>

   Being peer-to-peer, WebRTC represents a privacy-enabling technology,
   and therefore we want to avoid solutions that disable WebRTC or make
   it harder to use.  This means that WebRTC should be configured by
   default to only reveal the minimum amount of information needed to
   establish a performant WebRTC session, while providing options to
   reveal additional information upon user consent, or further limit
   this information if the user has specifically requested this.
   Specifically, WebRTC should:</pre><pre class=3D"gmail-newpage" style=3D"=
font-size:13.3333px;margin-top:0px;margin-bottom:0px"><br></pre><pre class=
=3D"gmail-newpage" style=3D"font-size:13.3333px;margin-top:0px;margin-botto=
m:0px"><pre class=3D"gmail-newpage" style=3D"font-size:13.3333px;margin-top=
:0px;margin-bottom:0px">   o  Provide a privacy-friendly default behavior w=
hich strikes the
      right balance between privacy and media performance for most users
      and use cases.

</pre><pre class=3D"gmail-newpage" style=3D"font-size:13.3333px;margin-top:=
0px;margin-bottom:0px">   o  For users who care more about one versus the o=
ther, provide a
      means to customize the experience.</pre><pre class=3D"gmail-newpage" =
style=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px"><br></pre><p=
re class=3D"gmail-newpage" style=3D"font-size:13.3333px;margin-top:0px;marg=
in-bottom:0px">[BA] This section might state whether it is a </pre><pre cla=
ss=3D"gmail-newpage" style=3D"font-size:13.3333px;margin-top:0px;margin-bot=
tom:0px">goal to enable WebRTC to be used with </pre><pre class=3D"gmail-ne=
wpage" style=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px">The O=
nion Routing network (Tor). There</pre><pre class=3D"gmail-newpage" style=
=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px">are a number of r=
easons why enabling Javascript</pre><pre class=3D"gmail-newpage" style=3D"f=
ont-size:13.3333px;margin-top:0px;margin-bottom:0px">for  use with Tor may =
represent a problem so that</pre><pre class=3D"gmail-newpage" style=3D"font=
-size:13.3333px;margin-top:0px;margin-bottom:0px">use of WebRTC might not b=
e advisable:</pre><pre class=3D"gmail-newpage" style=3D"font-size:13.3333px=
;margin-top:0px;margin-bottom:0px"><a href=3D"https://security.stackexchang=
e.com/questions/95046/why-disable-javascript-in-tor">https://security.stack=
exchange.com/questions/95046/why-disable-javascript-in-tor</a> </pre></pre>=
</pre><pre class=3D"gmail-newpage" style=3D"font-size:13.3333px;margin-top:=
0px;margin-bottom:0px"> </pre><pre class=3D"gmail-newpage" style=3D"font-si=
ze:13.3333px;margin-top:0px;margin-bottom:0px"><pre class=3D"gmail-newpage"=
 style=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px"><span class=
=3D"gmail-h2" style=3D"line-height:0pt;display:inline;font-size:1em;font-we=
ight:bold"><h2 style=3D"line-height:0pt;display:inline;font-size:1em"><a cl=
ass=3D"gmail-selflink" name=3D"section-4" href=3D"https://tools.ietf.org/ht=
ml/draft-ietf-rtcweb-ip-handling-03#section-4" style=3D"color:black;text-de=
coration-line:none">4</a>.  Detailed Design</h2></span></pre><pre class=3D"=
gmail-newpage" style=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0p=
x"><br></pre><pre class=3D"gmail-newpage" style=3D"font-size:13.3333px;marg=
in-top:0px;margin-bottom:0px"><pre class=3D"gmail-newpage" style=3D"font-si=
ze:13.3333px;margin-top:0px;margin-bottom:0px">   1.  By default, WebRTC sh=
ould follow normal IP routing rules, to the
       extent that this is easy to determine (i.e., not considering
       proxies).  This can be accomplished by binding local sockets to
       the wildcard addresses (0.0.0.0 for IPv4, :: for IPv6), which
       allows the OS to route WebRTC traffic the same way as it would
       HTTP traffic, and allows only the &#39;typical&#39; public addresses=
 to
       be discovered.
</pre><div><br></div><div>[BA] This paragraph discusses routing behavior an=
d</div><div>address binding together and leads me to wonder whether</div><d=
iv>some assumptions aren&#39;t being made (e.g. weak versus</div><div>stron=
g host model).  My suggestion is to simplify the</div><div>the paragraph an=
d leave details to later on. For example:</div><div><br></div><div>&quot;1.=
 By default WebRTC should only allow applications</div><div>to discover the=
 same address information made available</div><div>by an HTTP application: =
the &#39;typical&#39; public address.&quot;</div></pre></pre></pre></pre></=
pre></div></div>

--001a1143847cfc15ae054f1f15c0--


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In Section 3, please don't imply that WebRTC is only for privacy.  At
a minimum "Being peer-to-peer, WebRTC represents a technology that
could improve privacy,"  Though I'd avoid this sort of statement
altogether.  One interesting side effect of unintermediated or
peer-to-peer communications is that there is now network-level
metadata showing who was communicating.  There is a direct flow
between peers.  The privacy gain (no single middleman with access to
communications metadata) becomes a potential loss (passive network
observers can see who was talking more easily).

I think that Section 3 could be more simply stated as:

Provide a framework for understanding the problem so that controls
might be provided to make different trade-offs regarding performance
and privacy concerns with WebRTC.  Using that framework, define
settings that enable peer-to-peer communications each with a different
balance between performance and privacy.  Finally, provide
recommendations about settings that provide reasonable performance
without also exposing addressing information in a way that might
violate user expectations.

In Section 4, the mention of RETURN isn't really necessary and it will
stall progress of this draft.  The framing of RETURN ensures that it
falls into the "proxy" definition quite neatly, so it will work
without changing this draft.  Though RETURN would reduce the
performance downsides of the higher-numbered modes - significantly -
it doesn't really need a special mention here.

I'm not very enthusiastic about this bullet in Section 5:

   o  Future versions of browsers may present an indicator to signify
      that the page is using WebRTC to set up a peer-to-peer connection.
      Applications MUST only use WebRTC in a fashion that is consistent
      with user expectations.

The indicator thing is kinda speculative; based on my experience with
these things, I'd say that the chances that we did that would be close
to zero.  Also, the MUST here is completely toothless to the point
that I think it's dangerous.  Applications will care to the extent
that they care, and yelling at them isn't going to influence that one
iota.

Also, since the whole point of this document is to establish a
framework for thinking about this specific problem, I think that it's
safe to remove this sentence.  I would remove the entire bullet point.

Nits:

Many of the references need to have a space after the closing bracket.

Remove the reference to 8.8.8.8.  It's actually a good example, but
it's not necessary and it is likely to cause some people's heads to
explode.  As much as I might personally enjoy that, it's not my
document; you might be better avoiding this.


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Sean Turner <sean@sn3rd.com>
CC: Eric Rescorla <ekr@rtfm.com>, The IESG <iesg@ietf.org>, "rtcweb-chairs@ietf.org" <rtcweb-chairs@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>, "draft-ietf-rtcweb-overview@ietf.org" <draft-ietf-rtcweb-overview@ietf.org>
Thread-Topic: [rtcweb] Eric Rescorla's Discuss on draft-ietf-rtcweb-overview-18: (with DISCUSS and COMMENT)
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Hi,

What is the status regarding this?

Regards,

Christer

On 26/04/17 06:02, "Sean Turner" <sean@sn3rd.com> wrote:

>
>> On Apr 23, 2017, at 14:44, Christer Holmberg
>><christer.holmberg@ericsson.com> wrote:
>>=20
>> Hi,
>>=20
>>> ----------------------------------------------------------------------
>>> DISCUSS:
>>> ----------------------------------------------------------------------
>>>=20
>>> Your citation to ICE is to 5245-bis, but at least the JSEP editor
>>>consensus was that WebRTC depended on 5245, so this needs to be
>>>resolved one way or the other.
>>=20
>> Keep in mind that, no matter what draft-rtcweb-overview and
>>draft-rtcweb-jsep explicitly say, both specs reference 5245bis
>>*IMPLICITLY*, e.g., via draft-mmusic-bundle, draft-ice-trickle etc... As
>>I have indicated in the past, it would cause confusion to reference both.
>>=20
>> So, I think we shall reference 5245-bis everywhere (I also thought we
>>already decided no that in the past)-
>>=20
>> Regards,
>>=20
>> Christer
>
>/* bike shed alert:
>/*=20
>/* Assuming you=B9re of the mind that a bis/updates draft is
>/* signaling to all implementors of the original RFC that the
>/* intention is that all implementations be updated then it=B9s
>/* a bit more than implicit.
>
>spt
>


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From: Eric Rescorla <ekr@rtfm.com>
Date: Thu, 11 May 2017 06:43:51 -0700
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To: Christer Holmberg <christer.holmberg@ericsson.com>
Cc: Sean Turner <sean@sn3rd.com>, The IESG <iesg@ietf.org>,  "rtcweb-chairs@ietf.org" <rtcweb-chairs@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>,  "draft-ietf-rtcweb-overview@ietf.org" <draft-ietf-rtcweb-overview@ietf.org>
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Question for the chairs.

On Thu, May 11, 2017 at 12:44 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> Hi,
>
> What is the status regarding this?
>
> Regards,
>
> Christer
>
> On 26/04/17 06:02, "Sean Turner" <sean@sn3rd.com> wrote:
>
> >
> >> On Apr 23, 2017, at 14:44, Christer Holmberg
> >><christer.holmberg@ericsson.com> wrote:
> >>
> >> Hi,
> >>
> >>> ---------------------------------------------------------------------=
-
> >>> DISCUSS:
> >>> ---------------------------------------------------------------------=
-
> >>>
> >>> Your citation to ICE is to 5245-bis, but at least the JSEP editor
> >>>consensus was that WebRTC depended on 5245, so this needs to be
> >>>resolved one way or the other.
> >>
> >> Keep in mind that, no matter what draft-rtcweb-overview and
> >>draft-rtcweb-jsep explicitly say, both specs reference 5245bis
> >>*IMPLICITLY*, e.g., via draft-mmusic-bundle, draft-ice-trickle etc... A=
s
> >>I have indicated in the past, it would cause confusion to reference bot=
h.
> >>
> >> So, I think we shall reference 5245-bis everywhere (I also thought we
> >>already decided no that in the past)-
> >>
> >> Regards,
> >>
> >> Christer
> >
> >/* bike shed alert:
> >/*
> >/* Assuming you=C2=B9re of the mind that a bis/updates draft is
> >/* signaling to all implementors of the original RFC that the
> >/* intention is that all implementations be updated then it=C2=B9s
> >/* a bit more than implicit.
> >
> >spt
> >
>
>

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<div dir=3D"ltr">Question for the chairs.</div><div class=3D"gmail_extra"><=
br><div class=3D"gmail_quote">On Thu, May 11, 2017 at 12:44 AM, Christer Ho=
lmberg <span dir=3D"ltr">&lt;<a href=3D"mailto:christer.holmberg@ericsson.c=
om" target=3D"_blank">christer.holmberg@ericsson.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">Hi,<br>
<br>
What is the status regarding this?<br>
<br>
Regards,<br>
<br>
Christer<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
On 26/04/17 06:02, &quot;Sean Turner&quot; &lt;<a href=3D"mailto:sean@sn3rd=
.com">sean@sn3rd.com</a>&gt; wrote:<br>
<br>
&gt;<br>
&gt;&gt; On Apr 23, 2017, at 14:44, Christer Holmberg<br>
&gt;&gt;&lt;<a href=3D"mailto:christer.holmberg@ericsson.com">christer.holm=
berg@ericsson.<wbr>com</a>&gt; wrote:<br>
&gt;&gt;<br>
&gt;&gt; Hi,<br>
&gt;&gt;<br>
&gt;&gt;&gt; ------------------------------<wbr>---------------------------=
---<wbr>----------<br>
&gt;&gt;&gt; DISCUSS:<br>
&gt;&gt;&gt; ------------------------------<wbr>---------------------------=
---<wbr>----------<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Your citation to ICE is to 5245-bis, but at least the JSEP edi=
tor<br>
&gt;&gt;&gt;consensus was that WebRTC depended on 5245, so this needs to be=
<br>
&gt;&gt;&gt;resolved one way or the other.<br>
&gt;&gt;<br>
&gt;&gt; Keep in mind that, no matter what draft-rtcweb-overview and<br>
&gt;&gt;draft-rtcweb-jsep explicitly say, both specs reference 5245bis<br>
&gt;&gt;*IMPLICITLY*, e.g., via draft-mmusic-bundle, draft-ice-trickle etc.=
.. As<br>
&gt;&gt;I have indicated in the past, it would cause confusion to reference=
 both.<br>
&gt;&gt;<br>
&gt;&gt; So, I think we shall reference 5245-bis everywhere (I also thought=
 we<br>
&gt;&gt;already decided no that in the past)-<br>
&gt;&gt;<br>
&gt;&gt; Regards,<br>
&gt;&gt;<br>
&gt;&gt; Christer<br>
&gt;<br>
&gt;/* bike shed alert:<br>
&gt;/*<br>
&gt;/* Assuming you=C2=B9re of the mind that a bis/updates draft is<br>
&gt;/* signaling to all implementors of the original RFC that the<br>
&gt;/* intention is that all implementations be updated then it=C2=B9s<br>
&gt;/* a bit more than implicit.<br>
&gt;<br>
&gt;spt<br>
&gt;<br>
<br>
</div></div></blockquote></div><br></div>

--94eb2c087eeae49022054f3fc949--


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Date: Fri, 12 May 2017 17:54:55 -0600
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To: Eric Rescorla <ekr@rtfm.com>, Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [rtcweb] Eric Rescorla's Discuss on draft-ietf-rtcweb-overview-18: (with DISCUSS and COMMENT)
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Let me try and answer with the caveat that I may get this wrong and need =
to be corrected by my co-chairs...

TL;DR - The chairs recommend changing the ref in overview to point at =
5245 instead of 5245-bis


here is the longer version....=20

First a side note on how we got here. Some of the reasons we set up =
dependencies  like they are is that many years ago we made guess about =
what order work would get completed on and, shockingly, some predictions =
of standards developments timelines were less than perfect. We also =
assumed that some new work was going to require changes in ICE but as =
that work went progressed, we largely figured out ways to make it work =
with existing ICE implementations.=20

We are confident that overview does not actually depend on anything in =
5245-bis but instead just depends on 5245.=20

Next lets discuss trickle-ice. The WebRTC set of specs currently =
normatively depends on trickle ICE. There is some questions if trickle =
ICE might depend on 5245-bis. Some of the trickle ICE  authors do not =
think it does. One of the authors said the chairs asked them to ref 5245 =
instead of 5245bis as both tickle ICE and 5245bis would be done around =
the same time. In general, the webrtc chairs would prefer to make the =
WebRTC dependency cluster as small as possible. If trickle ice actually =
gets done before 5245bis, and it does not depend on any 5245bis =
features, then clearly is should be changed to just depend on 5245. The =
WG responsible for 5245bis and trickle ICE can figure out what they want =
to do as both theses drafts progress. Given there is a strong =
possibility that trickle ice will only reference 5245, we think it would =
be better if overview did not bring 5245bis into the WebRTC dependency =
cluster. If on the other hand, trickle ICE does end up depending on =
5245bis, there is no harm, and no need to change overview to point at =
5245.=20

There are other drafts that are normative dependencies of JSEP and the =
WebRTC cluster that also point at 5245bis. When we consider the =
technical things these drafts need, it seems likely they can also =
reference 5245 instead of 5245bis. For example =
draft-ietf-ice-dualstack-fairness, draft-ietf-mmusic-ice-sip-sdp, =
draft-ietf-mmusic-sdp-bundle-negotiation, and =
draft-ietf-rtcweb-transports (which is with the RFC Editor). The =
argument for the others is roughly the same as it is with trickle ICE.=20=


Cullen (without review from my co-chairs but trying to represent what we =
discussed on our chair call)


PS - if you are trying to figure out some of the dependencies for the =
WebRTC cluster, you might find =
https://datatracker.ietf.org/doc/draft-jennings-rtcweb-deps/?include_text=3D=
1 useful but it is not 100% accurate.=20



> On May 11, 2017, at 7:43 AM, Eric Rescorla <ekr@rtfm.com> wrote:
>=20
> Question for the chairs.
>=20
> On Thu, May 11, 2017 at 12:44 AM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:
> Hi,
>=20
> What is the status regarding this?
>=20
> Regards,
>=20
> Christer
>=20
> On 26/04/17 06:02, "Sean Turner" <sean@sn3rd.com> wrote:
>=20
> >
> >> On Apr 23, 2017, at 14:44, Christer Holmberg
> >><christer.holmberg@ericsson.com> wrote:
> >>
> >> Hi,
> >>
> >>> =
----------------------------------------------------------------------
> >>> DISCUSS:
> >>> =
----------------------------------------------------------------------
> >>>
> >>> Your citation to ICE is to 5245-bis, but at least the JSEP editor
> >>>consensus was that WebRTC depended on 5245, so this needs to be
> >>>resolved one way or the other.
> >>
> >> Keep in mind that, no matter what draft-rtcweb-overview and
> >>draft-rtcweb-jsep explicitly say, both specs reference 5245bis
> >>*IMPLICITLY*, e.g., via draft-mmusic-bundle, draft-ice-trickle =
etc... As
> >>I have indicated in the past, it would cause confusion to reference =
both.
> >>
> >> So, I think we shall reference 5245-bis everywhere (I also thought =
we
> >>already decided no that in the past)-
> >>
> >> Regards,
> >>
> >> Christer
> >
> >/* bike shed alert:
> >/*
> >/* Assuming you=C2=B9re of the mind that a bis/updates draft is
> >/* signaling to all implementors of the original RFC that the
> >/* intention is that all implementations be updated then it=C2=B9s
> >/* a bit more than implicit.
> >
> >spt
> >
>=20
>=20


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Cullen Jennings <fluffy@iii.ca>, Eric Rescorla <ekr@rtfm.com>
CC: Sean Turner <sean@sn3rd.com>, The IESG <iesg@ietf.org>, "rtcweb-chairs@ietf.org" <rtcweb-chairs@ietf.org>, RTCWeb IETF <rtcweb@ietf.org>, "draft-ietf-rtcweb-overview@ietf.org" <draft-ietf-rtcweb-overview@ietf.org>
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Subject: Re: [rtcweb] Eric Rescorla's Discuss on draft-ietf-rtcweb-overview-18: (with DISCUSS and COMMENT)
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From nobody Sat May 13 06:44:39 2017
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From: Cullen Jennings <fluffy@iii.ca>
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Date: Sat, 13 May 2017 07:42:04 -0600
Cc: Eric Rescorla <ekr@rtfm.com>, Sean Turner <sean@sn3rd.com>, The IESG <iesg@ietf.org>, "rtcweb-chairs@ietf.org" <rtcweb-chairs@ietf.org>, RTCWeb IETF <rtcweb@ietf.org>, "draft-ietf-rtcweb-overview@ietf.org" <draft-ietf-rtcweb-overview@ietf.org>
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Archived-At: <https://mailarchive.ietf.org/arch/msg/rtcweb/VKFUDHL5EKI7ZQVbWG8DyX7G-rc>
Subject: Re: [rtcweb] Eric Rescorla's Discuss on draft-ietf-rtcweb-overview-18: (with DISCUSS and COMMENT)
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OK, so let me try to give some really clear guidance here ... if you are =
writing a draft that does not need to normatively depend on another =
draft, it should not normatively depended on that draft.=20

I really doubt anyone is going to argue with that so lets make it so.=20

> On May 13, 2017, at 1:54 AM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:
>=20
> Hi,
>=20
> As co-chair of a number of the dependencies, we have been discussing =
what to reference a number of times, and we have also changed the =
references. We can't keep changing back and forth.=20
>=20
> In addition, I don't think WG X can decide what the drafts of WG Y =
reference. There needs to be a collaborative decision.
>=20
> "We also assumed that some new work was going to require changes in =
ICE but as that work went progressed, we largely figured out ways to =
make it work with existing ICE implementations."
>=20
> Is this "discovery" documented somewhere?
>=20
> "If trickle ice actually gets done before 5245bis, and it does not =
depend on any 5245bis features, then clearly is should be changed to =
just depend on 5245."
>=20
> First, we need to agree on whether trickle depends on 5245bis features =
or not.
>=20
> Second, as co-author of 5245bis, I have asked the chairs to initiate =
the road towards WGLC, so I would hope both 5245bis and trickle-ice =
could be done more or less at the same time.
>=20
> Regards,
>=20
> Christer
>=20
>=20
> -----Original Message-----
> From: Cullen Jennings [mailto:fluffy@iii.ca]=20
> Sent: 13 May 2017 01:55
> To: Eric Rescorla <ekr@rtfm.com>; Christer Holmberg =
<christer.holmberg@ericsson.com>
> Cc: Sean Turner <sean@sn3rd.com>; The IESG <iesg@ietf.org>; =
rtcweb-chairs@ietf.org; RTCWeb IETF <rtcweb@ietf.org>; =
draft-ietf-rtcweb-overview@ietf.org
> Subject: Re: [rtcweb] Eric Rescorla's Discuss on =
draft-ietf-rtcweb-overview-18: (with DISCUSS and COMMENT)
>=20
>=20
> Let me try and answer with the caveat that I may get this wrong and =
need to be corrected by my co-chairs...
>=20
> TL;DR - The chairs recommend changing the ref in overview to point at =
5245 instead of 5245-bis
>=20
>=20
> here is the longer version....=20
>=20
> First a side note on how we got here. Some of the reasons we set up =
dependencies  like they are is that many years ago we made guess about =
what order work would get completed on and, shockingly, some predictions =
of standards developments timelines were less than perfect. We also =
assumed that some new work was going to require changes in ICE but as =
that work went progressed, we largely figured out ways to make it work =
with existing ICE implementations.=20
>=20
> We are confident that overview does not actually depend on anything in =
5245-bis but instead just depends on 5245.=20
>=20
> Next lets discuss trickle-ice. The WebRTC set of specs currently =
normatively depends on trickle ICE. There is some questions if trickle =
ICE might depend on 5245-bis. Some of the trickle ICE  authors do not =
think it does. One of the authors said the chairs asked them to ref 5245 =
instead of 5245bis as both tickle ICE and 5245bis would be done around =
the same time. In general, the webrtc chairs would prefer to make the =
WebRTC dependency cluster as small as possible. If trickle ice actually =
gets done before 5245bis, and it does not depend on any 5245bis =
features, then clearly is should be changed to just depend on 5245. The =
WG responsible for 5245bis and trickle ICE can figure out what they want =
to do as both theses drafts progress. Given there is a strong =
possibility that trickle ice will only reference 5245, we think it would =
be better if overview did not bring 5245bis into the WebRTC dependency =
cluster. If on the other hand, trickle ICE does end up depending on =
5245bis, there is no harm, and no need to change overview to point at =
5245.=20
>=20
> There are other drafts that are normative dependencies of JSEP and the =
WebRTC cluster that also point at 5245bis. When we consider the =
technical things these drafts need, it seems likely they can also =
reference 5245 instead of 5245bis. For example =
draft-ietf-ice-dualstack-fairness, draft-ietf-mmusic-ice-sip-sdp, =
draft-ietf-mmusic-sdp-bundle-negotiation, and =
draft-ietf-rtcweb-transports (which is with the RFC Editor). The =
argument for the others is roughly the same as it is with trickle ICE.=20=

>=20
> Cullen (without review from my co-chairs but trying to represent what =
we discussed on our chair call)
>=20
>=20
> PS - if you are trying to figure out some of the dependencies for the =
WebRTC cluster, you might find =
https://datatracker.ietf.org/doc/draft-jennings-rtcweb-deps/?include_text=3D=
1 useful but it is not 100% accurate.=20
>=20
>=20
>=20
>> On May 11, 2017, at 7:43 AM, Eric Rescorla <ekr@rtfm.com> wrote:
>>=20
>> Question for the chairs.
>>=20
>> On Thu, May 11, 2017 at 12:44 AM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:
>> Hi,
>>=20
>> What is the status regarding this?
>>=20
>> Regards,
>>=20
>> Christer
>>=20
>> On 26/04/17 06:02, "Sean Turner" <sean@sn3rd.com> wrote:
>>=20
>>>=20
>>>> On Apr 23, 2017, at 14:44, Christer Holmberg=20
>>>> <christer.holmberg@ericsson.com> wrote:
>>>>=20
>>>> Hi,
>>>>=20
>>>>> ------------------------------------------------------------------
>>>>> ----
>>>>> DISCUSS:
>>>>> ------------------------------------------------------------------
>>>>> ----
>>>>>=20
>>>>> Your citation to ICE is to 5245-bis, but at least the JSEP editor=20=

>>>>> consensus was that WebRTC depended on 5245, so this needs to be=20
>>>>> resolved one way or the other.
>>>>=20
>>>> Keep in mind that, no matter what draft-rtcweb-overview and=20
>>>> draft-rtcweb-jsep explicitly say, both specs reference 5245bis=20
>>>> *IMPLICITLY*, e.g., via draft-mmusic-bundle, draft-ice-trickle=20
>>>> etc... As I have indicated in the past, it would cause confusion to =
reference both.
>>>>=20
>>>> So, I think we shall reference 5245-bis everywhere (I also thought=20=

>>>> we already decided no that in the past)-
>>>>=20
>>>> Regards,
>>>>=20
>>>> Christer
>>>=20
>>> /* bike shed alert:
>>> /*
>>> /* Assuming you=C2=B9re of the mind that a bis/updates draft is
>>> /* signaling to all implementors of the original RFC that the
>>> /* intention is that all implementations be updated then it=C2=B9s
>>> /* a bit more than implicit.
>>>=20
>>> spt
>>>=20
>>=20
>>=20
>=20


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Cullen Jennings <fluffy@iii.ca>
CC: Eric Rescorla <ekr@rtfm.com>, Sean Turner <sean@sn3rd.com>, The IESG <iesg@ietf.org>, "rtcweb-chairs@ietf.org" <rtcweb-chairs@ietf.org>, "RTCWeb IETF" <rtcweb@ietf.org>, "draft-ietf-rtcweb-overview@ietf.org" <draft-ietf-rtcweb-overview@ietf.org>
Thread-Topic: [rtcweb] Eric Rescorla's Discuss on draft-ietf-rtcweb-overview-18: (with DISCUSS and COMMENT)
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Subject: Re: [rtcweb] Eric Rescorla's Discuss on draft-ietf-rtcweb-overview-18: (with DISCUSS and COMMENT)
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From: Eric Rescorla <ekr@rtfm.com>
Date: Sat, 13 May 2017 10:49:03 -0700
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To: Christer Holmberg <christer.holmberg@ericsson.com>
Cc: Cullen Jennings <fluffy@iii.ca>, Sean Turner <sean@sn3rd.com>, The IESG <iesg@ietf.org>,  "rtcweb-chairs@ietf.org" <rtcweb-chairs@ietf.org>, RTCWeb IETF <rtcweb@ietf.org>,  "draft-ietf-rtcweb-overview@ietf.org" <draft-ietf-rtcweb-overview@ietf.org>
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Subject: Re: [rtcweb] Eric Rescorla's Discuss on draft-ietf-rtcweb-overview-18: (with DISCUSS and COMMENT)
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On Sat, May 13, 2017 at 10:31 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> Hi,
>
> So, if the changes done in 5245bis aren't needed, why are we working on
> 5245bis to begin with??? We did we spend all that time on modifying Ta
> etc???
>

I don't think anyone is saying that 5245 bis isn't a good thing, merely
that one can implement JSEP and trickle without reading it.

-Ekr


>
> Regards,
>
> Christer
>
>
> Sent from my iPhone
>
> > On 13 May 2017, at 16.42, Cullen Jennings <fluffy@iii.ca> wrote:
> >
> >
> > OK, so let me try to give some really clear guidance here ... if you ar=
e
> writing a draft that does not need to normatively depend on another draft=
,
> it should not normatively depended on that draft.
> >
> > I really doubt anyone is going to argue with that so lets make it so.
> >
> >> On May 13, 2017, at 1:54 AM, Christer Holmberg <
> christer.holmberg@ericsson.com> wrote:
> >>
> >> Hi,
> >>
> >> As co-chair of a number of the dependencies, we have been discussing
> what to reference a number of times, and we have also changed the
> references. We can't keep changing back and forth.
> >>
> >> In addition, I don't think WG X can decide what the drafts of WG Y
> reference. There needs to be a collaborative decision.
> >>
> >> "We also assumed that some new work was going to require changes in IC=
E
> but as that work went progressed, we largely figured out ways to make it
> work with existing ICE implementations."
> >>
> >> Is this "discovery" documented somewhere?
> >>
> >> "If trickle ice actually gets done before 5245bis, and it does not
> depend on any 5245bis features, then clearly is should be changed to just
> depend on 5245."
> >>
> >> First, we need to agree on whether trickle depends on 5245bis features
> or not.
> >>
> >> Second, as co-author of 5245bis, I have asked the chairs to initiate
> the road towards WGLC, so I would hope both 5245bis and trickle-ice could
> be done more or less at the same time.
> >>
> >> Regards,
> >>
> >> Christer
> >>
> >>
> >> -----Original Message-----
> >> From: Cullen Jennings [mailto:fluffy@iii.ca]
> >> Sent: 13 May 2017 01:55
> >> To: Eric Rescorla <ekr@rtfm.com>; Christer Holmberg <
> christer.holmberg@ericsson.com>
> >> Cc: Sean Turner <sean@sn3rd.com>; The IESG <iesg@ietf.org>;
> rtcweb-chairs@ietf.org; RTCWeb IETF <rtcweb@ietf.org>;
> draft-ietf-rtcweb-overview@ietf.org
> >> Subject: Re: [rtcweb] Eric Rescorla's Discuss on
> draft-ietf-rtcweb-overview-18: (with DISCUSS and COMMENT)
> >>
> >>
> >> Let me try and answer with the caveat that I may get this wrong and
> need to be corrected by my co-chairs...
> >>
> >> TL;DR - The chairs recommend changing the ref in overview to point at
> 5245 instead of 5245-bis
> >>
> >>
> >> here is the longer version....
> >>
> >> First a side note on how we got here. Some of the reasons we set up
> dependencies  like they are is that many years ago we made guess about wh=
at
> order work would get completed on and, shockingly, some predictions of
> standards developments timelines were less than perfect. We also assumed
> that some new work was going to require changes in ICE but as that work
> went progressed, we largely figured out ways to make it work with existin=
g
> ICE implementations.
> >>
> >> We are confident that overview does not actually depend on anything in
> 5245-bis but instead just depends on 5245.
> >>
> >> Next lets discuss trickle-ice. The WebRTC set of specs currently
> normatively depends on trickle ICE. There is some questions if trickle IC=
E
> might depend on 5245-bis. Some of the trickle ICE  authors do not think i=
t
> does. One of the authors said the chairs asked them to ref 5245 instead o=
f
> 5245bis as both tickle ICE and 5245bis would be done around the same time=
.
> In general, the webrtc chairs would prefer to make the WebRTC dependency
> cluster as small as possible. If trickle ice actually gets done before
> 5245bis, and it does not depend on any 5245bis features, then clearly is
> should be changed to just depend on 5245. The WG responsible for 5245bis
> and trickle ICE can figure out what they want to do as both theses drafts
> progress. Given there is a strong possibility that trickle ice will only
> reference 5245, we think it would be better if overview did not bring
> 5245bis into the WebRTC dependency cluster. If on the other hand, trickle
> ICE does end up depending on 5245bis, there is no harm, and no need to
> change overview to point at 5245.
> >>
> >> There are other drafts that are normative dependencies of JSEP and the
> WebRTC cluster that also point at 5245bis. When we consider the technical
> things these drafts need, it seems likely they can also reference 5245
> instead of 5245bis. For example draft-ietf-ice-dualstack-fairness,
> draft-ietf-mmusic-ice-sip-sdp, draft-ietf-mmusic-sdp-bundle-negotiation,
> and draft-ietf-rtcweb-transports (which is with the RFC Editor). The
> argument for the others is roughly the same as it is with trickle ICE.
> >>
> >> Cullen (without review from my co-chairs but trying to represent what
> we discussed on our chair call)
> >>
> >>
> >> PS - if you are trying to figure out some of the dependencies for the
> WebRTC cluster, you might find https://datatracker.ietf.org/
> doc/draft-jennings-rtcweb-deps/?include_text=3D1 useful but it is not 100=
%
> accurate.
> >>
> >>
> >>
> >>> On May 11, 2017, at 7:43 AM, Eric Rescorla <ekr@rtfm.com> wrote:
> >>>
> >>> Question for the chairs.
> >>>
> >>> On Thu, May 11, 2017 at 12:44 AM, Christer Holmberg <
> christer.holmberg@ericsson.com> wrote:
> >>> Hi,
> >>>
> >>> What is the status regarding this?
> >>>
> >>> Regards,
> >>>
> >>> Christer
> >>>
> >>>> On 26/04/17 06:02, "Sean Turner" <sean@sn3rd.com> wrote:
> >>>>
> >>>>
> >>>>> On Apr 23, 2017, at 14:44, Christer Holmberg
> >>>>> <christer.holmberg@ericsson.com> wrote:
> >>>>>
> >>>>> Hi,
> >>>>>
> >>>>>> ------------------------------------------------------------------
> >>>>>> ----
> >>>>>> DISCUSS:
> >>>>>> ------------------------------------------------------------------
> >>>>>> ----
> >>>>>>
> >>>>>> Your citation to ICE is to 5245-bis, but at least the JSEP editor
> >>>>>> consensus was that WebRTC depended on 5245, so this needs to be
> >>>>>> resolved one way or the other.
> >>>>>
> >>>>> Keep in mind that, no matter what draft-rtcweb-overview and
> >>>>> draft-rtcweb-jsep explicitly say, both specs reference 5245bis
> >>>>> *IMPLICITLY*, e.g., via draft-mmusic-bundle, draft-ice-trickle
> >>>>> etc... As I have indicated in the past, it would cause confusion to
> reference both.
> >>>>>
> >>>>> So, I think we shall reference 5245-bis everywhere (I also thought
> >>>>> we already decided no that in the past)-
> >>>>>
> >>>>> Regards,
> >>>>>
> >>>>> Christer
> >>>>
> >>>> /* bike shed alert:
> >>>> /*
> >>>> /* Assuming you=C2=B9re of the mind that a bis/updates draft is
> >>>> /* signaling to all implementors of the original RFC that the
> >>>> /* intention is that all implementations be updated then it=C2=B9s
> >>>> /* a bit more than implicit.
> >>>>
> >>>> spt
> >>>>
> >>>
> >>>
> >>
> >
>

--f403045c563277d578054f6b7287
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 Sat, May 13, 2017 at 10:31 AM, Christer Holmberg <span dir=3D"ltr">&=
lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">chri=
ster.holmberg@ericsson.com</a>&gt;</span> wrote:<br><blockquote class=3D"gm=
ail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-le=
ft:1ex">Hi,<br>
<br>
So, if the changes done in 5245bis aren&#39;t needed, why are we working on=
 5245bis to begin with??? We did we spend all that time on modifying Ta etc=
???<br></blockquote><div><br></div><div>I don&#39;t think anyone is saying =
that 5245 bis isn&#39;t a good thing, merely that one can implement JSEP an=
d trickle without reading it.</div><div><br></div><div>-Ekr</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">
<br>
Regards,<br>
<br>
Christer<br>
<br>
<br>
Sent from my iPhone<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
&gt; On 13 May 2017, at 16.42, Cullen Jennings &lt;<a href=3D"mailto:fluffy=
@iii.ca">fluffy@iii.ca</a>&gt; wrote:<br>
&gt;<br>
&gt;<br>
&gt; OK, so let me try to give some really clear guidance here ... if you a=
re writing a draft that does not need to normatively depend on another draf=
t, it should not normatively depended on that draft.<br>
&gt;<br>
&gt; I really doubt anyone is going to argue with that so lets make it so.<=
br>
&gt;<br>
&gt;&gt; On May 13, 2017, at 1:54 AM, Christer Holmberg &lt;<a href=3D"mail=
to:christer.holmberg@ericsson.com">christer.holmberg@ericsson.<wbr>com</a>&=
gt; wrote:<br>
&gt;&gt;<br>
&gt;&gt; Hi,<br>
&gt;&gt;<br>
&gt;&gt; As co-chair of a number of the dependencies, we have been discussi=
ng what to reference a number of times, and we have also changed the refere=
nces. We can&#39;t keep changing back and forth.<br>
&gt;&gt;<br>
&gt;&gt; In addition, I don&#39;t think WG X can decide what the drafts of =
WG Y reference. There needs to be a collaborative decision.<br>
&gt;&gt;<br>
&gt;&gt; &quot;We also assumed that some new work was going to require chan=
ges in ICE but as that work went progressed, we largely figured out ways to=
 make it work with existing ICE implementations.&quot;<br>
&gt;&gt;<br>
&gt;&gt; Is this &quot;discovery&quot; documented somewhere?<br>
&gt;&gt;<br>
&gt;&gt; &quot;If trickle ice actually gets done before 5245bis, and it doe=
s not depend on any 5245bis features, then clearly is should be changed to =
just depend on 5245.&quot;<br>
&gt;&gt;<br>
&gt;&gt; First, we need to agree on whether trickle depends on 5245bis feat=
ures or not.<br>
&gt;&gt;<br>
&gt;&gt; Second, as co-author of 5245bis, I have asked the chairs to initia=
te the road towards WGLC, so I would hope both 5245bis and trickle-ice coul=
d be done more or less at the same time.<br>
&gt;&gt;<br>
&gt;&gt; Regards,<br>
&gt;&gt;<br>
&gt;&gt; Christer<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; -----Original Message-----<br>
&gt;&gt; From: Cullen Jennings [mailto:<a href=3D"mailto:fluffy@iii.ca">flu=
ffy@iii.ca</a>]<br>
&gt;&gt; Sent: 13 May 2017 01:55<br>
&gt;&gt; To: Eric Rescorla &lt;<a href=3D"mailto:ekr@rtfm.com">ekr@rtfm.com=
</a>&gt;; Christer Holmberg &lt;<a href=3D"mailto:christer.holmberg@ericsso=
n.com">christer.holmberg@ericsson.<wbr>com</a>&gt;<br>
&gt;&gt; Cc: Sean Turner &lt;<a href=3D"mailto:sean@sn3rd.com">sean@sn3rd.c=
om</a>&gt;; The IESG &lt;<a href=3D"mailto:iesg@ietf.org">iesg@ietf.org</a>=
&gt;; <a href=3D"mailto:rtcweb-chairs@ietf.org">rtcweb-chairs@ietf.org</a>;=
 RTCWeb IETF &lt;<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>&gt;=
; <a href=3D"mailto:draft-ietf-rtcweb-overview@ietf.org">draft-ietf-rtcweb-=
overview@<wbr>ietf.org</a><br>
&gt;&gt; Subject: Re: [rtcweb] Eric Rescorla&#39;s Discuss on draft-ietf-rt=
cweb-overview-18: (with DISCUSS and COMMENT)<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; Let me try and answer with the caveat that I may get this wrong an=
d need to be corrected by my co-chairs...<br>
&gt;&gt;<br>
&gt;&gt; TL;DR - The chairs recommend changing the ref in overview to point=
 at 5245 instead of 5245-bis<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; here is the longer version....<br>
&gt;&gt;<br>
&gt;&gt; First a side note on how we got here. Some of the reasons we set u=
p dependencies=C2=A0 like they are is that many years ago we made guess abo=
ut what order work would get completed on and, shockingly, some predictions=
 of standards developments timelines were less than perfect. We also assume=
d that some new work was going to require changes in ICE but as that work w=
ent progressed, we largely figured out ways to make it work with existing I=
CE implementations.<br>
&gt;&gt;<br>
&gt;&gt; We are confident that overview does not actually depend on anythin=
g in 5245-bis but instead just depends on 5245.<br>
&gt;&gt;<br>
&gt;&gt; Next lets discuss trickle-ice. The WebRTC set of specs currently n=
ormatively depends on trickle ICE. There is some questions if trickle ICE m=
ight depend on 5245-bis. Some of the trickle ICE=C2=A0 authors do not think=
 it does. One of the authors said the chairs asked them to ref 5245 instead=
 of 5245bis as both tickle ICE and 5245bis would be done around the same ti=
me. In general, the webrtc chairs would prefer to make the WebRTC dependenc=
y cluster as small as possible. If trickle ice actually gets done before 52=
45bis, and it does not depend on any 5245bis features, then clearly is shou=
ld be changed to just depend on 5245. The WG responsible for 5245bis and tr=
ickle ICE can figure out what they want to do as both theses drafts progres=
s. Given there is a strong possibility that trickle ice will only reference=
 5245, we think it would be better if overview did not bring 5245bis into t=
he WebRTC dependency cluster. If on the other hand, trickle ICE does end up=
 depending on 5245bis, there is no harm, and no need to change overview to =
point at 5245.<br>
&gt;&gt;<br>
&gt;&gt; There are other drafts that are normative dependencies of JSEP and=
 the WebRTC cluster that also point at 5245bis. When we consider the techni=
cal things these drafts need, it seems likely they can also reference 5245 =
instead of 5245bis. For example draft-ietf-ice-dualstack-<wbr>fairness, dra=
ft-ietf-mmusic-ice-sip-sdp, draft-ietf-mmusic-sdp-bundle-<wbr>negotiation, =
and draft-ietf-rtcweb-transports (which is with the RFC Editor). The argume=
nt for the others is roughly the same as it is with trickle ICE.<br>
&gt;&gt;<br>
&gt;&gt; Cullen (without review from my co-chairs but trying to represent w=
hat we discussed on our chair call)<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; PS - if you are trying to figure out some of the dependencies for =
the WebRTC cluster, you might find <a href=3D"https://datatracker.ietf.org/=
doc/draft-jennings-rtcweb-deps/?include_text=3D1" rel=3D"noreferrer" target=
=3D"_blank">https://datatracker.ietf.org/<wbr>doc/draft-jennings-rtcweb-<wb=
r>deps/?include_text=3D1</a> useful but it is not 100% accurate.<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;&gt; On May 11, 2017, at 7:43 AM, Eric Rescorla &lt;<a href=3D"mail=
to:ekr@rtfm.com">ekr@rtfm.com</a>&gt; wrote:<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Question for the chairs.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; On Thu, May 11, 2017 at 12:44 AM, Christer Holmberg &lt;<a hre=
f=3D"mailto:christer.holmberg@ericsson.com">christer.holmberg@ericsson.<wbr=
>com</a>&gt; wrote:<br>
&gt;&gt;&gt; Hi,<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; What is the status regarding this?<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Regards,<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Christer<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; On 26/04/17 06:02, &quot;Sean Turner&quot; &lt;<a href=3D"=
mailto:sean@sn3rd.com">sean@sn3rd.com</a>&gt; wrote:<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt; On Apr 23, 2017, at 14:44, Christer Holmberg<br>
&gt;&gt;&gt;&gt;&gt; &lt;<a href=3D"mailto:christer.holmberg@ericsson.com">=
christer.holmberg@ericsson.<wbr>com</a>&gt; wrote:<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt; Hi,<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt; ------------------------------<wbr>---------------=
---------------<wbr>------<br>
&gt;&gt;&gt;&gt;&gt;&gt; ----<br>
&gt;&gt;&gt;&gt;&gt;&gt; DISCUSS:<br>
&gt;&gt;&gt;&gt;&gt;&gt; ------------------------------<wbr>---------------=
---------------<wbr>------<br>
&gt;&gt;&gt;&gt;&gt;&gt; ----<br>
&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt; Your citation to ICE is to 5245-bis, but at least =
the JSEP editor<br>
&gt;&gt;&gt;&gt;&gt;&gt; consensus was that WebRTC depended on 5245, so thi=
s needs to be<br>
&gt;&gt;&gt;&gt;&gt;&gt; resolved one way or the other.<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt; Keep in mind that, no matter what draft-rtcweb-overvie=
w and<br>
&gt;&gt;&gt;&gt;&gt; draft-rtcweb-jsep explicitly say, both specs reference=
 5245bis<br>
&gt;&gt;&gt;&gt;&gt; *IMPLICITLY*, e.g., via draft-mmusic-bundle, draft-ice=
-trickle<br>
&gt;&gt;&gt;&gt;&gt; etc... As I have indicated in the past, it would cause=
 confusion to reference both.<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt; So, I think we shall reference 5245-bis everywhere (I =
also thought<br>
&gt;&gt;&gt;&gt;&gt; we already decided no that in the past)-<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt; Regards,<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt; Christer<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; /* bike shed alert:<br>
&gt;&gt;&gt;&gt; /*<br>
&gt;&gt;&gt;&gt; /* Assuming you=C2=B9re of the mind that a bis/updates dra=
ft is<br>
&gt;&gt;&gt;&gt; /* signaling to all implementors of the original RFC that =
the<br>
&gt;&gt;&gt;&gt; /* intention is that all implementations be updated then i=
t=C2=B9s<br>
&gt;&gt;&gt;&gt; /* a bit more than implicit.<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; spt<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;<br>
&gt;&gt;<br>
&gt;<br>
</div></div></blockquote></div><br></div></div>

--f403045c563277d578054f6b7287--


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Eric Rescorla <ekr@rtfm.com>
CC: Cullen Jennings <fluffy@iii.ca>, Sean Turner <sean@sn3rd.com>, The IESG <iesg@ietf.org>, "rtcweb-chairs@ietf.org" <rtcweb-chairs@ietf.org>, "RTCWeb IETF" <rtcweb@ietf.org>, "draft-ietf-rtcweb-overview@ietf.org" <draft-ietf-rtcweb-overview@ietf.org>
Thread-Topic: [rtcweb] Eric Rescorla's Discuss on draft-ietf-rtcweb-overview-18: (with DISCUSS and COMMENT)
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From: Eric Rescorla <ekr@rtfm.com>
Date: Sat, 13 May 2017 11:00:28 -0700
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To: Christer Holmberg <christer.holmberg@ericsson.com>
Cc: Cullen Jennings <fluffy@iii.ca>, Sean Turner <sean@sn3rd.com>, The IESG <iesg@ietf.org>,  "rtcweb-chairs@ietf.org" <rtcweb-chairs@ietf.org>, RTCWeb IETF <rtcweb@ietf.org>,  "draft-ietf-rtcweb-overview@ietf.org" <draft-ietf-rtcweb-overview@ietf.org>
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Subject: Re: [rtcweb] Eric Rescorla's Discuss on draft-ietf-rtcweb-overview-18: (with DISCUSS and COMMENT)
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On Sat, May 13, 2017 at 10:57 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> Hi,
>
> >>So, if the changes done in 5245bis aren't needed, why are we working on
> 5245bis to begin with??? We did we spend all that time on modifying Ta
> etc???
> >
> >I don't think anyone is saying that 5245 bis isn't a good thing, merely
> that one can implement JSEP and trickle without reading it.
>
> This isn't about JSEP and trickle, it's about rtcweb-overview- which is
> supposed to give some overview of what people need to implement.
>

Yes, and rtcweb-overview references 5245-bis:

1. Directly
2. Transitively through other documents, which are primarily JSEP and
trickle.

And my point is that to implement those other documents, you need not
implement
5245-bis.


So, *IF* we decide to reference RFC 5245 in rtcweb-overview, I think that
> we at least should indicate that some dependencies might require 5245bis,
> and refer to individual dependencies for details. That way we can progres=
s
> rtcweb-overview, but still keep the door open for 5245bis where/if it's
> needed.
>

That just seems like kicking the can down the road. To the best of my
knowledge,
there is no technical reason why an rtcweb implementation needs to know
about
5245-bis (whatever the structure of the normative references in the
documents
happens to be). Are you aware of such a reason?

-Ekr


> Regards,
>
> Christer
>
>
>
>
> Sent from my iPhone
>
> > On 13 May 2017, at 16.42, Cullen Jennings <fluffy@iii.ca> wrote:
> >
> >
> > OK, so let me try to give some really clear guidance here ... if you ar=
e
> writing a draft that does not need to normatively depend on another draft=
,
> it should not normatively depended on that draft.
> >
> > I really doubt anyone is going to argue with that so lets make it so.
> >
> >> On May 13, 2017, at 1:54 AM, Christer Holmberg <
> christer.holmberg@ericsson.com> wrote:
> >>
> >> Hi,
> >>
> >> As co-chair of a number of the dependencies, we have been discussing
> what to reference a number of times, and we have also changed the
> references. We can't keep changing back and forth.
> >>
> >> In addition, I don't think WG X can decide what the drafts of WG Y
> reference. There needs to be a collaborative decision.
> >>
> >> "We also assumed that some new work was going to require changes in IC=
E
> but as that work went progressed, we largely figured out ways to make it
> work with existing ICE implementations."
> >>
> >> Is this "discovery" documented somewhere?
> >>
> >> "If trickle ice actually gets done before 5245bis, and it does not
> depend on any 5245bis features, then clearly is should be changed to just
> depend on 5245."
> >>
> >> First, we need to agree on whether trickle depends on 5245bis features
> or not.
> >>
> >> Second, as co-author of 5245bis, I have asked the chairs to initiate
> the road towards WGLC, so I would hope both 5245bis and trickle-ice could
> be done more or less at the same time.
> >>
> >> Regards,
> >>
> >> Christer
> >>
> >>
> >> -----Original Message-----
> >> From: Cullen Jennings [mailto:fluffy@iii.ca]
> >> Sent: 13 May 2017 01:55
> >> To: Eric Rescorla <ekr@rtfm.com>; Christer Holmberg <
> christer.holmberg@ericsson.com>
> >> Cc: Sean Turner <sean@sn3rd.com>; The IESG <iesg@ietf.org>;
> rtcweb-chairs@ietf.org; RTCWeb IETF <rtcweb@ietf.org>;
> draft-ietf-rtcweb-overview@ietf.org
> >> Subject: Re: [rtcweb] Eric Rescorla's Discuss on
> draft-ietf-rtcweb-overview-18: (with DISCUSS and COMMENT)
> >>
> >>
> >> Let me try and answer with the caveat that I may get this wrong and
> need to be corrected by my co-chairs...
> >>
> >> TL;DR - The chairs recommend changing the ref in overview to point at
> 5245 instead of 5245-bis
> >>
> >>
> >> here is the longer version....
> >>
> >> First a side note on how we got here. Some of the reasons we set up
> dependencies  like they are is that many years ago we made guess about wh=
at
> order work would get completed on and, shockingly, some predictions of
> standards developments timelines were less than perfect. We also assumed
> that some new work was going to require changes in ICE but as that work
> went progressed, we largely figured out ways to make it work with existin=
g
> ICE implementations.
> >>
> >> We are confident that overview does not actually depend on anything in
> 5245-bis but instead just depends on 5245.
> >>
> >> Next lets discuss trickle-ice. The WebRTC set of specs currently
> normatively depends on trickle ICE. There is some questions if trickle IC=
E
> might depend on 5245-bis. Some of the trickle ICE  authors do not think i=
t
> does. One of the authors said the chairs asked them to ref 5245 instead o=
f
> 5245bis as both tickle ICE and 5245bis would be done around the same time=
.
> In general, the webrtc chairs would prefer to make the WebRTC dependency
> cluster as small as possible. If trickle ice actually gets done before
> 5245bis, and it does not depend on any 5245bis features, then clearly is
> should be changed to just depend on 5245. The WG responsible for 5245bis
> and trickle ICE can figure out what they want to do as both theses drafts
> progress. Given there is a strong possibility that trickle ice will only
> reference 5245, we think it would be better if overview did not bring
> 5245bis into the WebRTC dependency cluster. If on the other hand, trickle
> ICE does end up depending on 5245bis, there is no harm, and no need to
> change overview to point at 5245.
> >>
> >> There are other drafts that are normative dependencies of JSEP and the
> WebRTC cluster that also point at 5245bis. When we consider the technical
> things these drafts need, it seems likely they can also reference 5245
> instead of 5245bis. For example draft-ietf-ice-dualstack-fairness,
> draft-ietf-mmusic-ice-sip-sdp, draft-ietf-mmusic-sdp-bundle-negotiation,
> and draft-ietf-rtcweb-transports (which is with the RFC Editor). The
> argument for the others is roughly the same as it is with trickle ICE.
> >>
> >> Cullen (without review from my co-chairs but trying to represent what
> we discussed on our chair call)
> >>
> >>
> >> PS - if you are trying to figure out some of the dependencies for the
> WebRTC cluster, you might find https://datatracker.ietf.org/
> doc/draft-jennings-rtcweb-deps/?include_text=3D1 useful but it is not 100=
%
> accurate.
> >>
> >>
> >>
> >>> On May 11, 2017, at 7:43 AM, Eric Rescorla <ekr@rtfm.com> wrote:
> >>>
> >>> Question for the chairs.
> >>>
> >>> On Thu, May 11, 2017 at 12:44 AM, Christer Holmberg <
> christer.holmberg@ericsson.com> wrote:
> >>> Hi,
> >>>
> >>> What is the status regarding this?
> >>>
> >>> Regards,
> >>>
> >>> Christer
> >>>
> >>>> On 26/04/17 06:02, "Sean Turner" <sean@sn3rd.com> wrote:
> >>>>
> >>>>
> >>>>> On Apr 23, 2017, at 14:44, Christer Holmberg
> >>>>> <christer.holmberg@ericsson.com> wrote:
> >>>>>
> >>>>> Hi,
> >>>>>
> >>>>>> ------------------------------------------------------------------
> >>>>>> ----
> >>>>>> DISCUSS:
> >>>>>> ------------------------------------------------------------------
> >>>>>> ----
> >>>>>>
> >>>>>> Your citation to ICE is to 5245-bis, but at least the JSEP editor
> >>>>>> consensus was that WebRTC depended on 5245, so this needs to be
> >>>>>> resolved one way or the other.
> >>>>>
> >>>>> Keep in mind that, no matter what draft-rtcweb-overview and
> >>>>> draft-rtcweb-jsep explicitly say, both specs reference 5245bis
> >>>>> *IMPLICITLY*, e.g., via draft-mmusic-bundle, draft-ice-trickle
> >>>>> etc... As I have indicated in the past, it would cause confusion to
> reference both.
> >>>>>
> >>>>> So, I think we shall reference 5245-bis everywhere (I also thought
> >>>>> we already decided no that in the past)-
> >>>>>
> >>>>> Regards,
> >>>>>
> >>>>> Christer
> >>>>
> >>>> /* bike shed alert:
> >>>> /*
> >>>> /* Assuming you=C2=B9re of the mind that a bis/updates draft is
> >>>> /* signaling to all implementors of the original RFC that the
> >>>> /* intention is that all implementations be updated then it=C2=B9s
> >>>> /* a bit more than implicit.
> >>>>
> >>>> spt
> >>>>
> >>>
> >>>
> >>
> >
>
>

--94eb2c08d3564dd71f054f6b9b97
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 Sat, May 13, 2017 at 10:57 AM, Christer Holmberg <span dir=3D"ltr">&=
lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">chri=
ster.holmberg@ericsson.com</a>&gt;</span> wrote:<br><blockquote class=3D"gm=
ail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-le=
ft:1ex"><span class=3D"">Hi,<br>
<br>
&gt;&gt;So, if the changes done in 5245bis aren&#39;t needed, why are we wo=
rking on 5245bis to begin with??? We did we spend all that time on modifyin=
g Ta etc???<br>
&gt;<br>
&gt;I don&#39;t think anyone is saying that 5245 bis isn&#39;t a good thing=
, merely that one can implement JSEP and trickle without reading it.<br>
<br>
</span>This isn&#39;t about JSEP and trickle, it&#39;s about rtcweb-overvie=
w- which is supposed to give some overview of what people need to implement=
.<br></blockquote><div><br></div><div>Yes, and rtcweb-overview references 5=
245-bis:</div><div><br></div><div>1. Directly</div><div>2. Transitively thr=
ough other documents, which are primarily JSEP and trickle.</div><div><br><=
/div><div>And my point is that to implement those other documents, you need=
 not implement</div><div>5245-bis.</div><div><br></div><div><br></div><bloc=
kquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #cc=
c solid;padding-left:1ex">
So, *IF* we decide to reference RFC 5245 in rtcweb-overview, I think that w=
e at least should indicate that some dependencies might require 5245bis, an=
d refer to individual dependencies for details. That way we can progress rt=
cweb-overview, but still keep the door open for 5245bis where/if it&#39;s n=
eeded.<br></blockquote><div><br></div><div>That just seems like kicking the=
 can down the road. To the best of my knowledge,</div><div>there is no tech=
nical reason why an rtcweb implementation needs to know about</div><div>524=
5-bis (whatever the structure of the normative references in the documents<=
/div><div>happens to be). Are you aware of such a reason?</div><div><br></d=
iv><div>-Ekr</div><div><br></div><blockquote class=3D"gmail_quote" style=3D=
"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div class=3D"HOEnZb"><div class=3D"h5"><br>
Regards,<br>
<br>
Christer<br>
<br>
=C2=A0<br>
<br>
<br>
Sent from my iPhone<br>
<br>
&gt; On 13 May 2017, at 16.42, Cullen Jennings &lt;<a href=3D"mailto:fluffy=
@iii.ca">fluffy@iii.ca</a>&gt; wrote:<br>
&gt;<br>
&gt;<br>
&gt; OK, so let me try to give some really clear guidance here ... if you a=
re writing a draft that does not need to normatively depend on another draf=
t, it should not normatively depended on that draft.<br>
&gt;<br>
&gt; I really doubt anyone is going to argue with that so lets make it so.<=
br>
&gt;<br>
&gt;&gt; On May 13, 2017, at 1:54 AM, Christer Holmberg &lt;<a href=3D"mail=
to:christer.holmberg@ericsson.com">christer.holmberg@ericsson.<wbr>com</a>&=
gt; wrote:<br>
&gt;&gt;<br>
&gt;&gt; Hi,<br>
&gt;&gt;<br>
&gt;&gt; As co-chair of a number of the dependencies, we have been discussi=
ng what to reference a number of times, and we have also changed the refere=
nces. We can&#39;t keep changing back and forth.<br>
&gt;&gt;<br>
&gt;&gt; In addition, I don&#39;t think WG X can decide what the drafts of =
WG Y reference. There needs to be a collaborative decision.<br>
&gt;&gt;<br>
&gt;&gt; &quot;We also assumed that some new work was going to require chan=
ges in ICE but as that work went progressed, we largely figured out ways to=
 make it work with existing ICE implementations.&quot;<br>
&gt;&gt;<br>
&gt;&gt; Is this &quot;discovery&quot; documented somewhere?<br>
&gt;&gt;<br>
&gt;&gt; &quot;If trickle ice actually gets done before 5245bis, and it doe=
s not depend on any 5245bis features, then clearly is should be changed to =
just depend on 5245.&quot;<br>
&gt;&gt;<br>
&gt;&gt; First, we need to agree on whether trickle depends on 5245bis feat=
ures or not.<br>
&gt;&gt;<br>
&gt;&gt; Second, as co-author of 5245bis, I have asked the chairs to initia=
te the road towards WGLC, so I would hope both 5245bis and trickle-ice coul=
d be done more or less at the same time.<br>
&gt;&gt;<br>
&gt;&gt; Regards,<br>
&gt;&gt;<br>
&gt;&gt; Christer<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; -----Original Message-----<br>
&gt;&gt; From: Cullen Jennings [mailto:<a href=3D"mailto:fluffy@iii.ca">flu=
ffy@iii.ca</a>]<br>
&gt;&gt; Sent: 13 May 2017 01:55<br>
&gt;&gt; To: Eric Rescorla &lt;<a href=3D"mailto:ekr@rtfm.com">ekr@rtfm.com=
</a>&gt;; Christer Holmberg &lt;<a href=3D"mailto:christer.holmberg@ericsso=
n.com">christer.holmberg@ericsson.<wbr>com</a>&gt;<br>
&gt;&gt; Cc: Sean Turner &lt;<a href=3D"mailto:sean@sn3rd.com">sean@sn3rd.c=
om</a>&gt;; The IESG &lt;<a href=3D"mailto:iesg@ietf.org">iesg@ietf.org</a>=
&gt;; <a href=3D"mailto:rtcweb-chairs@ietf.org">rtcweb-chairs@ietf.org</a>;=
 RTCWeb IETF &lt;<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>&gt;=
; <a href=3D"mailto:draft-ietf-rtcweb-overview@ietf.org">draft-ietf-rtcweb-=
overview@<wbr>ietf.org</a><br>
&gt;&gt; Subject: Re: [rtcweb] Eric Rescorla&#39;s Discuss on draft-ietf-rt=
cweb-overview-18: (with DISCUSS and COMMENT)<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; Let me try and answer with the caveat that I may get this wrong an=
d need to be corrected by my co-chairs...<br>
&gt;&gt;<br>
&gt;&gt; TL;DR - The chairs recommend changing the ref in overview to point=
 at 5245 instead of 5245-bis<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; here is the longer version....<br>
&gt;&gt;<br>
&gt;&gt; First a side note on how we got here. Some of the reasons we set u=
p dependencies=C2=A0 like they are is that many years ago we made guess abo=
ut what order work would get completed on and, shockingly, some predictions=
 of standards developments timelines were less than perfect. We also assume=
d that some new work was going to require changes in ICE but as that work w=
ent progressed, we largely figured out ways to make it work with existing I=
CE implementations.<br>
&gt;&gt;<br>
&gt;&gt; We are confident that overview does not actually depend on anythin=
g in 5245-bis but instead just depends on 5245.<br>
&gt;&gt;<br>
&gt;&gt; Next lets discuss trickle-ice. The WebRTC set of specs currently n=
ormatively depends on trickle ICE. There is some questions if trickle ICE m=
ight depend on 5245-bis. Some of the trickle ICE=C2=A0 authors do not think=
 it does. One of the authors said the chairs asked them to ref 5245 instead=
 of 5245bis as both tickle ICE and 5245bis would be done around the same ti=
me. In general, the webrtc chairs would prefer to make the WebRTC dependenc=
y cluster as small as possible. If trickle ice actually gets done before 52=
45bis, and it does not depend on any 5245bis features, then clearly is shou=
ld be changed to just depend on 5245. The WG responsible for 5245bis and tr=
ickle ICE can figure out what they want to do as both theses drafts progres=
s. Given there is a strong possibility that trickle ice will only reference=
 5245, we think it would be better if overview did not bring 5245bis into t=
he WebRTC dependency cluster. If on the other hand, trickle ICE does end up=
 depending on 5245bis, there is no harm, and no need to change overview to =
point at 5245.<br>
&gt;&gt;<br>
&gt;&gt; There are other drafts that are normative dependencies of JSEP and=
 the WebRTC cluster that also point at 5245bis. When we consider the techni=
cal things these drafts need, it seems likely they can also reference 5245 =
instead of 5245bis. For example draft-ietf-ice-dualstack-<wbr>fairness, dra=
ft-ietf-mmusic-ice-sip-sdp, draft-ietf-mmusic-sdp-bundle-<wbr>negotiation, =
and draft-ietf-rtcweb-transports (which is with the RFC Editor). The argume=
nt for the others is roughly the same as it is with trickle ICE.<br>
&gt;&gt;<br>
&gt;&gt; Cullen (without review from my co-chairs but trying to represent w=
hat we discussed on our chair call)<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; PS - if you are trying to figure out some of the dependencies for =
the WebRTC cluster, you might find <a href=3D"https://datatracker.ietf.org/=
doc/draft-jennings-rtcweb-deps/?include_text=3D1" rel=3D"noreferrer" target=
=3D"_blank">https://datatracker.ietf.org/<wbr>doc/draft-jennings-rtcweb-<wb=
r>deps/?include_text=3D1</a> useful but it is not 100% accurate.<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;&gt; On May 11, 2017, at 7:43 AM, Eric Rescorla &lt;<a href=3D"mail=
to:ekr@rtfm.com">ekr@rtfm.com</a>&gt; wrote:<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Question for the chairs.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; On Thu, May 11, 2017 at 12:44 AM, Christer Holmberg &lt;<a hre=
f=3D"mailto:christer.holmberg@ericsson.com">christer.holmberg@ericsson.<wbr=
>com</a>&gt; wrote:<br>
&gt;&gt;&gt; Hi,<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; What is the status regarding this?<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Regards,<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Christer<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; On 26/04/17 06:02, &quot;Sean Turner&quot; &lt;<a href=3D"=
mailto:sean@sn3rd.com">sean@sn3rd.com</a>&gt; wrote:<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt; On Apr 23, 2017, at 14:44, Christer Holmberg<br>
&gt;&gt;&gt;&gt;&gt; &lt;<a href=3D"mailto:christer.holmberg@ericsson.com">=
christer.holmberg@ericsson.<wbr>com</a>&gt; wrote:<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt; Hi,<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt; ------------------------------<wbr>---------------=
---------------<wbr>------<br>
&gt;&gt;&gt;&gt;&gt;&gt; ----<br>
&gt;&gt;&gt;&gt;&gt;&gt; DISCUSS:<br>
&gt;&gt;&gt;&gt;&gt;&gt; ------------------------------<wbr>---------------=
---------------<wbr>------<br>
&gt;&gt;&gt;&gt;&gt;&gt; ----<br>
&gt;&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt;&gt; Your citation to ICE is to 5245-bis, but at least =
the JSEP editor<br>
&gt;&gt;&gt;&gt;&gt;&gt; consensus was that WebRTC depended on 5245, so thi=
s needs to be<br>
&gt;&gt;&gt;&gt;&gt;&gt; resolved one way or the other.<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt; Keep in mind that, no matter what draft-rtcweb-overvie=
w and<br>
&gt;&gt;&gt;&gt;&gt; draft-rtcweb-jsep explicitly say, both specs reference=
 5245bis<br>
&gt;&gt;&gt;&gt;&gt; *IMPLICITLY*, e.g., via draft-mmusic-bundle, draft-ice=
-trickle<br>
&gt;&gt;&gt;&gt;&gt; etc... As I have indicated in the past, it would cause=
 confusion to reference both.<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt; So, I think we shall reference 5245-bis everywhere (I =
also thought<br>
&gt;&gt;&gt;&gt;&gt; we already decided no that in the past)-<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt; Regards,<br>
&gt;&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt;&gt; Christer<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; /* bike shed alert:<br>
&gt;&gt;&gt;&gt; /*<br>
&gt;&gt;&gt;&gt; /* Assuming you=C2=B9re of the mind that a bis/updates dra=
ft is<br>
&gt;&gt;&gt;&gt; /* signaling to all implementors of the original RFC that =
the<br>
&gt;&gt;&gt;&gt; /* intention is that all implementations be updated then i=
t=C2=B9s<br>
&gt;&gt;&gt;&gt; /* a bit more than implicit.<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; spt<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt;<br>
&gt;&gt;<br>
&gt;<br>
<br>
</div></div></blockquote></div><br></div></div>

--94eb2c08d3564dd71f054f6b9b97--


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Eric Rescorla <ekr@rtfm.com>
CC: Cullen Jennings <fluffy@iii.ca>, Sean Turner <sean@sn3rd.com>, The IESG <iesg@ietf.org>, "rtcweb-chairs@ietf.org" <rtcweb-chairs@ietf.org>, "RTCWeb IETF" <rtcweb@ietf.org>, "draft-ietf-rtcweb-overview@ietf.org" <draft-ietf-rtcweb-overview@ietf.org>
Thread-Topic: [rtcweb] Eric Rescorla's Discuss on draft-ietf-rtcweb-overview-18: (with DISCUSS and COMMENT)
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Subject: Re: [rtcweb] Eric Rescorla's Discuss on draft-ietf-rtcweb-overview-18: (with DISCUSS and COMMENT)
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From: Cullen Jennings <fluffy@iii.ca>
In-Reply-To: <7594FB04B1934943A5C02806D1A2204B4CBA0BF0@ESESSMB109.ericsson.se>
Date: Mon, 15 May 2017 06:48:40 -1000
Cc: Eric Rescorla <ekr@rtfm.com>, Sean Turner <sean@sn3rd.com>, The IESG <iesg@ietf.org>, "rtcweb-chairs@ietf.org" <rtcweb-chairs@ietf.org>, RTCWeb IETF <rtcweb@ietf.org>, "draft-ietf-rtcweb-overview@ietf.org" <draft-ietf-rtcweb-overview@ietf.org>
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Subject: Re: [rtcweb] Eric Rescorla's Discuss on draft-ietf-rtcweb-overview-18: (with DISCUSS and COMMENT)
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> On May 13, 2017, at 8:14 AM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:
>=20
>=20
> I am, however, aware that a number of dependencies do refer 5245bis. =
Maybe they don't need to, but my point is that I think all dependencies =
should reference the same ICE spec - RFC 5245 *OR* 5245bis. And I think =
RTCWEB, MMUSIC and ICE should agree on which spec. Since more or less =
the same people participate in all those groups, I don't think it should =
be a difficult thing to achieve.=20

I feel very strongly that unless one can show there is technical part of =
draft X you need to read to implement draft Y, then Y should not =
normatively reference X.=20

Perhaps there is something in ICEbis that is needed by one of the RTCweb =
specs and if so, we should discuss this but so far I'm not aware of any.=20=




--Apple-Mail=_6B1D4260-62AF-4B42-9143-3BA8C72DB2AC
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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dus-ascii"></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 May 13, 2017, at 8:14 AM, Christer Holmberg &lt;<a =
href=3D"mailto:christer.holmberg@ericsson.com" =
class=3D"">christer.holmberg@ericsson.com</a>&gt; wrote:</div><br =
class=3D"Apple-interchange-newline"><div class=3D""><br =
style=3D"font-family: Helvetica; font-size: 14px; font-style: normal; =
font-variant-caps: normal; font-weight: normal; letter-spacing: 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: 14px; font-style: normal; font-variant-caps: =
normal; font-weight: normal; letter-spacing: 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"">I am, however, =
aware that a number of dependencies do refer 5245bis. Maybe they don't =
need to, but my point is that I think all dependencies should reference =
the same ICE spec - RFC 5245 *OR* 5245bis. And I think RTCWEB, MMUSIC =
and ICE should agree on which spec. Since more or less the same people =
participate in all those groups, I don't think it should be a difficult =
thing to achieve.<span =
class=3D"Apple-converted-space">&nbsp;</span></span><br =
style=3D"font-family: Helvetica; font-size: 14px; font-style: normal; =
font-variant-caps: normal; font-weight: normal; letter-spacing: 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""></div></blockquote></div><br =
class=3D""><div class=3D"">I feel very strongly that unless one can show =
there is technical part of draft X you need to read to implement draft =
Y, then Y should not normatively reference X.&nbsp;</div><div =
class=3D""><br class=3D""></div><div class=3D"">Perhaps there is =
something in ICEbis that is needed by one of the RTCweb specs and if so, =
we should discuss this but so far I'm not aware of any.&nbsp;</div><div =
class=3D""><br class=3D""></div><div class=3D""><br =
class=3D""></div></body></html>=

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Cullen Jennings <fluffy@iii.ca>
CC: Eric Rescorla <ekr@rtfm.com>, Sean Turner <sean@sn3rd.com>, The IESG <iesg@ietf.org>, "rtcweb-chairs@ietf.org" <rtcweb-chairs@ietf.org>, "RTCWeb IETF" <rtcweb@ietf.org>, "draft-ietf-rtcweb-overview@ietf.org" <draft-ietf-rtcweb-overview@ietf.org>
Thread-Topic: [rtcweb] Eric Rescorla's Discuss on draft-ietf-rtcweb-overview-18: (with DISCUSS and COMMENT)
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Date: Mon, 15 May 2017 18:24:54 +0000
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Subject: Re: [rtcweb] Eric Rescorla's Discuss on draft-ietf-rtcweb-overview-18: (with DISCUSS and COMMENT)
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Hi,

So, we should check and confirm with the affected WGs whether 5245bis is re=
quired within the RTCWEB dependencies that currently reference 5245bis.

Regards,

Christer

From: Cullen Jennings [mailto:fluffy@iii.ca]
Sent: 15 May 2017 18:49
To: Christer Holmberg <christer.holmberg@ericsson.com>
Cc: Eric Rescorla <ekr@rtfm.com>; Sean Turner <sean@sn3rd.com>; The IESG <i=
esg@ietf.org>; rtcweb-chairs@ietf.org; RTCWeb IETF <rtcweb@ietf.org>; draft=
-ietf-rtcweb-overview@ietf.org
Subject: Re: [rtcweb] Eric Rescorla's Discuss on draft-ietf-rtcweb-overview=
-18: (with DISCUSS and COMMENT)


On May 13, 2017, at 8:14 AM, Christer Holmberg <christer.holmberg@ericsson.=
com<mailto:christer.holmberg@ericsson.com>> wrote:


I am, however, aware that a number of dependencies do refer 5245bis. Maybe =
they don't need to, but my point is that I think all dependencies should re=
ference the same ICE spec - RFC 5245 *OR* 5245bis. And I think RTCWEB, MMUS=
IC and ICE should agree on which spec. Since more or less the same people p=
articipate in all those groups, I don't think it should be a difficult thin=
g to achieve.

I feel very strongly that unless one can show there is technical part of dr=
aft X you need to read to implement draft Y, then Y should not normatively =
reference X.

Perhaps there is something in ICEbis that is needed by one of the RTCweb sp=
ecs and if so, we should discuss this but so far I'm not aware of any.



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<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,sans-serif;color:#1F497D;mso-fareast-language:EN-US">Hi,<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;mso-fareast-language:EN-US"><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;mso-fareast-language:EN-US">So, we sho=
uld check and confirm with the affected WGs whether 5245bis is required wit=
hin the RTCWEB dependencies that currently reference
 5245bis.<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;mso-fareast-language:EN-US"><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;mso-fareast-language:EN-US">Regards,<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;mso-fareast-language:EN-US"><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;mso-fareast-language:EN-US">Christer<o=
:p></o:p></span></p>
<p class=3D"MsoNormal"><a name=3D"_MailEndCompose"><span style=3D"font-size=
:11.0pt;font-family:&quot;Calibri&quot;,sans-serif;color:#1F497D;mso-fareas=
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0cm 0cm">
<p class=3D"MsoNormal"><b><span lang=3D"EN-US" style=3D"font-size:11.0pt;fo=
nt-family:&quot;Calibri&quot;,sans-serif">From:</span></b><span lang=3D"EN-=
US" style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,sans-serif"> =
Cullen Jennings [mailto:fluffy@iii.ca]
<br>
<b>Sent:</b> 15 May 2017 18:49<br>
<b>To:</b> Christer Holmberg &lt;christer.holmberg@ericsson.com&gt;<br>
<b>Cc:</b> Eric Rescorla &lt;ekr@rtfm.com&gt;; Sean Turner &lt;sean@sn3rd.c=
om&gt;; The IESG &lt;iesg@ietf.org&gt;; rtcweb-chairs@ietf.org; RTCWeb IETF=
 &lt;rtcweb@ietf.org&gt;; draft-ietf-rtcweb-overview@ietf.org<br>
<b>Subject:</b> Re: [rtcweb] Eric Rescorla's Discuss on draft-ietf-rtcweb-o=
verview-18: (with DISCUSS and COMMENT)<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<p class=3D"MsoNormal">On May 13, 2017, at 8:14 AM, Christer Holmberg &lt;<=
a href=3D"mailto:christer.holmberg@ericsson.com">christer.holmberg@ericsson=
.com</a>&gt; wrote:<o:p></o:p></p>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:&quot;He=
lvetica&quot;,sans-serif"><br>
I am, however, aware that a number of dependencies do refer 5245bis. Maybe =
they don't need to, but my point is that I think all dependencies should re=
ference the same ICE spec - RFC 5245 *OR* 5245bis. And I think RTCWEB, MMUS=
IC and ICE should agree on which
 spec. Since more or less the same people participate in all those groups, =
I don't think it should be a difficult thing to achieve.<span class=3D"appl=
e-converted-space">&nbsp;</span></span><o:p></o:p></p>
</div>
</blockquote>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">I feel very strongly that unless one can show there =
is technical part of draft X you need to read to implement draft Y, then Y =
should not normatively reference X.&nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Perhaps there is something in ICEbis that is needed =
by one of the RTCweb specs and if so, we should discuss this but so far I'm=
 not aware of any.&nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
</body>
</html>

--_000_7594FB04B1934943A5C02806D1A2204B4CBA36E6ESESSMB109erics_--


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From: Sean Turner <sean@sn3rd.com>
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Subject: Re: [rtcweb] Ben Campbell's Yes on draft-ietf-rtcweb-overview-18: (with COMMENT)
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> On Apr 24, 2017, at 22:37, Ben Campbell <ben@nostrum.com> wrote:
>=20
>=20
> ----------------------------------------------------------------------
> COMMENT:
> ----------------------------------------------------------------------
>=20
> I am balloting "yes", but I have a few minor comments:
>=20
> Substantive Comments:
>=20
> -2.2 :=20
> -- Why is a WebRTC gateway assumed to be a "compatible" endpoint =
rather
> than a full endpoint? I recognize a gateway is different from a =
typical
> end-user endpoint, but are there specific endpoint requirements that a
> gateway is not likely to meet? (Feel free to say "it's documented in =
the
> gateway draft...." :-) )

It=E2=80=99s documented in the (cough) now expired gateways draft:
https://www.ietf.org/archive/id/draft-ietf-rtcweb-gateways-02.txt
Ted=E2=80=99s pinged Harald to revive this one.

> -- "In this case, similar security considerations as for Javascript =
may
> be needed; however, since such APIs are not defined or referenced =
here,
> this document cannot give any specific rules for those interfaces."
> I am confused by this sentence, since I don't see any security
> considerations specific to Javascript in this draft, either.

RTCweb Javascript-related security considerations are addressed in the =
(cough) now expired security drafts:
https://www.ietf.org/archive/id/draft-ietf-rtcweb-security-arch-12.txt
https://www.ietf.org/archive/id/draft-ietf-rtcweb-security-08.txt
The Security Considerations points to draft-ietf-rtcweb-security, which =
will get you to the other draft as well.

> -7
> -- list item 2: Is it an open question whether a signaling gateway is
> needed for interacting with SIP devices?

I do not believe that item #2 (copied below) is an open question, it=E2=80=
=99s just pointing out that if you are going try to interoperate with =
legacy SIP devices you=E2=80=99re going to have to do more work.

  2.  It will be possible to gateway between legacy SIP devices that
      support ICE and appropriate RTP / SDP mechanisms, codecs and
      security mechanisms without using a media gateway.  A signaling
      gateway to convert between the signaling on the web side to the
      SIP signaling may be needed.

> -- Last paragraph: This is specifically about non-browser endpoints,
> right? As written, it seems to weaken the previous paragraph about
> browser endpoints, since the draft previously said the term "endpoint"
> includes both browsers and non-browsers.

Nope, it=E2=80=99s about all endpoints.  Look at it this way the =
penultimate paragraph says WebRTC browser implement the API and all =
WebRTC endpoints implement the protocols.  That=E2=80=99s in-line with =
the definitions defined earlier in s2.2


> -9, 2nd bullet: "Privacy concerns MUST be satisfied..."
> Is that MAY really intended as normative, or is a statement of fact? =
If
> normative, what actor(s) does it constrain? Also, if it is normative, =
the
> clause "the APIs should be available" seems to weaken the MUST.=20

This is supposed to be normative.  It=E2=80=99s constraining the =
=E2=80=9Clocal system=E2=80=9D :). I=E2=80=99m not sure it weakens it =
because it=E2=80=99s a should not a SHOULD.  But, since this draft also =
addresses endpoints that don=E2=80=99t have to have an API, we can=E2=80=99=
t really make it a SHOULD.

> Editorial Comments:
>=20
> -2.3, last paragraph: The paragraph is a single, convoluted sentence =
that
> is hard to parse. (It's also a comma splice). Please consider breaking
> into multiple simpler sentences.

https://github.com/rtcweb-wg/rtcweb-overview/pull/19

> -3, first paragraph: This is also convoluted and hard to parse.

https://github.com/rtcweb-wg/rtcweb-overview/pull/21

> -7, list item 1: The citiation to [3264] seems misplaced. It describes
> the offer/answer model, not SIP in general. I suggest moving the =
citation
> to after the word "semantics=E2=80=9D.

https://github.com/rtcweb-wg/rtcweb-overview/pull/35/files

spt


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> On Apr 26, 2017, at 15:22, Spencer Dawkins =
<spencerdawkins.ietf@gmail.com> wrote:
>=20
> ----------------------------------------------------------------------
> COMMENT:
> ----------------------------------------------------------------------
>=20
> I've been waiting for this one, for a while. Thanks for finishing it. =
I'm
> a Yes, with comments.
>=20
> I agree with EKR that there's a lot of general philosophy in this =
draft.
> I wouldn't ask that you pull all of it, but perhaps it could be =
trimmed
> down a bit.

The chairs discussed this and we=E2=80=99re going to not make a change =
here primarily because I fear we might go to far the other way and then =
we=E2=80=99re playing bring me rock.

> This is a nit, but in this text,
>=20
>   Other efforts, for instance the W3C WEBRTC, Web Applications and
>   Device API working groups, focus on making standardized APIs and
>   interfaces available, within or alongside the HTML5 effort,
>=20
> it would be nice to have the names here match what's on the W3C =
website.
> So, "Web Real-Time Communication", "Web Application Security", and
> "Device and Sensors", unless I'm guessing at the mapping wrong. It's =
also
> easy to read that text with "Web Applications and Device API" as a =
single
> working group, so using a comma after "Web Application Security" would =
be
> helpful.

I pulled the named from here:
https://www.w3.org/TR/tr-groups-all

The PR can be found here:
https://github.com/rtcweb-wg/rtcweb-overview/pull/23

> The term of art "floor control" is likely to be new to many readers in
> the future. Since it appears in a list of non-niche examples, maybe =
you
> don't need it at all?

I=E2=80=99m hoping people know how to google :) The wikipedia definition =
that popped up as the 1st result pretty much covers it.

> I'm not sure whether "let a thousand flowers bloom" is a reference to =
the
> Hundred Flowers campaign in 1956, but (1) that ended very badly for =
the
> bloomers, and (2) I could easily imagine the phrase tripping DPI
> filtering for a specific part of the Internet community. Maybe there's =
a
> better phrase?

There might be but that the phrase I=E2=80=99ve often heard to refer to =
=E2=80=9Clet the market sort it out".

> I'm not sure how tutorial you want section 4 to be, but I'd at least
> mention appropriate retransmission and in-order delivery, in addition =
to
> congestion control, since you get that with SCTP on the data channel.
>=20
> 4.  Data transport
>=20
>   Data transport refers to the sending and receiving of data over the
>   network interfaces, the choice of network-layer addresses at each
> end
>   of the communication, and the interaction with any intermediate
>   entities that handle the data, but do not modify it (such as TURN
>   relays).
>=20
>   It includes necessary functions for congestion control: When not to
>   send data.
>=20
> Or maybe you can just chop that sentence, because the next paragraph
> points to https://tools.ietf.org/html/draft-ietf-rtcweb-transports-06,
> anyway?

I just listed the additional SCTP functions:
https://github.com/rtcweb-wg/rtcweb-overview/pull/25/files

> I found the reference to MMUSIC WG in
>=20
>   3.  When a new codec is specified, and the SDP for the new codec is
>       specified in the MMUSIC WG, no other standardization should be
>       required for it to be possible to use that in the web browsers.
>=20
> to be odd. MMUSIC may be around forever, but this work might be
> refactored at some point in the future. Is the point that=20
>=20
>   3.  When SDP for a new codec is specified,=20
>       no other standardization should be
>       required for it to be used in the web browsers.
>=20
> Or is there another way to say this?

Fair enough, we can future proof the document with this change:
https://github.com/rtcweb-wg/rtcweb-overview/pull/27

> I'm also wondering if the statement is true for any WebRTC endpoint, =
not
> just browsers.

I=E2=80=99m not sure, but the main point of that entire bullet is that =
there=E2=80=99s no changes necessary to the API an not all endpoints =
need to implement the API on the browser does.

> In this text,
>=20
>   WebRTC endpoints MUST implement the functions described in that
>   document that relate to the network layer (for example Bundle
>   [I-D.ietf-mmusic-sdp-bundle-negotiation], RTCP-mux [RFC5761] and
>   Trickle ICE [I-D.ietf-ice-trickle]), but do not need to support the
>   API functionality described there.
>=20
> I would have thought these were related to the transport layer. No?

Maybe, but since these drafts/RFCs are about setting up the connection =
they do seem to fit in the =E2=80=9CConnection Management=E2=80=9D =
section ;)

spt


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From: Sean Turner <sean@sn3rd.com>
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Date: Tue, 16 May 2017 12:36:42 -0400
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To: =?utf-8?Q?Mirja_K=C3=BChlewind?= <ietf@kuehlewind.net>
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> On Apr 24, 2017, at 08:07, Mirja K=C3=BChlewind <ietf@kuehlewind.net> =
wrote:
>=20
>=20
> ----------------------------------------------------------------------
> COMMENT:
> ----------------------------------------------------------------------
>=20
> One high level comments on normative language:
> While I think this document is very useful to explain the relationship
> between the other webrtc documents and serves a a good starting point =
for
> an implementor, I'm not sure if the use of normative language is =
actually
> helpful. Most of the language is used to say that a webrtc endpoint =
MUST
> implement a certain other document. However, I believe this is =
inherently
> necessary to achieve interoperability. So I don't see a need to =
specify
> this normatively.

I can see that but I also think that as a =E2=80=9Ctreasure map=E2=80=9D =
document pointing normatively could work, i.e., it=E2=80=99s a style =
thing.

> In regard to the shepherd write-up, I just want to note that using
> normative language does not automatically make the document Standards
> Track; there are many informational docs that use normative language. =
As
> such, I don't want raise a big discussion on status now, but this
> document sounds more informational to me (giving pointers to other
> document). However, I don't object to publication on Standards Track.

Addressed via Ben=E2=80=99s response.

> minor comments:
> 1) I would not need all the text on the history of Internet =
communication
> in this doc (especially all text on page 3 in the intro as well as
> section 2.3 and the second to last paragraph in 3)... however, I guess =
it
> doesn't hurt

It is wordy, but I think Harald=E2=80=99s just providing background for =
folks who might note be so in the know.

> 2) Agree with Warren that 'Protocol' probably doesn't need to be
> (re)defined in this doc

I waffled around with deleting, but since this is a comment I=E2=80=99m =
leaning towards just leaving it based on the principle that it doesn=E2=80=
=99t really do any harm leaving it in.

> 3) section 3:=20
> "Data transport: TCP, UDP and the means to securely set up
>      connections between entities, as well as the functions for
>      deciding when to send data: Congestion management, bandwidth
>      estimation and so on."
> This seems to implicitly assume that only TCP or something =
encapsulated
> over UDP can be used. Even though that might be true, I assume this =
was
> not intentionally, maybe:
> NEW
> "Data transport: such as TCP or UDP and the means to securely set up
>      connections between entities, as well as the functions for
>      deciding when to send data: Congestion management, bandwidth
>      estimation and so on.=E2=80=9D

Fixed via PR:
https://github.com/rtcweb-wg/rtcweb-overview/pull/29

> nit:
> -"massage the signals": not sure if "massage" is actually a meaningful
> word here=E2=80=A6

manipulate might be a better word.  Fixed via PR:
https://github.com/rtcweb-wg/rtcweb-overview/pull/31

spt=


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> On Apr 27, 2017, at 10:24, Benoit Claise <bclaise@cisco.com> wrote:
>=20
>=20
> ----------------------------------------------------------------------
> COMMENT:
> ----------------------------------------------------------------------
>=20
> This topic below was discussed during the IESG telechat:
>=20
> Reading from the document objectives, from the abstract:
>=20
>   This document gives an overview and context of a protocol suite
>   intended for use with real-time applications that can be deployed in
>   browsers - "real time communication on the Web".
>=20
>   It intends to serve as a starting and coordination point to make
> sure
>   all the parts that are needed to achieve this goal are findable, and
>   that the parts that belong in the Internet protocol suite are fully
>   specified and on the right publication track.
>=20
> Reading this, I was thinking: great, I will have the full overview.
> With "deployed", "starting and coordination point to make sure that =
all
> the parts ..." I will have some  focus on the operational aspects,
> basically, how should operators operate theses browser-embedded
> applications.
> Now, reading further ...
>=20
>   This document is intended to serve as the roadmap to the WebRTC
>   specifications.  It defines terms used by other parts of the WebRTC
>   protocol specifications, lists references to other specifications
>   that don't need further elaboration in the WebRTC context, and gives
>   pointers to other documents that form part of the WebRTC suite.
>=20
> ... I thought: Ok, if not covered here, at least I will have a pointer =
to
> another operational document.
> But wait:
>=20
>   By reading this document and the documents it refers to, it should
> be
>   possible to have all information needed to implement an WebRTC
>   compatible implementation.
>=20
> So is this only about implementation?
>=20
> I like this document very much as it explains all the RTCWEB pieces in
> one location. However, there is one important piece missing: the =
network
> management considerations. See
> https://datatracker.ietf.org/doc/html/rfc5706#appendix-A
> This is where I'm coming from, discussing some more with Warren (this =
a
> cut and past from this ballot):
>=20
>    [ Edit: So, after more thought (and some discussion) I think that =
it
> would be useful for the document to at least note the fact that
> technologies like this mean that some of the existing operational
> practices may need to change. For example, many enterprises perform =
QoS
> based upon the fact that certain types of devices live in certain =
subnets
> (e.g many phones get placed in a specific VLAN using LLDP or CDP). =
With
> more real time content coming from browsers, these matching practices
> break, and so operators may not be able to QoS mark / prioritize =
traffic
> accordingly. Perhaps something like: "One of the implications of a
> solution like WebRTC is that more real-time traffic will be sourced =
from
> computers (and not dedicated devices like telephones or =
videoconferencing
> devices). This may have implications for operators performing QoS =
marking
> and prioritization" ? This isn't really specific to webrtc, but rather =
to
> a more general set of solutions like softphones and the like, but is
> accelerated by WebRTC. ]
>=20
> In light of the previous discussions about =
draft-mm-wg-effect-encrypt-11,
> the operators are used to manage voice, video, gaming a certain way, =
with
> their operational current practices. Now, their current practices =
might
> not work any longer. What should they do now in term of monitoring,
> troubleshooting, QoS, SLA monitoring, etc these days with WebRTC?
> While we should add this note (or a similar one) in the doc, I'm
> wondering: where are (should be) those operational aspects discussed, =
if
> not here?
> I've seen https://tools.ietf.org/html/draft-ietf-tsvwg-rtcweb-qos-18, =
not
> sure it's appropriate. Anyway, it's now in a RFC-editor state.
> I could have requested a specific manageability doc in the charter. =
Too
> late now.

To address Benoit/Warren/Kathleen=E2=80=99s comments on this, I=E2=80=99ve=
 submitted the following PR:
https://github.com/rtcweb-wg/rtcweb-overview/pull/15

spt=


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> On Apr 25, 2017, at 15:01, Kathleen Moriarty =
<kathleen.moriarty.ietf@gmail.com> wrote:
>=20
>=20
> ----------------------------------------------------------------------
> COMMENT:
> ----------------------------------------------------------------------
>=20
> Thanks for your work on this draft, it's a helpful overview.  I see =
the
> reasoning in the shepherd report for informational, but am curious if =
the
> standards track status is needed for other SDOs that might reference =
this
> document? =20

Addressed via discussions with Ben/Adam.

> In reading sections 8 & 9, I would think the presentation and control =
in
> section 8 would have the privacy implications of the second bullet in
> section 9.  As such, it seems odd that normative language is used in =
this
> bullet and not in section 8.  I'd be fine with no normative language =
in
> either as long as the protocol drafts cover that appropriately.  Some
> mention of privacy in section 8 could be helpful since it covers more
> ground than the example in section 9.

I=E2=80=99m looking at it this way: s8 is really also about local system =
support functions too.  We could reorder s8 and s9, but I=E2=80=99m =
hoping that thinking about s8 as a local support functions is =
sufficient.

> Security considerations: I don't see anything listed for security or
> privacy considerations in respect to the signaling channel to the
> web/application server.  Should there be considerations listed?  =
Security
> of the actual server and content on the server as well as =
vulnerabilities
> in listening protocols are just a few of the questions that come to =
mind.
> If it doesn't matter, please let me know.  I appreciate the comment on
> the browser being target rich as they have been in many attacks to =
gain
> entry into networks leveraging established outbound sessions.  Maybe =
this
> is covered in I-D.ietf-rtcweb-security and if so (have not had a =
chance
> to review it yet), a high-level mention of gateway security here might =
be
> helpful.

The security considerations and privacy considerations are addressed the =
two (cough) now expired security drafts:
(cough) now expired security drafts:
https://www.ietf.org/archive/id/draft-ietf-rtcweb-security-arch-12.txt
https://www.ietf.org/archive/id/draft-ietf-rtcweb-security-08.txt
The Security Considerations points to draft-ietf-rtcweb-security, which =
will get you to the other draft. Instead of recreating all of that here, =
I=E2=80=99m basically asking for your patience until these drafts make =
it to the IESG in the next couple of months.  There=E2=80=99s a huge, =
and I mean Donald Trump =E2=80=9Cyugeee=E2=80=9D, cluster of drafts that =
can=E2=80=99t go anywhere until those the security drafts get through =
the process so I hoping until then you=E2=80=99ll be satisfied =
sharpening your weapons of mass DISCUSSion until then ;)

> I agree with Warren's comment about the management aspects being =
covered
> here since it is an overview document.  It could be a very helpful
> consideration for protocol developers that may devise new ways to =
enable
> management as a result of understanding the issues.

Addressed via PR:
https://github.com/rtcweb-wg/rtcweb-overview/pull/15

> I had to look up jingle and BOSH, you may want to consider adding
> references to the XMPP specifications.

Hmm there are references to XMPP and Jingle in the draft.  BOSH, I had =
to look up too ;). Added a reference via the following PR:
https://github.com/rtcweb-wg/rtcweb-overview/pull/33

spt


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Subject: Re: [rtcweb] Warren Kumari's No Objection on draft-ietf-rtcweb-overview-18: (with COMMENT)
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> On Apr 25, 2017, at 11:11, Warren Kumari <warren@kumari.net> wrote:
>=20
>=20
> ----------------------------------------------------------------------
> COMMENT:
> ----------------------------------------------------------------------
>=20
> Thank you -- I like these sort of overview documents for complex =
things
> like WebRTC - they provide a newcomer to the technology a good place =
to
> start, and help describe some of the reasons why things look the way =
they
> do.
>=20
> [ Edit: So, after more thought (and some discussion) I think that it
> would be useful for the document to at least note the fact that
> technologies like this mean that some of the existing operational
> practices may need to change. For example, many enterprises perform =
QoS
> based upon the fact that certain types of devices live in certain =
subnets
> (e.g many phones get placed in a specific VLAN using LLDP or CDP). =
With
> more real time content coming from browsers, these matching practices
> break, and so operators may not be able to QoS mark / prioritize =
traffic
> accordingly. Perhaps something like: "One of the implications of a
> solution like WebRTC is that more real-time traffic will be sourced =
from
> computers (and not dedicated devices like telephones or=20
> videoconferencing devices). This may have implications for operators
> performing QoS marking and prioritization" ? This isn't really =
specific
> to webrtc, but rather to a more general set of solutions like =
softphones
> and the like, but is accelerated by WebRTC. ]=20

To address Benoit, Kathleen, and Warren=E2=80=99s concerns here=E2=80=99s =
a PR:
https://github.com/rtcweb-wg/rtcweb-overview/pull/15

> I do have a few comments on the document itself - there are all minor =
/
> bikeshedding and can be ignored if you choose:
> 1: "Development of The Universal Solution has proved hard, however, =
for
> all the usual reasons."
> -- this is cute, but leaves people wondering what "all the usual =
reasons
> are". Perhaps just "Development of The Universal Solution has, =
however,=20
> proved hard." (or just cut after the "however in the original").
>=20
> 2: I'm not sure why you have "Protocol" in the terminology section. It
> doesn't seem like it is useful for the document, and this document
> doesn't seem like the right place to (re) define it.
>=20
> 3: Acknowledgements:=20
> Funny spacing in "Olle E.     Johansson=E2=80=9D

To address #1 and #3, here=E2=80=99s a PR:
https://github.com/rtcweb-wg/rtcweb-overview/pull/17/files

Note I=E2=80=99m going to leave protocol in because I don=E2=80=99t =
think it really hurts to leave it in.

spt=


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> On Apr 22, 2017, at 07:16, Eric Rescorla <ekr@rtfm.com> wrote:
>=20
>=20
> ----------------------------------------------------------------------
> DISCUSS:
> ----------------------------------------------------------------------
>=20
> Your citation to ICE is to 5245-bis, but at least the JSEP editor
> consensus was that WebRTC depended on 5245, so this needs to be =
resolved
> one way or the other.

The chairs believe that the overview draft should refer to RFC5245.  =
There=E2=80=99s nothing in overview that requires a reference to =
5245-bis (aka ICE-bis); it=E2=80=99s a definition for an "ICE Agent=E2=80=9D=
.  And, let=E2=80=99s face it overview is going directly into cluster =
238.  So instead of arguing about whether to refer ICE or ICE-bis now =
when overview is going to get stuck anyway let=E2=80=99s let the process =
run it=E2=80=99s course: if trickle-ICE is going to get done about the =
same time as ICE-bis then great we can update the references in AUTH48 =
and if not we can figure out plan B.

I=E2=80=99ve submitted the following PR to change the reference to 5245:

https://github.com/rtcweb-wg/rtcweb-overview/pull/37

> ----------------------------------------------------------------------
> COMMENT:
> ----------------------------------------------------------------------


I haven=E2=80=99t yet completed the editorial changes as a result of =
ekr's comments so I=E2=80=99ll address that in another email.

spt=


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Subject: Re: [rtcweb] Kathleen Moriarty's No Objection on draft-ietf-rtcweb-overview-18: (with COMMENT)
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Hi Sean,

Thanks very much for your response to my comments.  The solutions all
look good and I appreciate the updates that were made.



On Tue, May 16, 2017 at 12:36 PM, Sean Turner <sean@sn3rd.com> wrote:
>
>> On Apr 25, 2017, at 15:01, Kathleen Moriarty <kathleen.moriarty.ietf@gma=
il.com> wrote:
>>
>>
>> ----------------------------------------------------------------------
>> COMMENT:
>> ----------------------------------------------------------------------
>>
>> Thanks for your work on this draft, it's a helpful overview.  I see the
>> reasoning in the shepherd report for informational, but am curious if th=
e
>> standards track status is needed for other SDOs that might reference thi=
s
>> document?
>
> Addressed via discussions with Ben/Adam.
>
>> In reading sections 8 & 9, I would think the presentation and control in
>> section 8 would have the privacy implications of the second bullet in
>> section 9.  As such, it seems odd that normative language is used in thi=
s
>> bullet and not in section 8.  I'd be fine with no normative language in
>> either as long as the protocol drafts cover that appropriately.  Some
>> mention of privacy in section 8 could be helpful since it covers more
>> ground than the example in section 9.
>
> I=E2=80=99m looking at it this way: s8 is really also about local system =
support functions too.  We could reorder s8 and s9, but I=E2=80=99m hoping =
that thinking about s8 as a local support functions is sufficient.

That's fine, this was just a comment anyway. Thanks.

>
>> Security considerations: I don't see anything listed for security or
>> privacy considerations in respect to the signaling channel to the
>> web/application server.  Should there be considerations listed?  Securit=
y
>> of the actual server and content on the server as well as vulnerabilitie=
s
>> in listening protocols are just a few of the questions that come to mind=
.
>> If it doesn't matter, please let me know.  I appreciate the comment on
>> the browser being target rich as they have been in many attacks to gain
>> entry into networks leveraging established outbound sessions.  Maybe thi=
s
>> is covered in I-D.ietf-rtcweb-security and if so (have not had a chance
>> to review it yet), a high-level mention of gateway security here might b=
e
>> helpful.
>
> The security considerations and privacy considerations are addressed the =
two (cough) now expired security drafts:
> (cough) now expired security drafts:
> https://www.ietf.org/archive/id/draft-ietf-rtcweb-security-arch-12.txt
> https://www.ietf.org/archive/id/draft-ietf-rtcweb-security-08.txt
> The Security Considerations points to draft-ietf-rtcweb-security, which w=
ill get you to the other draft. Instead of recreating all of that here, I=
=E2=80=99m basically asking for your patience until these drafts make it to=
 the IESG in the next couple of months.  There=E2=80=99s a huge, and I mean=
 Donald Trump =E2=80=9Cyugeee=E2=80=9D, cluster of drafts that can=E2=80=99=
t go anywhere until those the security drafts get through the process so I =
hoping until then you=E2=80=99ll be satisfied sharpening your weapons of ma=
ss DISCUSSion until then ;)

That sounds fine, thank you.

>
>> I agree with Warren's comment about the management aspects being covered
>> here since it is an overview document.  It could be a very helpful
>> consideration for protocol developers that may devise new ways to enable
>> management as a result of understanding the issues.
>
> Addressed via PR:
> https://github.com/rtcweb-wg/rtcweb-overview/pull/15

Thanks for that!

>
>> I had to look up jingle and BOSH, you may want to consider adding
>> references to the XMPP specifications.
>
> Hmm there are references to XMPP and Jingle in the draft.  BOSH, I had to=
 look up too ;). Added a reference via the following PR:
> https://github.com/rtcweb-wg/rtcweb-overview/pull/33

Thanks!
>
> spt
>



--=20

Best regards,
Kathleen


From nobody Tue May 16 12:51:20 2017
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From: Warren Kumari <warren@kumari.net>
Date: Tue, 16 May 2017 15:46:12 -0400
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To: Sean Turner <sean@sn3rd.com>
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Subject: Re: [rtcweb] Warren Kumari's No Objection on draft-ietf-rtcweb-overview-18: (with COMMENT)
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Thank you very much.

These work for me...

W

On Tue, May 16, 2017 at 12:36 PM, Sean Turner <sean@sn3rd.com> wrote:
>
>> On Apr 25, 2017, at 11:11, Warren Kumari <warren@kumari.net> wrote:
>>
>>
>> ----------------------------------------------------------------------
>> COMMENT:
>> ----------------------------------------------------------------------
>>
>> Thank you -- I like these sort of overview documents for complex things
>> like WebRTC - they provide a newcomer to the technology a good place to
>> start, and help describe some of the reasons why things look the way the=
y
>> do.
>>
>> [ Edit: So, after more thought (and some discussion) I think that it
>> would be useful for the document to at least note the fact that
>> technologies like this mean that some of the existing operational
>> practices may need to change. For example, many enterprises perform QoS
>> based upon the fact that certain types of devices live in certain subnet=
s
>> (e.g many phones get placed in a specific VLAN using LLDP or CDP). With
>> more real time content coming from browsers, these matching practices
>> break, and so operators may not be able to QoS mark / prioritize traffic
>> accordingly. Perhaps something like: "One of the implications of a
>> solution like WebRTC is that more real-time traffic will be sourced from
>> computers (and not dedicated devices like telephones or
>> videoconferencing devices). This may have implications for operators
>> performing QoS marking and prioritization" ? This isn't really specific
>> to webrtc, but rather to a more general set of solutions like softphones
>> and the like, but is accelerated by WebRTC. ]
>
> To address Benoit, Kathleen, and Warren=E2=80=99s concerns here=E2=80=99s=
 a PR:
> https://github.com/rtcweb-wg/rtcweb-overview/pull/15
>
>> I do have a few comments on the document itself - there are all minor /
>> bikeshedding and can be ignored if you choose:
>> 1: "Development of The Universal Solution has proved hard, however, for
>> all the usual reasons."
>> -- this is cute, but leaves people wondering what "all the usual reasons
>> are". Perhaps just "Development of The Universal Solution has, however,
>> proved hard." (or just cut after the "however in the original").
>>
>> 2: I'm not sure why you have "Protocol" in the terminology section. It
>> doesn't seem like it is useful for the document, and this document
>> doesn't seem like the right place to (re) define it.
>>
>> 3: Acknowledgements:
>> Funny spacing in "Olle E.     Johansson=E2=80=9D
>
> To address #1 and #3, here=E2=80=99s a PR:
> https://github.com/rtcweb-wg/rtcweb-overview/pull/17/files
>
> Note I=E2=80=99m going to leave protocol in because I don=E2=80=99t think=
 it really hurts to leave it in.
>
> spt



--=20
I don't think the execution is relevant when it was obviously a bad
idea in the first place.
This is like putting rabid weasels in your pants, and later expressing
regret at having chosen those particular rabid weasels and that pair
of pants.
   ---maf


From nobody Tue May 16 14:35:41 2017
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To: Sean Turner <sean@sn3rd.com>
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From: Benoit Claise <bclaise@cisco.com>
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Subject: Re: [rtcweb] Benoit Claise's No Objection on draft-ietf-rtcweb-overview-18: (with COMMENT)
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Thanks Sean.

Regards, B.
>> On Apr 27, 2017, at 10:24, Benoit Claise <bclaise@cisco.com> wrote:
>>
>>
>> ----------------------------------------------------------------------
>> COMMENT:
>> ----------------------------------------------------------------------
>>
>> This topic below was discussed during the IESG telechat:
>>
>> Reading from the document objectives, from the abstract:
>>
>>    This document gives an overview and context of a protocol suite
>>    intended for use with real-time applications that can be deployed in
>>    browsers - "real time communication on the Web".
>>
>>    It intends to serve as a starting and coordination point to make
>> sure
>>    all the parts that are needed to achieve this goal are findable, and
>>    that the parts that belong in the Internet protocol suite are fully
>>    specified and on the right publication track.
>>
>> Reading this, I was thinking: great, I will have the full overview.
>> With "deployed", "starting and coordination point to make sure that all
>> the parts ..." I will have some  focus on the operational aspects,
>> basically, how should operators operate theses browser-embedded
>> applications.
>> Now, reading further ...
>>
>>    This document is intended to serve as the roadmap to the WebRTC
>>    specifications.  It defines terms used by other parts of the WebRTC
>>    protocol specifications, lists references to other specifications
>>    that don't need further elaboration in the WebRTC context, and gives
>>    pointers to other documents that form part of the WebRTC suite.
>>
>> ... I thought: Ok, if not covered here, at least I will have a pointer to
>> another operational document.
>> But wait:
>>
>>    By reading this document and the documents it refers to, it should
>> be
>>    possible to have all information needed to implement an WebRTC
>>    compatible implementation.
>>
>> So is this only about implementation?
>>
>> I like this document very much as it explains all the RTCWEB pieces in
>> one location. However, there is one important piece missing: the network
>> management considerations. See
>> https://datatracker.ietf.org/doc/html/rfc5706#appendix-A
>> This is where I'm coming from, discussing some more with Warren (this a
>> cut and past from this ballot):
>>
>>     [ Edit: So, after more thought (and some discussion) I think that it
>> would be useful for the document to at least note the fact that
>> technologies like this mean that some of the existing operational
>> practices may need to change. For example, many enterprises perform QoS
>> based upon the fact that certain types of devices live in certain subnets
>> (e.g many phones get placed in a specific VLAN using LLDP or CDP). With
>> more real time content coming from browsers, these matching practices
>> break, and so operators may not be able to QoS mark / prioritize traffic
>> accordingly. Perhaps something like: "One of the implications of a
>> solution like WebRTC is that more real-time traffic will be sourced from
>> computers (and not dedicated devices like telephones or videoconferencing
>> devices). This may have implications for operators performing QoS marking
>> and prioritization" ? This isn't really specific to webrtc, but rather to
>> a more general set of solutions like softphones and the like, but is
>> accelerated by WebRTC. ]
>>
>> In light of the previous discussions about draft-mm-wg-effect-encrypt-11,
>> the operators are used to manage voice, video, gaming a certain way, with
>> their operational current practices. Now, their current practices might
>> not work any longer. What should they do now in term of monitoring,
>> troubleshooting, QoS, SLA monitoring, etc these days with WebRTC?
>> While we should add this note (or a similar one) in the doc, I'm
>> wondering: where are (should be) those operational aspects discussed, if
>> not here?
>> I've seen https://tools.ietf.org/html/draft-ietf-tsvwg-rtcweb-qos-18, not
>> sure it's appropriate. Anyway, it's now in a RFC-editor state.
>> I could have requested a specific manageability doc in the charter. Too
>> late now.
> To address Benoit/Warren/Kathleenâ€™s comments on this, Iâ€™ve submitted the following PR:
> https://github.com/rtcweb-wg/rtcweb-overview/pull/15
>
> spt.
>


From nobody Tue May 16 14:56:28 2017
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Subject: [rtcweb] Comments on draft-ietf-rtcweb-fec-04 (Opus FEC)
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--f403043eb7e86868ff054fab2bae
Content-Type: text/plain; charset="UTF-8"

Section 3.3

   For Opus, packets deemed as important are re-encoded at a lower
   bitrate and added to the subsequent packet, allowing partial recovery
   of a lost packet.  This scheme is fairly efficient; experiments
   performed indicate that when Opus FEC is used, the overhead imposed
   is about 20-30%, depending on the amount of protection needed.  Note
   that this mechanism can only carry redundancy information for the
   immediately preceding packet; as such the decoder cannot fully
   recover multiple consecutive lost packets, which can be a problem on
   wireless networks.  See [RFC6716], Section 2.1.7
<https://tools.ietf.org/html/rfc6716#section-2.1.7> for complete
   details.


Section 4.1


   When using the Opus codec, use of the built-in Opus FEC mechanism is
   RECOMMENDED.  This provides reasonable protection of the audio stream
   against typical losses, with modest overhead.  Note that as indicated
   above the built-in Opus FEC only provides single-frame redundancy; if
   multi-packet protection is needed, the built-in FEC should be
   combined with [RFC2198 <https://tools.ietf.org/html/rfc2198>]
redundancy to protect the N-2th, N-3rd, etc.
   packets.


[BA] While loss of a single packet is by far the most likely loss

mode, our tests do not indicate that there is much benefit from

Opus internal FEC.  For example, when we compared Opus internal

FEC against no FEC and concealment, we found little discernible

difference in MOS score. On the other hand, we did see a benefit

arising from RFC 2198 redundancy to protect against burst loss.

So based on our test results, we cannot recommend use of built-in

Opus FEC.

--f403043eb7e86868ff054fab2bae
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<div dir=3D"ltr">Section 3.3<div><br></div><div><pre class=3D"gmail-newpage=
" style=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px;color:rgb(0=
,0,0)">   For Opus, packets deemed as important are re-encoded at a lower
   bitrate and added to the subsequent packet, allowing partial recovery
   of a lost packet.  This scheme is fairly efficient; experiments
   performed indicate that when Opus FEC is used, the overhead imposed
   is about 20-30%, depending on the amount of protection needed.  Note
   that this mechanism can only carry redundancy information for the
   immediately preceding packet; as such the decoder cannot fully
   recover multiple consecutive lost packets, which can be a problem on
   wireless networks.  See <a href=3D"https://tools.ietf.org/html/rfc6716#s=
ection-2.1.7">[RFC6716], Section=C2=A02.1.7</a> for complete
   details.</pre><pre class=3D"gmail-newpage" style=3D"font-size:13.3333px;=
margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><br></pre><pre class=3D"=
gmail-newpage" style=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0p=
x;color:rgb(0,0,0)">Section 4.1</pre><pre class=3D"gmail-newpage" style=3D"=
font-size:13.3333px;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><br>=
</pre><pre class=3D"gmail-newpage" style=3D"font-size:13.3333px;margin-top:=
0px;margin-bottom:0px;color:rgb(0,0,0)"><pre class=3D"gmail-newpage" style=
=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px">   When using the=
 Opus codec, use of the built-in Opus FEC mechanism is
   RECOMMENDED.  This provides reasonable protection of the audio stream
   against typical losses, with modest overhead.  Note that as indicated
   above the built-in Opus FEC only provides single-frame redundancy; if
   multi-packet protection is needed, the built-in FEC should be
   combined with [<a href=3D"https://tools.ietf.org/html/rfc2198" title=3D"=
&quot;RTP Payload for Redundant Audio Data&quot;">RFC2198</a>] redundancy t=
o protect the N-2th, N-3rd, etc.
   packets.</pre></pre><pre class=3D"gmail-newpage" style=3D"font-size:13.3=
333px;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><br></pre><pre cla=
ss=3D"gmail-newpage" style=3D"font-size:13.3333px;margin-top:0px;margin-bot=
tom:0px;color:rgb(0,0,0)">[BA] While loss of a single packet is by far the =
most likely loss</pre><pre class=3D"gmail-newpage" style=3D"font-size:13.33=
33px;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)">mode, our tests do =
not indicate that there is much benefit from</pre><pre class=3D"gmail-newpa=
ge" style=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px;color:rgb=
(0,0,0)">Opus internal FEC.  For example, when we compared Opus internal</p=
re><pre class=3D"gmail-newpage" style=3D"font-size:13.3333px;margin-top:0px=
;margin-bottom:0px;color:rgb(0,0,0)">FEC against no FEC and concealment, we=
 found little discernible</pre><pre class=3D"gmail-newpage" style=3D"font-s=
ize:13.3333px;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)">difference=
 in MOS score. On the other hand, we did see a benefit</pre><pre class=3D"g=
mail-newpage" style=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px=
;color:rgb(0,0,0)">arising from RFC 2198 redundancy to protect against burs=
t loss.</pre><pre class=3D"gmail-newpage" style=3D"font-size:13.3333px;marg=
in-top:0px;margin-bottom:0px;color:rgb(0,0,0)">So based on our test results=
, we cannot recommend use of built-in</pre><pre class=3D"gmail-newpage" sty=
le=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)=
">Opus FEC. </pre></div></div>

--f403043eb7e86868ff054fab2bae--


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Subject: [rtcweb] Referring to 5245bis or 5245?
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ekr=E2=80=99s discuss on draft-ietf-rtcweb-overview has raised whether =
drafts should refer to RFC 5245 or draft-ietf-ice-rfc5245bis.  We only =
need to normatively refer to 5245bis if a technical part of 5245bis =
needs to be read and implemented in order to implement the referring =
draft.  We have 7 drafts that refer to RFC 5245 and 2  that refer to =
draft-ietf-rfc5245bis:

draft-ietf-rtcweb-overview: As noted in my response to ekr=E2=80=99s =
discuss position [0], the chairs believe that the reference to =E2=80=9CIC=
E=E2=80=9D in the ICE Agent definition should be to RFC 5245.

draft-ietf-rtcweb-transports: Likewise, the chairs believer that a =
reference to RFC 5245 is also appropriate in transports.  This draft was =
changed in version -17 to refer to 5245bis.  =46rom GH: "The drafts =
-bundle and -dualstack-fairness both depend on 5245bis according to =
Cullen's chart in 'rtcweb-deps-13', and we already have a normative =
dependency from -transport on these.  So consistency of the bundle is =
improved by referencing 5245bis."

spt

[0] =
https://mailarchive.ietf.org/arch/msg/rtcweb/GWdXRIO68FZwVtzzqugnELKeaY8=


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I'm really confused about the statement regarding -transports.  You
say that 5245 is sufficient, then follow with justification for the
opposite position.

If we have as large a dependency as bundle that refers to 5245bis,
then we are taking a transitive dependency on 5245bis and might as
well refer to that.

A lot of this comes down to what bundle says.  Now, I see that bundle
depends on both 5245 and its -bis, which seems pretty inconsistent.  I
don't immediately see any strong reason for bundle to refer to the
-bis, though it does refer to the ice-sip-sdp draft, which might be
sufficiently implicated as to make the change necessary.  We should
ask Christer to confirm this.

I think that if we clarify that either way, then the reference in
-dualstack-fairness seems less of a concern; that document doesn't
need to reference 5245bis, though it would be nice if it could.

On 18 May 2017 at 10:12, Sean Turner <sean@sn3rd.com> wrote:
> ekr=E2=80=99s discuss on draft-ietf-rtcweb-overview has raised whether dr=
afts should refer to RFC 5245 or draft-ietf-ice-rfc5245bis.  We only need t=
o normatively refer to 5245bis if a technical part of 5245bis needs to be r=
ead and implemented in order to implement the referring draft.  We have 7 d=
rafts that refer to RFC 5245 and 2  that refer to draft-ietf-rfc5245bis:
>
> draft-ietf-rtcweb-overview: As noted in my response to ekr=E2=80=99s disc=
uss position [0], the chairs believe that the reference to =E2=80=9CICE=E2=
=80=9D in the ICE Agent definition should be to RFC 5245.
>
> draft-ietf-rtcweb-transports: Likewise, the chairs believer that a refere=
nce to RFC 5245 is also appropriate in transports.  This draft was changed =
in version -17 to refer to 5245bis.  From GH: "The drafts -bundle and -dual=
stack-fairness both depend on 5245bis according to Cullen's chart in 'rtcwe=
b-deps-13', and we already have a normative dependency from -transport on t=
hese.  So consistency of the bundle is improved by referencing 5245bis."
>
> spt
>
> [0] https://mailarchive.ietf.org/arch/msg/rtcweb/GWdXRIO68FZwVtzzqugnELKe=
aY8
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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From: Sean Turner <sean@sn3rd.com>
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Subject: Re: [rtcweb] Referring to 5245bis or 5245?
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> On May 18, 2017, at 10:54, Martin Thomson <martin.thomson@gmail.com> =
wrote:
>=20
> I'm really confused about the statement regarding -transports.  You
> say that 5245 is sufficient, then follow with justification for the
> opposite position.

Sorry the change from 5425 to 5245bis was included in the latest version =
using that rationale.  This shows to me that it was =E2=80=9Cnice=E2=80=9D=
 to get alignment and point to 5245bis not that it is necessary to point =
5245.  I.e., it=E2=80=99d be just fine to switch it back to referring to =
5245.

> If we have as large a dependency as bundle that refers to 5245bis,
> then we are taking a transitive dependency on 5245bis and might as
> well refer to that.
>=20
> A lot of this comes down to what bundle says.  Now, I see that bundle
> depends on both 5245 and its -bis, which seems pretty inconsistent.  I
> don't immediately see any strong reason for bundle to refer to the
> -bis, though it does refer to the ice-sip-sdp draft, which might be
> sufficiently implicated as to make the change necessary.  We should
> ask Christer to confirm this.
>=20
> I think that if we clarify that either way, then the reference in
> -dualstack-fairness seems less of a concern; that document doesn't
> need to reference 5245bis, though it would be nice if it could.

Exactly!

spt

> On 18 May 2017 at 10:12, Sean Turner <sean@sn3rd.com> wrote:
>> ekr=E2=80=99s discuss on draft-ietf-rtcweb-overview has raised =
whether drafts should refer to RFC 5245 or draft-ietf-ice-rfc5245bis.  =
We only need to normatively refer to 5245bis if a technical part of =
5245bis needs to be read and implemented in order to implement the =
referring draft.  We have 7 drafts that refer to RFC 5245 and 2  that =
refer to draft-ietf-rfc5245bis:
>>=20
>> draft-ietf-rtcweb-overview: As noted in my response to ekr=E2=80=99s =
discuss position [0], the chairs believe that the reference to =E2=80=9CIC=
E=E2=80=9D in the ICE Agent definition should be to RFC 5245.
>>=20
>> draft-ietf-rtcweb-transports: Likewise, the chairs believer that a =
reference to RFC 5245 is also appropriate in transports.  This draft was =
changed in version -17 to refer to 5245bis.  =46rom GH: "The drafts =
-bundle and -dualstack-fairness both depend on 5245bis according to =
Cullen's chart in 'rtcweb-deps-13', and we already have a normative =
dependency from -transport on these.  So consistency of the bundle is =
improved by referencing 5245bis."
>>=20
>> spt
>>=20
>> [0] =
https://mailarchive.ietf.org/arch/msg/rtcweb/GWdXRIO68FZwVtzzqugnELKeaY8
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb


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Subject: Re: [rtcweb] Referring to 5245bis or 5245?
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> On May 18, 2017, at 11:35, Sean Turner <sean@sn3rd.com> wrote:
>=20
>=20
>> On May 18, 2017, at 10:54, Martin Thomson <martin.thomson@gmail.com> =
wrote:
>>=20
>> I'm really confused about the statement regarding -transports.  You
>> say that 5245 is sufficient, then follow with justification for the
>> opposite position.
>=20
> Sorry the change from 5425 to 5245bis was included in the latest =
version using that rationale.  This shows to me that it was =E2=80=9Cnice=E2=
=80=9D to get alignment and point to 5245bis not that it is necessary to =
point 5245.  I.e., it=E2=80=99d be just fine to switch it back to =
referring to 5245.

Whoops:

This shows to me that it was =E2=80=9Cnice=E2=80=9D to get alignment and =
point to 5245bis not that it is necessary to point 5245bis.  I.e., =
it=E2=80=99d be just fine to switch it back to referring to 5245.

>> If we have as large a dependency as bundle that refers to 5245bis,
>> then we are taking a transitive dependency on 5245bis and might as
>> well refer to that.
>>=20
>> A lot of this comes down to what bundle says.  Now, I see that bundle
>> depends on both 5245 and its -bis, which seems pretty inconsistent.  =
I
>> don't immediately see any strong reason for bundle to refer to the
>> -bis, though it does refer to the ice-sip-sdp draft, which might be
>> sufficiently implicated as to make the change necessary.  We should
>> ask Christer to confirm this.
>>=20
>> I think that if we clarify that either way, then the reference in
>> -dualstack-fairness seems less of a concern; that document doesn't
>> need to reference 5245bis, though it would be nice if it could.
>=20
> Exactly!
>=20
> spt
>=20
>> On 18 May 2017 at 10:12, Sean Turner <sean@sn3rd.com> wrote:
>>> ekr=E2=80=99s discuss on draft-ietf-rtcweb-overview has raised =
whether drafts should refer to RFC 5245 or draft-ietf-ice-rfc5245bis.  =
We only need to normatively refer to 5245bis if a technical part of =
5245bis needs to be read and implemented in order to implement the =
referring draft.  We have 7 drafts that refer to RFC 5245 and 2  that =
refer to draft-ietf-rfc5245bis:
>>>=20
>>> draft-ietf-rtcweb-overview: As noted in my response to ekr=E2=80=99s =
discuss position [0], the chairs believe that the reference to =E2=80=9CIC=
E=E2=80=9D in the ICE Agent definition should be to RFC 5245.
>>>=20
>>> draft-ietf-rtcweb-transports: Likewise, the chairs believer that a =
reference to RFC 5245 is also appropriate in transports.  This draft was =
changed in version -17 to refer to 5245bis.  =46rom GH: "The drafts =
-bundle and -dualstack-fairness both depend on 5245bis according to =
Cullen's chart in 'rtcweb-deps-13', and we already have a normative =
dependency from -transport on these.  So consistency of the bundle is =
improved by referencing 5245bis."
>>>=20
>>> spt
>>>=20
>>> [0] =
https://mailarchive.ietf.org/arch/msg/rtcweb/GWdXRIO68FZwVtzzqugnELKeaY8
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Martin Thomson <martin.thomson@gmail.com>, Sean Turner <sean@sn3rd.com>
CC: RTCWeb IETF <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Referring to 5245bis or 5245?
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Subject: Re: [rtcweb] Referring to 5245bis or 5245?
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From: Bernard Aboba <bernard.aboba@gmail.com>
Date: Thu, 18 May 2017 09:50:14 -0700
Message-ID: <CAOW+2duBrC3f=-XaKFvMmyQ_JU72eTsES-UZDYPjQg6yZhab8Q@mail.gmail.com>
To: Sean Turner <sean@sn3rd.com>
Cc: Martin Thomson <martin.thomson@gmail.com>, RTCWeb IETF <rtcweb@ietf.org>
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Archived-At: <https://mailarchive.ietf.org/arch/msg/rtcweb/mv9OcAA756RydGbBiB33CZWyPI0>
Subject: Re: [rtcweb] Referring to 5245bis or 5245?
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Sean said:

"draft-ietf-rtcweb-transports: Likewise, the chairs believer that a
reference to RFC 5245 is also appropriate in transports.  This draft was
changed in version -17 to refer to 5245bis.  From GH: "The drafts -bundle
and -dualstack-fairness both depend on 5245bis according to Cullen's chart
in 'rtcweb-deps-13', and we already have a normative dependency from
-transport on these.  So consistency of the bundle is improved by
referencing 5245bis.".

[BA] draft-ietf-rtcweb-overview has a normative reference to
draft-ietf-rtcweb-transports which has a normative reference to draft-
ietf-ice-dualstack-fairness which in turn has a normative reference to
draft-ietf-ice-rfc5245bis.  So even if you remove the normative reference
to RFC5245bis from overview and transports, publication of overview will
still be held up until publication of RFC 5245bis, which will obsolete RFC
5245.

On Thu, May 18, 2017 at 8:37 AM, Sean Turner <sean@sn3rd.com> wrote:

>
> > On May 18, 2017, at 11:35, Sean Turner <sean@sn3rd.com> wrote:
> >
> >
> >> On May 18, 2017, at 10:54, Martin Thomson <martin.thomson@gmail.com>
> wrote:
> >>
> >> I'm really confused about the statement regarding -transports.  You
> >> say that 5245 is sufficient, then follow with justification for the
> >> opposite position.
> >
> > Sorry the change from 5425 to 5245bis was included in the latest versio=
n
> using that rationale.  This shows to me that it was =E2=80=9Cnice=E2=80=
=9D to get alignment
> and point to 5245bis not that it is necessary to point 5245.  I.e., it=E2=
=80=99d be
> just fine to switch it back to referring to 5245.
>
> Whoops:
>
> This shows to me that it was =E2=80=9Cnice=E2=80=9D to get alignment and =
point to 5245bis
> not that it is necessary to point 5245bis.  I.e., it=E2=80=99d be just fi=
ne to
> switch it back to referring to 5245.
>
> >> If we have as large a dependency as bundle that refers to 5245bis,
> >> then we are taking a transitive dependency on 5245bis and might as
> >> well refer to that.
> >>
> >> A lot of this comes down to what bundle says.  Now, I see that bundle
> >> depends on both 5245 and its -bis, which seems pretty inconsistent.  I
> >> don't immediately see any strong reason for bundle to refer to the
> >> -bis, though it does refer to the ice-sip-sdp draft, which might be
> >> sufficiently implicated as to make the change necessary.  We should
> >> ask Christer to confirm this.
> >>
> >> I think that if we clarify that either way, then the reference in
> >> -dualstack-fairness seems less of a concern; that document doesn't
> >> need to reference 5245bis, though it would be nice if it could.
> >
> > Exactly!
> >
> > spt
> >
> >> On 18 May 2017 at 10:12, Sean Turner <sean@sn3rd.com> wrote:
> >>> ekr=E2=80=99s discuss on draft-ietf-rtcweb-overview has raised whethe=
r drafts
> should refer to RFC 5245 or draft-ietf-ice-rfc5245bis.  We only need to
> normatively refer to 5245bis if a technical part of 5245bis needs to be
> read and implemented in order to implement the referring draft.  We have =
7
> drafts that refer to RFC 5245 and 2  that refer to draft-ietf-rfc5245bis:
> >>>
> >>> draft-ietf-rtcweb-overview: As noted in my response to ekr=E2=80=99s =
discuss
> position [0], the chairs believe that the reference to =E2=80=9CICE=E2=80=
=9D in the ICE
> Agent definition should be to RFC 5245.
> >>>
> >>> draft-ietf-rtcweb-transports: Likewise, the chairs believer that a
> reference to RFC 5245 is also appropriate in transports.  This draft was
> changed in version -17 to refer to 5245bis.  From GH: "The drafts -bundle
> and -dualstack-fairness both depend on 5245bis according to Cullen's char=
t
> in 'rtcweb-deps-13', and we already have a normative dependency from
> -transport on these.  So consistency of the bundle is improved by
> referencing 5245bis."
> >>>
> >>> spt
> >>>
> >>> [0] https://mailarchive.ietf.org/arch/msg/rtcweb/
> GWdXRIO68FZwVtzzqugnELKeaY8
> >>> _______________________________________________
> >>> 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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Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Sean said:=C2=A0<div><br></div><div>&quot;<span style=3D"f=
ont-size:12.8px">draft-ietf-rtcweb-transports: Likewise, the chairs believe=
r that a reference to RFC 5245 is also appropriate in transports.=C2=A0 Thi=
s draft was changed in version -17 to refer to 5245bis.=C2=A0 From GH: &quo=
t;The drafts -bundle and -dualstack-fairness both depend on 5245bis accordi=
ng to Cullen&#39;s chart in &#39;rtcweb-deps-13&#39;, and we already have a=
 normative dependency from -transport on these.=C2=A0 So consistency of the=
 bundle is improved by referencing 5245bis.&quot;</span><span style=3D"font=
-size:12.8px">.</span><br style=3D"font-size:12.8px"><br style=3D"font-size=
:12.8px"><span style=3D"font-size:12.8px"></span></div><div>[BA] draft-ietf=
-rtcweb-overview has a normative reference to draft-ietf-rtcweb-transports =
which has a normative reference to draft-<a name=3D"ref-I-D.ietf-mmusic-ice=
-dualstack-fairness" id=3D"gmail-ref-I-D.ietf-mmusic-ice-dualstack-fairness=
" style=3D"font-size:13.3333px">ietf-ice-dualstack-fairness</a>=C2=A0which =
in turn has a normative reference to draft-<a name=3D"ref-I-D.ietf-ice-rfc5=
245bis" id=3D"gmail-ref-I-D.ietf-ice-rfc5245bis" style=3D"font-size:13.3333=
px">ietf-ice-rfc5245bis.=C2=A0 So even if you remove the normative referenc=
e to RFC5245bis from overview and transports, publication of overview will =
still be held up until publication of RFC 5245bis, which will obsolete RFC =
5245.=C2=A0</a></div></div><div class=3D"gmail_extra"><br><div class=3D"gma=
il_quote">On Thu, May 18, 2017 at 8:37 AM, Sean Turner <span dir=3D"ltr">&l=
t;<a href=3D"mailto:sean@sn3rd.com" target=3D"_blank">sean@sn3rd.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""><br>
&gt; On May 18, 2017, at 11:35, Sean Turner &lt;<a href=3D"mailto:sean@sn3r=
d.com">sean@sn3rd.com</a>&gt; wrote:<br>
&gt;<br>
&gt;<br>
&gt;&gt; On May 18, 2017, at 10:54, Martin Thomson &lt;<a href=3D"mailto:ma=
rtin.thomson@gmail.com">martin.thomson@gmail.com</a>&gt; wrote:<br>
&gt;&gt;<br>
&gt;&gt; I&#39;m really confused about the statement regarding -transports.=
=C2=A0 You<br>
&gt;&gt; say that 5245 is sufficient, then follow with justification for th=
e<br>
&gt;&gt; opposite position.<br>
&gt;<br>
&gt; Sorry the change from 5425 to 5245bis was included in the latest versi=
on using that rationale.=C2=A0 This shows to me that it was =E2=80=9Cnice=
=E2=80=9D to get alignment and point to 5245bis not that it is necessary to=
 point 5245.=C2=A0 I.e., it=E2=80=99d be just fine to switch it back to ref=
erring to 5245.<br>
<br>
</span>Whoops:<br>
<br>
This shows to me that it was =E2=80=9Cnice=E2=80=9D to get alignment and po=
int to 5245bis not that it is necessary to point 5245bis.=C2=A0 I.e., it=E2=
=80=99d be just fine to switch it back to referring to 5245.<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
&gt;&gt; If we have as large a dependency as bundle that refers to 5245bis,=
<br>
&gt;&gt; then we are taking a transitive dependency on 5245bis and might as=
<br>
&gt;&gt; well refer to that.<br>
&gt;&gt;<br>
&gt;&gt; A lot of this comes down to what bundle says.=C2=A0 Now, I see tha=
t bundle<br>
&gt;&gt; depends on both 5245 and its -bis, which seems pretty inconsistent=
.=C2=A0 I<br>
&gt;&gt; don&#39;t immediately see any strong reason for bundle to refer to=
 the<br>
&gt;&gt; -bis, though it does refer to the ice-sip-sdp draft, which might b=
e<br>
&gt;&gt; sufficiently implicated as to make the change necessary.=C2=A0 We =
should<br>
&gt;&gt; ask Christer to confirm this.<br>
&gt;&gt;<br>
&gt;&gt; I think that if we clarify that either way, then the reference in<=
br>
&gt;&gt; -dualstack-fairness seems less of a concern; that document doesn&#=
39;t<br>
&gt;&gt; need to reference 5245bis, though it would be nice if it could.<br=
>
&gt;<br>
&gt; Exactly!<br>
&gt;<br>
&gt; spt<br>
&gt;<br>
&gt;&gt; On 18 May 2017 at 10:12, Sean Turner &lt;<a href=3D"mailto:sean@sn=
3rd.com">sean@sn3rd.com</a>&gt; wrote:<br>
&gt;&gt;&gt; ekr=E2=80=99s discuss on draft-ietf-rtcweb-overview has raised=
 whether drafts should refer to RFC 5245 or draft-ietf-ice-rfc5245bis.=C2=
=A0 We only need to normatively refer to 5245bis if a technical part of 524=
5bis needs to be read and implemented in order to implement the referring d=
raft.=C2=A0 We have 7 drafts that refer to RFC 5245 and 2=C2=A0 that refer =
to draft-ietf-rfc5245bis:<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; draft-ietf-rtcweb-overview: As noted in my response to ekr=E2=
=80=99s discuss position [0], the chairs believe that the reference to =E2=
=80=9CICE=E2=80=9D in the ICE Agent definition should be to RFC 5245.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; draft-ietf-rtcweb-transports: Likewise, the chairs believer th=
at a reference to RFC 5245 is also appropriate in transports.=C2=A0 This dr=
aft was changed in version -17 to refer to 5245bis.=C2=A0 From GH: &quot;Th=
e drafts -bundle and -dualstack-fairness both depend on 5245bis according t=
o Cullen&#39;s chart in &#39;rtcweb-deps-13&#39;, and we already have a nor=
mative dependency from -transport on these.=C2=A0 So consistency of the bun=
dle is improved by referencing 5245bis.&quot;<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; spt<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; [0] <a href=3D"https://mailarchive.ietf.org/arch/msg/rtcweb/GW=
dXRIO68FZwVtzzqugnELKeaY8" rel=3D"noreferrer" target=3D"_blank">https://mai=
larchive.ietf.org/<wbr>arch/msg/rtcweb/<wbr>GWdXRIO68FZwVtzzqugnELKeaY8</a>=
<br>
&gt;&gt;&gt; ______________________________<wbr>_________________<br>
&gt;&gt;&gt; rtcweb mailing list<br>
&gt;&gt;&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt;&gt;&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" rel=
=3D"noreferrer" target=3D"_blank">https://www.ietf.org/mailman/<wbr>listinf=
o/rtcweb</a><br>
&gt;<br>
<br>
______________________________<wbr>_________________<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/<wbr>listinfo/rtcweb</a><br=
>
</div></div></blockquote></div><br></div>

--001a1141b3621c5a1e054fcf340f--


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Bernard Aboba <bernard.aboba@gmail.com>, Sean Turner <sean@sn3rd.com>
CC: RTCWeb IETF <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Referring to 5245bis or 5245?
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--_000_7594FB04B1934943A5C02806D1A2204B4CBA8FEFESESSMB109erics_--


From nobody Mon May 22 15:52:39 2017
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From: Cullen Jennings <fluffy@iii.ca>
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Date: Mon, 22 May 2017 16:52:23 -0600
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References: <4C1F0FE7-F7E6-47F7-922D-057E4E7FA466@sn3rd.com> <CABkgnnVhS07gUdw+MJT8dLH89=Y1HBhrrwh6wTGs5gyy8O5DWw@mail.gmail.com> <3CC0A416-5A81-46FA-886C-5F43BA5193A6@sn3rd.com> <6BD64B92-4DE2-4BAD-A23D-65E8F52E13B0@sn3rd.com> <CAOW+2duBrC3f=-XaKFvMmyQ_JU72eTsES-UZDYPjQg6yZhab8Q@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B4CBA8FEF@ESESSMB109.ericsson.se>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Archived-At: <https://mailarchive.ietf.org/arch/msg/rtcweb/AvJgsAK7yrBb0RvsR3BcEYu1pg8>
Subject: Re: [rtcweb] Referring to 5245bis or 5245?
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Note that I don't think the timeline is the major issue (it is an issue) =
... they key issue is that 5245bis does not seem to be needed for any =
technical reason by WebRTC.


> On May 18, 2017, at 11:03 AM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:
>=20
> Hi,
>=20
> =20
>=20
> In general, if people have issues with referencing 5245bis because =
they are afraid it will hold up publication of RTCWEB specs, note that I =
have indicated to the ICE WG chairs that I think 5245bis is getting =
ready for WGLC.
>=20
> =20
>=20
> Regards,
>=20
> =20
>=20
> Christer
>=20
> =20
>=20
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Bernard =
Aboba
> Sent: 18 May 2017 18:50
> To: Sean Turner <sean@sn3rd.com>
> Cc: RTCWeb IETF <rtcweb@ietf.org>
> Subject: Re: [rtcweb] Referring to 5245bis or 5245?
>=20
> =20
>=20
> Sean said:=20
>=20
> =20
>=20
> "draft-ietf-rtcweb-transports: Likewise, the chairs believer that a =
reference to RFC 5245 is also appropriate in transports.  This draft was =
changed in version -17 to refer to 5245bis.  =46rom GH: "The drafts =
-bundle and -dualstack-fairness both depend on 5245bis according to =
Cullen's chart in 'rtcweb-deps-13', and we already have a normative =
dependency from -transport on these.  So consistency of the bundle is =
improved by referencing 5245bis.".
>=20
> [BA] draft-ietf-rtcweb-overview has a normative reference to =
draft-ietf-rtcweb-transports which has a normative reference to =
draft-ietf-ice-dualstack-fairness which in turn has a normative =
reference to draft-ietf-ice-rfc5245bis.  So even if you remove the =
normative reference to RFC5245bis from overview and transports, =
publication of overview will still be held up until publication of RFC =
5245bis, which will obsolete RFC 5245.=20
>=20
> =20
>=20
> On Thu, May 18, 2017 at 8:37 AM, Sean Turner <sean@sn3rd.com> wrote:
>=20
>=20
> > On May 18, 2017, at 11:35, Sean Turner <sean@sn3rd.com> wrote:
> >
> >
> >> On May 18, 2017, at 10:54, Martin Thomson =
<martin.thomson@gmail.com> wrote:
> >>
> >> I'm really confused about the statement regarding -transports.  You
> >> say that 5245 is sufficient, then follow with justification for the
> >> opposite position.
> >
> > Sorry the change from 5425 to 5245bis was included in the latest =
version using that rationale.  This shows to me that it was =E2=80=9Cnice=E2=
=80=9D to get alignment and point to 5245bis not that it is necessary to =
point 5245.  I.e., it=E2=80=99d be just fine to switch it back to =
referring to 5245.
>=20
> Whoops:
>=20
> This shows to me that it was =E2=80=9Cnice=E2=80=9D to get alignment =
and point to 5245bis not that it is necessary to point 5245bis.  I.e., =
it=E2=80=99d be just fine to switch it back to referring to 5245.
>=20
>=20
> >> If we have as large a dependency as bundle that refers to 5245bis,
> >> then we are taking a transitive dependency on 5245bis and might as
> >> well refer to that.
> >>
> >> A lot of this comes down to what bundle says.  Now, I see that =
bundle
> >> depends on both 5245 and its -bis, which seems pretty inconsistent. =
 I
> >> don't immediately see any strong reason for bundle to refer to the
> >> -bis, though it does refer to the ice-sip-sdp draft, which might be
> >> sufficiently implicated as to make the change necessary.  We should
> >> ask Christer to confirm this.
> >>
> >> I think that if we clarify that either way, then the reference in
> >> -dualstack-fairness seems less of a concern; that document doesn't
> >> need to reference 5245bis, though it would be nice if it could.
> >
> > Exactly!
> >
> > spt
> >
> >> On 18 May 2017 at 10:12, Sean Turner <sean@sn3rd.com> wrote:
> >>> ekr=E2=80=99s discuss on draft-ietf-rtcweb-overview has raised =
whether drafts should refer to RFC 5245 or draft-ietf-ice-rfc5245bis.  =
We only need to normatively refer to 5245bis if a technical part of =
5245bis needs to be read and implemented in order to implement the =
referring draft.  We have 7 drafts that refer to RFC 5245 and 2  that =
refer to draft-ietf-rfc5245bis:
> >>>
> >>> draft-ietf-rtcweb-overview: As noted in my response to ekr=E2=80=99s=
 discuss position [0], the chairs believe that the reference to =
=E2=80=9CICE=E2=80=9D in the ICE Agent definition should be to RFC 5245.
> >>>
> >>> draft-ietf-rtcweb-transports: Likewise, the chairs believer that a =
reference to RFC 5245 is also appropriate in transports.  This draft was =
changed in version -17 to refer to 5245bis.  =46rom GH: "The drafts =
-bundle and -dualstack-fairness both depend on 5245bis according to =
Cullen's chart in 'rtcweb-deps-13', and we already have a normative =
dependency from -transport on these.  So consistency of the bundle is =
improved by referencing 5245bis."
> >>>
> >>> spt
> >>>
> >>> [0] =
https://mailarchive.ietf.org/arch/msg/rtcweb/GWdXRIO68FZwVtzzqugnELKeaY8
> >>> _______________________________________________
> >>> rtcweb mailing list
> >>> rtcweb@ietf.org
> >>> https://www.ietf.org/mailman/listinfo/rtcweb
> >
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
> =20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From nobody Tue May 23 22:38:27 2017
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Subject: [rtcweb] I-D Action: draft-ietf-rtcweb-fec-05.txt
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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 of the IETF.

        Title           : WebRTC Forward Error Correction Requirements
        Author          : Justin Uberti
	Filename        : draft-ietf-rtcweb-fec-05.txt
	Pages           : 10
	Date            : 2017-05-23

Abstract:
   This document provides information and requirements for how Forward
   Error Correction (FEC) should be used by WebRTC applications.


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

There are also htmlized versions available at:
https://tools.ietf.org/html/draft-ietf-rtcweb-fec-05
https://datatracker.ietf.org/doc/html/draft-ietf-rtcweb-fec-05

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-fec-05


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 Wed May 24 17:00:03 2017
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From: Justin Uberti <juberti@google.com>
Date: Wed, 24 May 2017 16:59:38 -0700
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-fec-05.txt
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Note that is simply a version bump for freshness.

On Tue, May 23, 2017 at 10:38 PM, <internet-drafts@ietf.org> wrote:

>
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
> This draft is a work item of the Real-Time Communication in WEB-browsers
> of the IETF.
>
>         Title           : WebRTC Forward Error Correction Requirements
>         Author          : Justin Uberti
>         Filename        : draft-ietf-rtcweb-fec-05.txt
>         Pages           : 10
>         Date            : 2017-05-23
>
> Abstract:
>    This document provides information and requirements for how Forward
>    Error Correction (FEC) should be used by WebRTC applications.
>
>
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-fec/
>
> There are also htmlized versions available at:
> https://tools.ietf.org/html/draft-ietf-rtcweb-fec-05
> https://datatracker.ietf.org/doc/html/draft-ietf-rtcweb-fec-05
>
> A diff from the previous version is available at:
> https://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-fec-05
>
>
> 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">Note that is simply a version bump for freshness.</div><di=
v class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Tue, May 23, 2017=
 at 10:38 PM,  <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"gmail_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>
This draft is a work item of the Real-Time Communication in WEB-browsers 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:=
 WebRTC Forward Error Correction Requirements<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Author=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 : Just=
in Uberti<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Filename=C2=A0 =C2=A0 =C2=A0 =C2=A0 : draft-iet=
f-rtcweb-fec-05.txt<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Pages=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0:=
 10<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Date=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 :=
 2017-05-23<br>
<br>
Abstract:<br>
=C2=A0 =C2=A0This document provides information and requirements for how Fo=
rward<br>
=C2=A0 =C2=A0Error Correction (FEC) should be used by WebRTC applications.<=
br>
<br>
<br>
The IETF datatracker status page for this draft is:<br>
<a href=3D"https://datatracker.ietf.org/doc/draft-ietf-rtcweb-fec/" rel=3D"=
noreferrer" target=3D"_blank">https://datatracker.ietf.org/<wbr>doc/draft-i=
etf-rtcweb-fec/</a><br>
<br>
There are also htmlized versions available at:<br>
<a href=3D"https://tools.ietf.org/html/draft-ietf-rtcweb-fec-05" rel=3D"nor=
eferrer" target=3D"_blank">https://tools.ietf.org/html/<wbr>draft-ietf-rtcw=
eb-fec-05</a><br>
<a href=3D"https://datatracker.ietf.org/doc/html/draft-ietf-rtcweb-fec-05" =
rel=3D"noreferrer" target=3D"_blank">https://datatracker.ietf.org/<wbr>doc/=
html/draft-ietf-rtcweb-<wbr>fec-05</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-fec-05" re=
l=3D"noreferrer" target=3D"_blank">https://www.ietf.org/rfcdiff?<wbr>url2=
=3Ddraft-ietf-rtcweb-fec-05</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-<wbr>drafts/</a><br>
<br>
______________________________<wbr>_________________<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/<wbr>listinfo/rtcweb</a><br=
>
</blockquote></div><br></div>

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From: Justin Uberti <juberti@google.com>
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Subject: Re: [rtcweb] Comments on draft-ietf-rtcweb-fec-04 (Opus FEC)
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This would be a pretty dramatic statement, basically indicating that the
FEC mechanism that has been built for Opus is not useful.

While I understand the rationale, I would like to have some more specific
results that we can point at before making such a claim in a RFC.

On Tue, May 16, 2017 at 2:50 PM, Bernard Aboba <bernard.aboba@gmail.com>
wrote:

> Section 3.3
>
>    For Opus, packets deemed as important are re-encoded at a lower
>    bitrate and added to the subsequent packet, allowing partial recovery
>    of a lost packet.  This scheme is fairly efficient; experiments
>    performed indicate that when Opus FEC is used, the overhead imposed
>    is about 20-30%, depending on the amount of protection needed.  Note
>    that this mechanism can only carry redundancy information for the
>    immediately preceding packet; as such the decoder cannot fully
>    recover multiple consecutive lost packets, which can be a problem on
>    wireless networks.  See [RFC6716], Section 2.1.7 <https://tools.ietf.org/html/rfc6716#section-2.1.7> for complete
>    details.
>
>
> Section 4.1
>
>
>    When using the Opus codec, use of the built-in Opus FEC mechanism is
>    RECOMMENDED.  This provides reasonable protection of the audio stream
>    against typical losses, with modest overhead.  Note that as indicated
>    above the built-in Opus FEC only provides single-frame redundancy; if
>    multi-packet protection is needed, the built-in FEC should be
>    combined with [RFC2198 <https://tools.ietf.org/html/rfc2198>] redundancy to protect the N-2th, N-3rd, etc.
>    packets.
>
>
> [BA] While loss of a single packet is by far the most likely loss
>
> mode, our tests do not indicate that there is much benefit from
>
> Opus internal FEC.  For example, when we compared Opus internal
>
> FEC against no FEC and concealment, we found little discernible
>
> difference in MOS score. On the other hand, we did see a benefit
>
> arising from RFC 2198 redundancy to protect against burst loss.
>
> So based on our test results, we cannot recommend use of built-in
>
> Opus FEC.
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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<div dir=3D"ltr">This would be a pretty dramatic statement, basically indic=
ating that the FEC mechanism that has been built for Opus is not useful.<di=
v><br></div><div>While I understand the rationale, I would like to have som=
e more specific results that we can point at before making such a claim in =
a RFC.</div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Tu=
e, May 16, 2017 at 2:50 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"ltr">Sec=
tion 3.3<div><br></div><div><pre class=3D"m_-2475104692027837020m_-79116263=
41986014241gmail-newpage" style=3D"font-size:13.3333px;margin-top:0px;margi=
n-bottom:0px;color:rgb(0,0,0)">   For Opus, packets deemed as important are=
 re-encoded at a lower
   bitrate and added to the subsequent packet, allowing partial recovery
   of a lost packet.  This scheme is fairly efficient; experiments
   performed indicate that when Opus FEC is used, the overhead imposed
   is about 20-30%, depending on the amount of protection needed.  Note
   that this mechanism can only carry redundancy information for the
   immediately preceding packet; as such the decoder cannot fully
   recover multiple consecutive lost packets, which can be a problem on
   wireless networks.  See <a href=3D"https://tools.ietf.org/html/rfc6716#s=
ection-2.1.7" target=3D"_blank">[RFC6716], Section=C2=A02.1.7</a> for compl=
ete
   details.</pre><pre class=3D"m_-2475104692027837020m_-7911626341986014241=
gmail-newpage" style=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0p=
x;color:rgb(0,0,0)"><br></pre><pre class=3D"m_-2475104692027837020m_-791162=
6341986014241gmail-newpage" style=3D"font-size:13.3333px;margin-top:0px;mar=
gin-bottom:0px;color:rgb(0,0,0)">Section 4.1</pre><pre class=3D"m_-24751046=
92027837020m_-7911626341986014241gmail-newpage" style=3D"font-size:13.3333p=
x;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><br></pre><pre class=
=3D"m_-2475104692027837020m_-7911626341986014241gmail-newpage" style=3D"fon=
t-size:13.3333px;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><pre cl=
ass=3D"m_-2475104692027837020m_-7911626341986014241gmail-newpage" style=3D"=
font-size:13.3333px;margin-top:0px;margin-bottom:0px">   When using the Opu=
s codec, use of the built-in Opus FEC mechanism is
   RECOMMENDED.  This provides reasonable protection of the audio stream
   against typical losses, with modest overhead.  Note that as indicated
   above the built-in Opus FEC only provides single-frame redundancy; if
   multi-packet protection is needed, the built-in FEC should be
   combined with [<a href=3D"https://tools.ietf.org/html/rfc2198" title=3D"=
&quot;RTP Payload for Redundant Audio Data&quot;" target=3D"_blank">RFC2198=
</a>] redundancy to protect the N-2th, N-3rd, etc.
   packets.</pre></pre><pre class=3D"m_-2475104692027837020m_-7911626341986=
014241gmail-newpage" style=3D"font-size:13.3333px;margin-top:0px;margin-bot=
tom:0px;color:rgb(0,0,0)"><br></pre><pre class=3D"m_-2475104692027837020m_-=
7911626341986014241gmail-newpage" style=3D"font-size:13.3333px;margin-top:0=
px;margin-bottom:0px;color:rgb(0,0,0)">[BA] While loss of a single packet i=
s by far the most likely loss</pre><pre class=3D"m_-2475104692027837020m_-7=
911626341986014241gmail-newpage" style=3D"font-size:13.3333px;margin-top:0p=
x;margin-bottom:0px;color:rgb(0,0,0)">mode, our tests do not indicate that =
there is much benefit from</pre><pre class=3D"m_-2475104692027837020m_-7911=
626341986014241gmail-newpage" style=3D"font-size:13.3333px;margin-top:0px;m=
argin-bottom:0px;color:rgb(0,0,0)">Opus internal FEC.  For example, when we=
 compared Opus internal</pre><pre class=3D"m_-2475104692027837020m_-7911626=
341986014241gmail-newpage" style=3D"font-size:13.3333px;margin-top:0px;marg=
in-bottom:0px;color:rgb(0,0,0)">FEC against no FEC and concealment, we foun=
d little discernible</pre><pre class=3D"m_-2475104692027837020m_-7911626341=
986014241gmail-newpage" style=3D"font-size:13.3333px;margin-top:0px;margin-=
bottom:0px;color:rgb(0,0,0)">difference in MOS score. On the other hand, we=
 did see a benefit</pre><pre class=3D"m_-2475104692027837020m_-791162634198=
6014241gmail-newpage" style=3D"font-size:13.3333px;margin-top:0px;margin-bo=
ttom:0px;color:rgb(0,0,0)">arising from RFC 2198 redundancy to protect agai=
nst burst loss.</pre><pre class=3D"m_-2475104692027837020m_-791162634198601=
4241gmail-newpage" style=3D"font-size:13.3333px;margin-top:0px;margin-botto=
m:0px;color:rgb(0,0,0)">So based on our test results, we cannot recommend u=
se of built-in</pre><pre class=3D"m_-2475104692027837020m_-7911626341986014=
241gmail-newpage" style=3D"font-size:13.3333px;margin-top:0px;margin-bottom=
:0px;color:rgb(0,0,0)">Opus FEC. </pre></div></div>
<br>______________________________<wbr>_________________<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/l<wbr>istinfo/rtcweb</a><br=
>
<br></blockquote></div><br></div></div>

--001a1135b7648fe0f105504df19c--


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Date: Wed, 24 May 2017 17:35:17 -0700
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--f403045e38a41e01a105504e6529
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To:  IETF RTCWeb WG

From:  Colin Gallagher
           https://www.linkedin.com/in/colingallagher
           https://lifeboat.com/ex/bios.colin.gallagher

Request for New Security Review of WebRTC based on observed vulnerabilities
in certain examples

Dear IETF RTCWeb WG,

This request relates to WebRTC.  The WebRTC is in draft at
https://w3c.github.io/webrtc-pc/ and is described also at
https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API#WebRTC_interfaces

There are certain examples where the use of WebRTC has presented security
vulnerabilities.  While not exhaustive, there are two I wanted to mention
that I had been exposed to recently.

1) The use of a video conferencing service which relies upon WebRTC (that
is a well established service)
2) The examination of a browser based WebRTC version of a decentralized
exchange (that is currently under development)

In both cases the security vulnerability which is described below was the
result of the use of WebRTC.

The problem under consideration is as follows:

As per the first example (the video conferencing service), recently I came
upon a vendor who shall remain unnamed who uses WebRTC as the backbone for
their video conferencing service.  As a result, I found I was unable to
access the service in question without disabling some security settings
which I would have preferred to leave intact.

My system is (based on what the aforementioned service detects) Mozilla/5.0
(X11; Ubuntu; Linux x86_64; rv:53.0) Gecko/20100101 Firefox/53.0.  This is
Ubuntu 16.10 64-bit with Firefox 53.0.2.

However, quite some *long* while back I disabled WebRTC because of the
security vulnerability (described in part here on information security
stack exchange
<https://security.stackexchange.com/questions/82129/what-is-webrtc-and-how-can-i-protect-myself-against-leaking-information-that-doe>).
So as to mitigate this vulnerability, I changed settings in Firefox as
follows:


   1. I typed *about:config* in the address bar
   2. I found the setting *media.peerconnection.enabled*
   3. I set it to *false*

Unfortunately, this setting does not allow Firefox to work with the video
conferencing service in question, which relies entirely on WebRTC.

So I had to go back in and change the media.peerconnection.enabled setting
to *true* in Firefox in order to get it to work with the service in
question. While this enabled me to conference with teams that desire to use
the video conferencing service that rely wholly upon WebRTC, it concerns me
because of the security vulnerability.

One suggested mitigation was having NoScript, but using NoScript was not
helpful (you generally need to disable it for the site you are viewing that
relies upon WebRTC, in order to obtain any functionality, and whether
NoScript is off or on, WebRTC still causes the vulnerability of leaking
information as previously described).

Furthermore, upon reload of the browser, even after setting
media.peerconnection.enabled to *true*, the video conferencing service
wouldn't work until I installed the WebRTC 0.1.3 extension for Firefox and
set it to expose my real IP address as well as allowing the following:

a. navigator.getUserMedia
b. window.MediaStreamTrack
c. window.RTCPeerConnection
d. window.RTCSessionDescription

That seemed to me to be a direct consequence of the persistent and ongoing
security vulnerability in WebRTC.

I contacted the video conferencing service provider and their solution was
simply to state that "If the peer to peer connection is of concern, you can
utilize our premium version which will route traffic through a forwarding
server with in our environment that handles the processing of the video and
sound of all users in the conference and send it to each user individually
rather than using a peer to peer connection."  In other words, they expect
that people should have to pay in order to mitigate this WebRTC security
vulnerability
<https://security.stackexchange.com/questions/82129/what-is-webrtc-and-how-can-i-protect-myself-against-leaking-information-that-doe>,
because they are unwilling to design to protect all users from it.

In a similar fashion, in the second example of the browser based WebRTC
version of a decentralized exchange, the user's financial information is
periodically exposed as a result of the information leakage which is
presently caused by WebRTC.  (Fortunately, this browser based WebRTC
decentralized exchange is under development and is not currently open to
actual usage.)

Although undoubtedly this has been discussed here before, I am asking that
WebRTC security issues be reconsidered and that a new security review be
done to ameliorate or eliminate this problem in terms of WebRTC leaking
this information.  It is my view that WebRTC should be further designed and
developed with an eye to mitigating this serious problem, so that users of
services which might rely partially or wholly upon WebRTC would not have to
be concerned about such information leakage.

Respectfully,
Colin Gallagher

--f403045e38a41e01a105504e6529
Content-Type: text/html; charset="UTF-8"
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div><div><div><div><div><div><div><div>To:=C2=A0 IETF RTC=
Web WG<br><br></div>From:=C2=A0 Colin Gallagher<br></div>=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 <a href=3D"https://www.linkedin.=
com/in/colingallagher">https://www.linkedin.com/in/colingallagher</a><br>=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 <a href=3D"htt=
ps://lifeboat.com/ex/bios.colin.gallagher">https://lifeboat.com/ex/bios.col=
in.gallagher</a><br><br></div>Request for New Security Review of WebRTC bas=
ed on observed vulnerabilities in certain examples<br><br></div>Dear IETF R=
TCWeb WG,<br><br></div><div>This request relates to WebRTC.=C2=A0 The WebRT=
C is in draft at <a href=3D"https://w3c.github.io/webrtc-pc/">https://w3c.g=
ithub.io/webrtc-pc/</a> and is=20
described also at=20
<a href=3D"https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API#WebR=
TC_interfaces">https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API#=
WebRTC_interfaces</a></div><div><br></div>There are certain examples where =
the use of WebRTC has presented security vulnerabilities.=C2=A0 While not e=
xhaustive, there are two I wanted to mention that I had been exposed to rec=
ently.<br></div><div><br></div>1) The use of a video conferencing service w=
hich relies upon WebRTC (that is a well established service)<br></div>2) Th=
e examination of a browser based WebRTC version of a decentralized exchange=
 (that is currently under development)<br><br></div>In both cases the secur=
ity vulnerability which is described below was the result of the use of Web=
RTC.<br><br>The problem under consideration is as follows:<br><br>As per th=
e first example (the video conferencing service), recently
 I came upon a vendor who shall remain unnamed who uses WebRTC as the=20
backbone for their video conferencing service.=C2=A0 As a result, I found I=
=20
was unable to access the service in question without disabling some=20
security settings which I would have preferred to leave intact. <br><br>My =
system is (based on what the aforementioned service detects) Mozilla/5.0 (X=
11; Ubuntu; Linux x86_64; rv:53.0)=20
Gecko/20100101 Firefox/53.0.=C2=A0 This is Ubuntu 16.10 64-bit with Firefox=
=20
53.0.2.<br><br>However, quite some <i>long</i> while back I disabled WebRTC=
 because of the security vulnerability (described in part <a href=3D"https:=
//security.stackexchange.com/questions/82129/what-is-webrtc-and-how-can-i-p=
rotect-myself-against-leaking-information-that-doe" target=3D"_blank">here =
on information security stack exchange</a>).=C2=A0 So as to mitigate this v=
ulnerability, I changed settings in Firefox as follows:<br><br><ol><li>I ty=
ped <em>about:config</em> in the address bar</li><li>I found the setting <e=
m>media.peerconnection.enabled</em></li><li>I set it to <strong>false</stro=
ng></li></ol><p><strong></strong>Unfortunately,
 this setting does not allow Firefox to work with the video conferencing
 service in question, which relies entirely on WebRTC.<br></p><p>So I had t=
o go back in and change the media.peerconnection.enabled setting to <b>true=
</b>
 in Firefox in order to get it to work with the service in question.=20
While this enabled me to conference with teams that desire to use the=20
video conferencing service that rely wholly upon WebRTC, it concerns me=20
because of the security vulnerability.</p><p>One suggested mitigation was h=
aving NoScript, but using NoScript was not helpful (you generally need to d=
isable it for the site you are viewing that relies upon WebRTC, in order to=
 obtain any functionality, and whether NoScript is off or on, WebRTC still =
causes the vulnerability of leaking information as previously described).<b=
r></p><p>Furthermore, upon reload of the browser, even after setting media.=
peerconnection.enabled to <b>true</b>,
 the video conferencing service wouldn&#39;t work until I installed the=20
WebRTC 0.1.3 extension for Firefox and set it to expose my real IP=20
address as well as allowing the following:</p><p>a. navigator.getUserMedia<=
br>b. window.MediaStreamTrack<br>c. window.RTCPeerConnection<br>d. window.R=
TCSessionDescription<br></p><p>That seemed to me to be a direct consequence=
 of the persistent and ongoing security vulnerability in WebRTC.<br></p><p>=
I
 contacted the video conferencing service provider and their solution=20
was simply to state that &quot;If the peer to peer connection is of concern=
,=20
you can utilize our=20
premium version which will route traffic through a forwarding server=20
with in our environment that handles the processing of the video and=20
sound of all users in the conference and send it to each user=20
individually rather than using a peer to peer connection.&quot;=C2=A0 In ot=
her=20
words, they expect that people should have to pay in order to mitigate <a h=
ref=3D"https://security.stackexchange.com/questions/82129/what-is-webrtc-an=
d-how-can-i-protect-myself-against-leaking-information-that-doe" target=3D"=
_blank">this WebRTC security vulnerability</a>, because they are unwilling =
to design to protect all users from it.</p><p>In a similar fashion, in the =
second example of the browser based WebRTC version of a decentralized excha=
nge, the user&#39;s financial information is periodically exposed as a resu=
lt of the information leakage which is presently caused by WebRTC.=C2=A0 (F=
ortunately, this browser based WebRTC decentralized exchange is under devel=
opment and is not currently open to actual usage.)<br></p><p>Although
 undoubtedly this has been discussed here before, I am asking that WebRTC s=
ecurity issues be
 reconsidered and that a new security review be done to ameliorate or=20
eliminate this problem in terms of WebRTC leaking this information.=C2=A0 I=
t is my view that WebRTC should be further designed and developed with an e=
ye to mitigating this serious problem, so that users of services which migh=
t rely partially or wholly upon WebRTC would not have to be concerned about=
 such information leakage.<br></p><p>Respectfully,</p>Colin Gallagher<br><b=
r></div>

--f403045e38a41e01a105504e6529--


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From: Justin Uberti <juberti@google.com>
Date: Thu, 25 May 2017 13:04:50 -0700
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Colin,

This is an important topic, and the group has reviewed this situation in
detail. The results have been written up in this document
<https://tools.ietf.org/html/draft-ietf-rtcweb-ip-handling-03>, and to my
knowledge most browsers that support WebRTC currently conform to its
recommendations.

However, I'm not totally clear on your specific technical concern. You
linked to an (now outdated) stackexchange page, but could you provide more
detail about the specific situation you are concerned about?

Justin

On Wed, May 24, 2017 at 5:35 PM, Colin Gallagher <
colingallagher.rpcv@gmail.com> wrote:

> To:  IETF RTCWeb WG
>
> From:  Colin Gallagher
>            https://www.linkedin.com/in/colingallagher
>            https://lifeboat.com/ex/bios.colin.gallagher
>
> Request for New Security Review of WebRTC based on observed
> vulnerabilities in certain examples
>
> Dear IETF RTCWeb WG,
>
> This request relates to WebRTC.  The WebRTC is in draft at
> https://w3c.github.io/webrtc-pc/ and is described also at
> https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API#
> WebRTC_interfaces
>
> There are certain examples where the use of WebRTC has presented security
> vulnerabilities.  While not exhaustive, there are two I wanted to mention
> that I had been exposed to recently.
>
> 1) The use of a video conferencing service which relies upon WebRTC (that
> is a well established service)
> 2) The examination of a browser based WebRTC version of a decentralized
> exchange (that is currently under development)
>
> In both cases the security vulnerability which is described below was the
> result of the use of WebRTC.
>
> The problem under consideration is as follows:
>
> As per the first example (the video conferencing service), recently I came
> upon a vendor who shall remain unnamed who uses WebRTC as the backbone for
> their video conferencing service.  As a result, I found I was unable to
> access the service in question without disabling some security settings
> which I would have preferred to leave intact.
>
> My system is (based on what the aforementioned service detects)
> Mozilla/5.0 (X11; Ubuntu; Linux x86_64; rv:53.0) Gecko/20100101
> Firefox/53.0.  This is Ubuntu 16.10 64-bit with Firefox 53.0.2.
>
> However, quite some *long* while back I disabled WebRTC because of the
> security vulnerability (described in part here on information security
> stack exchange
> <https://security.stackexchange.com/questions/82129/what-is-webrtc-and-how-can-i-protect-myself-against-leaking-information-that-doe>).
> So as to mitigate this vulnerability, I changed settings in Firefox as
> follows:
>
>
>    1. I typed *about:config* in the address bar
>    2. I found the setting *media.peerconnection.enabled*
>    3. I set it to *false*
>
> Unfortunately, this setting does not allow Firefox to work with the video
> conferencing service in question, which relies entirely on WebRTC.
>
> So I had to go back in and change the media.peerconnection.enabled setting
> to *true* in Firefox in order to get it to work with the service in
> question. While this enabled me to conference with teams that desire to use
> the video conferencing service that rely wholly upon WebRTC, it concerns me
> because of the security vulnerability.
>
> One suggested mitigation was having NoScript, but using NoScript was not
> helpful (you generally need to disable it for the site you are viewing that
> relies upon WebRTC, in order to obtain any functionality, and whether
> NoScript is off or on, WebRTC still causes the vulnerability of leaking
> information as previously described).
>
> Furthermore, upon reload of the browser, even after setting
> media.peerconnection.enabled to *true*, the video conferencing service
> wouldn't work until I installed the WebRTC 0.1.3 extension for Firefox and
> set it to expose my real IP address as well as allowing the following:
>
> a. navigator.getUserMedia
> b. window.MediaStreamTrack
> c. window.RTCPeerConnection
> d. window.RTCSessionDescription
>
> That seemed to me to be a direct consequence of the persistent and ongoing
> security vulnerability in WebRTC.
>
> I contacted the video conferencing service provider and their solution was
> simply to state that "If the peer to peer connection is of concern, you can
> utilize our premium version which will route traffic through a forwarding
> server with in our environment that handles the processing of the video and
> sound of all users in the conference and send it to each user individually
> rather than using a peer to peer connection."  In other words, they expect
> that people should have to pay in order to mitigate this WebRTC security
> vulnerability
> <https://security.stackexchange.com/questions/82129/what-is-webrtc-and-how-can-i-protect-myself-against-leaking-information-that-doe>,
> because they are unwilling to design to protect all users from it.
>
> In a similar fashion, in the second example of the browser based WebRTC
> version of a decentralized exchange, the user's financial information is
> periodically exposed as a result of the information leakage which is
> presently caused by WebRTC.  (Fortunately, this browser based WebRTC
> decentralized exchange is under development and is not currently open to
> actual usage.)
>
> Although undoubtedly this has been discussed here before, I am asking that
> WebRTC security issues be reconsidered and that a new security review be
> done to ameliorate or eliminate this problem in terms of WebRTC leaking
> this information.  It is my view that WebRTC should be further designed and
> developed with an eye to mitigating this serious problem, so that users of
> services which might rely partially or wholly upon WebRTC would not have to
> be concerned about such information leakage.
>
> Respectfully,
> Colin Gallagher
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--001a113ff9a6f5112b05505ebc94
Content-Type: text/html; charset="UTF-8"
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Colin,<div><br></div><div>This is an important topic, and =
the group has reviewed this situation in detail. The results have been writ=
ten up in <a href=3D"https://tools.ietf.org/html/draft-ietf-rtcweb-ip-handl=
ing-03">this document</a>, and to my knowledge most browsers that support W=
ebRTC currently conform to its recommendations.</div><div><br></div><div>Ho=
wever, I&#39;m not totally clear on your specific technical concern. You li=
nked to an (now outdated) stackexchange page, but could you provide more de=
tail about the specific situation you are concerned about?</div><div><br></=
div><div>Justin</div></div><div class=3D"gmail_extra"><br><div class=3D"gma=
il_quote">On Wed, May 24, 2017 at 5:35 PM, Colin Gallagher <span dir=3D"ltr=
">&lt;<a href=3D"mailto:colingallagher.rpcv@gmail.com" target=3D"_blank">co=
lingallagher.rpcv@gmail.com</a>&gt;</span> wrote:<br><blockquote class=3D"g=
mail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-l=
eft:1ex"><div dir=3D"ltr"><div><div><div><div><div><div><div><div>To:=C2=A0=
 IETF RTCWeb WG<br><br></div>From:=C2=A0 Colin Gallagher<br></div>=C2=A0=C2=
=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 <a href=3D"https://www.=
linkedin.com/in/colingallagher" target=3D"_blank">https://www.linkedin.com/=
in/<wbr>colingallagher</a><br>=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0 <a href=3D"https://lifeboat.com/ex/bios.colin.gallagher" ta=
rget=3D"_blank">https://lifeboat.com/ex/bios.<wbr>colin.gallagher</a><br><b=
r></div>Request for New Security Review of WebRTC based on observed vulnera=
bilities in certain examples<br><br></div>Dear IETF RTCWeb WG,<br><br></div=
><div>This request relates to WebRTC.=C2=A0 The WebRTC is in draft at <a hr=
ef=3D"https://w3c.github.io/webrtc-pc/" target=3D"_blank">https://w3c.githu=
b.io/webrtc-<wbr>pc/</a> and is=20
described also at=20
<a href=3D"https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API#WebR=
TC_interfaces" target=3D"_blank">https://developer.mozilla.org/<wbr>en-US/d=
ocs/Web/API/WebRTC_API#<wbr>WebRTC_interfaces</a></div><div><br></div>There=
 are certain examples where the use of WebRTC has presented security vulner=
abilities.=C2=A0 While not exhaustive, there are two I wanted to mention th=
at I had been exposed to recently.<br></div><div><br></div>1) The use of a =
video conferencing service which relies upon WebRTC (that is a well establi=
shed service)<br></div>2) The examination of a browser based WebRTC version=
 of a decentralized exchange (that is currently under development)<br><br><=
/div>In both cases the security vulnerability which is described below was =
the result of the use of WebRTC.<br><br>The problem under consideration is =
as follows:<br><br>As per the first example (the video conferencing service=
), recently
 I came upon a vendor who shall remain unnamed who uses WebRTC as the=20
backbone for their video conferencing service.=C2=A0 As a result, I found I=
=20
was unable to access the service in question without disabling some=20
security settings which I would have preferred to leave intact. <br><br>My =
system is (based on what the aforementioned service detects) Mozilla/5.0 (X=
11; Ubuntu; Linux x86_64; rv:53.0)=20
Gecko/20100101 Firefox/53.0.=C2=A0 This is Ubuntu 16.10 64-bit with Firefox=
=20
53.0.2.<br><br>However, quite some <i>long</i> while back I disabled WebRTC=
 because of the security vulnerability (described in part <a href=3D"https:=
//security.stackexchange.com/questions/82129/what-is-webrtc-and-how-can-i-p=
rotect-myself-against-leaking-information-that-doe" target=3D"_blank">here =
on information security stack exchange</a>).=C2=A0 So as to mitigate this v=
ulnerability, I changed settings in Firefox as follows:<br><br><ol><li>I ty=
ped <em>about:config</em> in the address bar</li><li>I found the setting <e=
m>media.peerconnection.enabled</em></li><li>I set it to <strong>false</stro=
ng></li></ol><p><strong></strong>Unfortunately,
 this setting does not allow Firefox to work with the video conferencing
 service in question, which relies entirely on WebRTC.<br></p><p>So I had t=
o go back in and change the media.peerconnection.enabled setting to <b>true=
</b>
 in Firefox in order to get it to work with the service in question.=20
While this enabled me to conference with teams that desire to use the=20
video conferencing service that rely wholly upon WebRTC, it concerns me=20
because of the security vulnerability.</p><p>One suggested mitigation was h=
aving NoScript, but using NoScript was not helpful (you generally need to d=
isable it for the site you are viewing that relies upon WebRTC, in order to=
 obtain any functionality, and whether NoScript is off or on, WebRTC still =
causes the vulnerability of leaking information as previously described).<b=
r></p><p>Furthermore, upon reload of the browser, even after setting media.=
peerconnection.enabled to <b>true</b>,
 the video conferencing service wouldn&#39;t work until I installed the=20
WebRTC 0.1.3 extension for Firefox and set it to expose my real IP=20
address as well as allowing the following:</p><p>a. navigator.getUserMedia<=
br>b. window.MediaStreamTrack<br>c. window.RTCPeerConnection<br>d. window.R=
TCSessionDescription<br></p><p>That seemed to me to be a direct consequence=
 of the persistent and ongoing security vulnerability in WebRTC.<br></p><p>=
I
 contacted the video conferencing service provider and their solution=20
was simply to state that &quot;If the peer to peer connection is of concern=
,=20
you can utilize our=20
premium version which will route traffic through a forwarding server=20
with in our environment that handles the processing of the video and=20
sound of all users in the conference and send it to each user=20
individually rather than using a peer to peer connection.&quot;=C2=A0 In ot=
her=20
words, they expect that people should have to pay in order to mitigate <a h=
ref=3D"https://security.stackexchange.com/questions/82129/what-is-webrtc-an=
d-how-can-i-protect-myself-against-leaking-information-that-doe" target=3D"=
_blank">this WebRTC security vulnerability</a>, because they are unwilling =
to design to protect all users from it.</p><p>In a similar fashion, in the =
second example of the browser based WebRTC version of a decentralized excha=
nge, the user&#39;s financial information is periodically exposed as a resu=
lt of the information leakage which is presently caused by WebRTC.=C2=A0 (F=
ortunately, this browser based WebRTC decentralized exchange is under devel=
opment and is not currently open to actual usage.)<br></p><p>Although
 undoubtedly this has been discussed here before, I am asking that WebRTC s=
ecurity issues be
 reconsidered and that a new security review be done to ameliorate or=20
eliminate this problem in terms of WebRTC leaking this information.=C2=A0 I=
t is my view that WebRTC should be further designed and developed with an e=
ye to mitigating this serious problem, so that users of services which migh=
t rely partially or wholly upon WebRTC would not have to be concerned about=
 such information leakage.<br></p><p>Respectfully,</p>Colin Gallagher<br><b=
r></div>
<br>______________________________<wbr>_________________<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/<wbr>listinfo/rtcweb</a><br=
>
<br></blockquote></div><br></div>

--001a113ff9a6f5112b05505ebc94--


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Justin,

It's my understanding that various browsers have made an attempt to conform
to the recommendations in the document that you have provided
<https://tools.ietf.org/html/draft-ietf-rtcweb-ip-handling-03>.
Unfortunately, at least in Firefox (the most current version) so far as I
can tell, there is a serious vulnerability.  To be more specific, the
problem was observed in my operating system with my current browser, which
is Ubuntu 16.10 64-bit with Firefox 53.0.2 (64-bit).  The browser does not
address the WebRTC problems by default.

The stackexchange page that I cited originally is actually not outdated in
the sense that its method of measuring whether or not the WebRTC
vulnerability exists on a given system is still intact and works.  The
stackexchange page suggests two pages to determine whether or not you have
leakage / WebRTC related vulnerability.  These pages are:

1) Browser leaks test for WebRTC https://browserleaks.com/webrtc
2) IP Leaks https://ipleak.net/

Currently for Browser leaks test for WebRTC (due to that I have again
changed settings in my system so that I will not have to be exposed to this
vulnerability) the settings show up on the page as:
RTCPeerConnection=C3=97False
RTCDataChannel=C3=97False
ORTC (Microsoft Edge)=C3=97False (for example....)

And under IP leaks, it currently shows "No leak, RTCPeerConnection not
available."

*That is because:*

(A) I have (again) changed my settings in about:config back to the followin=
g


   1. I typed *about:config* in the address bar
   2. I found the setting *media.peerconnection.enabled*
   3. I set it to *false  (it was previously set at true, so as to allow
   WebRTC to work, but this exposed information which I would not have want=
ed
   to be exposed)*

(B) And I also have changed my settings back to the following

in the WebRTC 0.1.3 extension for Firefox I have set it to no longer expose
my real IP address as well as no longer allowing the following:

a. navigator.getUserMedia
b. window.MediaStreamTrack
c. window.RTCPeerConnection
d. window.RTCSessionDescription

To be clear, *these mitigations that I have employed above would be
unnecessary if WebRTC had not been designed in such a manner that it so
clearly exposes so much user information.*

In the document you provided
<https://tools.ietf.org/html/draft-ietf-rtcweb-ip-handling-03>, titled
"WebRTC IP Address Handling Requirements -
draft-ietf-rtcweb-ip-handling-03," it is stated that:

*"1. If the client is behind a NAT, the client's private IP addresses,
typically [RFC1918 <https://tools.ietf.org/html/rfc1918>] addresses, can be
learned. *



* 2.  If the client tries to hide its physical location through a VPN,
       and the VPN and local OS support routing over multiple interfaces
       (i.e., a "split-tunnel" VPN), WebRTC will discover the public
       address for the VPN as well as the ISP public address that the
       VPN runs over.

   3.  If the client is behind a proxy (a client-configured "classical
       application proxy", as defined in [RFC1919], Section 3
<https://tools.ietf.org/html/rfc1919#section-3>), but
       direct access to the Internet is also supported, WebRTC's STUN
       [RFC5389 <https://tools.ietf.org/html/rfc5389>]checks will
bypass the proxy and reveal the public
       address of the client."*

All of these are obvious vulnerabilities and serious problems (not
only #2 as the document suggests).

It is stated in the document that,



*"we want to avoid solutions that disable WebRTC or make it harder to use."=
*

Yet, so far as my observations of WebRTC in the current version of
Firefox with Ubuntu 16.10 have shown,

there appears to be no other choice other than to disable WebRTC
because its use completely compromises

privacy and thus makes all other aspects of your system more difficult
generally.

It is stated in the document you cited that the goals here are to
"Provide a privacy-friendly default behavior which strikes the
      right balance between privacy and media performance for most users
      and use cases. (and) For users who care more about one versus
the other, provide a
      means to customize the experience."

This has clearlly not been accomplished in the context of Ubuntu 16.10
64-bit with Firefox 53.0.2 (64-bit).

Hence my request for a new security review of WebRTC based on observed
vulnerabilities.




On Thu, May 25, 2017 at 1:04 PM, Justin Uberti <juberti@google.com> wrote:

> Colin,
>
> This is an important topic, and the group has reviewed this situation in
> detail. The results have been written up in this document
> <https://tools.ietf.org/html/draft-ietf-rtcweb-ip-handling-03>, and to my
> knowledge most browsers that support WebRTC currently conform to its
> recommendations.
>
> However, I'm not totally clear on your specific technical concern. You
> linked to an (now outdated) stackexchange page, but could you provide mor=
e
> detail about the specific situation you are concerned about?
>
> Justin
>
> On Wed, May 24, 2017 at 5:35 PM, Colin Gallagher <
> colingallagher.rpcv@gmail.com> wrote:
>
>> To:  IETF RTCWeb WG
>>
>> From:  Colin Gallagher
>>            https://www.linkedin.com/in/colingallagher
>>            https://lifeboat.com/ex/bios.colin.gallagher
>>
>> Request for New Security Review of WebRTC based on observed
>> vulnerabilities in certain examples
>>
>> Dear IETF RTCWeb WG,
>>
>> This request relates to WebRTC.  The WebRTC is in draft at
>> https://w3c.github.io/webrtc-pc/ and is described also at
>> https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API#
>> WebRTC_interfaces
>>
>> There are certain examples where the use of WebRTC has presented securit=
y
>> vulnerabilities.  While not exhaustive, there are two I wanted to mentio=
n
>> that I had been exposed to recently.
>>
>> 1) The use of a video conferencing service which relies upon WebRTC (tha=
t
>> is a well established service)
>> 2) The examination of a browser based WebRTC version of a decentralized
>> exchange (that is currently under development)
>>
>> In both cases the security vulnerability which is described below was th=
e
>> result of the use of WebRTC.
>>
>> The problem under consideration is as follows:
>>
>> As per the first example (the video conferencing service), recently I
>> came upon a vendor who shall remain unnamed who uses WebRTC as the backb=
one
>> for their video conferencing service.  As a result, I found I was unable=
 to
>> access the service in question without disabling some security settings
>> which I would have preferred to leave intact.
>>
>> My system is (based on what the aforementioned service detects)
>> Mozilla/5.0 (X11; Ubuntu; Linux x86_64; rv:53.0) Gecko/20100101
>> Firefox/53.0.  This is Ubuntu 16.10 64-bit with Firefox 53.0.2.
>>
>> However, quite some *long* while back I disabled WebRTC because of the
>> security vulnerability (described in part here on information security
>> stack exchange
>> <https://security.stackexchange.com/questions/82129/what-is-webrtc-and-h=
ow-can-i-protect-myself-against-leaking-information-that-doe>).
>> So as to mitigate this vulnerability, I changed settings in Firefox as
>> follows:
>>
>>
>>    1. I typed *about:config* in the address bar
>>    2. I found the setting *media.peerconnection.enabled*
>>    3. I set it to *false*
>>
>> Unfortunately, this setting does not allow Firefox to work with the vide=
o
>> conferencing service in question, which relies entirely on WebRTC.
>>
>> So I had to go back in and change the media.peerconnection.enabled
>> setting to *true* in Firefox in order to get it to work with the service
>> in question. While this enabled me to conference with teams that desire =
to
>> use the video conferencing service that rely wholly upon WebRTC, it
>> concerns me because of the security vulnerability.
>>
>> One suggested mitigation was having NoScript, but using NoScript was not
>> helpful (you generally need to disable it for the site you are viewing t=
hat
>> relies upon WebRTC, in order to obtain any functionality, and whether
>> NoScript is off or on, WebRTC still causes the vulnerability of leaking
>> information as previously described).
>>
>> Furthermore, upon reload of the browser, even after setting
>> media.peerconnection.enabled to *true*, the video conferencing service
>> wouldn't work until I installed the WebRTC 0.1.3 extension for Firefox a=
nd
>> set it to expose my real IP address as well as allowing the following:
>>
>> a. navigator.getUserMedia
>> b. window.MediaStreamTrack
>> c. window.RTCPeerConnection
>> d. window.RTCSessionDescription
>>
>> That seemed to me to be a direct consequence of the persistent and
>> ongoing security vulnerability in WebRTC.
>>
>> I contacted the video conferencing service provider and their solution
>> was simply to state that "If the peer to peer connection is of concern, =
you
>> can utilize our premium version which will route traffic through a
>> forwarding server with in our environment that handles the processing of
>> the video and sound of all users in the conference and send it to each u=
ser
>> individually rather than using a peer to peer connection."  In other wor=
ds,
>> they expect that people should have to pay in order to mitigate this
>> WebRTC security vulnerability
>> <https://security.stackexchange.com/questions/82129/what-is-webrtc-and-h=
ow-can-i-protect-myself-against-leaking-information-that-doe>,
>> because they are unwilling to design to protect all users from it.
>>
>> In a similar fashion, in the second example of the browser based WebRTC
>> version of a decentralized exchange, the user's financial information is
>> periodically exposed as a result of the information leakage which is
>> presently caused by WebRTC.  (Fortunately, this browser based WebRTC
>> decentralized exchange is under development and is not currently open to
>> actual usage.)
>>
>> Although undoubtedly this has been discussed here before, I am asking
>> that WebRTC security issues be reconsidered and that a new security revi=
ew
>> be done to ameliorate or eliminate this problem in terms of WebRTC leaki=
ng
>> this information.  It is my view that WebRTC should be further designed =
and
>> developed with an eye to mitigating this serious problem, so that users =
of
>> services which might rely partially or wholly upon WebRTC would not have=
 to
>> be concerned about such information leakage.
>>
>> Respectfully,
>> Colin Gallagher
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>>
>

--94eb2c1b7fd6c912f5055068b2a2
Content-Type: text/html; charset="UTF-8"
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div><div><div><div><div><div><div><div>Justin,<br><br></d=
iv>It&#39;s my understanding that various browsers have made an attempt to =
conform to the recommendations in <a href=3D"https://tools.ietf.org/html/dr=
aft-ietf-rtcweb-ip-handling-03">the document that you have provided</a>.=C2=
=A0 Unfortunately, at least in Firefox (the most current version) so far as=
 I can tell, there is a serious vulnerability.=C2=A0 To be more specific, t=
he problem was observed in my operating system with my current browser, whi=
ch is Ubuntu 16.10 64-bit with Firefox=20
53.0.2 (64-bit).=C2=A0 The browser does not address the WebRTC problems by =
default.<br><br></div>The stackexchange page that I cited originally is act=
ually not outdated in the sense that its method of measuring whether or not=
 the WebRTC vulnerability exists on a given system is still intact and work=
s.=C2=A0 The stackexchange page suggests two pages to determine whether or =
not you have leakage / WebRTC related vulnerability.=C2=A0 These pages are:=
<br><br></div>1) Browser leaks test for WebRTC <a href=3D"https://browserle=
aks.com/webrtc">https://browserleaks.com/webrtc</a><br></div>2) IP Leaks <a=
 href=3D"https://ipleak.net/">https://ipleak.net/</a><br><br></div>Currentl=
y for Browser leaks test for WebRTC (due to that I have again changed setti=
ngs in my system so that I will not have to be exposed to this vulnerabilit=
y) the settings show up on the page as:<br>RTCPeerConnection<span class=3D"=
gmail-bad">=C3=97</span>False<br>RTCDataChannel<span class=3D"gmail-bad">=
=C3=97</span>False<br>ORTC (Microsoft Edge)<span class=3D"gmail-bad">=C3=97=
</span>False (for example....)<br><br></div>And under IP leaks, it currentl=
y shows &quot;No leak, RTCPeerConnection not available.&quot;=C2=A0 <br><br=
></div><i>That is because:</i><br><br></div>(A) I have (again) changed my s=
ettings in about:config back to the following<br><br><ol><li>I typed <em>ab=
out:config</em> in the address bar</li><li>I found the setting <em>media.pe=
erconnection.enabled</em></li><li>I set it to <strong>false=C2=A0 (it was p=
reviously set at true, so as to allow WebRTC to work, but this exposed info=
rmation which I would not have wanted to be exposed)</strong></li></ol><p>(=
B) And I also have changed my settings back to the following</p><p>in the W=
ebRTC 0.1.3 extension for Firefox I have set it to no longer expose my real=
 IP=20
address as well as no longer allowing the following:</p><p>a. navigator.get=
UserMedia<br>b. window.MediaStreamTrack<br>c. window.RTCPeerConnection<br>d=
. window.RTCSessionDescription</p><p>To be clear, <b>these mitigations that=
 I have employed above would be unnecessary if WebRTC had not been designed=
 in such a manner that it so clearly exposes so much user information.</b><=
/p><p>In <a href=3D"https://tools.ietf.org/html/draft-ietf-rtcweb-ip-handli=
ng-03">the document you provided</a>, titled &quot;WebRTC IP Address Handli=
ng Requirements - draft-ietf-rtcweb-ip-handling-03,&quot; it is stated that=
:</p><p><i>&quot;<span style=3D"font-family:arial,helvetica,sans-serif">1. =
 If the client is behind a NAT, the client&#39;s private IP addresses, typi=
cally [<a href=3D"https://tools.ietf.org/html/rfc1918" title=3D"&quot;Addre=
ss Allocation for Private Internets&quot;">RFC1918</a>] addresses, can be l=
earned.
</span></i></p><pre class=3D"gmail-newpage"><i><span style=3D"font-family:a=
rial,helvetica,sans-serif"> 2.  If the client tries to hide its physical lo=
cation through a VPN,
       and the VPN and local OS support routing over multiple interfaces
       (i.e., a &quot;split-tunnel&quot; VPN), WebRTC will discover the pub=
lic
       address for the VPN as well as the ISP public address that the
       VPN runs over.

   3.  If the client is behind a proxy (a client-configured &quot;classical
       application proxy&quot;, as defined in <a href=3D"https://tools.ietf=
.org/html/rfc1919#section-3">[RFC1919], Section=C2=A03</a>), but
       direct access to the Internet is also supported, WebRTC&#39;s STUN
       [<a href=3D"https://tools.ietf.org/html/rfc5389" title=3D"&quot;Sess=
ion Traversal Utilities for NAT (STUN)&quot;">RFC5389</a>]checks will bypas=
s the proxy and reveal the public
       address of the client.&quot;<br><br></span></i></pre><pre class=3D"g=
mail-newpage"><span style=3D"font-family:arial,helvetica,sans-serif">All of=
 these are obvious vulnerabilities and serious problems (not only #2 as the=
 document suggests).<br><br></span></pre><pre class=3D"gmail-newpage"><span=
 style=3D"font-family:arial,helvetica,sans-serif">It is stated in the docum=
ent that, <br><i><br>&quot;we want to avoid solutions that disable WebRTC o=
r make it harder to use.&quot;<br><br></i></span></pre><pre class=3D"gmail-=
newpage"><span style=3D"font-family:arial,helvetica,sans-serif">Yet, so far=
 as my observations of WebRTC in the current version of Firefox with Ubuntu=
 16.10 have shown,<br></span></pre><pre class=3D"gmail-newpage"><span style=
=3D"font-family:arial,helvetica,sans-serif">there appears to be no other ch=
oice other than to disable WebRTC because its use completely compromises <b=
r></span></pre><pre class=3D"gmail-newpage"><span style=3D"font-family:aria=
l,helvetica,sans-serif">privacy and thus makes all other aspects of your sy=
stem more difficult generally.  <br><br></span></pre><pre class=3D"gmail-ne=
wpage"><span style=3D"font-family:arial,helvetica,sans-serif">It is stated =
in the document you cited that the goals here are to &quot;</span>Provide a=
 privacy-friendly default behavior which strikes the
      right balance between privacy and media performance for most users
      and use cases. (and) For users who care more about one versus the oth=
er, provide a
      means to customize the experience.&quot;</pre><pre class=3D"gmail-new=
page"><span style=3D"font-family:arial,helvetica,sans-serif">This has clear=
lly not been accomplished in the context of Ubuntu 16.10 64-bit with Firefo=
x 53.0.2 (64-bit).<br><br></span></pre><pre class=3D"gmail-newpage"><span s=
tyle=3D"font-family:arial,helvetica,sans-serif">Hence my request for a new =
security review of WebRTC based on observed vulnerabilities.<br></span></pr=
e><p><span style=3D"font-family:arial,helvetica,sans-serif"></span><br></p>=
<strong></strong><div><div><div><div><div><br></div></div></div></div></div=
></div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Thu, Ma=
y 25, 2017 at 1:04 PM, Justin Uberti <span dir=3D"ltr">&lt;<a href=3D"mailt=
o:juberti@google.com" target=3D"_blank">juberti@google.com</a>&gt;</span> w=
rote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;borde=
r-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr">Colin,<div><br></d=
iv><div>This is an important topic, and the group has reviewed this situati=
on in detail. The results have been written up in <a href=3D"https://tools.=
ietf.org/html/draft-ietf-rtcweb-ip-handling-03" target=3D"_blank">this docu=
ment</a>, and to my knowledge most browsers that support WebRTC currently c=
onform to its recommendations.</div><div><br></div><div>However, I&#39;m no=
t totally clear on your specific technical concern. You linked to an (now o=
utdated) stackexchange page, but could you provide more detail about the sp=
ecific situation you are concerned about?</div><div><br></div><div>Justin</=
div></div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote"><div><d=
iv class=3D"h5">On Wed, May 24, 2017 at 5:35 PM, Colin Gallagher <span dir=
=3D"ltr">&lt;<a href=3D"mailto:colingallagher.rpcv@gmail.com" target=3D"_bl=
ank">colingallagher.rpcv@gmail.com</a><wbr>&gt;</span> wrote:<br></div></di=
v><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:=
1px #ccc solid;padding-left:1ex"><div><div class=3D"h5"><div dir=3D"ltr"><d=
iv><div><div><div><div><div><div><div>To:=C2=A0 IETF RTCWeb WG<br><br></div=
>From:=C2=A0 Colin Gallagher<br></div>=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0 <a href=3D"https://www.linkedin.com/in/colingallag=
her" target=3D"_blank">https://www.linkedin.com/in/co<wbr>lingallagher</a><=
br>=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 <a href=3D"=
https://lifeboat.com/ex/bios.colin.gallagher" target=3D"_blank">https://lif=
eboat.com/ex/bios.c<wbr>olin.gallagher</a><br><br></div>Request for New Sec=
urity Review of WebRTC based on observed vulnerabilities in certain example=
s<br><br></div>Dear IETF RTCWeb WG,<br><br></div><div>This request relates =
to WebRTC.=C2=A0 The WebRTC is in draft at <a href=3D"https://w3c.github.io=
/webrtc-pc/" target=3D"_blank">https://w3c.github.io/webrtc-p<wbr>c/</a> an=
d is=20
described also at=20
<a href=3D"https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API#WebR=
TC_interfaces" target=3D"_blank">https://developer.mozilla.org/<wbr>en-US/d=
ocs/Web/API/WebRTC_API#<wbr>WebRTC_interfaces</a></div><div><br></div>There=
 are certain examples where the use of WebRTC has presented security vulner=
abilities.=C2=A0 While not exhaustive, there are two I wanted to mention th=
at I had been exposed to recently.<br></div><div><br></div>1) The use of a =
video conferencing service which relies upon WebRTC (that is a well establi=
shed service)<br></div>2) The examination of a browser based WebRTC version=
 of a decentralized exchange (that is currently under development)<br><br><=
/div>In both cases the security vulnerability which is described below was =
the result of the use of WebRTC.<br><br>The problem under consideration is =
as follows:<br><br>As per the first example (the video conferencing service=
), recently
 I came upon a vendor who shall remain unnamed who uses WebRTC as the=20
backbone for their video conferencing service.=C2=A0 As a result, I found I=
=20
was unable to access the service in question without disabling some=20
security settings which I would have preferred to leave intact. <br><br>My =
system is (based on what the aforementioned service detects) Mozilla/5.0 (X=
11; Ubuntu; Linux x86_64; rv:53.0)=20
Gecko/20100101 Firefox/53.0.=C2=A0 This is Ubuntu 16.10 64-bit with Firefox=
=20
53.0.2.<br><br>However, quite some <i>long</i> while back I disabled WebRTC=
 because of the security vulnerability (described in part <a href=3D"https:=
//security.stackexchange.com/questions/82129/what-is-webrtc-and-how-can-i-p=
rotect-myself-against-leaking-information-that-doe" target=3D"_blank">here =
on information security stack exchange</a>).=C2=A0 So as to mitigate this v=
ulnerability, I changed settings in Firefox as follows:<br><br><ol><li>I ty=
ped <em>about:config</em> in the address bar</li><li>I found the setting <e=
m>media.peerconnection.enabled</em></li><li>I set it to <strong>false</stro=
ng></li></ol><p><strong></strong>Unfortunately,
 this setting does not allow Firefox to work with the video conferencing
 service in question, which relies entirely on WebRTC.<br></p><p>So I had t=
o go back in and change the media.peerconnection.enabled setting to <b>true=
</b>
 in Firefox in order to get it to work with the service in question.=20
While this enabled me to conference with teams that desire to use the=20
video conferencing service that rely wholly upon WebRTC, it concerns me=20
because of the security vulnerability.</p><p>One suggested mitigation was h=
aving NoScript, but using NoScript was not helpful (you generally need to d=
isable it for the site you are viewing that relies upon WebRTC, in order to=
 obtain any functionality, and whether NoScript is off or on, WebRTC still =
causes the vulnerability of leaking information as previously described).<b=
r></p><p>Furthermore, upon reload of the browser, even after setting media.=
peerconnection.enabled to <b>true</b>,
 the video conferencing service wouldn&#39;t work until I installed the=20
WebRTC 0.1.3 extension for Firefox and set it to expose my real IP=20
address as well as allowing the following:</p><p>a. navigator.getUserMedia<=
br>b. window.MediaStreamTrack<br>c. window.RTCPeerConnection<br>d. window.R=
TCSessionDescription<br></p><p>That seemed to me to be a direct consequence=
 of the persistent and ongoing security vulnerability in WebRTC.<br></p><p>=
I
 contacted the video conferencing service provider and their solution=20
was simply to state that &quot;If the peer to peer connection is of concern=
,=20
you can utilize our=20
premium version which will route traffic through a forwarding server=20
with in our environment that handles the processing of the video and=20
sound of all users in the conference and send it to each user=20
individually rather than using a peer to peer connection.&quot;=C2=A0 In ot=
her=20
words, they expect that people should have to pay in order to mitigate <a h=
ref=3D"https://security.stackexchange.com/questions/82129/what-is-webrtc-an=
d-how-can-i-protect-myself-against-leaking-information-that-doe" target=3D"=
_blank">this WebRTC security vulnerability</a>, because they are unwilling =
to design to protect all users from it.</p><p>In a similar fashion, in the =
second example of the browser based WebRTC version of a decentralized excha=
nge, the user&#39;s financial information is periodically exposed as a resu=
lt of the information leakage which is presently caused by WebRTC.=C2=A0 (F=
ortunately, this browser based WebRTC decentralized exchange is under devel=
opment and is not currently open to actual usage.)<br></p><p>Although
 undoubtedly this has been discussed here before, I am asking that WebRTC s=
ecurity issues be
 reconsidered and that a new security review be done to ameliorate or=20
eliminate this problem in terms of WebRTC leaking this information.=C2=A0 I=
t is my view that WebRTC should be further designed and developed with an e=
ye to mitigating this serious problem, so that users of services which migh=
t rely partially or wholly upon WebRTC would not have to be concerned about=
 such information leakage.<br></p><p>Respectfully,</p>Colin Gallagher<br><b=
r></div>
<br></div></div>______________________________<wbr>_________________<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/l<wbr>istinfo/rtcweb</a><br=
>
<br></blockquote></div><br></div>
</blockquote></div><br></div>

--94eb2c1b7fd6c912f5055068b2a2--


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Date: Fri, 26 May 2017 20:48:03 +0800
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To: Colin Gallagher <colingallagher.rpcv@gmail.com>
Cc: Justin Uberti <juberti@google.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
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Subject: Re: [rtcweb] Request for New Security Review of WebRTC based on observed vulnerabilities
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Colin,

I'm not really sure what you think the new information is here.

The draft that Justin referred to reflects the consensus view of the RTCWEB
WG
about the best balance between privacy and usability, and yes, it does
result in
the server learning your default RFC 1918 address, so it's not surprising
that
accounts for the results you are seeing with Firefox. If you think there's
a defect
where Firefox isn't conforming to the spec, then you should file a bug, but
otherwise I'm not really seeing what there is for the WG to reconsider

-Ekr


On Fri, May 26, 2017 at 3:58 PM, Colin Gallagher <
colingallagher.rpcv@gmail.com> wrote:

> Justin,
>
> It's my understanding that various browsers have made an attempt to
> conform to the recommendations in the document that you have provided
> <https://tools.ietf.org/html/draft-ietf-rtcweb-ip-handling-03>.
> Unfortunately, at least in Firefox (the most current version) so far as I
> can tell, there is a serious vulnerability.  To be more specific, the
> problem was observed in my operating system with my current browser, whic=
h
> is Ubuntu 16.10 64-bit with Firefox 53.0.2 (64-bit).  The browser does no=
t
> address the WebRTC problems by default.
>
> The stackexchange page that I cited originally is actually not outdated i=
n
> the sense that its method of measuring whether or not the WebRTC
> vulnerability exists on a given system is still intact and works.  The
> stackexchange page suggests two pages to determine whether or not you hav=
e
> leakage / WebRTC related vulnerability.  These pages are:
>
> 1) Browser leaks test for WebRTC https://browserleaks.com/webrtc
> 2) IP Leaks https://ipleak.net/
>
> Currently for Browser leaks test for WebRTC (due to that I have again
> changed settings in my system so that I will not have to be exposed to th=
is
> vulnerability) the settings show up on the page as:
> RTCPeerConnection=C3=97False
> RTCDataChannel=C3=97False
> ORTC (Microsoft Edge)=C3=97False (for example....)
>
> And under IP leaks, it currently shows "No leak, RTCPeerConnection not
> available."
>
> *That is because:*
>
> (A) I have (again) changed my settings in about:config back to the
> following
>
>
>    1. I typed *about:config* in the address bar
>    2. I found the setting *media.peerconnection.enabled*
>    3. I set it to *false  (it was previously set at true, so as to allow
>    WebRTC to work, but this exposed information which I would not have wa=
nted
>    to be exposed)*
>
> (B) And I also have changed my settings back to the following
>
> in the WebRTC 0.1.3 extension for Firefox I have set it to no longer
> expose my real IP address as well as no longer allowing the following:
>
> a. navigator.getUserMedia
> b. window.MediaStreamTrack
> c. window.RTCPeerConnection
> d. window.RTCSessionDescription
>
> To be clear, *these mitigations that I have employed above would be
> unnecessary if WebRTC had not been designed in such a manner that it so
> clearly exposes so much user information.*
>
> In the document you provided
> <https://tools.ietf.org/html/draft-ietf-rtcweb-ip-handling-03>, titled
> "WebRTC IP Address Handling Requirements - draft-ietf-rtcweb-ip-handling-=
03,"
> it is stated that:
>
> *"1. If the client is behind a NAT, the client's private IP addresses,
> typically [RFC1918 <https://tools.ietf.org/html/rfc1918>] addresses, can =
be
> learned. *
>
>
>
> * 2.  If the client tries to hide its physical location through a VPN,
>        and the VPN and local OS support routing over multiple interfaces
>        (i.e., a "split-tunnel" VPN), WebRTC will discover the public
>        address for the VPN as well as the ISP public address that the
>        VPN runs over.
>
>    3.  If the client is behind a proxy (a client-configured "classical
>        application proxy", as defined in [RFC1919], Section 3 <https://to=
ols.ietf.org/html/rfc1919#section-3>), but
>        direct access to the Internet is also supported, WebRTC's STUN
>        [RFC5389 <https://tools.ietf.org/html/rfc5389>]checks will bypass =
the proxy and reveal the public
>        address of the client."*
>
> All of these are obvious vulnerabilities and serious problems (not only #=
2 as the document suggests).
>
> It is stated in the document that,
>
>
>
> *"we want to avoid solutions that disable WebRTC or make it harder to use=
."*
>
> Yet, so far as my observations of WebRTC in the current version of Firefo=
x with Ubuntu 16.10 have shown,
>
> there appears to be no other choice other than to disable WebRTC because =
its use completely compromises
>
> privacy and thus makes all other aspects of your system more difficult ge=
nerally.
>
> It is stated in the document you cited that the goals here are to "Provid=
e a privacy-friendly default behavior which strikes the
>       right balance between privacy and media performance for most users
>       and use cases. (and) For users who care more about one versus the o=
ther, provide a
>       means to customize the experience."
>
> This has clearlly not been accomplished in the context of Ubuntu 16.10 64=
-bit with Firefox 53.0.2 (64-bit).
>
> Hence my request for a new security review of WebRTC based on observed vu=
lnerabilities.
>
>
>
>
> On Thu, May 25, 2017 at 1:04 PM, Justin Uberti <juberti@google.com> wrote=
:
>
>> Colin,
>>
>> This is an important topic, and the group has reviewed this situation in
>> detail. The results have been written up in this document
>> <https://tools.ietf.org/html/draft-ietf-rtcweb-ip-handling-03>, and to
>> my knowledge most browsers that support WebRTC currently conform to its
>> recommendations.
>>
>> However, I'm not totally clear on your specific technical concern. You
>> linked to an (now outdated) stackexchange page, but could you provide mo=
re
>> detail about the specific situation you are concerned about?
>>
>> Justin
>>
>> On Wed, May 24, 2017 at 5:35 PM, Colin Gallagher <
>> colingallagher.rpcv@gmail.com> wrote:
>>
>>> To:  IETF RTCWeb WG
>>>
>>> From:  Colin Gallagher
>>>            https://www.linkedin.com/in/colingallagher
>>>            https://lifeboat.com/ex/bios.colin.gallagher
>>>
>>> Request for New Security Review of WebRTC based on observed
>>> vulnerabilities in certain examples
>>>
>>> Dear IETF RTCWeb WG,
>>>
>>> This request relates to WebRTC.  The WebRTC is in draft at
>>> https://w3c.github.io/webrtc-pc/ and is described also at
>>> https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API#
>>> WebRTC_interfaces
>>>
>>> There are certain examples where the use of WebRTC has presented
>>> security vulnerabilities.  While not exhaustive, there are two I wanted=
 to
>>> mention that I had been exposed to recently.
>>>
>>> 1) The use of a video conferencing service which relies upon WebRTC
>>> (that is a well established service)
>>> 2) The examination of a browser based WebRTC version of a decentralized
>>> exchange (that is currently under development)
>>>
>>> In both cases the security vulnerability which is described below was
>>> the result of the use of WebRTC.
>>>
>>> The problem under consideration is as follows:
>>>
>>> As per the first example (the video conferencing service), recently I
>>> came upon a vendor who shall remain unnamed who uses WebRTC as the back=
bone
>>> for their video conferencing service.  As a result, I found I was unabl=
e to
>>> access the service in question without disabling some security settings
>>> which I would have preferred to leave intact.
>>>
>>> My system is (based on what the aforementioned service detects)
>>> Mozilla/5.0 (X11; Ubuntu; Linux x86_64; rv:53.0) Gecko/20100101
>>> Firefox/53.0.  This is Ubuntu 16.10 64-bit with Firefox 53.0.2.
>>>
>>> However, quite some *long* while back I disabled WebRTC because of the
>>> security vulnerability (described in part here on information security
>>> stack exchange
>>> <https://security.stackexchange.com/questions/82129/what-is-webrtc-and-=
how-can-i-protect-myself-against-leaking-information-that-doe>).
>>> So as to mitigate this vulnerability, I changed settings in Firefox as
>>> follows:
>>>
>>>
>>>    1. I typed *about:config* in the address bar
>>>    2. I found the setting *media.peerconnection.enabled*
>>>    3. I set it to *false*
>>>
>>> Unfortunately, this setting does not allow Firefox to work with the
>>> video conferencing service in question, which relies entirely on WebRTC=
.
>>>
>>> So I had to go back in and change the media.peerconnection.enabled
>>> setting to *true* in Firefox in order to get it to work with the
>>> service in question. While this enabled me to conference with teams tha=
t
>>> desire to use the video conferencing service that rely wholly upon WebR=
TC,
>>> it concerns me because of the security vulnerability.
>>>
>>> One suggested mitigation was having NoScript, but using NoScript was no=
t
>>> helpful (you generally need to disable it for the site you are viewing =
that
>>> relies upon WebRTC, in order to obtain any functionality, and whether
>>> NoScript is off or on, WebRTC still causes the vulnerability of leaking
>>> information as previously described).
>>>
>>> Furthermore, upon reload of the browser, even after setting
>>> media.peerconnection.enabled to *true*, the video conferencing service
>>> wouldn't work until I installed the WebRTC 0.1.3 extension for Firefox =
and
>>> set it to expose my real IP address as well as allowing the following:
>>>
>>> a. navigator.getUserMedia
>>> b. window.MediaStreamTrack
>>> c. window.RTCPeerConnection
>>> d. window.RTCSessionDescription
>>>
>>> That seemed to me to be a direct consequence of the persistent and
>>> ongoing security vulnerability in WebRTC.
>>>
>>> I contacted the video conferencing service provider and their solution
>>> was simply to state that "If the peer to peer connection is of concern,=
 you
>>> can utilize our premium version which will route traffic through a
>>> forwarding server with in our environment that handles the processing o=
f
>>> the video and sound of all users in the conference and send it to each =
user
>>> individually rather than using a peer to peer connection."  In other wo=
rds,
>>> they expect that people should have to pay in order to mitigate this
>>> WebRTC security vulnerability
>>> <https://security.stackexchange.com/questions/82129/what-is-webrtc-and-=
how-can-i-protect-myself-against-leaking-information-that-doe>,
>>> because they are unwilling to design to protect all users from it.
>>>
>>> In a similar fashion, in the second example of the browser based WebRTC
>>> version of a decentralized exchange, the user's financial information i=
s
>>> periodically exposed as a result of the information leakage which is
>>> presently caused by WebRTC.  (Fortunately, this browser based WebRTC
>>> decentralized exchange is under development and is not currently open t=
o
>>> actual usage.)
>>>
>>> Although undoubtedly this has been discussed here before, I am asking
>>> that WebRTC security issues be reconsidered and that a new security rev=
iew
>>> be done to ameliorate or eliminate this problem in terms of WebRTC leak=
ing
>>> this information.  It is my view that WebRTC should be further designed=
 and
>>> developed with an eye to mitigating this serious problem, so that users=
 of
>>> services which might rely partially or wholly upon WebRTC would not hav=
e to
>>> be concerned about such information leakage.
>>>
>>> Respectfully,
>>> Colin Gallagher
>>>
>>>
>>> _______________________________________________
>>> 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">Colin,<div><br></div><div>I&#39;m not really sure what you=
 think the new information is here.</div><div><br></div><div>The draft that=
 Justin referred to reflects the consensus view of the RTCWEB WG</div><div>=
about the best balance between privacy and usability, and yes, it does resu=
lt in</div><div>the server learning your default RFC 1918 address, so it&#3=
9;s not surprising that</div><div>accounts for the results you are seeing w=
ith Firefox. If you think there&#39;s a defect</div><div>where Firefox isn&=
#39;t conforming to the spec, then you should file a bug, but</div><div>oth=
erwise I&#39;m not really seeing what there is for the WG to reconsider</di=
v><div><br></div><div>-Ekr</div><div><br></div></div><div class=3D"gmail_ex=
tra"><br><div class=3D"gmail_quote">On Fri, May 26, 2017 at 3:58 PM, Colin =
Gallagher <span dir=3D"ltr">&lt;<a href=3D"mailto:colingallagher.rpcv@gmail=
.com" target=3D"_blank">colingallagher.rpcv@gmail.com</a>&gt;</span> wrote:=
<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-lef=
t:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div><div><div><div><di=
v><div><div><div>Justin,<br><br></div>It&#39;s my understanding that variou=
s browsers have made an attempt to conform to the recommendations in <a hre=
f=3D"https://tools.ietf.org/html/draft-ietf-rtcweb-ip-handling-03" target=
=3D"_blank">the document that you have provided</a>.=C2=A0 Unfortunately, a=
t least in Firefox (the most current version) so far as I can tell, there i=
s a serious vulnerability.=C2=A0 To be more specific, the problem was obser=
ved in my operating system with my current browser, which is Ubuntu 16.10 6=
4-bit with Firefox=20
53.0.2 (64-bit).=C2=A0 The browser does not address the WebRTC problems by =
default.<br><br></div>The stackexchange page that I cited originally is act=
ually not outdated in the sense that its method of measuring whether or not=
 the WebRTC vulnerability exists on a given system is still intact and work=
s.=C2=A0 The stackexchange page suggests two pages to determine whether or =
not you have leakage / WebRTC related vulnerability.=C2=A0 These pages are:=
<br><br></div>1) Browser leaks test for WebRTC <a href=3D"https://browserle=
aks.com/webrtc" target=3D"_blank">https://browserleaks.com/<wbr>webrtc</a><=
br></div>2) IP Leaks <a href=3D"https://ipleak.net/" target=3D"_blank">http=
s://ipleak.net/</a><br><br></div>Currently for Browser leaks test for WebRT=
C (due to that I have again changed settings in my system so that I will no=
t have to be exposed to this vulnerability) the settings show up on the pag=
e as:<br>RTCPeerConnection<span class=3D"m_3974075263048630147gmail-bad">=
=C3=97</span>False<br>RTCDataChannel<span class=3D"m_3974075263048630147gma=
il-bad">=C3=97</span>False<br>ORTC (Microsoft Edge)<span class=3D"m_3974075=
263048630147gmail-bad">=C3=97</span>False (for example....)<br><br></div>An=
d under IP leaks, it currently shows &quot;No leak, RTCPeerConnection not a=
vailable.&quot;=C2=A0 <br><br></div><i>That is because:</i><br><br></div>(A=
) I have (again) changed my settings in about:config back to the following<=
br><br><ol><span class=3D""><li>I typed <em>about:config</em> in the addres=
s bar</li><li>I found the setting <em>media.peerconnection.enabled</em></li=
></span><li>I set it to <strong>false=C2=A0 (it was previously set at true,=
 so as to allow WebRTC to work, but this exposed information which I would =
not have wanted to be exposed)</strong></li></ol><p>(B) And I also have cha=
nged my settings back to the following</p><p>in the WebRTC 0.1.3 extension =
for Firefox I have set it to no longer expose my real IP=20
address as well as no longer allowing the following:</p><span class=3D""><p=
>a. navigator.getUserMedia<br>b. window.MediaStreamTrack<br>c. window.RTCPe=
erConnection<br>d. window.RTCSessionDescription</p></span><p>To be clear, <=
b>these mitigations that I have employed above would be unnecessary if WebR=
TC had not been designed in such a manner that it so clearly exposes so muc=
h user information.</b></p><p>In <a href=3D"https://tools.ietf.org/html/dra=
ft-ietf-rtcweb-ip-handling-03" target=3D"_blank">the document you provided<=
/a>, titled &quot;WebRTC IP Address Handling Requirements - draft-ietf-rtcw=
eb-ip-handling-<wbr>03,&quot; it is stated that:</p><p><i>&quot;<span style=
=3D"font-family:arial,helvetica,sans-serif">1.  If the client is behind a N=
AT, the client&#39;s private IP addresses, typically [<a href=3D"https://to=
ols.ietf.org/html/rfc1918" title=3D"&quot;Address Allocation for Private In=
ternets&quot;" target=3D"_blank">RFC1918</a>] addresses, can be learned.
</span></i></p><pre class=3D"m_3974075263048630147gmail-newpage"><i><span s=
tyle=3D"font-family:arial,helvetica,sans-serif"> 2.  If the client tries to=
 hide its physical location through a VPN,
       and the VPN and local OS support routing over multiple interfaces
       (i.e., a &quot;split-tunnel&quot; VPN), WebRTC will discover the pub=
lic
       address for the VPN as well as the ISP public address that the
       VPN runs over.

   3.  If the client is behind a proxy (a client-configured &quot;classical
       application proxy&quot;, as defined in <a href=3D"https://tools.ietf=
.org/html/rfc1919#section-3" target=3D"_blank">[RFC1919], Section=C2=A03</a=
>), but
       direct access to the Internet is also supported, WebRTC&#39;s STUN
       [<a href=3D"https://tools.ietf.org/html/rfc5389" title=3D"&quot;Sess=
ion Traversal Utilities for NAT (STUN)&quot;" target=3D"_blank">RFC5389</a>=
]checks will bypass the proxy and reveal the public
       address of the client.&quot;<br><br></span></i></pre><pre class=3D"m=
_3974075263048630147gmail-newpage"><span style=3D"font-family:arial,helveti=
ca,sans-serif">All of these are obvious vulnerabilities and serious problem=
s (not only #2 as the document suggests).<br><br></span></pre><pre class=3D=
"m_3974075263048630147gmail-newpage"><span style=3D"font-family:arial,helve=
tica,sans-serif">It is stated in the document that, <br><i><br>&quot;we wan=
t to avoid solutions that disable WebRTC or make it harder to use.&quot;<br=
><br></i></span></pre><pre class=3D"m_3974075263048630147gmail-newpage"><sp=
an style=3D"font-family:arial,helvetica,sans-serif">Yet, so far as my obser=
vations of WebRTC in the current version of Firefox with Ubuntu 16.10 have =
shown,<br></span></pre><pre class=3D"m_3974075263048630147gmail-newpage"><s=
pan style=3D"font-family:arial,helvetica,sans-serif">there appears to be no=
 other choice other than to disable WebRTC because its use completely compr=
omises <br></span></pre><pre class=3D"m_3974075263048630147gmail-newpage"><=
span style=3D"font-family:arial,helvetica,sans-serif">privacy and thus make=
s all other aspects of your system more difficult generally.  <br><br></spa=
n></pre><pre class=3D"m_3974075263048630147gmail-newpage"><span style=3D"fo=
nt-family:arial,helvetica,sans-serif">It is stated in the document you cite=
d that the goals here are to &quot;</span>Provide a privacy-friendly defaul=
t behavior which strikes the
      right balance between privacy and media performance for most users
      and use cases. (and) For users who care more about one versus the oth=
er, provide a
      means to customize the experience.&quot;</pre><pre class=3D"m_3974075=
263048630147gmail-newpage"><span style=3D"font-family:arial,helvetica,sans-=
serif">This has clearlly not been accomplished in the context of Ubuntu 16.=
10 64-bit with Firefox 53.0.2 (64-bit).<br><br></span></pre><pre class=3D"m=
_3974075263048630147gmail-newpage"><span style=3D"font-family:arial,helveti=
ca,sans-serif">Hence my request for a new security review of WebRTC based o=
n observed vulnerabilities.<br></span></pre><p><span style=3D"font-family:a=
rial,helvetica,sans-serif"></span><br></p><strong></strong><div><div><div><=
div><div><br></div></div></div></div></div></div><div class=3D"HOEnZb"><div=
 class=3D"h5"><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On =
Thu, May 25, 2017 at 1:04 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">Colin,<di=
v><br></div><div>This is an important topic, and the group has reviewed thi=
s situation in detail. The results have been written up in <a href=3D"https=
://tools.ietf.org/html/draft-ietf-rtcweb-ip-handling-03" target=3D"_blank">=
this document</a>, and to my knowledge most browsers that support WebRTC cu=
rrently conform to its recommendations.</div><div><br></div><div>However, I=
&#39;m not totally clear on your specific technical concern. You linked to =
an (now outdated) stackexchange page, but could you provide more detail abo=
ut the specific situation you are concerned about?</div><div><br></div><div=
>Justin</div></div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote=
"><div><div class=3D"m_3974075263048630147h5">On Wed, May 24, 2017 at 5:35 =
PM, Colin Gallagher <span dir=3D"ltr">&lt;<a href=3D"mailto:colingallagher.=
rpcv@gmail.com" target=3D"_blank">colingallagher.rpcv@gmail.com</a><wbr>&gt=
;</span> wrote:<br></div></div><blockquote class=3D"gmail_quote" style=3D"m=
argin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div cla=
ss=3D"m_3974075263048630147h5"><div dir=3D"ltr"><div><div><div><div><div><d=
iv><div><div>To:=C2=A0 IETF RTCWeb WG<br><br></div>From:=C2=A0 Colin Gallag=
her<br></div>=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 <=
a href=3D"https://www.linkedin.com/in/colingallagher" target=3D"_blank">htt=
ps://www.linkedin.com/in/co<wbr>lingallagher</a><br>=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 <a href=3D"https://lifeboat.com/ex/=
bios.colin.gallagher" target=3D"_blank">https://lifeboat.com/ex/bios.c<wbr>=
olin.gallagher</a><br><br></div>Request for New Security Review of WebRTC b=
ased on observed vulnerabilities in certain examples<br><br></div>Dear IETF=
 RTCWeb WG,<br><br></div><div>This request relates to WebRTC.=C2=A0 The Web=
RTC is in draft at <a href=3D"https://w3c.github.io/webrtc-pc/" target=3D"_=
blank">https://w3c.github.io/webrtc-p<wbr>c/</a> and is=20
described also at=20
<a href=3D"https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API#WebR=
TC_interfaces" target=3D"_blank">https://developer.mozilla.org/<wbr>en-US/d=
ocs/Web/API/WebRTC_API#<wbr>WebRTC_interfaces</a></div><div><br></div>There=
 are certain examples where the use of WebRTC has presented security vulner=
abilities.=C2=A0 While not exhaustive, there are two I wanted to mention th=
at I had been exposed to recently.<br></div><div><br></div>1) The use of a =
video conferencing service which relies upon WebRTC (that is a well establi=
shed service)<br></div>2) The examination of a browser based WebRTC version=
 of a decentralized exchange (that is currently under development)<br><br><=
/div>In both cases the security vulnerability which is described below was =
the result of the use of WebRTC.<br><br>The problem under consideration is =
as follows:<br><br>As per the first example (the video conferencing service=
), recently
 I came upon a vendor who shall remain unnamed who uses WebRTC as the=20
backbone for their video conferencing service.=C2=A0 As a result, I found I=
=20
was unable to access the service in question without disabling some=20
security settings which I would have preferred to leave intact. <br><br>My =
system is (based on what the aforementioned service detects) Mozilla/5.0 (X=
11; Ubuntu; Linux x86_64; rv:53.0)=20
Gecko/20100101 Firefox/53.0.=C2=A0 This is Ubuntu 16.10 64-bit with Firefox=
=20
53.0.2.<br><br>However, quite some <i>long</i> while back I disabled WebRTC=
 because of the security vulnerability (described in part <a href=3D"https:=
//security.stackexchange.com/questions/82129/what-is-webrtc-and-how-can-i-p=
rotect-myself-against-leaking-information-that-doe" target=3D"_blank">here =
on information security stack exchange</a>).=C2=A0 So as to mitigate this v=
ulnerability, I changed settings in Firefox as follows:<br><br><ol><li>I ty=
ped <em>about:config</em> in the address bar</li><li>I found the setting <e=
m>media.peerconnection.enabled</em></li><li>I set it to <strong>false</stro=
ng></li></ol><p><strong></strong>Unfortunately,
 this setting does not allow Firefox to work with the video conferencing
 service in question, which relies entirely on WebRTC.<br></p><p>So I had t=
o go back in and change the media.peerconnection.enabled setting to <b>true=
</b>
 in Firefox in order to get it to work with the service in question.=20
While this enabled me to conference with teams that desire to use the=20
video conferencing service that rely wholly upon WebRTC, it concerns me=20
because of the security vulnerability.</p><p>One suggested mitigation was h=
aving NoScript, but using NoScript was not helpful (you generally need to d=
isable it for the site you are viewing that relies upon WebRTC, in order to=
 obtain any functionality, and whether NoScript is off or on, WebRTC still =
causes the vulnerability of leaking information as previously described).<b=
r></p><p>Furthermore, upon reload of the browser, even after setting media.=
peerconnection.enabled to <b>true</b>,
 the video conferencing service wouldn&#39;t work until I installed the=20
WebRTC 0.1.3 extension for Firefox and set it to expose my real IP=20
address as well as allowing the following:</p><p>a. navigator.getUserMedia<=
br>b. window.MediaStreamTrack<br>c. window.RTCPeerConnection<br>d. window.R=
TCSessionDescription<br></p><p>That seemed to me to be a direct consequence=
 of the persistent and ongoing security vulnerability in WebRTC.<br></p><p>=
I
 contacted the video conferencing service provider and their solution=20
was simply to state that &quot;If the peer to peer connection is of concern=
,=20
you can utilize our=20
premium version which will route traffic through a forwarding server=20
with in our environment that handles the processing of the video and=20
sound of all users in the conference and send it to each user=20
individually rather than using a peer to peer connection.&quot;=C2=A0 In ot=
her=20
words, they expect that people should have to pay in order to mitigate <a h=
ref=3D"https://security.stackexchange.com/questions/82129/what-is-webrtc-an=
d-how-can-i-protect-myself-against-leaking-information-that-doe" target=3D"=
_blank">this WebRTC security vulnerability</a>, because they are unwilling =
to design to protect all users from it.</p><p>In a similar fashion, in the =
second example of the browser based WebRTC version of a decentralized excha=
nge, the user&#39;s financial information is periodically exposed as a resu=
lt of the information leakage which is presently caused by WebRTC.=C2=A0 (F=
ortunately, this browser based WebRTC decentralized exchange is under devel=
opment and is not currently open to actual usage.)<br></p><p>Although
 undoubtedly this has been discussed here before, I am asking that WebRTC s=
ecurity issues be
 reconsidered and that a new security review be done to ameliorate or=20
eliminate this problem in terms of WebRTC leaking this information.=C2=A0 I=
t is my view that WebRTC should be further designed and developed with an e=
ye to mitigating this serious problem, so that users of services which migh=
t rely partially or wholly upon WebRTC would not have to be concerned about=
 such information leakage.<br></p><p>Respectfully,</p>Colin Gallagher<br><b=
r></div>
<br></div></div>______________________________<wbr>_________________<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/l<wbr>istinfo/rtcweb</a><br=
>
<br></blockquote></div><br></div>
</blockquote></div><br></div>
</div></div><br>______________________________<wbr>_________________<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/<wbr>listinfo/rtcweb</a><br=
>
<br></blockquote></div><br></div>

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Fri, 26 May 2017 14:53:14 +0200
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It seems to me that you don't want WebRTC. For WebRTC to work IPs must be
exposed. So not sure what you are asking for.

El 26/5/2017 9:58, "Colin Gallagher" <colingallagher.rpcv@gmail.com>
escribi=C3=B3:

> Justin,
>
> It's my understanding that various browsers have made an attempt to
> conform to the recommendations in the document that you have provided
> <https://tools.ietf.org/html/draft-ietf-rtcweb-ip-handling-03>.
> Unfortunately, at least in Firefox (the most current version) so far as I
> can tell, there is a serious vulnerability.  To be more specific, the
> problem was observed in my operating system with my current browser, whic=
h
> is Ubuntu 16.10 64-bit with Firefox 53.0.2 (64-bit).  The browser does no=
t
> address the WebRTC problems by default.
>
> The stackexchange page that I cited originally is actually not outdated i=
n
> the sense that its method of measuring whether or not the WebRTC
> vulnerability exists on a given system is still intact and works.  The
> stackexchange page suggests two pages to determine whether or not you hav=
e
> leakage / WebRTC related vulnerability.  These pages are:
>
> 1) Browser leaks test for WebRTC https://browserleaks.com/webrtc
> 2) IP Leaks https://ipleak.net/
>
> Currently for Browser leaks test for WebRTC (due to that I have again
> changed settings in my system so that I will not have to be exposed to th=
is
> vulnerability) the settings show up on the page as:
> RTCPeerConnection=C3=97False
> RTCDataChannel=C3=97False
> ORTC (Microsoft Edge)=C3=97False (for example....)
>
> And under IP leaks, it currently shows "No leak, RTCPeerConnection not
> available."
>
> *That is because:*
>
> (A) I have (again) changed my settings in about:config back to the
> following
>
>
>    1. I typed *about:config* in the address bar
>    2. I found the setting *media.peerconnection.enabled*
>    3. I set it to *false  (it was previously set at true, so as to allow
>    WebRTC to work, but this exposed information which I would not have wa=
nted
>    to be exposed)*
>
> (B) And I also have changed my settings back to the following
>
> in the WebRTC 0.1.3 extension for Firefox I have set it to no longer
> expose my real IP address as well as no longer allowing the following:
>
> a. navigator.getUserMedia
> b. window.MediaStreamTrack
> c. window.RTCPeerConnection
> d. window.RTCSessionDescription
>
> To be clear, *these mitigations that I have employed above would be
> unnecessary if WebRTC had not been designed in such a manner that it so
> clearly exposes so much user information.*
>
> In the document you provided
> <https://tools.ietf.org/html/draft-ietf-rtcweb-ip-handling-03>, titled
> "WebRTC IP Address Handling Requirements - draft-ietf-rtcweb-ip-handling-=
03,"
> it is stated that:
>
> *"1. If the client is behind a NAT, the client's private IP addresses,
> typically [RFC1918 <https://tools.ietf.org/html/rfc1918>] addresses, can =
be
> learned. *
>
>
>
> * 2.  If the client tries to hide its physical location through a VPN,
>        and the VPN and local OS support routing over multiple interfaces
>        (i.e., a "split-tunnel" VPN), WebRTC will discover the public
>        address for the VPN as well as the ISP public address that the
>        VPN runs over.
>
>    3.  If the client is behind a proxy (a client-configured "classical
>        application proxy", as defined in [RFC1919], Section 3 <https://to=
ols.ietf.org/html/rfc1919#section-3>), but
>        direct access to the Internet is also supported, WebRTC's STUN
>        [RFC5389 <https://tools.ietf.org/html/rfc5389>]checks will bypass =
the proxy and reveal the public
>        address of the client."*
>
> All of these are obvious vulnerabilities and serious problems (not only #=
2 as the document suggests).
>
> It is stated in the document that,
>
>
>
> *"we want to avoid solutions that disable WebRTC or make it harder to use=
."*
>
> Yet, so far as my observations of WebRTC in the current version of Firefo=
x with Ubuntu 16.10 have shown,
>
> there appears to be no other choice other than to disable WebRTC because =
its use completely compromises
>
> privacy and thus makes all other aspects of your system more difficult ge=
nerally.
>
> It is stated in the document you cited that the goals here are to "Provid=
e a privacy-friendly default behavior which strikes the
>       right balance between privacy and media performance for most users
>       and use cases. (and) For users who care more about one versus the o=
ther, provide a
>       means to customize the experience."
>
> This has clearlly not been accomplished in the context of Ubuntu 16.10 64=
-bit with Firefox 53.0.2 (64-bit).
>
> Hence my request for a new security review of WebRTC based on observed vu=
lnerabilities.
>
>
>
>
> On Thu, May 25, 2017 at 1:04 PM, Justin Uberti <juberti@google.com> wrote=
:
>
>> Colin,
>>
>> This is an important topic, and the group has reviewed this situation in
>> detail. The results have been written up in this document
>> <https://tools.ietf.org/html/draft-ietf-rtcweb-ip-handling-03>, and to
>> my knowledge most browsers that support WebRTC currently conform to its
>> recommendations.
>>
>> However, I'm not totally clear on your specific technical concern. You
>> linked to an (now outdated) stackexchange page, but could you provide mo=
re
>> detail about the specific situation you are concerned about?
>>
>> Justin
>>
>> On Wed, May 24, 2017 at 5:35 PM, Colin Gallagher <
>> colingallagher.rpcv@gmail.com> wrote:
>>
>>> To:  IETF RTCWeb WG
>>>
>>> From:  Colin Gallagher
>>>            https://www.linkedin.com/in/colingallagher
>>>            https://lifeboat.com/ex/bios.colin.gallagher
>>>
>>> Request for New Security Review of WebRTC based on observed
>>> vulnerabilities in certain examples
>>>
>>> Dear IETF RTCWeb WG,
>>>
>>> This request relates to WebRTC.  The WebRTC is in draft at
>>> https://w3c.github.io/webrtc-pc/ and is described also at
>>> https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API#
>>> WebRTC_interfaces
>>>
>>> There are certain examples where the use of WebRTC has presented
>>> security vulnerabilities.  While not exhaustive, there are two I wanted=
 to
>>> mention that I had been exposed to recently.
>>>
>>> 1) The use of a video conferencing service which relies upon WebRTC
>>> (that is a well established service)
>>> 2) The examination of a browser based WebRTC version of a decentralized
>>> exchange (that is currently under development)
>>>
>>> In both cases the security vulnerability which is described below was
>>> the result of the use of WebRTC.
>>>
>>> The problem under consideration is as follows:
>>>
>>> As per the first example (the video conferencing service), recently I
>>> came upon a vendor who shall remain unnamed who uses WebRTC as the back=
bone
>>> for their video conferencing service.  As a result, I found I was unabl=
e to
>>> access the service in question without disabling some security settings
>>> which I would have preferred to leave intact.
>>>
>>> My system is (based on what the aforementioned service detects)
>>> Mozilla/5.0 (X11; Ubuntu; Linux x86_64; rv:53.0) Gecko/20100101
>>> Firefox/53.0.  This is Ubuntu 16.10 64-bit with Firefox 53.0.2.
>>>
>>> However, quite some *long* while back I disabled WebRTC because of the
>>> security vulnerability (described in part here on information security
>>> stack exchange
>>> <https://security.stackexchange.com/questions/82129/what-is-webrtc-and-=
how-can-i-protect-myself-against-leaking-information-that-doe>).
>>> So as to mitigate this vulnerability, I changed settings in Firefox as
>>> follows:
>>>
>>>
>>>    1. I typed *about:config* in the address bar
>>>    2. I found the setting *media.peerconnection.enabled*
>>>    3. I set it to *false*
>>>
>>> Unfortunately, this setting does not allow Firefox to work with the
>>> video conferencing service in question, which relies entirely on WebRTC=
.
>>>
>>> So I had to go back in and change the media.peerconnection.enabled
>>> setting to *true* in Firefox in order to get it to work with the
>>> service in question. While this enabled me to conference with teams tha=
t
>>> desire to use the video conferencing service that rely wholly upon WebR=
TC,
>>> it concerns me because of the security vulnerability.
>>>
>>> One suggested mitigation was having NoScript, but using NoScript was no=
t
>>> helpful (you generally need to disable it for the site you are viewing =
that
>>> relies upon WebRTC, in order to obtain any functionality, and whether
>>> NoScript is off or on, WebRTC still causes the vulnerability of leaking
>>> information as previously described).
>>>
>>> Furthermore, upon reload of the browser, even after setting
>>> media.peerconnection.enabled to *true*, the video conferencing service
>>> wouldn't work until I installed the WebRTC 0.1.3 extension for Firefox =
and
>>> set it to expose my real IP address as well as allowing the following:
>>>
>>> a. navigator.getUserMedia
>>> b. window.MediaStreamTrack
>>> c. window.RTCPeerConnection
>>> d. window.RTCSessionDescription
>>>
>>> That seemed to me to be a direct consequence of the persistent and
>>> ongoing security vulnerability in WebRTC.
>>>
>>> I contacted the video conferencing service provider and their solution
>>> was simply to state that "If the peer to peer connection is of concern,=
 you
>>> can utilize our premium version which will route traffic through a
>>> forwarding server with in our environment that handles the processing o=
f
>>> the video and sound of all users in the conference and send it to each =
user
>>> individually rather than using a peer to peer connection."  In other wo=
rds,
>>> they expect that people should have to pay in order to mitigate this
>>> WebRTC security vulnerability
>>> <https://security.stackexchange.com/questions/82129/what-is-webrtc-and-=
how-can-i-protect-myself-against-leaking-information-that-doe>,
>>> because they are unwilling to design to protect all users from it.
>>>
>>> In a similar fashion, in the second example of the browser based WebRTC
>>> version of a decentralized exchange, the user's financial information i=
s
>>> periodically exposed as a result of the information leakage which is
>>> presently caused by WebRTC.  (Fortunately, this browser based WebRTC
>>> decentralized exchange is under development and is not currently open t=
o
>>> actual usage.)
>>>
>>> Although undoubtedly this has been discussed here before, I am asking
>>> that WebRTC security issues be reconsidered and that a new security rev=
iew
>>> be done to ameliorate or eliminate this problem in terms of WebRTC leak=
ing
>>> this information.  It is my view that WebRTC should be further designed=
 and
>>> developed with an eye to mitigating this serious problem, so that users=
 of
>>> services which might rely partially or wholly upon WebRTC would not hav=
e to
>>> be concerned about such information leakage.
>>>
>>> Respectfully,
>>> Colin Gallagher
>>>
>>>
>>> _______________________________________________
>>> 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
>
>

--f403045c1c801d801105506cd253
Content-Type: text/html; charset="UTF-8"
Content-Transfer-Encoding: quoted-printable

<div dir=3D"auto">It seems to me that you don&#39;t want WebRTC. For WebRTC=
 to work IPs must be exposed. So not sure what you are asking for.</div><di=
v class=3D"gmail_extra"><br><div class=3D"gmail_quote">El 26/5/2017 9:58, &=
quot;Colin Gallagher&quot; &lt;<a href=3D"mailto:colingallagher.rpcv@gmail.=
com">colingallagher.rpcv@gmail.com</a>&gt; escribi=C3=B3:<br type=3D"attrib=
ution"><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><div><div><div>=
<div><div><div><div>Justin,<br><br></div>It&#39;s my understanding that var=
ious browsers have made an attempt to conform to the recommendations in <a =
href=3D"https://tools.ietf.org/html/draft-ietf-rtcweb-ip-handling-03" targe=
t=3D"_blank">the document that you have provided</a>.=C2=A0 Unfortunately, =
at least in Firefox (the most current version) so far as I can tell, there =
is a serious vulnerability.=C2=A0 To be more specific, the problem was obse=
rved in my operating system with my current browser, which is Ubuntu 16.10 =
64-bit with Firefox=20
53.0.2 (64-bit).=C2=A0 The browser does not address the WebRTC problems by =
default.<br><br></div>The stackexchange page that I cited originally is act=
ually not outdated in the sense that its method of measuring whether or not=
 the WebRTC vulnerability exists on a given system is still intact and work=
s.=C2=A0 The stackexchange page suggests two pages to determine whether or =
not you have leakage / WebRTC related vulnerability.=C2=A0 These pages are:=
<br><br></div>1) Browser leaks test for WebRTC <a href=3D"https://browserle=
aks.com/webrtc" target=3D"_blank">https://browserleaks.com/<wbr>webrtc</a><=
br></div>2) IP Leaks <a href=3D"https://ipleak.net/" target=3D"_blank">http=
s://ipleak.net/</a><br><br></div>Currently for Browser leaks test for WebRT=
C (due to that I have again changed settings in my system so that I will no=
t have to be exposed to this vulnerability) the settings show up on the pag=
e as:<br>RTCPeerConnection<span class=3D"m_-284210964436143041gmail-bad">=
=C3=97</span>False<br>RTCDataChannel<span class=3D"m_-284210964436143041gma=
il-bad">=C3=97</span>False<br>ORTC (Microsoft Edge)<span class=3D"m_-284210=
964436143041gmail-bad">=C3=97</span>False (for example....)<br><br></div>An=
d under IP leaks, it currently shows &quot;No leak, RTCPeerConnection not a=
vailable.&quot;=C2=A0 <br><br></div><i>That is because:</i><br><br></div>(A=
) I have (again) changed my settings in about:config back to the following<=
br><br><ol><li>I typed <em>about:config</em> in the address bar</li><li>I f=
ound the setting <em>media.peerconnection.enabled</em></li><li>I set it to =
<strong>false=C2=A0 (it was previously set at true, so as to allow WebRTC t=
o work, but this exposed information which I would not have wanted to be ex=
posed)</strong></li></ol><p>(B) And I also have changed my settings back to=
 the following</p><p>in the WebRTC 0.1.3 extension for Firefox I have set i=
t to no longer expose my real IP=20
address as well as no longer allowing the following:</p><p>a. navigator.get=
UserMedia<br>b. window.MediaStreamTrack<br>c. window.RTCPeerConnection<br>d=
. window.RTCSessionDescription</p><p>To be clear, <b>these mitigations that=
 I have employed above would be unnecessary if WebRTC had not been designed=
 in such a manner that it so clearly exposes so much user information.</b><=
/p><p>In <a href=3D"https://tools.ietf.org/html/draft-ietf-rtcweb-ip-handli=
ng-03" target=3D"_blank">the document you provided</a>, titled &quot;WebRTC=
 IP Address Handling Requirements - draft-ietf-rtcweb-ip-handling-<wbr>03,&=
quot; it is stated that:</p><p><i>&quot;<span style=3D"font-family:arial,he=
lvetica,sans-serif">1.  If the client is behind a NAT, the client&#39;s pri=
vate IP addresses, typically [<a href=3D"https://tools.ietf.org/html/rfc191=
8" title=3D"&quot;Address Allocation for Private Internets&quot;" target=3D=
"_blank">RFC1918</a>] addresses, can be learned.
</span></i></p><pre class=3D"m_-284210964436143041gmail-newpage"><i><span s=
tyle=3D"font-family:arial,helvetica,sans-serif"> 2.  If the client tries to=
 hide its physical location through a VPN,
       and the VPN and local OS support routing over multiple interfaces
       (i.e., a &quot;split-tunnel&quot; VPN), WebRTC will discover the pub=
lic
       address for the VPN as well as the ISP public address that the
       VPN runs over.

   3.  If the client is behind a proxy (a client-configured &quot;classical
       application proxy&quot;, as defined in <a href=3D"https://tools.ietf=
.org/html/rfc1919#section-3" target=3D"_blank">[RFC1919], Section=C2=A03</a=
>), but
       direct access to the Internet is also supported, WebRTC&#39;s STUN
       [<a href=3D"https://tools.ietf.org/html/rfc5389" title=3D"&quot;Sess=
ion Traversal Utilities for NAT (STUN)&quot;" target=3D"_blank">RFC5389</a>=
]checks will bypass the proxy and reveal the public
       address of the client.&quot;<br><br></span></i></pre><pre class=3D"m=
_-284210964436143041gmail-newpage"><span style=3D"font-family:arial,helveti=
ca,sans-serif">All of these are obvious vulnerabilities and serious problem=
s (not only #2 as the document suggests).<br><br></span></pre><pre class=3D=
"m_-284210964436143041gmail-newpage"><span style=3D"font-family:arial,helve=
tica,sans-serif">It is stated in the document that, <br><i><br>&quot;we wan=
t to avoid solutions that disable WebRTC or make it harder to use.&quot;<br=
><br></i></span></pre><pre class=3D"m_-284210964436143041gmail-newpage"><sp=
an style=3D"font-family:arial,helvetica,sans-serif">Yet, so far as my obser=
vations of WebRTC in the current version of Firefox with Ubuntu 16.10 have =
shown,<br></span></pre><pre class=3D"m_-284210964436143041gmail-newpage"><s=
pan style=3D"font-family:arial,helvetica,sans-serif">there appears to be no=
 other choice other than to disable WebRTC because its use completely compr=
omises <br></span></pre><pre class=3D"m_-284210964436143041gmail-newpage"><=
span style=3D"font-family:arial,helvetica,sans-serif">privacy and thus make=
s all other aspects of your system more difficult generally.  <br><br></spa=
n></pre><pre class=3D"m_-284210964436143041gmail-newpage"><span style=3D"fo=
nt-family:arial,helvetica,sans-serif">It is stated in the document you cite=
d that the goals here are to &quot;</span>Provide a privacy-friendly defaul=
t behavior which strikes the
      right balance between privacy and media performance for most users
      and use cases. (and) For users who care more about one versus the oth=
er, provide a
      means to customize the experience.&quot;</pre><pre class=3D"m_-284210=
964436143041gmail-newpage"><span style=3D"font-family:arial,helvetica,sans-=
serif">This has clearlly not been accomplished in the context of Ubuntu 16.=
10 64-bit with Firefox 53.0.2 (64-bit).<br><br></span></pre><pre class=3D"m=
_-284210964436143041gmail-newpage"><span style=3D"font-family:arial,helveti=
ca,sans-serif">Hence my request for a new security review of WebRTC based o=
n observed vulnerabilities.<br></span></pre><p><span style=3D"font-family:a=
rial,helvetica,sans-serif"></span><br></p><strong></strong><div><div><div><=
div><div><br></div></div></div></div></div></div><div class=3D"gmail_extra"=
><br><div class=3D"gmail_quote">On Thu, May 25, 2017 at 1:04 PM, Justin Ube=
rti <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"gma=
il_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-lef=
t:1ex"><div dir=3D"ltr">Colin,<div><br></div><div>This is an important topi=
c, and the group has reviewed this situation in detail. The results have be=
en written up in <a href=3D"https://tools.ietf.org/html/draft-ietf-rtcweb-i=
p-handling-03" target=3D"_blank">this document</a>, and to my knowledge mos=
t browsers that support WebRTC currently conform to its recommendations.</d=
iv><div><br></div><div>However, I&#39;m not totally clear on your specific =
technical concern. You linked to an (now outdated) stackexchange page, but =
could you provide more detail about the specific situation you are concerne=
d about?</div><div><br></div><div>Justin</div></div><div class=3D"gmail_ext=
ra"><br><div class=3D"gmail_quote"><div><div class=3D"m_-284210964436143041=
h5">On Wed, May 24, 2017 at 5:35 PM, Colin Gallagher <span dir=3D"ltr">&lt;=
<a href=3D"mailto:colingallagher.rpcv@gmail.com" target=3D"_blank">colingal=
lagher.rpcv@gmail.com</a><wbr>&gt;</span> wrote:<br></div></div><blockquote=
 class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc soli=
d;padding-left:1ex"><div><div class=3D"m_-284210964436143041h5"><div dir=3D=
"ltr"><div><div><div><div><div><div><div><div>To:=C2=A0 IETF RTCWeb WG<br><=
br></div>From:=C2=A0 Colin Gallagher<br></div>=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 <a href=3D"https://www.linkedin.com/in/co=
lingallagher" target=3D"_blank">https://www.linkedin.com/in/co<wbr>lingalla=
gher</a><br>=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 <a=
 href=3D"https://lifeboat.com/ex/bios.colin.gallagher" target=3D"_blank">ht=
tps://lifeboat.com/ex/bios.c<wbr>olin.gallagher</a><br><br></div>Request fo=
r New Security Review of WebRTC based on observed vulnerabilities in certai=
n examples<br><br></div>Dear IETF RTCWeb WG,<br><br></div><div>This request=
 relates to WebRTC.=C2=A0 The WebRTC is in draft at <a href=3D"https://w3c.=
github.io/webrtc-pc/" target=3D"_blank">https://w3c.github.io/webrtc-p<wbr>=
c/</a> and is=20
described also at=20
<a href=3D"https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API#WebR=
TC_interfaces" target=3D"_blank">https://developer.mozilla.org/<wbr>en-US/d=
ocs/Web/API/WebRTC_API#<wbr>WebRTC_interfaces</a></div><div><br></div>There=
 are certain examples where the use of WebRTC has presented security vulner=
abilities.=C2=A0 While not exhaustive, there are two I wanted to mention th=
at I had been exposed to recently.<br></div><div><br></div>1) The use of a =
video conferencing service which relies upon WebRTC (that is a well establi=
shed service)<br></div>2) The examination of a browser based WebRTC version=
 of a decentralized exchange (that is currently under development)<br><br><=
/div>In both cases the security vulnerability which is described below was =
the result of the use of WebRTC.<br><br>The problem under consideration is =
as follows:<br><br>As per the first example (the video conferencing service=
), recently
 I came upon a vendor who shall remain unnamed who uses WebRTC as the=20
backbone for their video conferencing service.=C2=A0 As a result, I found I=
=20
was unable to access the service in question without disabling some=20
security settings which I would have preferred to leave intact. <br><br>My =
system is (based on what the aforementioned service detects) Mozilla/5.0 (X=
11; Ubuntu; Linux x86_64; rv:53.0)=20
Gecko/20100101 Firefox/53.0.=C2=A0 This is Ubuntu 16.10 64-bit with Firefox=
=20
53.0.2.<br><br>However, quite some <i>long</i> while back I disabled WebRTC=
 because of the security vulnerability (described in part <a href=3D"https:=
//security.stackexchange.com/questions/82129/what-is-webrtc-and-how-can-i-p=
rotect-myself-against-leaking-information-that-doe" target=3D"_blank">here =
on information security stack exchange</a>).=C2=A0 So as to mitigate this v=
ulnerability, I changed settings in Firefox as follows:<br><br><ol><li>I ty=
ped <em>about:config</em> in the address bar</li><li>I found the setting <e=
m>media.peerconnection.enabled</em></li><li>I set it to <strong>false</stro=
ng></li></ol><p><strong></strong>Unfortunately,
 this setting does not allow Firefox to work with the video conferencing
 service in question, which relies entirely on WebRTC.<br></p><p>So I had t=
o go back in and change the media.peerconnection.enabled setting to <b>true=
</b>
 in Firefox in order to get it to work with the service in question.=20
While this enabled me to conference with teams that desire to use the=20
video conferencing service that rely wholly upon WebRTC, it concerns me=20
because of the security vulnerability.</p><p>One suggested mitigation was h=
aving NoScript, but using NoScript was not helpful (you generally need to d=
isable it for the site you are viewing that relies upon WebRTC, in order to=
 obtain any functionality, and whether NoScript is off or on, WebRTC still =
causes the vulnerability of leaking information as previously described).<b=
r></p><p>Furthermore, upon reload of the browser, even after setting media.=
peerconnection.enabled to <b>true</b>,
 the video conferencing service wouldn&#39;t work until I installed the=20
WebRTC 0.1.3 extension for Firefox and set it to expose my real IP=20
address as well as allowing the following:</p><p>a. navigator.getUserMedia<=
br>b. window.MediaStreamTrack<br>c. window.RTCPeerConnection<br>d. window.R=
TCSessionDescription<br></p><p>That seemed to me to be a direct consequence=
 of the persistent and ongoing security vulnerability in WebRTC.<br></p><p>=
I
 contacted the video conferencing service provider and their solution=20
was simply to state that &quot;If the peer to peer connection is of concern=
,=20
you can utilize our=20
premium version which will route traffic through a forwarding server=20
with in our environment that handles the processing of the video and=20
sound of all users in the conference and send it to each user=20
individually rather than using a peer to peer connection.&quot;=C2=A0 In ot=
her=20
words, they expect that people should have to pay in order to mitigate <a h=
ref=3D"https://security.stackexchange.com/questions/82129/what-is-webrtc-an=
d-how-can-i-protect-myself-against-leaking-information-that-doe" target=3D"=
_blank">this WebRTC security vulnerability</a>, because they are unwilling =
to design to protect all users from it.</p><p>In a similar fashion, in the =
second example of the browser based WebRTC version of a decentralized excha=
nge, the user&#39;s financial information is periodically exposed as a resu=
lt of the information leakage which is presently caused by WebRTC.=C2=A0 (F=
ortunately, this browser based WebRTC decentralized exchange is under devel=
opment and is not currently open to actual usage.)<br></p><p>Although
 undoubtedly this has been discussed here before, I am asking that WebRTC s=
ecurity issues be
 reconsidered and that a new security review be done to ameliorate or=20
eliminate this problem in terms of WebRTC leaking this information.=C2=A0 I=
t is my view that WebRTC should be further designed and developed with an e=
ye to mitigating this serious problem, so that users of services which migh=
t rely partially or wholly upon WebRTC would not have to be concerned about=
 such information leakage.<br></p><p>Respectfully,</p>Colin Gallagher<br><b=
r></div>
<br></div></div>______________________________<wbr>_________________<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/l<wbr>istinfo/rtcweb</a><br=
>
<br></blockquote></div><br></div>
</blockquote></div><br></div>
<br>______________________________<wbr>_________________<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/<wbr>listinfo/rtcweb</a><br=
>
<br></blockquote></div></div>

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It is quite clear what I am asking for.  I am asking for a new security
review of WebRTC.  To quote my original request as posted:

"I am asking that WebRTC security issues be reconsidered and that a new
security review be done to ameliorate or eliminate this problem in terms of
WebRTC leaking this information.  It is my view that WebRTC should be
further designed and developed with an eye to mitigating this serious
problem, so that users of services which might rely partially or wholly
upon WebRTC would not have to be concerned about such information leakage."

In other words, I would like WebRTC to be reviewed for this very obvious
security issue and redesigned so that it does not leak this information
when utilized in browsers.

Some developers who have considered using WebRTC (to develop a browser
based WebRTC version of decentralized currency exchange) have observed this
persistent security vulnerability and rather than accept the problems it
would pose for users, the developers have instead contemplated
incorporating onion routing among peers to mitigate the problem of IP being
revealed or associated with messages.

However, it should not only fall to developers who are considering using
WebRTC to find some elaborate workaround or fix to the privacy problem with
their particular application.  WebRTC itself needs to be fixed as well.

Respectfully,

Colin Gallagher

On Fri, May 26, 2017 at 5:53 AM, I=C3=B1aki Baz Castillo <ibc@aliax.net> wr=
ote:

> It seems to me that you don't want WebRTC. For WebRTC to work IPs must be
> exposed. So not sure what you are asking for.
>
> El 26/5/2017 9:58, "Colin Gallagher" <colingallagher.rpcv@gmail.com>
> escribi=C3=B3:
>
>> Justin,
>>
>> It's my understanding that various browsers have made an attempt to
>> conform to the recommendations in the document that you have provided
>> <https://tools.ietf.org/html/draft-ietf-rtcweb-ip-handling-03>.
>> Unfortunately, at least in Firefox (the most current version) so far as =
I
>> can tell, there is a serious vulnerability.  To be more specific, the
>> problem was observed in my operating system with my current browser, whi=
ch
>> is Ubuntu 16.10 64-bit with Firefox 53.0.2 (64-bit).  The browser does n=
ot
>> address the WebRTC problems by default.
>>
>> The stackexchange page that I cited originally is actually not outdated
>> in the sense that its method of measuring whether or not the WebRTC
>> vulnerability exists on a given system is still intact and works.  The
>> stackexchange page suggests two pages to determine whether or not you ha=
ve
>> leakage / WebRTC related vulnerability.  These pages are:
>>
>> 1) Browser leaks test for WebRTC https://browserleaks.com/webrtc
>> 2) IP Leaks https://ipleak.net/
>>
>> Currently for Browser leaks test for WebRTC (due to that I have again
>> changed settings in my system so that I will not have to be exposed to t=
his
>> vulnerability) the settings show up on the page as:
>> RTCPeerConnection=C3=97False
>> RTCDataChannel=C3=97False
>> ORTC (Microsoft Edge)=C3=97False (for example....)
>>
>> And under IP leaks, it currently shows "No leak, RTCPeerConnection not
>> available."
>>
>> *That is because:*
>>
>> (A) I have (again) changed my settings in about:config back to the
>> following
>>
>>
>>    1. I typed *about:config* in the address bar
>>    2. I found the setting *media.peerconnection.enabled*
>>    3. I set it to *false  (it was previously set at true, so as to allow
>>    WebRTC to work, but this exposed information which I would not have w=
anted
>>    to be exposed)*
>>
>> (B) And I also have changed my settings back to the following
>>
>> in the WebRTC 0.1.3 extension for Firefox I have set it to no longer
>> expose my real IP address as well as no longer allowing the following:
>>
>> a. navigator.getUserMedia
>> b. window.MediaStreamTrack
>> c. window.RTCPeerConnection
>> d. window.RTCSessionDescription
>>
>> To be clear, *these mitigations that I have employed above would be
>> unnecessary if WebRTC had not been designed in such a manner that it so
>> clearly exposes so much user information.*
>>
>> In the document you provided
>> <https://tools.ietf.org/html/draft-ietf-rtcweb-ip-handling-03>, titled
>> "WebRTC IP Address Handling Requirements - draft-ietf-rtcweb-ip-handling=
-03,"
>> it is stated that:
>>
>> *"1. If the client is behind a NAT, the client's private IP addresses,
>> typically [RFC1918 <https://tools.ietf.org/html/rfc1918>] addresses, can=
 be
>> learned. *
>>
>>
>>
>> * 2.  If the client tries to hide its physical location through a VPN,
>>        and the VPN and local OS support routing over multiple interfaces
>>        (i.e., a "split-tunnel" VPN), WebRTC will discover the public
>>        address for the VPN as well as the ISP public address that the
>>        VPN runs over.
>>
>>    3.  If the client is behind a proxy (a client-configured "classical
>>        application proxy", as defined in [RFC1919], Section 3 <https://t=
ools.ietf.org/html/rfc1919#section-3>), but
>>        direct access to the Internet is also supported, WebRTC's STUN
>>        [RFC5389 <https://tools.ietf.org/html/rfc5389>]checks will bypass=
 the proxy and reveal the public
>>        address of the client."*
>>
>> All of these are obvious vulnerabilities and serious problems (not only =
#2 as the document suggests).
>>
>> It is stated in the document that,
>>
>>
>>
>> *"we want to avoid solutions that disable WebRTC or make it harder to us=
e."*
>>
>> Yet, so far as my observations of WebRTC in the current version of Firef=
ox with Ubuntu 16.10 have shown,
>>
>> there appears to be no other choice other than to disable WebRTC because=
 its use completely compromises
>>
>> privacy and thus makes all other aspects of your system more difficult g=
enerally.
>>
>> It is stated in the document you cited that the goals here are to "Provi=
de a privacy-friendly default behavior which strikes the
>>       right balance between privacy and media performance for most users
>>       and use cases. (and) For users who care more about one versus the =
other, provide a
>>       means to customize the experience."
>>
>> This has clearlly not been accomplished in the context of Ubuntu 16.10 6=
4-bit with Firefox 53.0.2 (64-bit).
>>
>> Hence my request for a new security review of WebRTC based on observed v=
ulnerabilities.
>>
>>
>>
>>
>> On Thu, May 25, 2017 at 1:04 PM, Justin Uberti <juberti@google.com>
>> wrote:
>>
>>> Colin,
>>>
>>> This is an important topic, and the group has reviewed this situation i=
n
>>> detail. The results have been written up in this document
>>> <https://tools.ietf.org/html/draft-ietf-rtcweb-ip-handling-03>, and to
>>> my knowledge most browsers that support WebRTC currently conform to its
>>> recommendations.
>>>
>>> However, I'm not totally clear on your specific technical concern. You
>>> linked to an (now outdated) stackexchange page, but could you provide m=
ore
>>> detail about the specific situation you are concerned about?
>>>
>>> Justin
>>>
>>> On Wed, May 24, 2017 at 5:35 PM, Colin Gallagher <
>>> colingallagher.rpcv@gmail.com> wrote:
>>>
>>>> To:  IETF RTCWeb WG
>>>>
>>>> From:  Colin Gallagher
>>>>            https://www.linkedin.com/in/colingallagher
>>>>            https://lifeboat.com/ex/bios.colin.gallagher
>>>>
>>>> Request for New Security Review of WebRTC based on observed
>>>> vulnerabilities in certain examples
>>>>
>>>> Dear IETF RTCWeb WG,
>>>>
>>>> This request relates to WebRTC.  The WebRTC is in draft at
>>>> https://w3c.github.io/webrtc-pc/ and is described also at
>>>> https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API#
>>>> WebRTC_interfaces
>>>>
>>>> There are certain examples where the use of WebRTC has presented
>>>> security vulnerabilities.  While not exhaustive, there are two I wante=
d to
>>>> mention that I had been exposed to recently.
>>>>
>>>> 1) The use of a video conferencing service which relies upon WebRTC
>>>> (that is a well established service)
>>>> 2) The examination of a browser based WebRTC version of a decentralize=
d
>>>> exchange (that is currently under development)
>>>>
>>>> In both cases the security vulnerability which is described below was
>>>> the result of the use of WebRTC.
>>>>
>>>> The problem under consideration is as follows:
>>>>
>>>> As per the first example (the video conferencing service), recently I
>>>> came upon a vendor who shall remain unnamed who uses WebRTC as the bac=
kbone
>>>> for their video conferencing service.  As a result, I found I was unab=
le to
>>>> access the service in question without disabling some security setting=
s
>>>> which I would have preferred to leave intact.
>>>>
>>>> My system is (based on what the aforementioned service detects)
>>>> Mozilla/5.0 (X11; Ubuntu; Linux x86_64; rv:53.0) Gecko/20100101
>>>> Firefox/53.0.  This is Ubuntu 16.10 64-bit with Firefox 53.0.2.
>>>>
>>>> However, quite some *long* while back I disabled WebRTC because of the
>>>> security vulnerability (described in part here on information security
>>>> stack exchange
>>>> <https://security.stackexchange.com/questions/82129/what-is-webrtc-and=
-how-can-i-protect-myself-against-leaking-information-that-doe>).
>>>> So as to mitigate this vulnerability, I changed settings in Firefox as
>>>> follows:
>>>>
>>>>
>>>>    1. I typed *about:config* in the address bar
>>>>    2. I found the setting *media.peerconnection.enabled*
>>>>    3. I set it to *false*
>>>>
>>>> Unfortunately, this setting does not allow Firefox to work with the
>>>> video conferencing service in question, which relies entirely on WebRT=
C.
>>>>
>>>> So I had to go back in and change the media.peerconnection.enabled
>>>> setting to *true* in Firefox in order to get it to work with the
>>>> service in question. While this enabled me to conference with teams th=
at
>>>> desire to use the video conferencing service that rely wholly upon Web=
RTC,
>>>> it concerns me because of the security vulnerability.
>>>>
>>>> One suggested mitigation was having NoScript, but using NoScript was
>>>> not helpful (you generally need to disable it for the site you are vie=
wing
>>>> that relies upon WebRTC, in order to obtain any functionality, and whe=
ther
>>>> NoScript is off or on, WebRTC still causes the vulnerability of leakin=
g
>>>> information as previously described).
>>>>
>>>> Furthermore, upon reload of the browser, even after setting
>>>> media.peerconnection.enabled to *true*, the video conferencing service
>>>> wouldn't work until I installed the WebRTC 0.1.3 extension for Firefox=
 and
>>>> set it to expose my real IP address as well as allowing the following:
>>>>
>>>> a. navigator.getUserMedia
>>>> b. window.MediaStreamTrack
>>>> c. window.RTCPeerConnection
>>>> d. window.RTCSessionDescription
>>>>
>>>> That seemed to me to be a direct consequence of the persistent and
>>>> ongoing security vulnerability in WebRTC.
>>>>
>>>> I contacted the video conferencing service provider and their solution
>>>> was simply to state that "If the peer to peer connection is of concern=
, you
>>>> can utilize our premium version which will route traffic through a
>>>> forwarding server with in our environment that handles the processing =
of
>>>> the video and sound of all users in the conference and send it to each=
 user
>>>> individually rather than using a peer to peer connection."  In other w=
ords,
>>>> they expect that people should have to pay in order to mitigate this
>>>> WebRTC security vulnerability
>>>> <https://security.stackexchange.com/questions/82129/what-is-webrtc-and=
-how-can-i-protect-myself-against-leaking-information-that-doe>,
>>>> because they are unwilling to design to protect all users from it.
>>>>
>>>> In a similar fashion, in the second example of the browser based WebRT=
C
>>>> version of a decentralized exchange, the user's financial information =
is
>>>> periodically exposed as a result of the information leakage which is
>>>> presently caused by WebRTC.  (Fortunately, this browser based WebRTC
>>>> decentralized exchange is under development and is not currently open =
to
>>>> actual usage.)
>>>>
>>>> Although undoubtedly this has been discussed here before, I am asking
>>>> that WebRTC security issues be reconsidered and that a new security re=
view
>>>> be done to ameliorate or eliminate this problem in terms of WebRTC lea=
king
>>>> this information.  It is my view that WebRTC should be further designe=
d and
>>>> developed with an eye to mitigating this serious problem, so that user=
s of
>>>> services which might rely partially or wholly upon WebRTC would not ha=
ve to
>>>> be concerned about such information leakage.
>>>>
>>>> Respectfully,
>>>> Colin Gallagher
>>>>
>>>>
>>>> _______________________________________________
>>>> 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
>>
>>

--94eb2c193f667b663305507074bb
Content-Type: text/html; charset="UTF-8"
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div><div><div><div><div>It is quite clear what I am askin=
g for.=C2=A0 I am asking for a new security review of WebRTC.=C2=A0 To quot=
e my original request as posted:<br><br>&quot;I am asking that WebRTC secur=
ity issues be
 reconsidered and that a new security review be done to ameliorate or=20
eliminate this problem in terms of WebRTC leaking this information.=C2=A0 I=
t=20
is my view that WebRTC should be further designed and developed with an=20
eye to mitigating this serious problem, so that users of services which=20
might rely partially or wholly upon WebRTC would not have to be=20
concerned about such information leakage.&quot;<br><br></div>In other words=
, I would like WebRTC to be reviewed for this very obvious security issue a=
nd redesigned so that it does not leak this information when utilized in br=
owsers.<br><br></div>Some developers who have considered using WebRTC (to d=
evelop a browser based WebRTC version of decentralized currency exchange) h=
ave observed this persistent security vulnerability and rather than accept =
the problems it would pose for users, the developers have instead contempla=
ted incorporating onion routing among peers to mitigate the problem of IP b=
eing revealed or associated with messages.<br><br></div>However, it should =
not only fall to developers who are considering using WebRTC to find some e=
laborate workaround or fix to the privacy problem with their particular app=
lication.=C2=A0 WebRTC itself needs to be fixed as well.<br><br></div>Respe=
ctfully,<br><br></div>Colin Gallagher<br></div><div class=3D"gmail_extra"><=
br><div class=3D"gmail_quote">On Fri, May 26, 2017 at 5:53 AM, I=C3=B1aki B=
az Castillo <span dir=3D"ltr">&lt;<a href=3D"mailto:ibc@aliax.net" target=
=3D"_blank">ibc@aliax.net</a>&gt;</span> wrote:<br><blockquote class=3D"gma=
il_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-lef=
t:1ex"><div dir=3D"auto">It seems to me that you don&#39;t want WebRTC. For=
 WebRTC to work IPs must be exposed. So not sure what you are asking for.</=
div><div class=3D"HOEnZb"><div class=3D"h5"><div class=3D"gmail_extra"><br>=
<div class=3D"gmail_quote">El 26/5/2017 9:58, &quot;Colin Gallagher&quot; &=
lt;<a href=3D"mailto:colingallagher.rpcv@gmail.com" target=3D"_blank">colin=
gallagher.rpcv@gmail.com</a><wbr>&gt; escribi=C3=B3:<br type=3D"attribution=
"><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><div><div><div><div>=
<div><div><div>Justin,<br><br></div>It&#39;s my understanding that various =
browsers have made an attempt to conform to the recommendations in <a href=
=3D"https://tools.ietf.org/html/draft-ietf-rtcweb-ip-handling-03" target=3D=
"_blank">the document that you have provided</a>.=C2=A0 Unfortunately, at l=
east in Firefox (the most current version) so far as I can tell, there is a=
 serious vulnerability.=C2=A0 To be more specific, the problem was observed=
 in my operating system with my current browser, which is Ubuntu 16.10 64-b=
it with Firefox=20
53.0.2 (64-bit).=C2=A0 The browser does not address the WebRTC problems by =
default.<br><br></div>The stackexchange page that I cited originally is act=
ually not outdated in the sense that its method of measuring whether or not=
 the WebRTC vulnerability exists on a given system is still intact and work=
s.=C2=A0 The stackexchange page suggests two pages to determine whether or =
not you have leakage / WebRTC related vulnerability.=C2=A0 These pages are:=
<br><br></div>1) Browser leaks test for WebRTC <a href=3D"https://browserle=
aks.com/webrtc" target=3D"_blank">https://browserleaks.com/webrt<wbr>c</a><=
br></div>2) IP Leaks <a href=3D"https://ipleak.net/" target=3D"_blank">http=
s://ipleak.net/</a><br><br></div>Currently for Browser leaks test for WebRT=
C (due to that I have again changed settings in my system so that I will no=
t have to be exposed to this vulnerability) the settings show up on the pag=
e as:<br>RTCPeerConnection<span class=3D"m_5234864893797269998m_-2842109644=
36143041gmail-bad">=C3=97</span>False<br>RTCDataChannel<span class=3D"m_523=
4864893797269998m_-284210964436143041gmail-bad">=C3=97</span>False<br>ORTC =
(Microsoft Edge)<span class=3D"m_5234864893797269998m_-284210964436143041gm=
ail-bad">=C3=97</span>False (for example....)<br><br></div>And under IP lea=
ks, it currently shows &quot;No leak, RTCPeerConnection not available.&quot=
;=C2=A0 <br><br></div><i>That is because:</i><br><br></div>(A) I have (agai=
n) changed my settings in about:config back to the following<br><br><ol><li=
>I typed <em>about:config</em> in the address bar</li><li>I found the setti=
ng <em>media.peerconnection.enabled</em></li><li>I set it to <strong>false=
=C2=A0 (it was previously set at true, so as to allow WebRTC to work, but t=
his exposed information which I would not have wanted to be exposed)</stron=
g></li></ol><p>(B) And I also have changed my settings back to the followin=
g</p><p>in the WebRTC 0.1.3 extension for Firefox I have set it to no longe=
r expose my real IP=20
address as well as no longer allowing the following:</p><p>a. navigator.get=
UserMedia<br>b. window.MediaStreamTrack<br>c. window.RTCPeerConnection<br>d=
. window.RTCSessionDescription</p><p>To be clear, <b>these mitigations that=
 I have employed above would be unnecessary if WebRTC had not been designed=
 in such a manner that it so clearly exposes so much user information.</b><=
/p><p>In <a href=3D"https://tools.ietf.org/html/draft-ietf-rtcweb-ip-handli=
ng-03" target=3D"_blank">the document you provided</a>, titled &quot;WebRTC=
 IP Address Handling Requirements - draft-ietf-rtcweb-ip-handling-<wbr>03,&=
quot; it is stated that:</p><p><i>&quot;<span style=3D"font-family:arial,he=
lvetica,sans-serif">1.  If the client is behind a NAT, the client&#39;s pri=
vate IP addresses, typically [<a href=3D"https://tools.ietf.org/html/rfc191=
8" title=3D"&quot;Address Allocation for Private Internets&quot;" target=3D=
"_blank">RFC1918</a>] addresses, can be learned.
</span></i></p><pre class=3D"m_5234864893797269998m_-284210964436143041gmai=
l-newpage"><i><span style=3D"font-family:arial,helvetica,sans-serif"> 2.  I=
f the client tries to hide its physical location through a VPN,
       and the VPN and local OS support routing over multiple interfaces
       (i.e., a &quot;split-tunnel&quot; VPN), WebRTC will discover the pub=
lic
       address for the VPN as well as the ISP public address that the
       VPN runs over.

   3.  If the client is behind a proxy (a client-configured &quot;classical
       application proxy&quot;, as defined in <a href=3D"https://tools.ietf=
.org/html/rfc1919#section-3" target=3D"_blank">[RFC1919], Section=C2=A03</a=
>), but
       direct access to the Internet is also supported, WebRTC&#39;s STUN
       [<a href=3D"https://tools.ietf.org/html/rfc5389" title=3D"&quot;Sess=
ion Traversal Utilities for NAT (STUN)&quot;" target=3D"_blank">RFC5389</a>=
]checks will bypass the proxy and reveal the public
       address of the client.&quot;<br><br></span></i></pre><pre class=3D"m=
_5234864893797269998m_-284210964436143041gmail-newpage"><span style=3D"font=
-family:arial,helvetica,sans-serif">All of these are obvious vulnerabilitie=
s and serious problems (not only #2 as the document suggests).<br><br></spa=
n></pre><pre class=3D"m_5234864893797269998m_-284210964436143041gmail-newpa=
ge"><span style=3D"font-family:arial,helvetica,sans-serif">It is stated in =
the document that, <br><i><br>&quot;we want to avoid solutions that disable=
 WebRTC or make it harder to use.&quot;<br><br></i></span></pre><pre class=
=3D"m_5234864893797269998m_-284210964436143041gmail-newpage"><span style=3D=
"font-family:arial,helvetica,sans-serif">Yet, so far as my observations of =
WebRTC in the current version of Firefox with Ubuntu 16.10 have shown,<br><=
/span></pre><pre class=3D"m_5234864893797269998m_-284210964436143041gmail-n=
ewpage"><span style=3D"font-family:arial,helvetica,sans-serif">there appear=
s to be no other choice other than to disable WebRTC because its use comple=
tely compromises <br></span></pre><pre class=3D"m_5234864893797269998m_-284=
210964436143041gmail-newpage"><span style=3D"font-family:arial,helvetica,sa=
ns-serif">privacy and thus makes all other aspects of your system more diff=
icult generally.  <br><br></span></pre><pre class=3D"m_5234864893797269998m=
_-284210964436143041gmail-newpage"><span style=3D"font-family:arial,helveti=
ca,sans-serif">It is stated in the document you cited that the goals here a=
re to &quot;</span>Provide a privacy-friendly default behavior which strike=
s the
      right balance between privacy and media performance for most users
      and use cases. (and) For users who care more about one versus the oth=
er, provide a
      means to customize the experience.&quot;</pre><pre class=3D"m_5234864=
893797269998m_-284210964436143041gmail-newpage"><span style=3D"font-family:=
arial,helvetica,sans-serif">This has clearlly not been accomplished in the =
context of Ubuntu 16.10 64-bit with Firefox 53.0.2 (64-bit).<br><br></span>=
</pre><pre class=3D"m_5234864893797269998m_-284210964436143041gmail-newpage=
"><span style=3D"font-family:arial,helvetica,sans-serif">Hence my request f=
or a new security review of WebRTC based on observed vulnerabilities.<br></=
span></pre><p><span style=3D"font-family:arial,helvetica,sans-serif"></span=
><br></p><strong></strong><div><div><div><div><div><br></div></div></div></=
div></div></div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">O=
n Thu, May 25, 2017 at 1:04 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">Colin,<di=
v><br></div><div>This is an important topic, and the group has reviewed thi=
s situation in detail. The results have been written up in <a href=3D"https=
://tools.ietf.org/html/draft-ietf-rtcweb-ip-handling-03" target=3D"_blank">=
this document</a>, and to my knowledge most browsers that support WebRTC cu=
rrently conform to its recommendations.</div><div><br></div><div>However, I=
&#39;m not totally clear on your specific technical concern. You linked to =
an (now outdated) stackexchange page, but could you provide more detail abo=
ut the specific situation you are concerned about?</div><div><br></div><div=
>Justin</div></div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote=
"><div><div class=3D"m_5234864893797269998m_-284210964436143041h5">On Wed, =
May 24, 2017 at 5:35 PM, Colin Gallagher <span dir=3D"ltr">&lt;<a href=3D"m=
ailto:colingallagher.rpcv@gmail.com" target=3D"_blank">colingallagher.rpcv@=
gmail.com</a><wbr>&gt;</span> wrote:<br></div></div><blockquote class=3D"gm=
ail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-le=
ft:1ex"><div><div class=3D"m_5234864893797269998m_-284210964436143041h5"><d=
iv dir=3D"ltr"><div><div><div><div><div><div><div><div>To:=C2=A0 IETF RTCWe=
b WG<br><br></div>From:=C2=A0 Colin Gallagher<br></div>=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 <a href=3D"https://www.linkedin.=
com/in/colingallagher" target=3D"_blank">https://www.linkedin.com/in/co<wbr=
>lingallagher</a><br>=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0 <a href=3D"https://lifeboat.com/ex/bios.colin.gallagher" target=3D"_=
blank">https://lifeboat.com/ex/bios.c<wbr>olin.gallagher</a><br><br></div>R=
equest for New Security Review of WebRTC based on observed vulnerabilities =
in certain examples<br><br></div>Dear IETF RTCWeb WG,<br><br></div><div>Thi=
s request relates to WebRTC.=C2=A0 The WebRTC is in draft at <a href=3D"htt=
ps://w3c.github.io/webrtc-pc/" target=3D"_blank">https://w3c.github.io/webr=
tc-p<wbr>c/</a> and is=20
described also at=20
<a href=3D"https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API#WebR=
TC_interfaces" target=3D"_blank">https://developer.mozilla.org/<wbr>en-US/d=
ocs/Web/API/WebRTC_API#<wbr>WebRTC_interfaces</a></div><div><br></div>There=
 are certain examples where the use of WebRTC has presented security vulner=
abilities.=C2=A0 While not exhaustive, there are two I wanted to mention th=
at I had been exposed to recently.<br></div><div><br></div>1) The use of a =
video conferencing service which relies upon WebRTC (that is a well establi=
shed service)<br></div>2) The examination of a browser based WebRTC version=
 of a decentralized exchange (that is currently under development)<br><br><=
/div>In both cases the security vulnerability which is described below was =
the result of the use of WebRTC.<br><br>The problem under consideration is =
as follows:<br><br>As per the first example (the video conferencing service=
), recently
 I came upon a vendor who shall remain unnamed who uses WebRTC as the=20
backbone for their video conferencing service.=C2=A0 As a result, I found I=
=20
was unable to access the service in question without disabling some=20
security settings which I would have preferred to leave intact. <br><br>My =
system is (based on what the aforementioned service detects) Mozilla/5.0 (X=
11; Ubuntu; Linux x86_64; rv:53.0)=20
Gecko/20100101 Firefox/53.0.=C2=A0 This is Ubuntu 16.10 64-bit with Firefox=
=20
53.0.2.<br><br>However, quite some <i>long</i> while back I disabled WebRTC=
 because of the security vulnerability (described in part <a href=3D"https:=
//security.stackexchange.com/questions/82129/what-is-webrtc-and-how-can-i-p=
rotect-myself-against-leaking-information-that-doe" target=3D"_blank">here =
on information security stack exchange</a>).=C2=A0 So as to mitigate this v=
ulnerability, I changed settings in Firefox as follows:<br><br><ol><li>I ty=
ped <em>about:config</em> in the address bar</li><li>I found the setting <e=
m>media.peerconnection.enabled</em></li><li>I set it to <strong>false</stro=
ng></li></ol><p><strong></strong>Unfortunately,
 this setting does not allow Firefox to work with the video conferencing
 service in question, which relies entirely on WebRTC.<br></p><p>So I had t=
o go back in and change the media.peerconnection.enabled setting to <b>true=
</b>
 in Firefox in order to get it to work with the service in question.=20
While this enabled me to conference with teams that desire to use the=20
video conferencing service that rely wholly upon WebRTC, it concerns me=20
because of the security vulnerability.</p><p>One suggested mitigation was h=
aving NoScript, but using NoScript was not helpful (you generally need to d=
isable it for the site you are viewing that relies upon WebRTC, in order to=
 obtain any functionality, and whether NoScript is off or on, WebRTC still =
causes the vulnerability of leaking information as previously described).<b=
r></p><p>Furthermore, upon reload of the browser, even after setting media.=
peerconnection.enabled to <b>true</b>,
 the video conferencing service wouldn&#39;t work until I installed the=20
WebRTC 0.1.3 extension for Firefox and set it to expose my real IP=20
address as well as allowing the following:</p><p>a. navigator.getUserMedia<=
br>b. window.MediaStreamTrack<br>c. window.RTCPeerConnection<br>d. window.R=
TCSessionDescription<br></p><p>That seemed to me to be a direct consequence=
 of the persistent and ongoing security vulnerability in WebRTC.<br></p><p>=
I
 contacted the video conferencing service provider and their solution=20
was simply to state that &quot;If the peer to peer connection is of concern=
,=20
you can utilize our=20
premium version which will route traffic through a forwarding server=20
with in our environment that handles the processing of the video and=20
sound of all users in the conference and send it to each user=20
individually rather than using a peer to peer connection.&quot;=C2=A0 In ot=
her=20
words, they expect that people should have to pay in order to mitigate <a h=
ref=3D"https://security.stackexchange.com/questions/82129/what-is-webrtc-an=
d-how-can-i-protect-myself-against-leaking-information-that-doe" target=3D"=
_blank">this WebRTC security vulnerability</a>, because they are unwilling =
to design to protect all users from it.</p><p>In a similar fashion, in the =
second example of the browser based WebRTC version of a decentralized excha=
nge, the user&#39;s financial information is periodically exposed as a resu=
lt of the information leakage which is presently caused by WebRTC.=C2=A0 (F=
ortunately, this browser based WebRTC decentralized exchange is under devel=
opment and is not currently open to actual usage.)<br></p><p>Although
 undoubtedly this has been discussed here before, I am asking that WebRTC s=
ecurity issues be
 reconsidered and that a new security review be done to ameliorate or=20
eliminate this problem in terms of WebRTC leaking this information.=C2=A0 I=
t is my view that WebRTC should be further designed and developed with an e=
ye to mitigating this serious problem, so that users of services which migh=
t rely partially or wholly upon WebRTC would not have to be concerned about=
 such information leakage.<br></p><p>Respectfully,</p>Colin Gallagher<br><b=
r></div>
<br></div></div>______________________________<wbr>_________________<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/l<wbr>istinfo/rtcweb</a><br=
>
<br></blockquote></div><br></div>
</blockquote></div><br></div>
<br>______________________________<wbr>_________________<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/l<wbr>istinfo/rtcweb</a><br=
>
<br></blockquote></div></div>
</div></div></blockquote></div><br></div>

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From: Justin Uberti <juberti@google.com>
Date: Fri, 26 May 2017 10:49:56 -0700
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Colin,

As Eric mentioned, the group has reviewed this exact issue, and the
aforementioned document represents the result of the review.

I am still not quite clear on the exact technical issue you would like to
see addressed and why it is, as you claim, a serious vulnerability.

Comments inline.

On Fri, May 26, 2017 at 12:58 AM, Colin Gallagher <
colingallagher.rpcv@gmail.com> wrote:

> Justin,
>
> It's my understanding that various browsers have made an attempt to
> conform to the recommendations in the document that you have provided
> <https://tools.ietf.org/html/draft-ietf-rtcweb-ip-handling-03>.
> Unfortunately, at least in Firefox (the most current version) so far as I
> can tell, there is a serious vulnerability.  To be more specific, the
> problem was observed in my operating system with my current browser, whic=
h
> is Ubuntu 16.10 64-bit with Firefox 53.0.2 (64-bit).  The browser does no=
t
> address the WebRTC problems by default.
>
> The stackexchange page that I cited originally is actually not outdated i=
n
> the sense that its method of measuring whether or not the WebRTC
> vulnerability exists on a given system is still intact and works.  The
> stackexchange page suggests two pages to determine whether or not you hav=
e
> leakage / WebRTC related vulnerability.  These pages are:
>
> 1) Browser leaks test for WebRTC https://browserleaks.com/webrtc
> 2) IP Leaks https://ipleak.net/
>
> Currently for Browser leaks test for WebRTC (due to that I have again
> changed settings in my system so that I will not have to be exposed to th=
is
> vulnerability) the settings show up on the page as:
> RTCPeerConnection=C3=97False
> RTCDataChannel=C3=97False
> ORTC (Microsoft Edge)=C3=97False (for example....)
>
> And under IP leaks, it currently shows "No leak, RTCPeerConnection not
> available."
>
> *That is because:*
>
> (A) I have (again) changed my settings in about:config back to the
> following
>
>
>    1. I typed *about:config* in the address bar
>    2. I found the setting *media.peerconnection.enabled*
>    3. I set it to *false  (it was previously set at true, so as to allow
>    WebRTC to work, but this exposed information which I would not have wa=
nted
>    to be exposed)*
>
> (B) And I also have changed my settings back to the following
>
> in the WebRTC 0.1.3 extension for Firefox I have set it to no longer
> expose my real IP address as well as no longer allowing the following:
>
> a. navigator.getUserMedia
> b. window.MediaStreamTrack
> c. window.RTCPeerConnection
> d. window.RTCSessionDescription
>
> To be clear, *these mitigations that I have employed above would be
> unnecessary if WebRTC had not been designed in such a manner that it so
> clearly exposes so much user information.*
>
> In the document you provided
> <https://tools.ietf.org/html/draft-ietf-rtcweb-ip-handling-03>, titled
> "WebRTC IP Address Handling Requirements - draft-ietf-rtcweb-ip-handling-=
03,"
> it is stated that:
>
> *"1. If the client is behind a NAT, the client's private IP addresses,
> typically [RFC1918 <https://tools.ietf.org/html/rfc1918>] addresses, can =
be
> learned. *
>
>
>
> * 2.  If the client tries to hide its physical location through a VPN,
>        and the VPN and local OS support routing over multiple interfaces
>        (i.e., a "split-tunnel" VPN), WebRTC will discover the public
>        address for the VPN as well as the ISP public address that the
>        VPN runs over.
>
>    3.  If the client is behind a proxy (a client-configured "classical
>        application proxy", as defined in [RFC1919], Section 3 <https://to=
ols.ietf.org/html/rfc1919#section-3>), but
>        direct access to the Internet is also supported, WebRTC's STUN
>        [RFC5389 <https://tools.ietf.org/html/rfc5389>]checks will bypass =
the proxy and reveal the public
>        address of the client."*
>
> All of these are obvious vulnerabilities and serious problems (not only #=
2 as the document suggests).
>
>
The document discusses #1 and #3 in detail, and outlines mechanisms that
can be used to address these cases if needed.

If, despite the findings in the document, you think that these are indeed
serious problems, it would be good to outline exactly why.

>
> It is stated in the document that,
>
>
>
> *"we want to avoid solutions that disable WebRTC or make it harder to use=
."*
>
> Yet, so far as my observations of WebRTC in the current version of Firefo=
x with Ubuntu 16.10 have shown,
>
> there appears to be no other choice other than to disable WebRTC because =
its use completely compromises
>
> privacy and thus makes all other aspects of your system more difficult ge=
nerally.
>
>
> It is stated in the document you cited that the goals here are to "Provid=
e a privacy-friendly default behavior which strikes the
>       right balance between privacy and media performance for most users
>       and use cases. (and) For users who care more about one versus the o=
ther, provide a
>       means to customize the experience."
>
> This has clearlly not been accomplished in the context of Ubuntu 16.10 64=
-bit with Firefox 53.0.2 (64-bit).
>
> Hence my request for a new security review of WebRTC based on observed vu=
lnerabilities.
>
>
To my knowledge, the document addresses these goals. If you believe Firefox
is not conforming, please file a bug; if you think the document is in
error, please point out where and why.

>
>
>
> On Thu, May 25, 2017 at 1:04 PM, Justin Uberti <juberti@google.com> wrote=
:
>
>> Colin,
>>
>> This is an important topic, and the group has reviewed this situation in
>> detail. The results have been written up in this document
>> <https://tools.ietf.org/html/draft-ietf-rtcweb-ip-handling-03>, and to
>> my knowledge most browsers that support WebRTC currently conform to its
>> recommendations.
>>
>> However, I'm not totally clear on your specific technical concern. You
>> linked to an (now outdated) stackexchange page, but could you provide mo=
re
>> detail about the specific situation you are concerned about?
>>
>> Justin
>>
>> On Wed, May 24, 2017 at 5:35 PM, Colin Gallagher <
>> colingallagher.rpcv@gmail.com> wrote:
>>
>>> To:  IETF RTCWeb WG
>>>
>>> From:  Colin Gallagher
>>>            https://www.linkedin.com/in/colingallagher
>>>            https://lifeboat.com/ex/bios.colin.gallagher
>>>
>>> Request for New Security Review of WebRTC based on observed
>>> vulnerabilities in certain examples
>>>
>>> Dear IETF RTCWeb WG,
>>>
>>> This request relates to WebRTC.  The WebRTC is in draft at
>>> https://w3c.github.io/webrtc-pc/ and is described also at
>>> https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API#
>>> WebRTC_interfaces
>>>
>>> There are certain examples where the use of WebRTC has presented
>>> security vulnerabilities.  While not exhaustive, there are two I wanted=
 to
>>> mention that I had been exposed to recently.
>>>
>>> 1) The use of a video conferencing service which relies upon WebRTC
>>> (that is a well established service)
>>> 2) The examination of a browser based WebRTC version of a decentralized
>>> exchange (that is currently under development)
>>>
>>> In both cases the security vulnerability which is described below was
>>> the result of the use of WebRTC.
>>>
>>> The problem under consideration is as follows:
>>>
>>> As per the first example (the video conferencing service), recently I
>>> came upon a vendor who shall remain unnamed who uses WebRTC as the back=
bone
>>> for their video conferencing service.  As a result, I found I was unabl=
e to
>>> access the service in question without disabling some security settings
>>> which I would have preferred to leave intact.
>>>
>>> My system is (based on what the aforementioned service detects)
>>> Mozilla/5.0 (X11; Ubuntu; Linux x86_64; rv:53.0) Gecko/20100101
>>> Firefox/53.0.  This is Ubuntu 16.10 64-bit with Firefox 53.0.2.
>>>
>>> However, quite some *long* while back I disabled WebRTC because of the
>>> security vulnerability (described in part here on information security
>>> stack exchange
>>> <https://security.stackexchange.com/questions/82129/what-is-webrtc-and-=
how-can-i-protect-myself-against-leaking-information-that-doe>).
>>> So as to mitigate this vulnerability, I changed settings in Firefox as
>>> follows:
>>>
>>>
>>>    1. I typed *about:config* in the address bar
>>>    2. I found the setting *media.peerconnection.enabled*
>>>    3. I set it to *false*
>>>
>>> Unfortunately, this setting does not allow Firefox to work with the
>>> video conferencing service in question, which relies entirely on WebRTC=
.
>>>
>>> So I had to go back in and change the media.peerconnection.enabled
>>> setting to *true* in Firefox in order to get it to work with the
>>> service in question. While this enabled me to conference with teams tha=
t
>>> desire to use the video conferencing service that rely wholly upon WebR=
TC,
>>> it concerns me because of the security vulnerability.
>>>
>>> One suggested mitigation was having NoScript, but using NoScript was no=
t
>>> helpful (you generally need to disable it for the site you are viewing =
that
>>> relies upon WebRTC, in order to obtain any functionality, and whether
>>> NoScript is off or on, WebRTC still causes the vulnerability of leaking
>>> information as previously described).
>>>
>>> Furthermore, upon reload of the browser, even after setting
>>> media.peerconnection.enabled to *true*, the video conferencing service
>>> wouldn't work until I installed the WebRTC 0.1.3 extension for Firefox =
and
>>> set it to expose my real IP address as well as allowing the following:
>>>
>>> a. navigator.getUserMedia
>>> b. window.MediaStreamTrack
>>> c. window.RTCPeerConnection
>>> d. window.RTCSessionDescription
>>>
>>> That seemed to me to be a direct consequence of the persistent and
>>> ongoing security vulnerability in WebRTC.
>>>
>>> I contacted the video conferencing service provider and their solution
>>> was simply to state that "If the peer to peer connection is of concern,=
 you
>>> can utilize our premium version which will route traffic through a
>>> forwarding server with in our environment that handles the processing o=
f
>>> the video and sound of all users in the conference and send it to each =
user
>>> individually rather than using a peer to peer connection."  In other wo=
rds,
>>> they expect that people should have to pay in order to mitigate this
>>> WebRTC security vulnerability
>>> <https://security.stackexchange.com/questions/82129/what-is-webrtc-and-=
how-can-i-protect-myself-against-leaking-information-that-doe>,
>>> because they are unwilling to design to protect all users from it.
>>>
>>> In a similar fashion, in the second example of the browser based WebRTC
>>> version of a decentralized exchange, the user's financial information i=
s
>>> periodically exposed as a result of the information leakage which is
>>> presently caused by WebRTC.  (Fortunately, this browser based WebRTC
>>> decentralized exchange is under development and is not currently open t=
o
>>> actual usage.)
>>>
>>> Although undoubtedly this has been discussed here before, I am asking
>>> that WebRTC security issues be reconsidered and that a new security rev=
iew
>>> be done to ameliorate or eliminate this problem in terms of WebRTC leak=
ing
>>> this information.  It is my view that WebRTC should be further designed=
 and
>>> developed with an eye to mitigating this serious problem, so that users=
 of
>>> services which might rely partially or wholly upon WebRTC would not hav=
e to
>>> be concerned about such information leakage.
>>>
>>> Respectfully,
>>> Colin Gallagher
>>>
>>>
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>
>>>
>>
>

--001a113ff9a66765aa055070f882
Content-Type: text/html; charset="UTF-8"
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Colin,<div><br></div><div>As Eric mentioned, the group has=
 reviewed this exact issue, and the aforementioned document represents the =
result of the review.</div><div><br></div><div>I am still not quite clear o=
n the exact technical issue you would like to see addressed and why it is, =
as you claim, a serious vulnerability.</div><div><br></div><div>Comments in=
line.</div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Fri=
, May 26, 2017 at 12:58 AM, Colin Gallagher <span dir=3D"ltr">&lt;<a href=
=3D"mailto:colingallagher.rpcv@gmail.com" target=3D"_blank">colingallagher.=
rpcv@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"><div><div><div><div><div><div><div><div>Justin,<br><br></div>I=
t&#39;s my understanding that various browsers have made an attempt to conf=
orm to the recommendations in <a href=3D"https://tools.ietf.org/html/draft-=
ietf-rtcweb-ip-handling-03" target=3D"_blank">the document that you have pr=
ovided</a>.=C2=A0 Unfortunately, at least in Firefox (the most current vers=
ion) so far as I can tell, there is a serious vulnerability.=C2=A0 To be mo=
re specific, the problem was observed in my operating system with my curren=
t browser, which is Ubuntu 16.10 64-bit with Firefox=20
53.0.2 (64-bit).=C2=A0 The browser does not address the WebRTC problems by =
default.<br><br></div>The stackexchange page that I cited originally is act=
ually not outdated in the sense that its method of measuring whether or not=
 the WebRTC vulnerability exists on a given system is still intact and work=
s.=C2=A0 The stackexchange page suggests two pages to determine whether or =
not you have leakage / WebRTC related vulnerability.=C2=A0 These pages are:=
<br><br></div>1) Browser leaks test for WebRTC <a href=3D"https://browserle=
aks.com/webrtc" target=3D"_blank">https://browserleaks.com/<wbr>webrtc</a><=
br></div>2) IP Leaks <a href=3D"https://ipleak.net/" target=3D"_blank">http=
s://ipleak.net/</a><br><br></div>Currently for Browser leaks test for WebRT=
C (due to that I have again changed settings in my system so that I will no=
t have to be exposed to this vulnerability) the settings show up on the pag=
e as:<br>RTCPeerConnection<span class=3D"m_3620305583784542015gmail-bad">=
=C3=97</span>False<br>RTCDataChannel<span class=3D"m_3620305583784542015gma=
il-bad">=C3=97</span>False<br>ORTC (Microsoft Edge)<span class=3D"m_3620305=
583784542015gmail-bad">=C3=97</span>False (for example....)<br><br></div>An=
d under IP leaks, it currently shows &quot;No leak, RTCPeerConnection not a=
vailable.&quot;=C2=A0 <br><br></div><i>That is because:</i><br><br></div>(A=
) I have (again) changed my settings in about:config back to the following<=
br><br><ol><span class=3D""><li>I typed <em>about:config</em> in the addres=
s bar</li><li>I found the setting <em>media.peerconnection.enabled</em></li=
></span><li>I set it to <strong>false=C2=A0 (it was previously set at true,=
 so as to allow WebRTC to work, but this exposed information which I would =
not have wanted to be exposed)</strong></li></ol><p>(B) And I also have cha=
nged my settings back to the following</p><p>in the WebRTC 0.1.3 extension =
for Firefox I have set it to no longer expose my real IP=20
address as well as no longer allowing the following:</p><span class=3D""><p=
>a. navigator.getUserMedia<br>b. window.MediaStreamTrack<br>c. window.RTCPe=
erConnection<br>d. window.RTCSessionDescription</p></span><p>To be clear, <=
b>these mitigations that I have employed above would be unnecessary if WebR=
TC had not been designed in such a manner that it so clearly exposes so muc=
h user information.</b></p><p>In <a href=3D"https://tools.ietf.org/html/dra=
ft-ietf-rtcweb-ip-handling-03" target=3D"_blank">the document you provided<=
/a>, titled &quot;WebRTC IP Address Handling Requirements - draft-ietf-rtcw=
eb-ip-handling-<wbr>03,&quot; it is stated that:</p><p><i>&quot;<span style=
=3D"font-family:arial,helvetica,sans-serif">1.  If the client is behind a N=
AT, the client&#39;s private IP addresses, typically [<a href=3D"https://to=
ols.ietf.org/html/rfc1918" title=3D"&quot;Address Allocation for Private In=
ternets&quot;" target=3D"_blank">RFC1918</a>] addresses, can be learned.
</span></i></p><pre class=3D"m_3620305583784542015gmail-newpage"><i><span s=
tyle=3D"font-family:arial,helvetica,sans-serif"> 2.  If the client tries to=
 hide its physical location through a VPN,
       and the VPN and local OS support routing over multiple interfaces
       (i.e., a &quot;split-tunnel&quot; VPN), WebRTC will discover the pub=
lic
       address for the VPN as well as the ISP public address that the
       VPN runs over.

   3.  If the client is behind a proxy (a client-configured &quot;classical
       application proxy&quot;, as defined in <a href=3D"https://tools.ietf=
.org/html/rfc1919#section-3" target=3D"_blank">[RFC1919], Section=C2=A03</a=
>), but
       direct access to the Internet is also supported, WebRTC&#39;s STUN
       [<a href=3D"https://tools.ietf.org/html/rfc5389" title=3D"&quot;Sess=
ion Traversal Utilities for NAT (STUN)&quot;" target=3D"_blank">RFC5389</a>=
]checks will bypass the proxy and reveal the public
       address of the client.&quot;<br><br></span></i></pre><pre class=3D"m=
_3620305583784542015gmail-newpage"><span style=3D"font-family:arial,helveti=
ca,sans-serif">All of these are obvious vulnerabilities and serious problem=
s (not only #2 as the document suggests).<br></span></pre></div></blockquot=
e><div><br></div><div>The document discusses #1 and #3 in detail, and outli=
nes mechanisms that can be used to address these cases if needed.</div><div=
><br></div><div>If, despite the findings in the document, you think that th=
ese are indeed serious problems, it would be good to outline exactly why.</=
div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-lef=
t:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><pre class=3D"m_3620305=
583784542015gmail-newpage"><span style=3D"font-family:arial,helvetica,sans-=
serif"><br></span></pre><pre class=3D"m_3620305583784542015gmail-newpage"><=
span style=3D"font-family:arial,helvetica,sans-serif">It is stated in the d=
ocument that, <br><i><br>&quot;we want to avoid solutions that disable WebR=
TC or make it harder to use.&quot;<br><br></i></span></pre><pre class=3D"m_=
3620305583784542015gmail-newpage"><span style=3D"font-family:arial,helvetic=
a,sans-serif">Yet, so far as my observations of WebRTC in the current versi=
on of Firefox with Ubuntu 16.10 have shown,<br></span></pre><pre class=3D"m=
_3620305583784542015gmail-newpage"><span style=3D"font-family:arial,helveti=
ca,sans-serif">there appears to be no other choice other than to disable We=
bRTC because its use completely compromises <br></span></pre><pre class=3D"=
m_3620305583784542015gmail-newpage"><span style=3D"font-family:arial,helvet=
ica,sans-serif">privacy and thus makes all other aspects of your system mor=
e difficult generally.</span></pre></div></blockquote><blockquote class=3D"=
gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-=
left:1ex"><div dir=3D"ltr"><pre class=3D"m_3620305583784542015gmail-newpage=
"><span style=3D"font-family:arial,helvetica,sans-serif"><br></span></pre><=
pre class=3D"m_3620305583784542015gmail-newpage"><span style=3D"font-family=
:arial,helvetica,sans-serif">It is stated in the document you cited that th=
e goals here are to &quot;</span>Provide a privacy-friendly default behavio=
r which strikes the
      right balance between privacy and media performance for most users
      and use cases. (and) For users who care more about one versus the oth=
er, provide a
      means to customize the experience.&quot;</pre><pre class=3D"m_3620305=
583784542015gmail-newpage"><span style=3D"font-family:arial,helvetica,sans-=
serif">This has clearlly not been accomplished in the context of Ubuntu 16.=
10 64-bit with Firefox 53.0.2 (64-bit).<br><br></span></pre><pre class=3D"m=
_3620305583784542015gmail-newpage"><span style=3D"font-family:arial,helveti=
ca,sans-serif">Hence my request for a new security review of WebRTC based o=
n observed vulnerabilities.<br></span></pre></div></blockquote><div><br></d=
iv><div>To my knowledge, the document addresses these goals. If you believe=
 Firefox is not conforming, please file a bug; if you think the document is=
 in error, please point out where and why.</div><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"><pre class=3D"m_3620305583784542015gmail-newpage"><spa=
n style=3D"font-family:arial,helvetica,sans-serif"></span></pre><p><span st=
yle=3D"font-family:arial,helvetica,sans-serif"></span><br></p><strong></str=
ong><div><div><div><div><div><br></div></div></div></div></div></div><div c=
lass=3D"HOEnZb"><div class=3D"h5"><div class=3D"gmail_extra"><br><div class=
=3D"gmail_quote">On Thu, May 25, 2017 at 1:04 PM, Justin Uberti <span dir=
=3D"ltr">&lt;<a href=3D"mailto:juberti@google.com" target=3D"_blank">jubert=
i@google.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" sty=
le=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div d=
ir=3D"ltr">Colin,<div><br></div><div>This is an important topic, and the gr=
oup has reviewed this situation in detail. The results have been written up=
 in <a href=3D"https://tools.ietf.org/html/draft-ietf-rtcweb-ip-handling-03=
" target=3D"_blank">this document</a>, and to my knowledge most browsers th=
at support WebRTC currently conform to its recommendations.</div><div><br><=
/div><div>However, I&#39;m not totally clear on your specific technical con=
cern. You linked to an (now outdated) stackexchange page, but could you pro=
vide more detail about the specific situation you are concerned about?</div=
><div><br></div><div>Justin</div></div><div class=3D"gmail_extra"><br><div =
class=3D"gmail_quote"><div><div class=3D"m_3620305583784542015h5">On Wed, M=
ay 24, 2017 at 5:35 PM, Colin Gallagher <span dir=3D"ltr">&lt;<a href=3D"ma=
ilto:colingallagher.rpcv@gmail.com" target=3D"_blank">colingallagher.rpcv@g=
mail.com</a><wbr>&gt;</span> wrote:<br></div></div><blockquote class=3D"gma=
il_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-lef=
t:1ex"><div><div class=3D"m_3620305583784542015h5"><div dir=3D"ltr"><div><d=
iv><div><div><div><div><div><div>To:=C2=A0 IETF RTCWeb WG<br><br></div>From=
:=C2=A0 Colin Gallagher<br></div>=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0 <a href=3D"https://www.linkedin.com/in/colingallagher" t=
arget=3D"_blank">https://www.linkedin.com/in/co<wbr>lingallagher</a><br>=C2=
=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 <a href=3D"https:=
//lifeboat.com/ex/bios.colin.gallagher" target=3D"_blank">https://lifeboat.=
com/ex/bios.c<wbr>olin.gallagher</a><br><br></div>Request for New Security =
Review of WebRTC based on observed vulnerabilities in certain examples<br><=
br></div>Dear IETF RTCWeb WG,<br><br></div><div>This request relates to Web=
RTC.=C2=A0 The WebRTC is in draft at <a href=3D"https://w3c.github.io/webrt=
c-pc/" target=3D"_blank">https://w3c.github.io/webrtc-p<wbr>c/</a> and is=
=20
described also at=20
<a href=3D"https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API#WebR=
TC_interfaces" target=3D"_blank">https://developer.mozilla.org/<wbr>en-US/d=
ocs/Web/API/WebRTC_API#<wbr>WebRTC_interfaces</a></div><div><br></div>There=
 are certain examples where the use of WebRTC has presented security vulner=
abilities.=C2=A0 While not exhaustive, there are two I wanted to mention th=
at I had been exposed to recently.<br></div><div><br></div>1) The use of a =
video conferencing service which relies upon WebRTC (that is a well establi=
shed service)<br></div>2) The examination of a browser based WebRTC version=
 of a decentralized exchange (that is currently under development)<br><br><=
/div>In both cases the security vulnerability which is described below was =
the result of the use of WebRTC.<br><br>The problem under consideration is =
as follows:<br><br>As per the first example (the video conferencing service=
), recently
 I came upon a vendor who shall remain unnamed who uses WebRTC as the=20
backbone for their video conferencing service.=C2=A0 As a result, I found I=
=20
was unable to access the service in question without disabling some=20
security settings which I would have preferred to leave intact. <br><br>My =
system is (based on what the aforementioned service detects) Mozilla/5.0 (X=
11; Ubuntu; Linux x86_64; rv:53.0)=20
Gecko/20100101 Firefox/53.0.=C2=A0 This is Ubuntu 16.10 64-bit with Firefox=
=20
53.0.2.<br><br>However, quite some <i>long</i> while back I disabled WebRTC=
 because of the security vulnerability (described in part <a href=3D"https:=
//security.stackexchange.com/questions/82129/what-is-webrtc-and-how-can-i-p=
rotect-myself-against-leaking-information-that-doe" target=3D"_blank">here =
on information security stack exchange</a>).=C2=A0 So as to mitigate this v=
ulnerability, I changed settings in Firefox as follows:<br><br><ol><li>I ty=
ped <em>about:config</em> in the address bar</li><li>I found the setting <e=
m>media.peerconnection.enabled</em></li><li>I set it to <strong>false</stro=
ng></li></ol><p><strong></strong>Unfortunately,
 this setting does not allow Firefox to work with the video conferencing
 service in question, which relies entirely on WebRTC.<br></p><p>So I had t=
o go back in and change the media.peerconnection.enabled setting to <b>true=
</b>
 in Firefox in order to get it to work with the service in question.=20
While this enabled me to conference with teams that desire to use the=20
video conferencing service that rely wholly upon WebRTC, it concerns me=20
because of the security vulnerability.</p><p>One suggested mitigation was h=
aving NoScript, but using NoScript was not helpful (you generally need to d=
isable it for the site you are viewing that relies upon WebRTC, in order to=
 obtain any functionality, and whether NoScript is off or on, WebRTC still =
causes the vulnerability of leaking information as previously described).<b=
r></p><p>Furthermore, upon reload of the browser, even after setting media.=
peerconnection.enabled to <b>true</b>,
 the video conferencing service wouldn&#39;t work until I installed the=20
WebRTC 0.1.3 extension for Firefox and set it to expose my real IP=20
address as well as allowing the following:</p><p>a. navigator.getUserMedia<=
br>b. window.MediaStreamTrack<br>c. window.RTCPeerConnection<br>d. window.R=
TCSessionDescription<br></p><p>That seemed to me to be a direct consequence=
 of the persistent and ongoing security vulnerability in WebRTC.<br></p><p>=
I
 contacted the video conferencing service provider and their solution=20
was simply to state that &quot;If the peer to peer connection is of concern=
,=20
you can utilize our=20
premium version which will route traffic through a forwarding server=20
with in our environment that handles the processing of the video and=20
sound of all users in the conference and send it to each user=20
individually rather than using a peer to peer connection.&quot;=C2=A0 In ot=
her=20
words, they expect that people should have to pay in order to mitigate <a h=
ref=3D"https://security.stackexchange.com/questions/82129/what-is-webrtc-an=
d-how-can-i-protect-myself-against-leaking-information-that-doe" target=3D"=
_blank">this WebRTC security vulnerability</a>, because they are unwilling =
to design to protect all users from it.</p><p>In a similar fashion, in the =
second example of the browser based WebRTC version of a decentralized excha=
nge, the user&#39;s financial information is periodically exposed as a resu=
lt of the information leakage which is presently caused by WebRTC.=C2=A0 (F=
ortunately, this browser based WebRTC decentralized exchange is under devel=
opment and is not currently open to actual usage.)<br></p><p>Although
 undoubtedly this has been discussed here before, I am asking that WebRTC s=
ecurity issues be
 reconsidered and that a new security review be done to ameliorate or=20
eliminate this problem in terms of WebRTC leaking this information.=C2=A0 I=
t is my view that WebRTC should be further designed and developed with an e=
ye to mitigating this serious problem, so that users of services which migh=
t rely partially or wholly upon WebRTC would not have to be concerned about=
 such information leakage.<br></p><p>Respectfully,</p>Colin Gallagher<br><b=
r></div>
<br></div></div>______________________________<wbr>_________________<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/l<wbr>istinfo/rtcweb</a><br=
>
<br></blockquote></div><br></div>
</blockquote></div><br></div>
</div></div></blockquote></div><br></div></div>

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It seems the consensus of the group here is essentially a refusal to
further review and potentially redesign WebRTC to remove the serious
security vulnerabilities that it presents to its users.

Thank you for your time.  I will not further respond to this thread.



On Fri, May 26, 2017 at 10:49 AM, Justin Uberti <juberti@google.com> wrote:

> Colin,
>
> As Eric mentioned, the group has reviewed this exact issue, and the
> aforementioned document represents the result of the review.
>
> I am still not quite clear on the exact technical issue you would like to
> see addressed and why it is, as you claim, a serious vulnerability.
>
> Comments inline.
>
> On Fri, May 26, 2017 at 12:58 AM, Colin Gallagher <
> colingallagher.rpcv@gmail.com> wrote:
>
>> Justin,
>>
>> It's my understanding that various browsers have made an attempt to
>> conform to the recommendations in the document that you have provided
>> <https://tools.ietf.org/html/draft-ietf-rtcweb-ip-handling-03>.
>> Unfortunately, at least in Firefox (the most current version) so far as =
I
>> can tell, there is a serious vulnerability.  To be more specific, the
>> problem was observed in my operating system with my current browser, whi=
ch
>> is Ubuntu 16.10 64-bit with Firefox 53.0.2 (64-bit).  The browser does n=
ot
>> address the WebRTC problems by default.
>>
>> The stackexchange page that I cited originally is actually not outdated
>> in the sense that its method of measuring whether or not the WebRTC
>> vulnerability exists on a given system is still intact and works.  The
>> stackexchange page suggests two pages to determine whether or not you ha=
ve
>> leakage / WebRTC related vulnerability.  These pages are:
>>
>> 1) Browser leaks test for WebRTC https://browserleaks.com/webrtc
>> 2) IP Leaks https://ipleak.net/
>>
>> Currently for Browser leaks test for WebRTC (due to that I have again
>> changed settings in my system so that I will not have to be exposed to t=
his
>> vulnerability) the settings show up on the page as:
>> RTCPeerConnection=C3=97False
>> RTCDataChannel=C3=97False
>> ORTC (Microsoft Edge)=C3=97False (for example....)
>>
>> And under IP leaks, it currently shows "No leak, RTCPeerConnection not
>> available."
>>
>> *That is because:*
>>
>> (A) I have (again) changed my settings in about:config back to the
>> following
>>
>>
>>    1. I typed *about:config* in the address bar
>>    2. I found the setting *media.peerconnection.enabled*
>>    3. I set it to *false  (it was previously set at true, so as to allow
>>    WebRTC to work, but this exposed information which I would not have w=
anted
>>    to be exposed)*
>>
>> (B) And I also have changed my settings back to the following
>>
>> in the WebRTC 0.1.3 extension for Firefox I have set it to no longer
>> expose my real IP address as well as no longer allowing the following:
>>
>> a. navigator.getUserMedia
>> b. window.MediaStreamTrack
>> c. window.RTCPeerConnection
>> d. window.RTCSessionDescription
>>
>> To be clear, *these mitigations that I have employed above would be
>> unnecessary if WebRTC had not been designed in such a manner that it so
>> clearly exposes so much user information.*
>>
>> In the document you provided
>> <https://tools.ietf.org/html/draft-ietf-rtcweb-ip-handling-03>, titled
>> "WebRTC IP Address Handling Requirements - draft-ietf-rtcweb-ip-handling=
-03,"
>> it is stated that:
>>
>> *"1. If the client is behind a NAT, the client's private IP addresses,
>> typically [RFC1918 <https://tools.ietf.org/html/rfc1918>] addresses, can=
 be
>> learned. *
>>
>>
>>
>> * 2.  If the client tries to hide its physical location through a VPN,
>>        and the VPN and local OS support routing over multiple interfaces
>>        (i.e., a "split-tunnel" VPN), WebRTC will discover the public
>>        address for the VPN as well as the ISP public address that the
>>        VPN runs over.
>>
>>    3.  If the client is behind a proxy (a client-configured "classical
>>        application proxy", as defined in [RFC1919], Section 3 <https://t=
ools.ietf.org/html/rfc1919#section-3>), but
>>        direct access to the Internet is also supported, WebRTC's STUN
>>        [RFC5389 <https://tools.ietf.org/html/rfc5389>]checks will bypass=
 the proxy and reveal the public
>>        address of the client."*
>>
>> All of these are obvious vulnerabilities and serious problems (not only =
#2 as the document suggests).
>>
>>
> The document discusses #1 and #3 in detail, and outlines mechanisms that
> can be used to address these cases if needed.
>
> If, despite the findings in the document, you think that these are indeed
> serious problems, it would be good to outline exactly why.
>
>>
>> It is stated in the document that,
>>
>>
>>
>> *"we want to avoid solutions that disable WebRTC or make it harder to us=
e."*
>>
>> Yet, so far as my observations of WebRTC in the current version of Firef=
ox with Ubuntu 16.10 have shown,
>>
>> there appears to be no other choice other than to disable WebRTC because=
 its use completely compromises
>>
>> privacy and thus makes all other aspects of your system more difficult g=
enerally.
>>
>>
>> It is stated in the document you cited that the goals here are to "Provi=
de a privacy-friendly default behavior which strikes the
>>       right balance between privacy and media performance for most users
>>       and use cases. (and) For users who care more about one versus the =
other, provide a
>>       means to customize the experience."
>>
>> This has clearlly not been accomplished in the context of Ubuntu 16.10 6=
4-bit with Firefox 53.0.2 (64-bit).
>>
>> Hence my request for a new security review of WebRTC based on observed v=
ulnerabilities.
>>
>>
> To my knowledge, the document addresses these goals. If you believe
> Firefox is not conforming, please file a bug; if you think the document i=
s
> in error, please point out where and why.
>
>>
>>
>>
>> On Thu, May 25, 2017 at 1:04 PM, Justin Uberti <juberti@google.com>
>> wrote:
>>
>>> Colin,
>>>
>>> This is an important topic, and the group has reviewed this situation i=
n
>>> detail. The results have been written up in this document
>>> <https://tools.ietf.org/html/draft-ietf-rtcweb-ip-handling-03>, and to
>>> my knowledge most browsers that support WebRTC currently conform to its
>>> recommendations.
>>>
>>> However, I'm not totally clear on your specific technical concern. You
>>> linked to an (now outdated) stackexchange page, but could you provide m=
ore
>>> detail about the specific situation you are concerned about?
>>>
>>> Justin
>>>
>>> On Wed, May 24, 2017 at 5:35 PM, Colin Gallagher <
>>> colingallagher.rpcv@gmail.com> wrote:
>>>
>>>> To:  IETF RTCWeb WG
>>>>
>>>> From:  Colin Gallagher
>>>>            https://www.linkedin.com/in/colingallagher
>>>>            https://lifeboat.com/ex/bios.colin.gallagher
>>>>
>>>> Request for New Security Review of WebRTC based on observed
>>>> vulnerabilities in certain examples
>>>>
>>>> Dear IETF RTCWeb WG,
>>>>
>>>> This request relates to WebRTC.  The WebRTC is in draft at
>>>> https://w3c.github.io/webrtc-pc/ and is described also at
>>>> https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API#
>>>> WebRTC_interfaces
>>>>
>>>> There are certain examples where the use of WebRTC has presented
>>>> security vulnerabilities.  While not exhaustive, there are two I wante=
d to
>>>> mention that I had been exposed to recently.
>>>>
>>>> 1) The use of a video conferencing service which relies upon WebRTC
>>>> (that is a well established service)
>>>> 2) The examination of a browser based WebRTC version of a decentralize=
d
>>>> exchange (that is currently under development)
>>>>
>>>> In both cases the security vulnerability which is described below was
>>>> the result of the use of WebRTC.
>>>>
>>>> The problem under consideration is as follows:
>>>>
>>>> As per the first example (the video conferencing service), recently I
>>>> came upon a vendor who shall remain unnamed who uses WebRTC as the bac=
kbone
>>>> for their video conferencing service.  As a result, I found I was unab=
le to
>>>> access the service in question without disabling some security setting=
s
>>>> which I would have preferred to leave intact.
>>>>
>>>> My system is (based on what the aforementioned service detects)
>>>> Mozilla/5.0 (X11; Ubuntu; Linux x86_64; rv:53.0) Gecko/20100101
>>>> Firefox/53.0.  This is Ubuntu 16.10 64-bit with Firefox 53.0.2.
>>>>
>>>> However, quite some *long* while back I disabled WebRTC because of the
>>>> security vulnerability (described in part here on information security
>>>> stack exchange
>>>> <https://security.stackexchange.com/questions/82129/what-is-webrtc-and=
-how-can-i-protect-myself-against-leaking-information-that-doe>).
>>>> So as to mitigate this vulnerability, I changed settings in Firefox as
>>>> follows:
>>>>
>>>>
>>>>    1. I typed *about:config* in the address bar
>>>>    2. I found the setting *media.peerconnection.enabled*
>>>>    3. I set it to *false*
>>>>
>>>> Unfortunately, this setting does not allow Firefox to work with the
>>>> video conferencing service in question, which relies entirely on WebRT=
C.
>>>>
>>>> So I had to go back in and change the media.peerconnection.enabled
>>>> setting to *true* in Firefox in order to get it to work with the
>>>> service in question. While this enabled me to conference with teams th=
at
>>>> desire to use the video conferencing service that rely wholly upon Web=
RTC,
>>>> it concerns me because of the security vulnerability.
>>>>
>>>> One suggested mitigation was having NoScript, but using NoScript was
>>>> not helpful (you generally need to disable it for the site you are vie=
wing
>>>> that relies upon WebRTC, in order to obtain any functionality, and whe=
ther
>>>> NoScript is off or on, WebRTC still causes the vulnerability of leakin=
g
>>>> information as previously described).
>>>>
>>>> Furthermore, upon reload of the browser, even after setting
>>>> media.peerconnection.enabled to *true*, the video conferencing service
>>>> wouldn't work until I installed the WebRTC 0.1.3 extension for Firefox=
 and
>>>> set it to expose my real IP address as well as allowing the following:
>>>>
>>>> a. navigator.getUserMedia
>>>> b. window.MediaStreamTrack
>>>> c. window.RTCPeerConnection
>>>> d. window.RTCSessionDescription
>>>>
>>>> That seemed to me to be a direct consequence of the persistent and
>>>> ongoing security vulnerability in WebRTC.
>>>>
>>>> I contacted the video conferencing service provider and their solution
>>>> was simply to state that "If the peer to peer connection is of concern=
, you
>>>> can utilize our premium version which will route traffic through a
>>>> forwarding server with in our environment that handles the processing =
of
>>>> the video and sound of all users in the conference and send it to each=
 user
>>>> individually rather than using a peer to peer connection."  In other w=
ords,
>>>> they expect that people should have to pay in order to mitigate this
>>>> WebRTC security vulnerability
>>>> <https://security.stackexchange.com/questions/82129/what-is-webrtc-and=
-how-can-i-protect-myself-against-leaking-information-that-doe>,
>>>> because they are unwilling to design to protect all users from it.
>>>>
>>>> In a similar fashion, in the second example of the browser based WebRT=
C
>>>> version of a decentralized exchange, the user's financial information =
is
>>>> periodically exposed as a result of the information leakage which is
>>>> presently caused by WebRTC.  (Fortunately, this browser based WebRTC
>>>> decentralized exchange is under development and is not currently open =
to
>>>> actual usage.)
>>>>
>>>> Although undoubtedly this has been discussed here before, I am asking
>>>> that WebRTC security issues be reconsidered and that a new security re=
view
>>>> be done to ameliorate or eliminate this problem in terms of WebRTC lea=
king
>>>> this information.  It is my view that WebRTC should be further designe=
d and
>>>> developed with an eye to mitigating this serious problem, so that user=
s of
>>>> services which might rely partially or wholly upon WebRTC would not ha=
ve to
>>>> be concerned about such information leakage.
>>>>
>>>> Respectfully,
>>>> Colin Gallagher
>>>>
>>>>
>>>> _______________________________________________
>>>> rtcweb mailing list
>>>> rtcweb@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>>
>>>>
>>>
>>
>

--001a11462f7cda6be80550714e65
Content-Type: text/html; charset="UTF-8"
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div>It seems the consensus of the group here is essential=
ly a refusal to further review and potentially redesign WebRTC to remove th=
e serious security vulnerabilities that it presents to its users.<br><br></=
div>Thank you for your time.=C2=A0 I will not further respond to this threa=
d.<br><div><br><br></div></div><div class=3D"gmail_extra"><br><div class=3D=
"gmail_quote">On Fri, May 26, 2017 at 10:49 AM, Justin Uberti <span dir=3D"=
ltr">&lt;<a href=3D"mailto:juberti@google.com" target=3D"_blank">juberti@go=
ogle.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">Colin,<div><br></div><div>As Eric mentioned, the group has reviewe=
d this exact issue, and the aforementioned document represents the result o=
f the review.</div><div><br></div><div>I am still not quite clear on the ex=
act technical issue you would like to see addressed and why it is, as you c=
laim, a serious vulnerability.</div><div><br></div><div>Comments inline.</d=
iv><div class=3D"gmail_extra"><br><div class=3D"gmail_quote"><div><div clas=
s=3D"h5">On Fri, May 26, 2017 at 12:58 AM, Colin Gallagher <span dir=3D"ltr=
">&lt;<a href=3D"mailto:colingallagher.rpcv@gmail.com" target=3D"_blank">co=
lingallagher.rpcv@gmail.com</a><wbr>&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><div><div><div><div><div><div><div>Just=
in,<br><br></div>It&#39;s my understanding that various browsers have made =
an attempt to conform to the recommendations in <a href=3D"https://tools.ie=
tf.org/html/draft-ietf-rtcweb-ip-handling-03" target=3D"_blank">the documen=
t that you have provided</a>.=C2=A0 Unfortunately, at least in Firefox (the=
 most current version) so far as I can tell, there is a serious vulnerabili=
ty.=C2=A0 To be more specific, the problem was observed in my operating sys=
tem with my current browser, which is Ubuntu 16.10 64-bit with Firefox=20
53.0.2 (64-bit).=C2=A0 The browser does not address the WebRTC problems by =
default.<br><br></div>The stackexchange page that I cited originally is act=
ually not outdated in the sense that its method of measuring whether or not=
 the WebRTC vulnerability exists on a given system is still intact and work=
s.=C2=A0 The stackexchange page suggests two pages to determine whether or =
not you have leakage / WebRTC related vulnerability.=C2=A0 These pages are:=
<br><br></div>1) Browser leaks test for WebRTC <a href=3D"https://browserle=
aks.com/webrtc" target=3D"_blank">https://browserleaks.com/webrt<wbr>c</a><=
br></div>2) IP Leaks <a href=3D"https://ipleak.net/" target=3D"_blank">http=
s://ipleak.net/</a><br><br></div>Currently for Browser leaks test for WebRT=
C (due to that I have again changed settings in my system so that I will no=
t have to be exposed to this vulnerability) the settings show up on the pag=
e as:<br>RTCPeerConnection<span class=3D"m_3602413254325098192m_36203055837=
84542015gmail-bad">=C3=97</span>False<br>RTCDataChannel<span class=3D"m_360=
2413254325098192m_3620305583784542015gmail-bad">=C3=97</span>False<br>ORTC =
(Microsoft Edge)<span class=3D"m_3602413254325098192m_3620305583784542015gm=
ail-bad">=C3=97</span>False (for example....)<br><br></div>And under IP lea=
ks, it currently shows &quot;No leak, RTCPeerConnection not available.&quot=
;=C2=A0 <br><br></div><i>That is because:</i><br><br></div>(A) I have (agai=
n) changed my settings in about:config back to the following<br><br><ol><sp=
an><li>I typed <em>about:config</em> in the address bar</li><li>I found the=
 setting <em>media.peerconnection.enabled</em></li></span><li>I set it to <=
strong>false=C2=A0 (it was previously set at true, so as to allow WebRTC to=
 work, but this exposed information which I would not have wanted to be exp=
osed)</strong></li></ol><p>(B) And I also have changed my settings back to =
the following</p><p>in the WebRTC 0.1.3 extension for Firefox I have set it=
 to no longer expose my real IP=20
address as well as no longer allowing the following:</p><span><p>a. navigat=
or.getUserMedia<br>b. window.MediaStreamTrack<br>c. window.RTCPeerConnectio=
n<br>d. window.RTCSessionDescription</p></span><p>To be clear, <b>these mit=
igations that I have employed above would be unnecessary if WebRTC had not =
been designed in such a manner that it so clearly exposes so much user info=
rmation.</b></p><p>In <a href=3D"https://tools.ietf.org/html/draft-ietf-rtc=
web-ip-handling-03" target=3D"_blank">the document you provided</a>, titled=
 &quot;WebRTC IP Address Handling Requirements - draft-ietf-rtcweb-ip-handl=
ing-<wbr>03,&quot; it is stated that:</p><p><i>&quot;<span style=3D"font-fa=
mily:arial,helvetica,sans-serif">1.  If the client is behind a NAT, the cli=
ent&#39;s private IP addresses, typically [<a href=3D"https://tools.ietf.or=
g/html/rfc1918" title=3D"&quot;Address Allocation for Private Internets&quo=
t;" target=3D"_blank">RFC1918</a>] addresses, can be learned.
</span></i></p><pre class=3D"m_3602413254325098192m_3620305583784542015gmai=
l-newpage"><i><span style=3D"font-family:arial,helvetica,sans-serif"> 2.  I=
f the client tries to hide its physical location through a VPN,
       and the VPN and local OS support routing over multiple interfaces
       (i.e., a &quot;split-tunnel&quot; VPN), WebRTC will discover the pub=
lic
       address for the VPN as well as the ISP public address that the
       VPN runs over.

   3.  If the client is behind a proxy (a client-configured &quot;classical
       application proxy&quot;, as defined in <a href=3D"https://tools.ietf=
.org/html/rfc1919#section-3" target=3D"_blank">[RFC1919], Section=C2=A03</a=
>), but
       direct access to the Internet is also supported, WebRTC&#39;s STUN
       [<a href=3D"https://tools.ietf.org/html/rfc5389" title=3D"&quot;Sess=
ion Traversal Utilities for NAT (STUN)&quot;" target=3D"_blank">RFC5389</a>=
]checks will bypass the proxy and reveal the public
       address of the client.&quot;<br><br></span></i></pre><pre class=3D"m=
_3602413254325098192m_3620305583784542015gmail-newpage"><span style=3D"font=
-family:arial,helvetica,sans-serif">All of these are obvious vulnerabilitie=
s and serious problems (not only #2 as the document suggests).<br></span></=
pre></div></blockquote><div><br></div></div></div><div>The document discuss=
es #1 and #3 in detail, and outlines mechanisms that can be used to address=
 these cases if needed.</div><div><br></div><div>If, despite the findings i=
n the document, you think that these are indeed serious problems, it would =
be good to outline exactly why.</div><span class=3D""><blockquote class=3D"=
gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-=
left:1ex"><div dir=3D"ltr"><pre class=3D"m_3602413254325098192m_36203055837=
84542015gmail-newpage"><span style=3D"font-family:arial,helvetica,sans-seri=
f"><br></span></pre><pre class=3D"m_3602413254325098192m_362030558378454201=
5gmail-newpage"><span style=3D"font-family:arial,helvetica,sans-serif">It i=
s stated in the document that, <br><i><br>&quot;we want to avoid solutions =
that disable WebRTC or make it harder to use.&quot;<br><br></i></span></pre=
><pre class=3D"m_3602413254325098192m_3620305583784542015gmail-newpage"><sp=
an style=3D"font-family:arial,helvetica,sans-serif">Yet, so far as my obser=
vations of WebRTC in the current version of Firefox with Ubuntu 16.10 have =
shown,<br></span></pre><pre class=3D"m_3602413254325098192m_362030558378454=
2015gmail-newpage"><span style=3D"font-family:arial,helvetica,sans-serif">t=
here appears to be no other choice other than to disable WebRTC because its=
 use completely compromises <br></span></pre><pre class=3D"m_36024132543250=
98192m_3620305583784542015gmail-newpage"><span style=3D"font-family:arial,h=
elvetica,sans-serif">privacy and thus makes all other aspects of your syste=
m more difficult generally.</span></pre></div></blockquote><blockquote clas=
s=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;pad=
ding-left:1ex"><div dir=3D"ltr"><pre class=3D"m_3602413254325098192m_362030=
5583784542015gmail-newpage"><span style=3D"font-family:arial,helvetica,sans=
-serif"><br></span></pre><pre class=3D"m_3602413254325098192m_3620305583784=
542015gmail-newpage"><span style=3D"font-family:arial,helvetica,sans-serif"=
>It is stated in the document you cited that the goals here are to &quot;</=
span>Provide a privacy-friendly default behavior which strikes the
      right balance between privacy and media performance for most users
      and use cases. (and) For users who care more about one versus the oth=
er, provide a
      means to customize the experience.&quot;</pre><pre class=3D"m_3602413=
254325098192m_3620305583784542015gmail-newpage"><span style=3D"font-family:=
arial,helvetica,sans-serif">This has clearlly not been accomplished in the =
context of Ubuntu 16.10 64-bit with Firefox 53.0.2 (64-bit).<br><br></span>=
</pre><pre class=3D"m_3602413254325098192m_3620305583784542015gmail-newpage=
"><span style=3D"font-family:arial,helvetica,sans-serif">Hence my request f=
or a new security review of WebRTC based on observed vulnerabilities.<br></=
span></pre></div></blockquote><div><br></div></span><div>To my knowledge, t=
he document addresses these goals. If you believe Firefox is not conforming=
, please file a bug; if you think the document is in error, please point ou=
t where and why.</div><div><div class=3D"h5"><blockquote class=3D"gmail_quo=
te" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"=
><div dir=3D"ltr"><pre class=3D"m_3602413254325098192m_3620305583784542015g=
mail-newpage"><span style=3D"font-family:arial,helvetica,sans-serif"></span=
></pre><p><span style=3D"font-family:arial,helvetica,sans-serif"></span><br=
></p><strong></strong><div><div><div><div><div><br></div></div></div></div>=
</div></div><div class=3D"m_3602413254325098192HOEnZb"><div class=3D"m_3602=
413254325098192h5"><div class=3D"gmail_extra"><br><div class=3D"gmail_quote=
">On Thu, May 25, 2017 at 1:04 PM, Justin Uberti <span dir=3D"ltr">&lt;<a h=
ref=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">Colin,=
<div><br></div><div>This is an important topic, and the group has reviewed =
this situation in detail. The results have been written up in <a href=3D"ht=
tps://tools.ietf.org/html/draft-ietf-rtcweb-ip-handling-03" target=3D"_blan=
k">this document</a>, and to my knowledge most browsers that support WebRTC=
 currently conform to its recommendations.</div><div><br></div><div>However=
, I&#39;m not totally clear on your specific technical concern. You linked =
to an (now outdated) stackexchange page, but could you provide more detail =
about the specific situation you are concerned about?</div><div><br></div><=
div>Justin</div></div><div class=3D"gmail_extra"><br><div class=3D"gmail_qu=
ote"><div><div class=3D"m_3602413254325098192m_3620305583784542015h5">On We=
d, May 24, 2017 at 5:35 PM, Colin Gallagher <span dir=3D"ltr">&lt;<a href=
=3D"mailto:colingallagher.rpcv@gmail.com" target=3D"_blank">colingallagher.=
rpcv@gmail.com</a><wbr>&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"m_3602413254325098192m_3620305583784542015=
h5"><div dir=3D"ltr"><div><div><div><div><div><div><div><div>To:=C2=A0 IETF=
 RTCWeb WG<br><br></div>From:=C2=A0 Colin Gallagher<br></div>=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 <a href=3D"https://www.lin=
kedin.com/in/colingallagher" target=3D"_blank">https://www.linkedin.com/in/=
co<wbr>lingallagher</a><br>=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0 <a href=3D"https://lifeboat.com/ex/bios.colin.gallagher" targe=
t=3D"_blank">https://lifeboat.com/ex/bios.c<wbr>olin.gallagher</a><br><br><=
/div>Request for New Security Review of WebRTC based on observed vulnerabil=
ities in certain examples<br><br></div>Dear IETF RTCWeb WG,<br><br></div><d=
iv>This request relates to WebRTC.=C2=A0 The WebRTC is in draft at <a href=
=3D"https://w3c.github.io/webrtc-pc/" target=3D"_blank">https://w3c.github.=
io/webrtc-p<wbr>c/</a> and is=20
described also at=20
<a href=3D"https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API#WebR=
TC_interfaces" target=3D"_blank">https://developer.mozilla.org/<wbr>en-US/d=
ocs/Web/API/WebRTC_API#<wbr>WebRTC_interfaces</a></div><div><br></div>There=
 are certain examples where the use of WebRTC has presented security vulner=
abilities.=C2=A0 While not exhaustive, there are two I wanted to mention th=
at I had been exposed to recently.<br></div><div><br></div>1) The use of a =
video conferencing service which relies upon WebRTC (that is a well establi=
shed service)<br></div>2) The examination of a browser based WebRTC version=
 of a decentralized exchange (that is currently under development)<br><br><=
/div>In both cases the security vulnerability which is described below was =
the result of the use of WebRTC.<br><br>The problem under consideration is =
as follows:<br><br>As per the first example (the video conferencing service=
), recently
 I came upon a vendor who shall remain unnamed who uses WebRTC as the=20
backbone for their video conferencing service.=C2=A0 As a result, I found I=
=20
was unable to access the service in question without disabling some=20
security settings which I would have preferred to leave intact. <br><br>My =
system is (based on what the aforementioned service detects) Mozilla/5.0 (X=
11; Ubuntu; Linux x86_64; rv:53.0)=20
Gecko/20100101 Firefox/53.0.=C2=A0 This is Ubuntu 16.10 64-bit with Firefox=
=20
53.0.2.<br><br>However, quite some <i>long</i> while back I disabled WebRTC=
 because of the security vulnerability (described in part <a href=3D"https:=
//security.stackexchange.com/questions/82129/what-is-webrtc-and-how-can-i-p=
rotect-myself-against-leaking-information-that-doe" target=3D"_blank">here =
on information security stack exchange</a>).=C2=A0 So as to mitigate this v=
ulnerability, I changed settings in Firefox as follows:<br><br><ol><li>I ty=
ped <em>about:config</em> in the address bar</li><li>I found the setting <e=
m>media.peerconnection.enabled</em></li><li>I set it to <strong>false</stro=
ng></li></ol><p><strong></strong>Unfortunately,
 this setting does not allow Firefox to work with the video conferencing
 service in question, which relies entirely on WebRTC.<br></p><p>So I had t=
o go back in and change the media.peerconnection.enabled setting to <b>true=
</b>
 in Firefox in order to get it to work with the service in question.=20
While this enabled me to conference with teams that desire to use the=20
video conferencing service that rely wholly upon WebRTC, it concerns me=20
because of the security vulnerability.</p><p>One suggested mitigation was h=
aving NoScript, but using NoScript was not helpful (you generally need to d=
isable it for the site you are viewing that relies upon WebRTC, in order to=
 obtain any functionality, and whether NoScript is off or on, WebRTC still =
causes the vulnerability of leaking information as previously described).<b=
r></p><p>Furthermore, upon reload of the browser, even after setting media.=
peerconnection.enabled to <b>true</b>,
 the video conferencing service wouldn&#39;t work until I installed the=20
WebRTC 0.1.3 extension for Firefox and set it to expose my real IP=20
address as well as allowing the following:</p><p>a. navigator.getUserMedia<=
br>b. window.MediaStreamTrack<br>c. window.RTCPeerConnection<br>d. window.R=
TCSessionDescription<br></p><p>That seemed to me to be a direct consequence=
 of the persistent and ongoing security vulnerability in WebRTC.<br></p><p>=
I
 contacted the video conferencing service provider and their solution=20
was simply to state that &quot;If the peer to peer connection is of concern=
,=20
you can utilize our=20
premium version which will route traffic through a forwarding server=20
with in our environment that handles the processing of the video and=20
sound of all users in the conference and send it to each user=20
individually rather than using a peer to peer connection.&quot;=C2=A0 In ot=
her=20
words, they expect that people should have to pay in order to mitigate <a h=
ref=3D"https://security.stackexchange.com/questions/82129/what-is-webrtc-an=
d-how-can-i-protect-myself-against-leaking-information-that-doe" target=3D"=
_blank">this WebRTC security vulnerability</a>, because they are unwilling =
to design to protect all users from it.</p><p>In a similar fashion, in the =
second example of the browser based WebRTC version of a decentralized excha=
nge, the user&#39;s financial information is periodically exposed as a resu=
lt of the information leakage which is presently caused by WebRTC.=C2=A0 (F=
ortunately, this browser based WebRTC decentralized exchange is under devel=
opment and is not currently open to actual usage.)<br></p><p>Although
 undoubtedly this has been discussed here before, I am asking that WebRTC s=
ecurity issues be
 reconsidered and that a new security review be done to ameliorate or=20
eliminate this problem in terms of WebRTC leaking this information.=C2=A0 I=
t is my view that WebRTC should be further designed and developed with an e=
ye to mitigating this serious problem, so that users of services which migh=
t rely partially or wholly upon WebRTC would not have to be concerned about=
 such information leakage.<br></p><p>Respectfully,</p>Colin Gallagher<br><b=
r></div>
<br></div></div>______________________________<wbr>_________________<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/l<wbr>istinfo/rtcweb</a><br=
>
<br></blockquote></div><br></div>
</blockquote></div><br></div>
</div></div></blockquote></div></div></div><br></div></div>
</blockquote></div><br></div>

--001a11462f7cda6be80550714e65--


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From: Justin Uberti <juberti@google.com>
Date: Fri, 26 May 2017 14:07:58 -0700
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--001a113ff9a692cd96055073bc66
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Colin,

The group is absolutely willing to consider new evidence and arguments.

However it is not enough to say that there are "obvious security issues"
without any explanation of why these issues are serious and/or why the
existing document's treatment of them is flawed.

Help us understand your specific technical concerns.

Justin

On Fri, May 26, 2017 at 11:14 AM, Colin Gallagher <
colingallagher.rpcv@gmail.com> wrote:

> It seems the consensus of the group here is essentially a refusal to
> further review and potentially redesign WebRTC to remove the serious
> security vulnerabilities that it presents to its users.
>
> Thank you for your time.  I will not further respond to this thread.
>
>
>
> On Fri, May 26, 2017 at 10:49 AM, Justin Uberti <juberti@google.com>
> wrote:
>
>> Colin,
>>
>> As Eric mentioned, the group has reviewed this exact issue, and the
>> aforementioned document represents the result of the review.
>>
>> I am still not quite clear on the exact technical issue you would like t=
o
>> see addressed and why it is, as you claim, a serious vulnerability.
>>
>> Comments inline.
>>
>> On Fri, May 26, 2017 at 12:58 AM, Colin Gallagher <
>> colingallagher.rpcv@gmail.com> wrote:
>>
>>> Justin,
>>>
>>> It's my understanding that various browsers have made an attempt to
>>> conform to the recommendations in the document that you have provided
>>> <https://tools.ietf.org/html/draft-ietf-rtcweb-ip-handling-03>.
>>> Unfortunately, at least in Firefox (the most current version) so far as=
 I
>>> can tell, there is a serious vulnerability.  To be more specific, the
>>> problem was observed in my operating system with my current browser, wh=
ich
>>> is Ubuntu 16.10 64-bit with Firefox 53.0.2 (64-bit).  The browser does =
not
>>> address the WebRTC problems by default.
>>>
>>> The stackexchange page that I cited originally is actually not outdated
>>> in the sense that its method of measuring whether or not the WebRTC
>>> vulnerability exists on a given system is still intact and works.  The
>>> stackexchange page suggests two pages to determine whether or not you h=
ave
>>> leakage / WebRTC related vulnerability.  These pages are:
>>>
>>> 1) Browser leaks test for WebRTC https://browserleaks.com/webrtc
>>> 2) IP Leaks https://ipleak.net/
>>>
>>> Currently for Browser leaks test for WebRTC (due to that I have again
>>> changed settings in my system so that I will not have to be exposed to =
this
>>> vulnerability) the settings show up on the page as:
>>> RTCPeerConnection=C3=97False
>>> RTCDataChannel=C3=97False
>>> ORTC (Microsoft Edge)=C3=97False (for example....)
>>>
>>> And under IP leaks, it currently shows "No leak, RTCPeerConnection not
>>> available."
>>>
>>> *That is because:*
>>>
>>> (A) I have (again) changed my settings in about:config back to the
>>> following
>>>
>>>
>>>    1. I typed *about:config* in the address bar
>>>    2. I found the setting *media.peerconnection.enabled*
>>>    3. I set it to *false  (it was previously set at true, so as to
>>>    allow WebRTC to work, but this exposed information which I would not=
 have
>>>    wanted to be exposed)*
>>>
>>> (B) And I also have changed my settings back to the following
>>>
>>> in the WebRTC 0.1.3 extension for Firefox I have set it to no longer
>>> expose my real IP address as well as no longer allowing the following:
>>>
>>> a. navigator.getUserMedia
>>> b. window.MediaStreamTrack
>>> c. window.RTCPeerConnection
>>> d. window.RTCSessionDescription
>>>
>>> To be clear, *these mitigations that I have employed above would be
>>> unnecessary if WebRTC had not been designed in such a manner that it so
>>> clearly exposes so much user information.*
>>>
>>> In the document you provided
>>> <https://tools.ietf.org/html/draft-ietf-rtcweb-ip-handling-03>, titled
>>> "WebRTC IP Address Handling Requirements - draft-ietf-rtcweb-ip-handlin=
g-03,"
>>> it is stated that:
>>>
>>> *"1. If the client is behind a NAT, the client's private IP addresses,
>>> typically [RFC1918 <https://tools.ietf.org/html/rfc1918>] addresses, ca=
n be
>>> learned. *
>>>
>>>
>>>
>>> * 2.  If the client tries to hide its physical location through a VPN,
>>>        and the VPN and local OS support routing over multiple interface=
s
>>>        (i.e., a "split-tunnel" VPN), WebRTC will discover the public
>>>        address for the VPN as well as the ISP public address that the
>>>        VPN runs over.
>>>
>>>    3.  If the client is behind a proxy (a client-configured "classical
>>>        application proxy", as defined in [RFC1919], Section 3 <https://=
tools.ietf.org/html/rfc1919#section-3>), but
>>>        direct access to the Internet is also supported, WebRTC's STUN
>>>        [RFC5389 <https://tools.ietf.org/html/rfc5389>]checks will bypas=
s the proxy and reveal the public
>>>        address of the client."*
>>>
>>> All of these are obvious vulnerabilities and serious problems (not only=
 #2 as the document suggests).
>>>
>>>
>> The document discusses #1 and #3 in detail, and outlines mechanisms that
>> can be used to address these cases if needed.
>>
>> If, despite the findings in the document, you think that these are indee=
d
>> serious problems, it would be good to outline exactly why.
>>
>>>
>>> It is stated in the document that,
>>>
>>>
>>>
>>> *"we want to avoid solutions that disable WebRTC or make it harder to u=
se."*
>>>
>>> Yet, so far as my observations of WebRTC in the current version of Fire=
fox with Ubuntu 16.10 have shown,
>>>
>>> there appears to be no other choice other than to disable WebRTC becaus=
e its use completely compromises
>>>
>>> privacy and thus makes all other aspects of your system more difficult =
generally.
>>>
>>>
>>> It is stated in the document you cited that the goals here are to "Prov=
ide a privacy-friendly default behavior which strikes the
>>>       right balance between privacy and media performance for most user=
s
>>>       and use cases. (and) For users who care more about one versus the=
 other, provide a
>>>       means to customize the experience."
>>>
>>> This has clearlly not been accomplished in the context of Ubuntu 16.10 =
64-bit with Firefox 53.0.2 (64-bit).
>>>
>>> Hence my request for a new security review of WebRTC based on observed =
vulnerabilities.
>>>
>>>
>> To my knowledge, the document addresses these goals. If you believe
>> Firefox is not conforming, please file a bug; if you think the document =
is
>> in error, please point out where and why.
>>
>>>
>>>
>>>
>>> On Thu, May 25, 2017 at 1:04 PM, Justin Uberti <juberti@google.com>
>>> wrote:
>>>
>>>> Colin,
>>>>
>>>> This is an important topic, and the group has reviewed this situation
>>>> in detail. The results have been written up in this document
>>>> <https://tools.ietf.org/html/draft-ietf-rtcweb-ip-handling-03>, and to
>>>> my knowledge most browsers that support WebRTC currently conform to it=
s
>>>> recommendations.
>>>>
>>>> However, I'm not totally clear on your specific technical concern. You
>>>> linked to an (now outdated) stackexchange page, but could you provide =
more
>>>> detail about the specific situation you are concerned about?
>>>>
>>>> Justin
>>>>
>>>> On Wed, May 24, 2017 at 5:35 PM, Colin Gallagher <
>>>> colingallagher.rpcv@gmail.com> wrote:
>>>>
>>>>> To:  IETF RTCWeb WG
>>>>>
>>>>> From:  Colin Gallagher
>>>>>            https://www.linkedin.com/in/colingallagher
>>>>>            https://lifeboat.com/ex/bios.colin.gallagher
>>>>>
>>>>> Request for New Security Review of WebRTC based on observed
>>>>> vulnerabilities in certain examples
>>>>>
>>>>> Dear IETF RTCWeb WG,
>>>>>
>>>>> This request relates to WebRTC.  The WebRTC is in draft at
>>>>> https://w3c.github.io/webrtc-pc/ and is described also at
>>>>> https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API#
>>>>> WebRTC_interfaces
>>>>>
>>>>> There are certain examples where the use of WebRTC has presented
>>>>> security vulnerabilities.  While not exhaustive, there are two I want=
ed to
>>>>> mention that I had been exposed to recently.
>>>>>
>>>>> 1) The use of a video conferencing service which relies upon WebRTC
>>>>> (that is a well established service)
>>>>> 2) The examination of a browser based WebRTC version of a
>>>>> decentralized exchange (that is currently under development)
>>>>>
>>>>> In both cases the security vulnerability which is described below was
>>>>> the result of the use of WebRTC.
>>>>>
>>>>> The problem under consideration is as follows:
>>>>>
>>>>> As per the first example (the video conferencing service), recently I
>>>>> came upon a vendor who shall remain unnamed who uses WebRTC as the ba=
ckbone
>>>>> for their video conferencing service.  As a result, I found I was una=
ble to
>>>>> access the service in question without disabling some security settin=
gs
>>>>> which I would have preferred to leave intact.
>>>>>
>>>>> My system is (based on what the aforementioned service detects)
>>>>> Mozilla/5.0 (X11; Ubuntu; Linux x86_64; rv:53.0) Gecko/20100101
>>>>> Firefox/53.0.  This is Ubuntu 16.10 64-bit with Firefox 53.0.2.
>>>>>
>>>>> However, quite some *long* while back I disabled WebRTC because of
>>>>> the security vulnerability (described in part here on information
>>>>> security stack exchange
>>>>> <https://security.stackexchange.com/questions/82129/what-is-webrtc-an=
d-how-can-i-protect-myself-against-leaking-information-that-doe>).
>>>>> So as to mitigate this vulnerability, I changed settings in Firefox a=
s
>>>>> follows:
>>>>>
>>>>>
>>>>>    1. I typed *about:config* in the address bar
>>>>>    2. I found the setting *media.peerconnection.enabled*
>>>>>    3. I set it to *false*
>>>>>
>>>>> Unfortunately, this setting does not allow Firefox to work with the
>>>>> video conferencing service in question, which relies entirely on WebR=
TC.
>>>>>
>>>>> So I had to go back in and change the media.peerconnection.enabled
>>>>> setting to *true* in Firefox in order to get it to work with the
>>>>> service in question. While this enabled me to conference with teams t=
hat
>>>>> desire to use the video conferencing service that rely wholly upon We=
bRTC,
>>>>> it concerns me because of the security vulnerability.
>>>>>
>>>>> One suggested mitigation was having NoScript, but using NoScript was
>>>>> not helpful (you generally need to disable it for the site you are vi=
ewing
>>>>> that relies upon WebRTC, in order to obtain any functionality, and wh=
ether
>>>>> NoScript is off or on, WebRTC still causes the vulnerability of leaki=
ng
>>>>> information as previously described).
>>>>>
>>>>> Furthermore, upon reload of the browser, even after setting
>>>>> media.peerconnection.enabled to *true*, the video conferencing
>>>>> service wouldn't work until I installed the WebRTC 0.1.3 extension fo=
r
>>>>> Firefox and set it to expose my real IP address as well as allowing t=
he
>>>>> following:
>>>>>
>>>>> a. navigator.getUserMedia
>>>>> b. window.MediaStreamTrack
>>>>> c. window.RTCPeerConnection
>>>>> d. window.RTCSessionDescription
>>>>>
>>>>> That seemed to me to be a direct consequence of the persistent and
>>>>> ongoing security vulnerability in WebRTC.
>>>>>
>>>>> I contacted the video conferencing service provider and their solutio=
n
>>>>> was simply to state that "If the peer to peer connection is of concer=
n, you
>>>>> can utilize our premium version which will route traffic through a
>>>>> forwarding server with in our environment that handles the processing=
 of
>>>>> the video and sound of all users in the conference and send it to eac=
h user
>>>>> individually rather than using a peer to peer connection."  In other =
words,
>>>>> they expect that people should have to pay in order to mitigate this
>>>>> WebRTC security vulnerability
>>>>> <https://security.stackexchange.com/questions/82129/what-is-webrtc-an=
d-how-can-i-protect-myself-against-leaking-information-that-doe>,
>>>>> because they are unwilling to design to protect all users from it.
>>>>>
>>>>> In a similar fashion, in the second example of the browser based
>>>>> WebRTC version of a decentralized exchange, the user's financial
>>>>> information is periodically exposed as a result of the information le=
akage
>>>>> which is presently caused by WebRTC.  (Fortunately, this browser base=
d
>>>>> WebRTC decentralized exchange is under development and is not current=
ly
>>>>> open to actual usage.)
>>>>>
>>>>> Although undoubtedly this has been discussed here before, I am asking
>>>>> that WebRTC security issues be reconsidered and that a new security r=
eview
>>>>> be done to ameliorate or eliminate this problem in terms of WebRTC le=
aking
>>>>> this information.  It is my view that WebRTC should be further design=
ed and
>>>>> developed with an eye to mitigating this serious problem, so that use=
rs of
>>>>> services which might rely partially or wholly upon WebRTC would not h=
ave to
>>>>> be concerned about such information leakage.
>>>>>
>>>>> Respectfully,
>>>>> Colin Gallagher
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> rtcweb mailing list
>>>>> rtcweb@ietf.org
>>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>>>
>>>>>
>>>>
>>>
>>
>

--001a113ff9a692cd96055073bc66
Content-Type: text/html; charset="UTF-8"
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Colin,<div><br></div><div>The group is absolutely willing =
to consider new evidence and arguments.=C2=A0</div><div><br></div><div>Howe=
ver it is not enough to say that there are &quot;obvious security issues&qu=
ot; without any explanation of why these issues are serious and/or why the =
existing document&#39;s treatment of them is flawed.<br></div><div><br></di=
v><div>Help us understand your specific technical concerns.</div><div><br><=
/div><div>Justin</div></div><div class=3D"gmail_extra"><br><div class=3D"gm=
ail_quote">On Fri, May 26, 2017 at 11:14 AM, Colin Gallagher <span dir=3D"l=
tr">&lt;<a href=3D"mailto:colingallagher.rpcv@gmail.com" target=3D"_blank">=
colingallagher.rpcv@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>It seems the consensus of the group here i=
s essentially a refusal to further review and potentially redesign WebRTC t=
o remove the serious security vulnerabilities that it presents to its users=
.<br><br></div>Thank you for your time.=C2=A0 I will not further respond to=
 this thread.<br><div><br><br></div></div><div class=3D"HOEnZb"><div class=
=3D"h5"><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Fri, M=
ay 26, 2017 at 10:49 AM, Justin Uberti <span dir=3D"ltr">&lt;<a href=3D"mai=
lto: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;bor=
der-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr">Colin,<div><br><=
/div><div>As Eric mentioned, the group has reviewed this exact issue, and t=
he aforementioned document represents the result of the review.</div><div><=
br></div><div>I am still not quite clear on the exact technical issue you w=
ould like to see addressed and why it is, as you claim, a serious vulnerabi=
lity.</div><div><br></div><div>Comments inline.</div><div class=3D"gmail_ex=
tra"><br><div class=3D"gmail_quote"><div><div class=3D"m_-26590354475518785=
08h5">On Fri, May 26, 2017 at 12:58 AM, Colin Gallagher <span dir=3D"ltr">&=
lt;<a href=3D"mailto:colingallagher.rpcv@gmail.com" target=3D"_blank">colin=
gallagher.rpcv@gmail.com</a><wbr>&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><div><div><div><div><div><div><div>Justin,=
<br><br></div>It&#39;s my understanding that various browsers have made an =
attempt to conform to the recommendations in <a href=3D"https://tools.ietf.=
org/html/draft-ietf-rtcweb-ip-handling-03" target=3D"_blank">the document t=
hat you have provided</a>.=C2=A0 Unfortunately, at least in Firefox (the mo=
st current version) so far as I can tell, there is a serious vulnerability.=
=C2=A0 To be more specific, the problem was observed in my operating system=
 with my current browser, which is Ubuntu 16.10 64-bit with Firefox=20
53.0.2 (64-bit).=C2=A0 The browser does not address the WebRTC problems by =
default.<br><br></div>The stackexchange page that I cited originally is act=
ually not outdated in the sense that its method of measuring whether or not=
 the WebRTC vulnerability exists on a given system is still intact and work=
s.=C2=A0 The stackexchange page suggests two pages to determine whether or =
not you have leakage / WebRTC related vulnerability.=C2=A0 These pages are:=
<br><br></div>1) Browser leaks test for WebRTC <a href=3D"https://browserle=
aks.com/webrtc" target=3D"_blank">https://browserleaks.com/webrt<wbr>c</a><=
br></div>2) IP Leaks <a href=3D"https://ipleak.net/" target=3D"_blank">http=
s://ipleak.net/</a><br><br></div>Currently for Browser leaks test for WebRT=
C (due to that I have again changed settings in my system so that I will no=
t have to be exposed to this vulnerability) the settings show up on the pag=
e as:<br>RTCPeerConnection<span class=3D"m_-2659035447551878508m_3602413254=
325098192m_3620305583784542015gmail-bad">=C3=97</span>False<br>RTCDataChann=
el<span class=3D"m_-2659035447551878508m_3602413254325098192m_3620305583784=
542015gmail-bad">=C3=97</span>False<br>ORTC (Microsoft Edge)<span class=3D"=
m_-2659035447551878508m_3602413254325098192m_3620305583784542015gmail-bad">=
=C3=97</span>False (for example....)<br><br></div>And under IP leaks, it cu=
rrently shows &quot;No leak, RTCPeerConnection not available.&quot;=C2=A0 <=
br><br></div><i>That is because:</i><br><br></div>(A) I have (again) change=
d my settings in about:config back to the following<br><br><ol><span><li>I =
typed <em>about:config</em> in the address bar</li><li>I found the setting =
<em>media.peerconnection.enabled</em></li></span><li>I set it to <strong>fa=
lse=C2=A0 (it was previously set at true, so as to allow WebRTC to work, bu=
t this exposed information which I would not have wanted to be exposed)</st=
rong></li></ol><p>(B) And I also have changed my settings back to the follo=
wing</p><p>in the WebRTC 0.1.3 extension for Firefox I have set it to no lo=
nger expose my real IP=20
address as well as no longer allowing the following:</p><span><p>a. navigat=
or.getUserMedia<br>b. window.MediaStreamTrack<br>c. window.RTCPeerConnectio=
n<br>d. window.RTCSessionDescription</p></span><p>To be clear, <b>these mit=
igations that I have employed above would be unnecessary if WebRTC had not =
been designed in such a manner that it so clearly exposes so much user info=
rmation.</b></p><p>In <a href=3D"https://tools.ietf.org/html/draft-ietf-rtc=
web-ip-handling-03" target=3D"_blank">the document you provided</a>, titled=
 &quot;WebRTC IP Address Handling Requirements - draft-ietf-rtcweb-ip-handl=
ing-<wbr>03,&quot; it is stated that:</p><p><i>&quot;<span style=3D"font-fa=
mily:arial,helvetica,sans-serif">1.  If the client is behind a NAT, the cli=
ent&#39;s private IP addresses, typically [<a href=3D"https://tools.ietf.or=
g/html/rfc1918" title=3D"&quot;Address Allocation for Private Internets&quo=
t;" target=3D"_blank">RFC1918</a>] addresses, can be learned.
</span></i></p><pre class=3D"m_-2659035447551878508m_3602413254325098192m_3=
620305583784542015gmail-newpage"><i><span style=3D"font-family:arial,helvet=
ica,sans-serif"> 2.  If the client tries to hide its physical location thro=
ugh a VPN,
       and the VPN and local OS support routing over multiple interfaces
       (i.e., a &quot;split-tunnel&quot; VPN), WebRTC will discover the pub=
lic
       address for the VPN as well as the ISP public address that the
       VPN runs over.

   3.  If the client is behind a proxy (a client-configured &quot;classical
       application proxy&quot;, as defined in <a href=3D"https://tools.ietf=
.org/html/rfc1919#section-3" target=3D"_blank">[RFC1919], Section=C2=A03</a=
>), but
       direct access to the Internet is also supported, WebRTC&#39;s STUN
       [<a href=3D"https://tools.ietf.org/html/rfc5389" title=3D"&quot;Sess=
ion Traversal Utilities for NAT (STUN)&quot;" target=3D"_blank">RFC5389</a>=
]checks will bypass the proxy and reveal the public
       address of the client.&quot;<br><br></span></i></pre><pre class=3D"m=
_-2659035447551878508m_3602413254325098192m_3620305583784542015gmail-newpag=
e"><span style=3D"font-family:arial,helvetica,sans-serif">All of these are =
obvious vulnerabilities and serious problems (not only #2 as the document s=
uggests).<br></span></pre></div></blockquote><div><br></div></div></div><di=
v>The document discusses #1 and #3 in detail, and outlines mechanisms that =
can be used to address these cases if needed.</div><div><br></div><div>If, =
despite the findings in the document, you think that these are indeed serio=
us problems, it would be good to outline exactly why.</div><span><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"><pre class=3D"m_-2659035447551878508m=
_3602413254325098192m_3620305583784542015gmail-newpage"><span style=3D"font=
-family:arial,helvetica,sans-serif"><br></span></pre><pre class=3D"m_-26590=
35447551878508m_3602413254325098192m_3620305583784542015gmail-newpage"><spa=
n style=3D"font-family:arial,helvetica,sans-serif">It is stated in the docu=
ment that, <br><i><br>&quot;we want to avoid solutions that disable WebRTC =
or make it harder to use.&quot;<br><br></i></span></pre><pre class=3D"m_-26=
59035447551878508m_3602413254325098192m_3620305583784542015gmail-newpage"><=
span style=3D"font-family:arial,helvetica,sans-serif">Yet, so far as my obs=
ervations of WebRTC in the current version of Firefox with Ubuntu 16.10 hav=
e shown,<br></span></pre><pre class=3D"m_-2659035447551878508m_360241325432=
5098192m_3620305583784542015gmail-newpage"><span style=3D"font-family:arial=
,helvetica,sans-serif">there appears to be no other choice other than to di=
sable WebRTC because its use completely compromises <br></span></pre><pre c=
lass=3D"m_-2659035447551878508m_3602413254325098192m_3620305583784542015gma=
il-newpage"><span style=3D"font-family:arial,helvetica,sans-serif">privacy =
and thus makes all other aspects of your system more difficult generally.</=
span></pre></div></blockquote><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"><pre class=3D"m_-2659035447551878508m_3602413254325098192m_3620305583784=
542015gmail-newpage"><span style=3D"font-family:arial,helvetica,sans-serif"=
><br></span></pre><pre class=3D"m_-2659035447551878508m_3602413254325098192=
m_3620305583784542015gmail-newpage"><span style=3D"font-family:arial,helvet=
ica,sans-serif">It is stated in the document you cited that the goals here =
are to &quot;</span>Provide a privacy-friendly default behavior which strik=
es the
      right balance between privacy and media performance for most users
      and use cases. (and) For users who care more about one versus the oth=
er, provide a
      means to customize the experience.&quot;</pre><pre class=3D"m_-265903=
5447551878508m_3602413254325098192m_3620305583784542015gmail-newpage"><span=
 style=3D"font-family:arial,helvetica,sans-serif">This has clearlly not bee=
n accomplished in the context of Ubuntu 16.10 64-bit with Firefox 53.0.2 (6=
4-bit).<br><br></span></pre><pre class=3D"m_-2659035447551878508m_360241325=
4325098192m_3620305583784542015gmail-newpage"><span style=3D"font-family:ar=
ial,helvetica,sans-serif">Hence my request for a new security review of Web=
RTC based on observed vulnerabilities.<br></span></pre></div></blockquote><=
div><br></div></span><div>To my knowledge, the document addresses these goa=
ls. If you believe Firefox is not conforming, please file a bug; if you thi=
nk the document is in error, please point out where and why.</div><div><div=
 class=3D"m_-2659035447551878508h5"><blockquote class=3D"gmail_quote" style=
=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=
=3D"ltr"><pre class=3D"m_-2659035447551878508m_3602413254325098192m_3620305=
583784542015gmail-newpage"><span style=3D"font-family:arial,helvetica,sans-=
serif"></span></pre><p><span style=3D"font-family:arial,helvetica,sans-seri=
f"></span><br></p><strong></strong><div><div><div><div><div><br></div></div=
></div></div></div></div><div class=3D"m_-2659035447551878508m_360241325432=
5098192HOEnZb"><div class=3D"m_-2659035447551878508m_3602413254325098192h5"=
><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Thu, May 25, =
2017 at 1:04 PM, Justin Uberti <span dir=3D"ltr">&lt;<a href=3D"mailto:jube=
rti@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">Colin,<div><br></div><di=
v>This is an important topic, and the group has reviewed this situation in =
detail. The results have been written up in <a href=3D"https://tools.ietf.o=
rg/html/draft-ietf-rtcweb-ip-handling-03" target=3D"_blank">this document</=
a>, and to my knowledge most browsers that support WebRTC currently conform=
 to its recommendations.</div><div><br></div><div>However, I&#39;m not tota=
lly clear on your specific technical concern. You linked to an (now outdate=
d) stackexchange page, but could you provide more detail about the specific=
 situation you are concerned about?</div><div><br></div><div>Justin</div></=
div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote"><div><div cla=
ss=3D"m_-2659035447551878508m_3602413254325098192m_3620305583784542015h5">O=
n Wed, May 24, 2017 at 5:35 PM, Colin Gallagher <span dir=3D"ltr">&lt;<a hr=
ef=3D"mailto:colingallagher.rpcv@gmail.com" target=3D"_blank">colingallaghe=
r.rpcv@gmail.com</a><wbr>&gt;</span> wrote:<br></div></div><blockquote clas=
s=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;pad=
ding-left:1ex"><div><div class=3D"m_-2659035447551878508m_36024132543250981=
92m_3620305583784542015h5"><div dir=3D"ltr"><div><div><div><div><div><div><=
div><div>To:=C2=A0 IETF RTCWeb WG<br><br></div>From:=C2=A0 Colin Gallagher<=
br></div>=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 <a hr=
ef=3D"https://www.linkedin.com/in/colingallagher" target=3D"_blank">https:/=
/www.linkedin.com/in/co<wbr>lingallagher</a><br>=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 <a href=3D"https://lifeboat.com/ex/bios.c=
olin.gallagher" target=3D"_blank">https://lifeboat.com/ex/bios.c<wbr>olin.g=
allagher</a><br><br></div>Request for New Security Review of WebRTC based o=
n observed vulnerabilities in certain examples<br><br></div>Dear IETF RTCWe=
b WG,<br><br></div><div>This request relates to WebRTC.=C2=A0 The WebRTC is=
 in draft at <a href=3D"https://w3c.github.io/webrtc-pc/" target=3D"_blank"=
>https://w3c.github.io/webrtc-p<wbr>c/</a> and is=20
described also at=20
<a href=3D"https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API#WebR=
TC_interfaces" target=3D"_blank">https://developer.mozilla.org/<wbr>en-US/d=
ocs/Web/API/WebRTC_API#<wbr>WebRTC_interfaces</a></div><div><br></div>There=
 are certain examples where the use of WebRTC has presented security vulner=
abilities.=C2=A0 While not exhaustive, there are two I wanted to mention th=
at I had been exposed to recently.<br></div><div><br></div>1) The use of a =
video conferencing service which relies upon WebRTC (that is a well establi=
shed service)<br></div>2) The examination of a browser based WebRTC version=
 of a decentralized exchange (that is currently under development)<br><br><=
/div>In both cases the security vulnerability which is described below was =
the result of the use of WebRTC.<br><br>The problem under consideration is =
as follows:<br><br>As per the first example (the video conferencing service=
), recently
 I came upon a vendor who shall remain unnamed who uses WebRTC as the=20
backbone for their video conferencing service.=C2=A0 As a result, I found I=
=20
was unable to access the service in question without disabling some=20
security settings which I would have preferred to leave intact. <br><br>My =
system is (based on what the aforementioned service detects) Mozilla/5.0 (X=
11; Ubuntu; Linux x86_64; rv:53.0)=20
Gecko/20100101 Firefox/53.0.=C2=A0 This is Ubuntu 16.10 64-bit with Firefox=
=20
53.0.2.<br><br>However, quite some <i>long</i> while back I disabled WebRTC=
 because of the security vulnerability (described in part <a href=3D"https:=
//security.stackexchange.com/questions/82129/what-is-webrtc-and-how-can-i-p=
rotect-myself-against-leaking-information-that-doe" target=3D"_blank">here =
on information security stack exchange</a>).=C2=A0 So as to mitigate this v=
ulnerability, I changed settings in Firefox as follows:<br><br><ol><li>I ty=
ped <em>about:config</em> in the address bar</li><li>I found the setting <e=
m>media.peerconnection.enabled</em></li><li>I set it to <strong>false</stro=
ng></li></ol><p><strong></strong>Unfortunately,
 this setting does not allow Firefox to work with the video conferencing
 service in question, which relies entirely on WebRTC.<br></p><p>So I had t=
o go back in and change the media.peerconnection.enabled setting to <b>true=
</b>
 in Firefox in order to get it to work with the service in question.=20
While this enabled me to conference with teams that desire to use the=20
video conferencing service that rely wholly upon WebRTC, it concerns me=20
because of the security vulnerability.</p><p>One suggested mitigation was h=
aving NoScript, but using NoScript was not helpful (you generally need to d=
isable it for the site you are viewing that relies upon WebRTC, in order to=
 obtain any functionality, and whether NoScript is off or on, WebRTC still =
causes the vulnerability of leaking information as previously described).<b=
r></p><p>Furthermore, upon reload of the browser, even after setting media.=
peerconnection.enabled to <b>true</b>,
 the video conferencing service wouldn&#39;t work until I installed the=20
WebRTC 0.1.3 extension for Firefox and set it to expose my real IP=20
address as well as allowing the following:</p><p>a. navigator.getUserMedia<=
br>b. window.MediaStreamTrack<br>c. window.RTCPeerConnection<br>d. window.R=
TCSessionDescription<br></p><p>That seemed to me to be a direct consequence=
 of the persistent and ongoing security vulnerability in WebRTC.<br></p><p>=
I
 contacted the video conferencing service provider and their solution=20
was simply to state that &quot;If the peer to peer connection is of concern=
,=20
you can utilize our=20
premium version which will route traffic through a forwarding server=20
with in our environment that handles the processing of the video and=20
sound of all users in the conference and send it to each user=20
individually rather than using a peer to peer connection.&quot;=C2=A0 In ot=
her=20
words, they expect that people should have to pay in order to mitigate <a h=
ref=3D"https://security.stackexchange.com/questions/82129/what-is-webrtc-an=
d-how-can-i-protect-myself-against-leaking-information-that-doe" target=3D"=
_blank">this WebRTC security vulnerability</a>, because they are unwilling =
to design to protect all users from it.</p><p>In a similar fashion, in the =
second example of the browser based WebRTC version of a decentralized excha=
nge, the user&#39;s financial information is periodically exposed as a resu=
lt of the information leakage which is presently caused by WebRTC.=C2=A0 (F=
ortunately, this browser based WebRTC decentralized exchange is under devel=
opment and is not currently open to actual usage.)<br></p><p>Although
 undoubtedly this has been discussed here before, I am asking that WebRTC s=
ecurity issues be
 reconsidered and that a new security review be done to ameliorate or=20
eliminate this problem in terms of WebRTC leaking this information.=C2=A0 I=
t is my view that WebRTC should be further designed and developed with an e=
ye to mitigating this serious problem, so that users of services which migh=
t rely partially or wholly upon WebRTC would not have to be concerned about=
 such information leakage.<br></p><p>Respectfully,</p>Colin Gallagher<br><b=
r></div>
<br></div></div>______________________________<wbr>_________________<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/l<wbr>istinfo/rtcweb</a><br=
>
<br></blockquote></div><br></div>
</blockquote></div><br></div>
</div></div></blockquote></div></div></div><br></div></div>
</blockquote></div><br></div>
</div></div></blockquote></div><br></div>

--001a113ff9a692cd96055073bc66--


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To: rtcweb@ietf.org
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Subject: Re: [rtcweb] Request for New Security Review of WebRTC based on observed vulnerabilities
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Colin -=20

Are you claiming that the recommendations of the document Justin refers to a=
re inadequate, or just that some browsers have not fully or correctly implem=
ented it?=20

If the former, what would you propose to change?

> On May 26, 2017, at 10:49 AM, Justin Uberti <juberti@google.com> wrote:
>=20
> Colin,
>=20
> As Eric mentioned, the group has reviewed this exact issue, and the aforem=
entioned document represents the result of the review.
>=20
> I am still not quite clear on the exact technical issue you would like to s=
ee addressed and why it is, as you claim, a serious vulnerability.
>=20
> Comments inline.
>=20
>> On Fri, May 26, 2017 at 12:58 AM, Colin Gallagher <colingallagher.rpcv@gm=
ail.com> wrote:
>> Justin,
>>=20
>> It's my understanding that various browsers have made an attempt to confo=
rm to the recommendations in the document that you have provided.  Unfortuna=
tely, at least in Firefox (the most current version) so far as I can tell, t=
here is a serious vulnerability.  To be more specific, the problem was obser=
ved in my operating system with my current browser, which is Ubuntu 16.10 64=
-bit with Firefox 53.0.2 (64-bit).  The browser does not address the WebRTC p=
roblems by default.
>>=20
>> The stackexchange page that I cited originally is actually not outdated i=
n the sense that its method of measuring whether or not the WebRTC vulnerabi=
lity exists on a given system is still intact and works.  The stackexchange p=
age suggests two pages to determine whether or not you have leakage / WebRTC=
 related vulnerability.  These pages are:
>>=20
>> 1) Browser leaks test for WebRTC https://browserleaks.com/webrtc
>> 2) IP Leaks https://ipleak.net/
>>=20
>> Currently for Browser leaks test for WebRTC (due to that I have again cha=
nged settings in my system so that I will not have to be exposed to this vul=
nerability) the settings show up on the page as:
>> RTCPeerConnection=C3=97False
>> RTCDataChannel=C3=97False
>> ORTC (Microsoft Edge)=C3=97False (for example....)
>>=20
>> And under IP leaks, it currently shows "No leak, RTCPeerConnection not av=
ailable." =20
>>=20
>> That is because:
>>=20
>> (A) I have (again) changed my settings in about:config back to the follow=
ing
>>=20
>> I typed about:config in the address bar
>> I found the setting media.peerconnection.enabled
>> I set it to false  (it was previously set at true, so as to allow WebRTC t=
o work, but this exposed information which I would not have wanted to be exp=
osed)
>> (B) And I also have changed my settings back to the following
>>=20
>> in the WebRTC 0.1.3 extension for Firefox I have set it to no longer expo=
se my real IP address as well as no longer allowing the following:
>>=20
>> a. navigator.getUserMedia
>> b. window.MediaStreamTrack
>> c. window.RTCPeerConnection
>> d. window.RTCSessionDescription
>>=20
>> To be clear, these mitigations that I have employed above would be unnece=
ssary if WebRTC had not been designed in such a manner that it so clearly ex=
poses so much user information.
>>=20
>> In the document you provided, titled "WebRTC IP Address Handling Requirem=
ents - draft-ietf-rtcweb-ip-handling-03," it is stated that:
>>=20
>> "1. If the client is behind a NAT, the client's private IP addresses, typ=
ically [RFC1918] addresses, can be learned.
>>=20
>>  2.  If the client tries to hide its physical location through a VPN,
>>        and the VPN and local OS support routing over multiple interfaces
>>        (i.e., a "split-tunnel" VPN), WebRTC will discover the public
>>        address for the VPN as well as the ISP public address that the
>>        VPN runs over.
>>=20
>>    3.  If the client is behind a proxy (a client-configured "classical
>>        application proxy", as defined in [RFC1919], Section 3), but
>>        direct access to the Internet is also supported, WebRTC's STUN
>>        [RFC5389]checks will bypass the proxy and reveal the public
>>        address of the client."
>>=20
>> All of these are obvious vulnerabilities and serious problems (not only #=
2 as the document suggests).
>=20
> The document discusses #1 and #3 in detail, and outlines mechanisms that c=
an be used to address these cases if needed.
>=20
> If, despite the findings in the document, you think that these are indeed s=
erious problems, it would be good to outline exactly why.
>>=20
>> It is stated in the document that,=20
>>=20
>> "we want to avoid solutions that disable WebRTC or make it harder to use.=
"
>>=20
>> Yet, so far as my observations of WebRTC in the current version of Firefo=
x with Ubuntu 16.10 have shown,
>> there appears to be no other choice other than to disable WebRTC because i=
ts use completely compromises=20
>> privacy and thus makes all other aspects of your system more difficult ge=
nerally.
>>=20
>> It is stated in the document you cited that the goals here are to "Provid=
e a privacy-friendly default behavior which strikes the
>>       right balance between privacy and media performance for most users
>>       and use cases. (and) For users who care more about one versus the o=
ther, provide a
>>       means to customize the experience."
>> This has clearlly not been accomplished in the context of Ubuntu 16.10 64=
-bit with Firefox 53.0.2 (64-bit).
>>=20
>> Hence my request for a new security review of WebRTC based on observed vu=
lnerabilities.
>=20
> To my knowledge, the document addresses these goals. If you believe Firefo=
x is not conforming, please file a bug; if you think the document is in erro=
r, please point out where and why.
>>=20
>>=20
>>=20
>>> On Thu, May 25, 2017 at 1:04 PM, Justin Uberti <juberti@google.com> wrot=
e:
>>> Colin,
>>>=20
>>> This is an important topic, and the group has reviewed this situation in=
 detail. The results have been written up in this document, and to my knowle=
dge most browsers that support WebRTC currently conform to its recommendatio=
ns.
>>>=20
>>> However, I'm not totally clear on your specific technical concern. You l=
inked to an (now outdated) stackexchange page, but could you provide more de=
tail about the specific situation you are concerned about?
>>>=20
>>> Justin
>>>=20
>>>> On Wed, May 24, 2017 at 5:35 PM, Colin Gallagher <colingallagher.rpcv@g=
mail.com> wrote:
>>>> To:  IETF RTCWeb WG
>>>>=20
>>>> From:  Colin Gallagher
>>>>            https://www.linkedin.com/in/colingallagher
>>>>            https://lifeboat.com/ex/bios.colin.gallagher
>>>>=20
>>>> Request for New Security Review of WebRTC based on observed vulnerabili=
ties in certain examples
>>>>=20
>>>> Dear IETF RTCWeb WG,
>>>>=20
>>>> This request relates to WebRTC.  The WebRTC is in draft at https://w3c.=
github.io/webrtc-pc/ and is described also at https://developer.mozilla.org/=
en-US/docs/Web/API/WebRTC_API#WebRTC_interfaces
>>>>=20
>>>> There are certain examples where the use of WebRTC has presented securi=
ty vulnerabilities.  While not exhaustive, there are two I wanted to mention=
 that I had been exposed to recently.
>>>>=20
>>>> 1) The use of a video conferencing service which relies upon WebRTC (th=
at is a well established service)
>>>> 2) The examination of a browser based WebRTC version of a decentralized=
 exchange (that is currently under development)
>>>>=20
>>>> In both cases the security vulnerability which is described below was t=
he result of the use of WebRTC.
>>>>=20
>>>> The problem under consideration is as follows:
>>>>=20
>>>> As per the first example (the video conferencing service), recently I c=
ame upon a vendor who shall remain unnamed who uses WebRTC as the backbone f=
or their video conferencing service.  As a result, I found I was unable to a=
ccess the service in question without disabling some security settings which=
 I would have preferred to leave intact.=20
>>>>=20
>>>> My system is (based on what the aforementioned service detects) Mozilla=
/5.0 (X11; Ubuntu; Linux x86_64; rv:53.0) Gecko/20100101 Firefox/53.0.  This=
 is Ubuntu 16.10 64-bit with Firefox 53.0.2.
>>>>=20
>>>> However, quite some long while back I disabled WebRTC because of the se=
curity vulnerability (described in part here on information security stack e=
xchange).  So as to mitigate this vulnerability, I changed settings in Firef=
ox as follows:
>>>>=20
>>>> I typed about:config in the address bar
>>>> I found the setting media.peerconnection.enabled
>>>> I set it to false
>>>> Unfortunately, this setting does not allow Firefox to work with the vid=
eo conferencing service in question, which relies entirely on WebRTC.
>>>>=20
>>>> So I had to go back in and change the media.peerconnection.enabled sett=
ing to true in Firefox in order to get it to work with the service in questi=
on. While this enabled me to conference with teams that desire to use the vi=
deo conferencing service that rely wholly upon WebRTC, it concerns me becaus=
e of the security vulnerability.
>>>>=20
>>>> One suggested mitigation was having NoScript, but using NoScript was no=
t helpful (you generally need to disable it for the site you are viewing tha=
t relies upon WebRTC, in order to obtain any functionality, and whether NoSc=
ript is off or on, WebRTC still causes the vulnerability of leaking informat=
ion as previously described).
>>>>=20
>>>> Furthermore, upon reload of the browser, even after setting media.peerc=
onnection.enabled to true, the video conferencing service wouldn't work unti=
l I installed the WebRTC 0.1.3 extension for Firefox and set it to expose my=
 real IP address as well as allowing the following:
>>>>=20
>>>> a. navigator.getUserMedia
>>>> b. window.MediaStreamTrack
>>>> c. window.RTCPeerConnection
>>>> d. window.RTCSessionDescription
>>>>=20
>>>> That seemed to me to be a direct consequence of the persistent and ongo=
ing security vulnerability in WebRTC.
>>>>=20
>>>> I contacted the video conferencing service provider and their solution w=
as simply to state that "If the peer to peer connection is of concern, you c=
an utilize our premium version which will route traffic through a forwarding=
 server with in our environment that handles the processing of the video and=
 sound of all users in the conference and send it to each user individually r=
ather than using a peer to peer connection."  In other words, they expect th=
at people should have to pay in order to mitigate this WebRTC security vulne=
rability, because they are unwilling to design to protect all users from it.=

>>>>=20
>>>> In a similar fashion, in the second example of the browser based WebRTC=
 version of a decentralized exchange, the user's financial information is pe=
riodically exposed as a result of the information leakage which is presently=
 caused by WebRTC.  (Fortunately, this browser based WebRTC decentralized ex=
change is under development and is not currently open to actual usage.)
>>>>=20
>>>> Although undoubtedly this has been discussed here before, I am asking t=
hat WebRTC security issues be reconsidered and that a new security review be=
 done to ameliorate or eliminate this problem in terms of WebRTC leaking thi=
s information.  It is my view that WebRTC should be further designed and dev=
eloped with an eye to mitigating this serious problem, so that users of serv=
ices which might rely partially or wholly upon WebRTC would not have to be c=
oncerned about such information leakage.
>>>>=20
>>>> Respectfully,
>>>>=20
>>>> Colin Gallagher
>>>>=20
>>>>=20
>>>> _______________________________________________
>>>> rtcweb mailing list
>>>> rtcweb@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>>=20
>>>=20
>>=20
>=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=3D=
utf-8"></head><body dir=3D"auto"><div></div><div>Colin -&nbsp;</div><div><br=
></div><div>Are you claiming that the recommendations of the document Justin=
 refers to are inadequate, or just that some browsers have not fully or corr=
ectly implemented it?&nbsp;</div><div><br></div><div>If the former, what wou=
ld you propose to change?</div><div><br>On May 26, 2017, at 10:49 AM, Justin=
 Uberti &lt;<a href=3D"mailto:juberti@google.com">juberti@google.com</a>&gt;=
 wrote:<br><br></div><blockquote type=3D"cite"><div><div dir=3D"ltr">Colin,<=
div><br></div><div>As Eric mentioned, the group has reviewed this exact issu=
e, and the aforementioned document represents the result of the review.</div=
><div><br></div><div>I am still not quite clear on the exact technical issue=
 you would like to see addressed and why it is, as you claim, a serious vuln=
erability.</div><div><br></div><div>Comments inline.</div><div class=3D"gmai=
l_extra"><br><div class=3D"gmail_quote">On Fri, May 26, 2017 at 12:58 AM, Co=
lin Gallagher <span dir=3D"ltr">&lt;<a href=3D"mailto:colingallagher.rpcv@gm=
ail.com" target=3D"_blank">colingallagher.rpcv@gmail.com</a>&gt;</span> wrot=
e:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-le=
ft:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div><div><div><div><di=
v><div><div><div>Justin,<br><br></div>It's my understanding that various bro=
wsers have made an attempt to conform to the recommendations in <a href=3D"h=
ttps://tools.ietf.org/html/draft-ietf-rtcweb-ip-handling-03" target=3D"_blan=
k">the document that you have provided</a>.&nbsp; Unfortunately, at least in=
 Firefox (the most current version) so far as I can tell, there is a serious=
 vulnerability.&nbsp; To be more specific, the problem was observed in my op=
erating system with my current browser, which is Ubuntu 16.10 64-bit with Fi=
refox=20
53.0.2 (64-bit).&nbsp; The browser does not address the WebRTC problems by d=
efault.<br><br></div>The stackexchange page that I cited originally is actua=
lly not outdated in the sense that its method of measuring whether or not th=
e WebRTC vulnerability exists on a given system is still intact and works.&n=
bsp; The stackexchange page suggests two pages to determine whether or not y=
ou have leakage / WebRTC related vulnerability.&nbsp; These pages are:<br><b=
r></div>1) Browser leaks test for WebRTC <a href=3D"https://browserleaks.com=
/webrtc" target=3D"_blank">https://browserleaks.com/<wbr>webrtc</a><br></div=
>2) IP Leaks <a href=3D"https://ipleak.net/" target=3D"_blank">https://iplea=
k.net/</a><br><br></div>Currently for Browser leaks test for WebRTC (due to t=
hat I have again changed settings in my system so that I will not have to be=
 exposed to this vulnerability) the settings show up on the page as:<br>RTCP=
eerConnection<span class=3D"m_3620305583784542015gmail-bad">=C3=97</span>Fal=
se<br>RTCDataChannel<span class=3D"m_3620305583784542015gmail-bad">=C3=97</s=
pan>False<br>ORTC (Microsoft Edge)<span class=3D"m_3620305583784542015gmail-=
bad">=C3=97</span>False (for example....)<br><br></div>And under IP leaks, i=
t currently shows "No leak, RTCPeerConnection not available."&nbsp; <br><br>=
</div><i>That is because:</i><br><br></div>(A) I have (again) changed my set=
tings in about:config back to the following<br><br><ol><span class=3D""><li>=
I typed <em>about:config</em> in the address bar</li><li>I found the setting=
 <em>media.peerconnection.enabled</em></li></span><li>I set it to <strong>fa=
lse&nbsp; (it was previously set at true, so as to allow WebRTC to work, but=
 this exposed information which I would not have wanted to be exposed)</stro=
ng></li></ol><p>(B) And I also have changed my settings back to the followin=
g</p><p>in the WebRTC 0.1.3 extension for Firefox I have set it to no longer=
 expose my real IP=20
address as well as no longer allowing the following:</p><span class=3D""><p>=
a. navigator.getUserMedia<br>b. window.MediaStreamTrack<br>c. window.RTCPeer=
Connection<br>d. window.RTCSessionDescription</p></span><p>To be clear, <b>t=
hese mitigations that I have employed above would be unnecessary if WebRTC h=
ad not been designed in such a manner that it so clearly exposes so much use=
r information.</b></p><p>In <a href=3D"https://tools.ietf.org/html/draft-iet=
f-rtcweb-ip-handling-03" target=3D"_blank">the document you provided</a>, ti=
tled "WebRTC IP Address Handling Requirements - draft-ietf-rtcweb-ip-handlin=
g-<wbr>03," it is stated that:</p><p><i>"<span style=3D"font-family:arial,he=
lvetica,sans-serif">1.  If the client is behind a NAT, the client's private I=
P addresses, typically [<a href=3D"https://tools.ietf.org/html/rfc1918" titl=
e=3D"&quot;Address Allocation for Private Internets&quot;" target=3D"_blank"=
>RFC1918</a>] addresses, can be learned.
</span></i></p><pre class=3D"m_3620305583784542015gmail-newpage"><i><span st=
yle=3D"font-family:arial,helvetica,sans-serif"> 2.  If the client tries to h=
ide its physical location through a VPN,
       and the VPN and local OS support routing over multiple interfaces
       (i.e., a "split-tunnel" VPN), WebRTC will discover the public
       address for the VPN as well as the ISP public address that the
       VPN runs over.

   3.  If the client is behind a proxy (a client-configured "classical
       application proxy", as defined in <a href=3D"https://tools.ietf.org/h=
tml/rfc1919#section-3" target=3D"_blank">[RFC1919], Section&nbsp;3</a>), but=

       direct access to the Internet is also supported, WebRTC's STUN
       [<a href=3D"https://tools.ietf.org/html/rfc5389" title=3D"&quot;Sessi=
on Traversal Utilities for NAT (STUN)&quot;" target=3D"_blank">RFC5389</a>]c=
hecks will bypass the proxy and reveal the public
       address of the client."<br><br></span></i></pre><pre class=3D"m_36203=
05583784542015gmail-newpage"><span style=3D"font-family:arial,helvetica,sans=
-serif">All of these are obvious vulnerabilities and serious problems (not o=
nly #2 as the document suggests).<br></span></pre></div></blockquote><div><b=
r></div><div>The document discusses #1 and #3 in detail, and outlines mechan=
isms that can be used to address these cases if needed.</div><div><br></div>=
<div>If, despite the findings in the document, you think that these are inde=
ed serious problems, it would be good to outline exactly why.</div><blockquo=
te class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc sol=
id;padding-left:1ex"><div dir=3D"ltr"><pre class=3D"m_3620305583784542015gma=
il-newpage"><span style=3D"font-family:arial,helvetica,sans-serif"><br></spa=
n></pre><pre class=3D"m_3620305583784542015gmail-newpage"><span style=3D"fon=
t-family:arial,helvetica,sans-serif">It is stated in the document that, <br>=
<i><br>"we want to avoid solutions that disable WebRTC or make it harder to u=
se."<br><br></i></span></pre><pre class=3D"m_3620305583784542015gmail-newpag=
e"><span style=3D"font-family:arial,helvetica,sans-serif">Yet, so far as my o=
bservations of WebRTC in the current version of Firefox with Ubuntu 16.10 ha=
ve shown,<br></span></pre><pre class=3D"m_3620305583784542015gmail-newpage">=
<span style=3D"font-family:arial,helvetica,sans-serif">there appears to be n=
o other choice other than to disable WebRTC because its use completely compr=
omises <br></span></pre><pre class=3D"m_3620305583784542015gmail-newpage"><s=
pan style=3D"font-family:arial,helvetica,sans-serif">privacy and thus makes a=
ll other aspects of your system more difficult generally.</span></pre></div>=
</blockquote><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bo=
rder-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><pre class=3D"m_=
3620305583784542015gmail-newpage"><span style=3D"font-family:arial,helvetica=
,sans-serif"><br></span></pre><pre class=3D"m_3620305583784542015gmail-newpa=
ge"><span style=3D"font-family:arial,helvetica,sans-serif">It is stated in t=
he document you cited that the goals here are to "</span>Provide a privacy-f=
riendly default behavior which strikes the
      right balance between privacy and media performance for most users
      and use cases. (and) For users who care more about one versus the othe=
r, provide a
      means to customize the experience."</pre><pre class=3D"m_3620305583784=
542015gmail-newpage"><span style=3D"font-family:arial,helvetica,sans-serif">=
This has clearlly not been accomplished in the context of Ubuntu 16.10 64-bi=
t with Firefox 53.0.2 (64-bit).<br><br></span></pre><pre class=3D"m_36203055=
83784542015gmail-newpage"><span style=3D"font-family:arial,helvetica,sans-se=
rif">Hence my request for a new security review of WebRTC based on observed v=
ulnerabilities.<br></span></pre></div></blockquote><div><br></div><div>To my=
 knowledge, the document addresses these goals. If you believe Firefox is no=
t conforming, please file a bug; if you think the document is in error, plea=
se point out where and why.</div><blockquote class=3D"gmail_quote" style=3D"=
margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=3D"l=
tr"><pre class=3D"m_3620305583784542015gmail-newpage"><span style=3D"font-fa=
mily:arial,helvetica,sans-serif"></span></pre><p><span style=3D"font-family:=
arial,helvetica,sans-serif"></span><br></p><strong></strong><div><div><div><=
div><div><br></div></div></div></div></div></div><div class=3D"HOEnZb"><div c=
lass=3D"h5"><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Thu=
, May 25, 2017 at 1:04 PM, Justin Uberti <span dir=3D"ltr">&lt;<a href=3D"ma=
ilto: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;bord=
er-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr">Colin,<div><br></d=
iv><div>This is an important topic, and the group has reviewed this situatio=
n in detail. The results have been written up in <a href=3D"https://tools.ie=
tf.org/html/draft-ietf-rtcweb-ip-handling-03" target=3D"_blank">this documen=
t</a>, and to my knowledge most browsers that support WebRTC currently confo=
rm to its recommendations.</div><div><br></div><div>However, I'm not totally=
 clear on your specific technical concern. You linked to an (now outdated) s=
tackexchange page, but could you provide more detail about the specific situ=
ation you are concerned about?</div><div><br></div><div>Justin</div></div><d=
iv class=3D"gmail_extra"><br><div class=3D"gmail_quote"><div><div class=3D"m=
_3620305583784542015h5">On Wed, May 24, 2017 at 5:35 PM, Colin Gallagher <sp=
an dir=3D"ltr">&lt;<a href=3D"mailto:colingallagher.rpcv@gmail.com" target=3D=
"_blank">colingallagher.rpcv@gmail.com</a><wbr>&gt;</span> wrote:<br></div><=
/div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-lef=
t:1px #ccc solid;padding-left:1ex"><div><div class=3D"m_3620305583784542015h=
5"><div dir=3D"ltr"><div><div><div><div><div><div><div><div>To:&nbsp; IETF R=
TCWeb WG<br><br></div>From:&nbsp; Colin Gallagher<br></div>&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <a href=3D"https://www.linkedin.=
com/in/colingallagher" target=3D"_blank">https://www.linkedin.com/in/co<wbr>=
lingallagher</a><br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp; <a href=3D"https://lifeboat.com/ex/bios.colin.gallagher" target=3D"_bla=
nk">https://lifeboat.com/ex/bios.c<wbr>olin.gallagher</a><br><br></div>Reque=
st for New Security Review of WebRTC based on observed vulnerabilities in ce=
rtain examples<br><br></div>Dear IETF RTCWeb WG,<br><br></div><div>This requ=
est relates to WebRTC.&nbsp; The WebRTC is in draft at <a href=3D"https://w3=
c.github.io/webrtc-pc/" target=3D"_blank">https://w3c.github.io/webrtc-p<wbr=
>c/</a> and is=20
described also at=20
<a href=3D"https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API#WebRT=
C_interfaces" target=3D"_blank">https://developer.mozilla.org/<wbr>en-US/doc=
s/Web/API/WebRTC_API#<wbr>WebRTC_interfaces</a></div><div><br></div>There ar=
e certain examples where the use of WebRTC has presented security vulnerabil=
ities.&nbsp; While not exhaustive, there are two I wanted to mention that I h=
ad been exposed to recently.<br></div><div><br></div>1) The use of a video c=
onferencing service which relies upon WebRTC (that is a well established ser=
vice)<br></div>2) The examination of a browser based WebRTC version of a dec=
entralized exchange (that is currently under development)<br><br></div>In bo=
th cases the security vulnerability which is described below was the result o=
f the use of WebRTC.<br><br>The problem under consideration is as follows:<b=
r><br>As per the first example (the video conferencing service), recently
 I came upon a vendor who shall remain unnamed who uses WebRTC as the=20
backbone for their video conferencing service.&nbsp; As a result, I found I=20=

was unable to access the service in question without disabling some=20
security settings which I would have preferred to leave intact. <br><br>My s=
ystem is (based on what the aforementioned service detects) Mozilla/5.0 (X11=
; Ubuntu; Linux x86_64; rv:53.0)=20
Gecko/20100101 Firefox/53.0.&nbsp; This is Ubuntu 16.10 64-bit with Firefox=20=

53.0.2.<br><br>However, quite some <i>long</i> while back I disabled WebRTC b=
ecause of the security vulnerability (described in part <a href=3D"https://s=
ecurity.stackexchange.com/questions/82129/what-is-webrtc-and-how-can-i-prote=
ct-myself-against-leaking-information-that-doe" target=3D"_blank">here on in=
formation security stack exchange</a>).&nbsp; So as to mitigate this vulnera=
bility, I changed settings in Firefox as follows:<br><br><ol><li>I typed <em=
>about:config</em> in the address bar</li><li>I found the setting <em>media.=
peerconnection.enabled</em></li><li>I set it to <strong>false</strong></li><=
/ol><p><strong></strong>Unfortunately,
 this setting does not allow Firefox to work with the video conferencing
 service in question, which relies entirely on WebRTC.<br></p><p>So I had to=
 go back in and change the media.peerconnection.enabled setting to <b>true</=
b>
 in Firefox in order to get it to work with the service in question.=20
While this enabled me to conference with teams that desire to use the=20
video conferencing service that rely wholly upon WebRTC, it concerns me=20
because of the security vulnerability.</p><p>One suggested mitigation was ha=
ving NoScript, but using NoScript was not helpful (you generally need to dis=
able it for the site you are viewing that relies upon WebRTC, in order to ob=
tain any functionality, and whether NoScript is off or on, WebRTC still caus=
es the vulnerability of leaking information as previously described).<br></p=
><p>Furthermore, upon reload of the browser, even after setting media.peerco=
nnection.enabled to <b>true</b>,
 the video conferencing service wouldn't work until I installed the=20
WebRTC 0.1.3 extension for Firefox and set it to expose my real IP=20
address as well as allowing the following:</p><p>a. navigator.getUserMedia<b=
r>b. window.MediaStreamTrack<br>c. window.RTCPeerConnection<br>d. window.RTC=
SessionDescription<br></p><p>That seemed to me to be a direct consequence of=
 the persistent and ongoing security vulnerability in WebRTC.<br></p><p>I
 contacted the video conferencing service provider and their solution=20
was simply to state that "If the peer to peer connection is of concern,=20
you can utilize our=20
premium version which will route traffic through a forwarding server=20
with in our environment that handles the processing of the video and=20
sound of all users in the conference and send it to each user=20
individually rather than using a peer to peer connection."&nbsp; In other=20=

words, they expect that people should have to pay in order to mitigate <a hr=
ef=3D"https://security.stackexchange.com/questions/82129/what-is-webrtc-and-=
how-can-i-protect-myself-against-leaking-information-that-doe" target=3D"_bl=
ank">this WebRTC security vulnerability</a>, because they are unwilling to d=
esign to protect all users from it.</p><p>In a similar fashion, in the secon=
d example of the browser based WebRTC version of a decentralized exchange, t=
he user's financial information is periodically exposed as a result of the i=
nformation leakage which is presently caused by WebRTC.&nbsp; (Fortunately, t=
his browser based WebRTC decentralized exchange is under development and is n=
ot currently open to actual usage.)<br></p><p>Although
 undoubtedly this has been discussed here before, I am asking that WebRTC se=
curity issues be
 reconsidered and that a new security review be done to ameliorate or=20
eliminate this problem in terms of WebRTC leaking this information.&nbsp; It=
 is my view that WebRTC should be further designed and developed with an eye=
 to mitigating this serious problem, so that users of services which might r=
ely partially or wholly upon WebRTC would not have to be concerned about suc=
h information leakage.<br></p><p>Respectfully,</p>Colin Gallagher<br><br></d=
iv>
<br></div></div>______________________________<wbr>_________________<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" t=
arget=3D"_blank">https://www.ietf.org/mailman/l<wbr>istinfo/rtcweb</a><br>
<br></blockquote></div><br></div>
</blockquote></div><br></div>
</div></div></blockquote></div><br></div></div>
</div></blockquote><blockquote type=3D"cite"><div><span>____________________=
___________________________</span><br><span>rtcweb mailing list</span><br><s=
pan><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: Bernard Aboba <bernard.aboba@gmail.com>
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Subject: Re: [rtcweb] Request for New Security Review of WebRTC based on observed vulnerabilities
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Based on your messages so far I am not clear what you are asking for the IET=
F RTCWEB WG to do. You have not filed any issues against the IP Handling doc=
ument so it is not clear if you feel it needs to be changed or even what it i=
s may sdonf. Nor are you posting your request to the SDO (W3C) that actually=
 owns the API and reviews its privacy and security implications.=20

> On May 26, 2017, at 11:14 AM, Colin Gallagher <colingallagher.rpcv@gmail.c=
om> wrote:
>=20
> It seems the consensus of the group here is essentially a refusal to furth=
er review and potentially redesign WebRTC to remove the serious security vul=
nerabilities that it presents to its users.
>=20
> Thank you for your time.  I will not further respond to this thread.
>=20
>> On Fri, May 26, 2017 at 10:49 AM, Justin Uberti <juberti@google.com> wrot=
e:
>> Colin,
>>=20
>> As Eric mentioned, the group has reviewed this exact issue, and the afore=
mentioned document represents the result of the review.
>>=20
>> I am still not quite clear on the exact technical issue you would like to=
 see addressed and why it is, as you claim, a serious vulnerability.
>>=20
>> Comments inline.
>>=20
>>> On Fri, May 26, 2017 at 12:58 AM, Colin Gallagher <colingallagher.rpcv@g=
mail.com> wrote:
>>> Justin,
>>>=20
>>> It's my understanding that various browsers have made an attempt to conf=
orm to the recommendations in the document that you have provided.  Unfortun=
ately, at least in Firefox (the most current version) so far as I can tell, t=
here is a serious vulnerability.  To be more specific, the problem was obser=
ved in my operating system with my current browser, which is Ubuntu 16.10 64=
-bit with Firefox 53.0.2 (64-bit).  The browser does not address the WebRTC p=
roblems by default.
>>>=20
>>> The stackexchange page that I cited originally is actually not outdated i=
n the sense that its method of measuring whether or not the WebRTC vulnerabi=
lity exists on a given system is still intact and works.  The stackexchange p=
age suggests two pages to determine whether or not you have leakage / WebRTC=
 related vulnerability.  These pages are:
>>>=20
>>> 1) Browser leaks test for WebRTC https://browserleaks.com/webrtc
>>> 2) IP Leaks https://ipleak.net/
>>>=20
>>> Currently for Browser leaks test for WebRTC (due to that I have again ch=
anged settings in my system so that I will not have to be exposed to this vu=
lnerability) the settings show up on the page as:
>>> RTCPeerConnection=C3=97False
>>> RTCDataChannel=C3=97False
>>> ORTC (Microsoft Edge)=C3=97False (for example....)
>>>=20
>>> And under IP leaks, it currently shows "No leak, RTCPeerConnection not a=
vailable." =20
>>>=20
>>> That is because:
>>>=20
>>> (A) I have (again) changed my settings in about:config back to the follo=
wing
>>>=20
>>> I typed about:config in the address bar
>>> I found the setting media.peerconnection.enabled
>>> I set it to false  (it was previously set at true, so as to allow WebRTC=
 to work, but this exposed information which I would not have wanted to be e=
xposed)
>>> (B) And I also have changed my settings back to the following
>>>=20
>>> in the WebRTC 0.1.3 extension for Firefox I have set it to no longer exp=
ose my real IP address as well as no longer allowing the following:
>>>=20
>>> a. navigator.getUserMedia
>>> b. window.MediaStreamTrack
>>> c. window.RTCPeerConnection
>>> d. window.RTCSessionDescription
>>>=20
>>> To be clear, these mitigations that I have employed above would be unnec=
essary if WebRTC had not been designed in such a manner that it so clearly e=
xposes so much user information.
>>>=20
>>> In the document you provided, titled "WebRTC IP Address Handling Require=
ments - draft-ietf-rtcweb-ip-handling-03," it is stated that:
>>>=20
>>> "1. If the client is behind a NAT, the client's private IP addresses, ty=
pically [RFC1918] addresses, can be learned.
>>>=20
>>>  2.  If the client tries to hide its physical location through a VPN,
>>>        and the VPN and local OS support routing over multiple interfaces=

>>>        (i.e., a "split-tunnel" VPN), WebRTC will discover the public
>>>        address for the VPN as well as the ISP public address that the
>>>        VPN runs over.
>>>=20
>>>    3.  If the client is behind a proxy (a client-configured "classical
>>>        application proxy", as defined in [RFC1919], Section 3), but
>>>        direct access to the Internet is also supported, WebRTC's STUN
>>>        [RFC5389]checks will bypass the proxy and reveal the public
>>>        address of the client."
>>>=20
>>> All of these are obvious vulnerabilities and serious problems (not only #=
2 as the document suggests).
>>=20
>> The document discusses #1 and #3 in detail, and outlines mechanisms that c=
an be used to address these cases if needed.
>>=20
>> If, despite the findings in the document, you think that these are indeed=
 serious problems, it would be good to outline exactly why.
>>>=20
>>> It is stated in the document that,=20
>>>=20
>>> "we want to avoid solutions that disable WebRTC or make it harder to use=
."
>>>=20
>>> Yet, so far as my observations of WebRTC in the current version of Firef=
ox with Ubuntu 16.10 have shown,
>>> there appears to be no other choice other than to disable WebRTC because=
 its use completely compromises=20
>>> privacy and thus makes all other aspects of your system more difficult g=
enerally.
>>>=20
>>> It is stated in the document you cited that the goals here are to "Provi=
de a privacy-friendly default behavior which strikes the
>>>       right balance between privacy and media performance for most users=

>>>       and use cases. (and) For users who care more about one versus the o=
ther, provide a
>>>       means to customize the experience."
>>> This has clearlly not been accomplished in the context of Ubuntu 16.10 6=
4-bit with Firefox 53.0.2 (64-bit).
>>>=20
>>> Hence my request for a new security review of WebRTC based on observed v=
ulnerabilities.
>>=20
>> To my knowledge, the document addresses these goals. If you believe Firef=
ox is not conforming, please file a bug; if you think the document is in err=
or, please point out where and why.
>>>=20
>>>=20
>>>=20
>>>> On Thu, May 25, 2017 at 1:04 PM, Justin Uberti <juberti@google.com> wro=
te:
>>>> Colin,
>>>>=20
>>>> This is an important topic, and the group has reviewed this situation i=
n detail. The results have been written up in this document, and to my knowl=
edge most browsers that support WebRTC currently conform to its recommendati=
ons.
>>>>=20
>>>> However, I'm not totally clear on your specific technical concern. You l=
inked to an (now outdated) stackexchange page, but could you provide more de=
tail about the specific situation you are concerned about?
>>>>=20
>>>> Justin
>>>>=20
>>>>> On Wed, May 24, 2017 at 5:35 PM, Colin Gallagher <colingallagher.rpcv@=
gmail.com> wrote:
>>>>> To:  IETF RTCWeb WG
>>>>>=20
>>>>> From:  Colin Gallagher
>>>>>            https://www.linkedin.com/in/colingallagher
>>>>>            https://lifeboat.com/ex/bios.colin.gallagher
>>>>>=20
>>>>> Request for New Security Review of WebRTC based on observed vulnerabil=
ities in certain examples
>>>>>=20
>>>>> Dear IETF RTCWeb WG,
>>>>>=20
>>>>> This request relates to WebRTC.  The WebRTC is in draft at https://w3c=
.github.io/webrtc-pc/ and is described also at https://developer.mozilla.org=
/en-US/docs/Web/API/WebRTC_API#WebRTC_interfaces
>>>>>=20
>>>>> There are certain examples where the use of WebRTC has presented secur=
ity vulnerabilities.  While not exhaustive, there are two I wanted to mentio=
n that I had been exposed to recently.
>>>>>=20
>>>>> 1) The use of a video conferencing service which relies upon WebRTC (t=
hat is a well established service)
>>>>> 2) The examination of a browser based WebRTC version of a decentralize=
d exchange (that is currently under development)
>>>>>=20
>>>>> In both cases the security vulnerability which is described below was t=
he result of the use of WebRTC.
>>>>>=20
>>>>> The problem under consideration is as follows:
>>>>>=20
>>>>> As per the first example (the video conferencing service), recently I c=
ame upon a vendor who shall remain unnamed who uses WebRTC as the backbone f=
or their video conferencing service.  As a result, I found I was unable to a=
ccess the service in question without disabling some security settings which=
 I would have preferred to leave intact.=20
>>>>>=20
>>>>> My system is (based on what the aforementioned service detects) Mozill=
a/5.0 (X11; Ubuntu; Linux x86_64; rv:53.0) Gecko/20100101 Firefox/53.0.  Thi=
s is Ubuntu 16.10 64-bit with Firefox 53.0.2.
>>>>>=20
>>>>> However, quite some long while back I disabled WebRTC because of the s=
ecurity vulnerability (described in part here on information security stack e=
xchange).  So as to mitigate this vulnerability, I changed settings in Firef=
ox as follows:
>>>>>=20
>>>>> I typed about:config in the address bar
>>>>> I found the setting media.peerconnection.enabled
>>>>> I set it to false
>>>>> Unfortunately, this setting does not allow Firefox to work with the vi=
deo conferencing service in question, which relies entirely on WebRTC.
>>>>>=20
>>>>> So I had to go back in and change the media.peerconnection.enabled set=
ting to true  in Firefox in order to get it to work with the service in ques=
tion. While this enabled me to conference with teams that desire to use the v=
ideo conferencing service that rely wholly upon WebRTC, it concerns me becau=
se of the security vulnerability.
>>>>>=20
>>>>> One suggested mitigation was having NoScript, but using NoScript was n=
ot helpful (you generally need to disable it for the site you are viewing th=
at relies upon WebRTC, in order to obtain any functionality, and whether NoS=
cript is off or on, WebRTC still causes the vulnerability of leaking informa=
tion as previously described).
>>>>>=20
>>>>> Furthermore, upon reload of the browser, even after setting media.peer=
connection.enabled to true, the video conferencing service wouldn't work unt=
il I installed the WebRTC 0.1.3 extension for Firefox and set it to expose m=
y real IP address as well as allowing the following:
>>>>>=20
>>>>> a. navigator.getUserMedia
>>>>> b. window.MediaStreamTrack
>>>>> c. window.RTCPeerConnection
>>>>> d. window.RTCSessionDescription
>>>>>=20
>>>>> That seemed to me to be a direct consequence of the persistent and ong=
oing security vulnerability in WebRTC.
>>>>>=20
>>>>> I contacted the video conferencing service provider and their solution=
 was simply to state that "If the peer to peer connection is of concern, you=
 can utilize our premium version which will route traffic through a forwardi=
ng server with in our environment that handles the processing of the video a=
nd sound of all users in the conference and send it to each user individuall=
y rather than using a peer to peer connection."  In other words, they expect=
 that people should have to pay in order to mitigate this WebRTC security vu=
lnerability, because they are unwilling to design to protect all users from i=
t.
>>>>>=20
>>>>> In a similar fashion, in the second example of the browser based WebRT=
C version of a decentralized exchange, the user's financial information is p=
eriodically exposed as a result of the information leakage which is presentl=
y caused by WebRTC.  (Fortunately, this browser based WebRTC decentralized e=
xchange is under development and is not currently open to actual usage.)
>>>>>=20
>>>>> Although undoubtedly this has been discussed here before, I am asking t=
hat WebRTC security issues be reconsidered and that a new security review be=
 done to ameliorate or eliminate this problem in terms of WebRTC leaking thi=
s information.  It is my view that WebRTC should be further designed and dev=
eloped with an eye to mitigating this serious problem, so that users of serv=
ices which might rely partially or wholly upon WebRTC would not have to be c=
oncerned about such information leakage.
>>>>>=20
>>>>> Respectfully,
>>>>>=20
>>>>> Colin Gallagher
>>>>>=20
>>>>>=20
>>>>> _______________________________________________
>>>>> rtcweb mailing list
>>>>> rtcweb@ietf.org
>>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>>>=20
>>>>=20
>>>=20
>>=20
>=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=3D=
utf-8"></head><body dir=3D"auto"><div></div><div>Based on your messages so f=
ar I am not clear what you are asking for the IETF RTCWEB WG to do. You have=
 not filed any issues against the IP Handling document so it is not clear if=
 you feel it needs to be changed or even what it is may sdonf. Nor are you p=
osting your request to the SDO (W3C) that actually owns the API and reviews i=
ts privacy and security implications.&nbsp;</div><div><br>On May 26, 2017, a=
t 11:14 AM, Colin Gallagher &lt;<a href=3D"mailto:colingallagher.rpcv@gmail.=
com">colingallagher.rpcv@gmail.com</a>&gt; wrote:<br><br></div><blockquote t=
ype=3D"cite"><div><div dir=3D"ltr"><div>It seems the consensus of the group h=
ere is essentially a refusal to further review and potentially redesign WebR=
TC to remove the serious security vulnerabilities that it presents to its us=
ers.<br><br></div>Thank you for your time.&nbsp; I will not further respond t=
o this thread.<br><div><br></div></div><div class=3D"gmail_extra"><div class=
=3D"gmail_quote">On Fri, May 26, 2017 at 10:49 AM, Justin Uberti <span dir=3D=
"ltr">&lt;<a href=3D"mailto:juberti@google.com" target=3D"_blank">juberti@go=
ogle.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">Colin,<div><br></div><div>As Eric mentioned, the group has reviewed thi=
s exact issue, and the aforementioned document represents the result of the r=
eview.</div><div><br></div><div>I am still not quite clear on the exact tech=
nical issue you would like to see addressed and why it is, as you claim, a s=
erious vulnerability.</div><div><br></div><div>Comments inline.</div><div cl=
ass=3D"gmail_extra"><br><div class=3D"gmail_quote"><div><div class=3D"h5">On=
 Fri, May 26, 2017 at 12:58 AM, Colin Gallagher <span dir=3D"ltr">&lt;<a hre=
f=3D"mailto:colingallagher.rpcv@gmail.com" target=3D"_blank">colingallagher.=
rpcv@gmail.com</a><wbr>&gt;</span> wrote:<br><blockquote class=3D"gmail_quot=
e" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><=
div dir=3D"ltr"><div><div><div><div><div><div><div><div>Justin,<br><br></div=
>It's my understanding that various browsers have made an attempt to conform=
 to the recommendations in <a href=3D"https://tools.ietf.org/html/draft-ietf=
-rtcweb-ip-handling-03" target=3D"_blank">the document that you have provide=
d</a>.&nbsp; Unfortunately, at least in Firefox (the most current version) s=
o far as I can tell, there is a serious vulnerability.&nbsp; To be more spec=
ific, the problem was observed in my operating system with my current browse=
r, which is Ubuntu 16.10 64-bit with Firefox=20
53.0.2 (64-bit).&nbsp; The browser does not address the WebRTC problems by d=
efault.<br><br></div>The stackexchange page that I cited originally is actua=
lly not outdated in the sense that its method of measuring whether or not th=
e WebRTC vulnerability exists on a given system is still intact and works.&n=
bsp; The stackexchange page suggests two pages to determine whether or not y=
ou have leakage / WebRTC related vulnerability.&nbsp; These pages are:<br><b=
r></div>1) Browser leaks test for WebRTC <a href=3D"https://browserleaks.com=
/webrtc" target=3D"_blank">https://browserleaks.com/webrt<wbr>c</a><br></div=
>2) IP Leaks <a href=3D"https://ipleak.net/" target=3D"_blank">https://iplea=
k.net/</a><br><br></div>Currently for Browser leaks test for WebRTC (due to t=
hat I have again changed settings in my system so that I will not have to be=
 exposed to this vulnerability) the settings show up on the page as:<br>RTCP=
eerConnection<span class=3D"m_3602413254325098192m_3620305583784542015gmail-=
bad">=C3=97</span>False<br>RTCDataChannel<span class=3D"m_360241325432509819=
2m_3620305583784542015gmail-bad">=C3=97</span>False<br>ORTC (Microsoft Edge)=
<span class=3D"m_3602413254325098192m_3620305583784542015gmail-bad">=C3=97</=
span>False (for example....)<br><br></div>And under IP leaks, it currently s=
hows "No leak, RTCPeerConnection not available."&nbsp; <br><br></div><i>That=
 is because:</i><br><br></div>(A) I have (again) changed my settings in abou=
t:config back to the following<br><br><ol><span><li>I typed <em>about:config=
</em> in the address bar</li><li>I found the setting <em>media.peerconnectio=
n.enabled</em></li></span><li>I set it to <strong>false&nbsp; (it was previo=
usly set at true, so as to allow WebRTC to work, but this exposed informatio=
n which I would not have wanted to be exposed)</strong></li></ol><p>(B) And I=
 also have changed my settings back to the following</p><p>in the WebRTC 0.1=
.3 extension for Firefox I have set it to no longer expose my real IP=20
address as well as no longer allowing the following:</p><span><p>a. navigato=
r.getUserMedia<br>b. window.MediaStreamTrack<br>c. window.RTCPeerConnection<=
br>d. window.RTCSessionDescription</p></span><p>To be clear, <b>these mitiga=
tions that I have employed above would be unnecessary if WebRTC had not been=
 designed in such a manner that it so clearly exposes so much user informati=
on.</b></p><p>In <a href=3D"https://tools.ietf.org/html/draft-ietf-rtcweb-ip=
-handling-03" target=3D"_blank">the document you provided</a>, titled "WebRT=
C IP Address Handling Requirements - draft-ietf-rtcweb-ip-handling-<wbr>03,"=
 it is stated that:</p><p><i>"<span style=3D"font-family:arial,helvetica,san=
s-serif">1.  If the client is behind a NAT, the client's private IP addresse=
s, typically [<a href=3D"https://tools.ietf.org/html/rfc1918" title=3D"&quot=
;Address Allocation for Private Internets&quot;" target=3D"_blank">RFC1918</=
a>] addresses, can be learned.
</span></i></p><pre class=3D"m_3602413254325098192m_3620305583784542015gmail=
-newpage"><i><span style=3D"font-family:arial,helvetica,sans-serif"> 2.  If t=
he client tries to hide its physical location through a VPN,
       and the VPN and local OS support routing over multiple interfaces
       (i.e., a "split-tunnel" VPN), WebRTC will discover the public
       address for the VPN as well as the ISP public address that the
       VPN runs over.

   3.  If the client is behind a proxy (a client-configured "classical
       application proxy", as defined in <a href=3D"https://tools.ietf.org/h=
tml/rfc1919#section-3" target=3D"_blank">[RFC1919], Section&nbsp;3</a>), but=

       direct access to the Internet is also supported, WebRTC's STUN
       [<a href=3D"https://tools.ietf.org/html/rfc5389" title=3D"&quot;Sessi=
on Traversal Utilities for NAT (STUN)&quot;" target=3D"_blank">RFC5389</a>]c=
hecks will bypass the proxy and reveal the public
       address of the client."<br><br></span></i></pre><pre class=3D"m_36024=
13254325098192m_3620305583784542015gmail-newpage"><span style=3D"font-family=
:arial,helvetica,sans-serif">All of these are obvious vulnerabilities and se=
rious problems (not only #2 as the document suggests).<br></span></pre></div=
></blockquote><div><br></div></div></div><div>The document discusses #1 and #=
3 in detail, and outlines mechanisms that can be used to address these cases=
 if needed.</div><div><br></div><div>If, despite the findings in the documen=
t, you think that these are indeed serious problems, it would be good to out=
line exactly why.</div><span class=3D""><blockquote class=3D"gmail_quote" st=
yle=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div d=
ir=3D"ltr"><pre class=3D"m_3602413254325098192m_3620305583784542015gmail-new=
page"><span style=3D"font-family:arial,helvetica,sans-serif"><br></span></pr=
e><pre class=3D"m_3602413254325098192m_3620305583784542015gmail-newpage"><sp=
an style=3D"font-family:arial,helvetica,sans-serif">It is stated in the docu=
ment that, <br><i><br>"we want to avoid solutions that disable WebRTC or mak=
e it harder to use."<br><br></i></span></pre><pre class=3D"m_360241325432509=
8192m_3620305583784542015gmail-newpage"><span style=3D"font-family:arial,hel=
vetica,sans-serif">Yet, so far as my observations of WebRTC in the current v=
ersion of Firefox with Ubuntu 16.10 have shown,<br></span></pre><pre class=3D=
"m_3602413254325098192m_3620305583784542015gmail-newpage"><span style=3D"fon=
t-family:arial,helvetica,sans-serif">there appears to be no other choice oth=
er than to disable WebRTC because its use completely compromises <br></span>=
</pre><pre class=3D"m_3602413254325098192m_3620305583784542015gmail-newpage"=
><span style=3D"font-family:arial,helvetica,sans-serif">privacy and thus mak=
es all other aspects of your system more difficult generally.</span></pre></=
div></blockquote><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"><pre class=3D=
"m_3602413254325098192m_3620305583784542015gmail-newpage"><span style=3D"fon=
t-family:arial,helvetica,sans-serif"><br></span></pre><pre class=3D"m_360241=
3254325098192m_3620305583784542015gmail-newpage"><span style=3D"font-family:=
arial,helvetica,sans-serif">It is stated in the document you cited that the g=
oals here are to "</span>Provide a privacy-friendly default behavior which s=
trikes the
      right balance between privacy and media performance for most users
      and use cases. (and) For users who care more about one versus the othe=
r, provide a
      means to customize the experience."</pre><pre class=3D"m_3602413254325=
098192m_3620305583784542015gmail-newpage"><span style=3D"font-family:arial,h=
elvetica,sans-serif">This has clearlly not been accomplished in the context o=
f Ubuntu 16.10 64-bit with Firefox 53.0.2 (64-bit).<br><br></span></pre><pre=
 class=3D"m_3602413254325098192m_3620305583784542015gmail-newpage"><span sty=
le=3D"font-family:arial,helvetica,sans-serif">Hence my request for a new sec=
urity review of WebRTC based on observed vulnerabilities.<br></span></pre></=
div></blockquote><div><br></div></span><div>To my knowledge, the document ad=
dresses these goals. If you believe Firefox is not conforming, please file a=
 bug; if you think the document is in error, please point out where and why.=
</div><div><div class=3D"h5"><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">=
<pre class=3D"m_3602413254325098192m_3620305583784542015gmail-newpage"><span=
 style=3D"font-family:arial,helvetica,sans-serif"></span></pre><p><span styl=
e=3D"font-family:arial,helvetica,sans-serif"></span><br></p><strong></strong=
><div><div><div><div><div><br></div></div></div></div></div></div><div class=
=3D"m_3602413254325098192HOEnZb"><div class=3D"m_3602413254325098192h5"><div=
 class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Thu, May 25, 2017 a=
t 1:04 PM, Justin Uberti <span dir=3D"ltr">&lt;<a href=3D"mailto:juberti@goo=
gle.com" target=3D"_blank">juberti@google.com</a>&gt;</span> wrote:<br><bloc=
kquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc=
 solid;padding-left:1ex"><div dir=3D"ltr">Colin,<div><br></div><div>This is a=
n important topic, and the group has reviewed this situation in detail. The r=
esults have been written up in <a href=3D"https://tools.ietf.org/html/draft-=
ietf-rtcweb-ip-handling-03" target=3D"_blank">this document</a>, and to my k=
nowledge most browsers that support WebRTC currently conform to its recommen=
dations.</div><div><br></div><div>However, I'm not totally clear on your spe=
cific technical concern. You linked to an (now outdated) stackexchange page,=
 but could you provide more detail about the specific situation you are conc=
erned about?</div><div><br></div><div>Justin</div></div><div class=3D"gmail_=
extra"><br><div class=3D"gmail_quote"><div><div class=3D"m_36024132543250981=
92m_3620305583784542015h5">On Wed, May 24, 2017 at 5:35 PM, Colin Gallagher <=
span dir=3D"ltr">&lt;<a href=3D"mailto:colingallagher.rpcv@gmail.com" target=
=3D"_blank">colingallagher.rpcv@gmail.com</a><wbr>&gt;</span> wrote:<br></di=
v></div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-=
left:1px #ccc solid;padding-left:1ex"><div><div class=3D"m_36024132543250981=
92m_3620305583784542015h5"><div dir=3D"ltr"><div><div><div><div><div><div><d=
iv><div>To:&nbsp; IETF RTCWeb WG<br><br></div>From:&nbsp; Colin Gallagher<br=
></div>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <a href=3D=
"https://www.linkedin.com/in/colingallagher" target=3D"_blank">https://www.l=
inkedin.com/in/co<wbr>lingallagher</a><br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp; <a href=3D"https://lifeboat.com/ex/bios.colin.gal=
lagher" target=3D"_blank">https://lifeboat.com/ex/bios.c<wbr>olin.gallagher<=
/a><br><br></div>Request for New Security Review of WebRTC based on observed=
 vulnerabilities in certain examples<br><br></div>Dear IETF RTCWeb WG,<br><b=
r></div><div>This request relates to WebRTC.&nbsp; The WebRTC is in draft at=
 <a href=3D"https://w3c.github.io/webrtc-pc/" target=3D"_blank">https://w3c.=
github.io/webrtc-p<wbr>c/</a> and is=20
described also at=20
<a href=3D"https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API#WebRT=
C_interfaces" target=3D"_blank">https://developer.mozilla.org/<wbr>en-US/doc=
s/Web/API/WebRTC_API#<wbr>WebRTC_interfaces</a></div><div><br></div>There ar=
e certain examples where the use of WebRTC has presented security vulnerabil=
ities.&nbsp; While not exhaustive, there are two I wanted to mention that I h=
ad been exposed to recently.<br></div><div><br></div>1) The use of a video c=
onferencing service which relies upon WebRTC (that is a well established ser=
vice)<br></div>2) The examination of a browser based WebRTC version of a dec=
entralized exchange (that is currently under development)<br><br></div>In bo=
th cases the security vulnerability which is described below was the result o=
f the use of WebRTC.<br><br>The problem under consideration is as follows:<b=
r><br>As per the first example (the video conferencing service), recently
 I came upon a vendor who shall remain unnamed who uses WebRTC as the=20
backbone for their video conferencing service.&nbsp; As a result, I found I=20=

was unable to access the service in question without disabling some=20
security settings which I would have preferred to leave intact. <br><br>My s=
ystem is (based on what the aforementioned service detects) Mozilla/5.0 (X11=
; Ubuntu; Linux x86_64; rv:53.0)=20
Gecko/20100101 Firefox/53.0.&nbsp; This is Ubuntu 16.10 64-bit with Firefox=20=

53.0.2.<br><br>However, quite some <i>long</i> while back I disabled WebRTC b=
ecause of the security vulnerability (described in part <a href=3D"https://s=
ecurity.stackexchange.com/questions/82129/what-is-webrtc-and-how-can-i-prote=
ct-myself-against-leaking-information-that-doe" target=3D"_blank">here on in=
formation security stack exchange</a>).&nbsp; So as to mitigate this vulnera=
bility, I changed settings in Firefox as follows:<br><br><ol><li>I typed <em=
>about:config</em> in the address bar</li><li>I found the setting <em>media.=
peerconnection.enabled</em></li><li>I set it to <strong>false</strong></li><=
/ol><p><strong></strong>Unfortunately,
 this setting does not allow Firefox to work with the video conferencing
 service in question, which relies entirely on WebRTC.<br></p><p>So I had to=
 go back in and change the media.peerconnection.enabled setting to <b>true</=
b>
 in Firefox in order to get it to work with the service in question.=20
While this enabled me to conference with teams that desire to use the=20
video conferencing service that rely wholly upon WebRTC, it concerns me=20
because of the security vulnerability.</p><p>One suggested mitigation was ha=
ving NoScript, but using NoScript was not helpful (you generally need to dis=
able it for the site you are viewing that relies upon WebRTC, in order to ob=
tain any functionality, and whether NoScript is off or on, WebRTC still caus=
es the vulnerability of leaking information as previously described).<br></p=
><p>Furthermore, upon reload of the browser, even after setting media.peerco=
nnection.enabled to <b>true</b>,
 the video conferencing service wouldn't work until I installed the=20
WebRTC 0.1.3 extension for Firefox and set it to expose my real IP=20
address as well as allowing the following:</p><p>a. navigator.getUserMedia<b=
r>b. window.MediaStreamTrack<br>c. window.RTCPeerConnection<br>d. window.RTC=
SessionDescription<br></p><p>That seemed to me to be a direct consequence of=
 the persistent and ongoing security vulnerability in WebRTC.<br></p><p>I
 contacted the video conferencing service provider and their solution=20
was simply to state that "If the peer to peer connection is of concern,=20
you can utilize our=20
premium version which will route traffic through a forwarding server=20
with in our environment that handles the processing of the video and=20
sound of all users in the conference and send it to each user=20
individually rather than using a peer to peer connection."&nbsp; In other=20=

words, they expect that people should have to pay in order to mitigate <a hr=
ef=3D"https://security.stackexchange.com/questions/82129/what-is-webrtc-and-=
how-can-i-protect-myself-against-leaking-information-that-doe" target=3D"_bl=
ank">this WebRTC security vulnerability</a>, because they are unwilling to d=
esign to protect all users from it.</p><p>In a similar fashion, in the secon=
d example of the browser based WebRTC version of a decentralized exchange, t=
he user's financial information is periodically exposed as a result of the i=
nformation leakage which is presently caused by WebRTC.&nbsp; (Fortunately, t=
his browser based WebRTC decentralized exchange is under development and is n=
ot currently open to actual usage.)<br></p><p>Although
 undoubtedly this has been discussed here before, I am asking that WebRTC se=
curity issues be
 reconsidered and that a new security review be done to ameliorate or=20
eliminate this problem in terms of WebRTC leaking this information.&nbsp; It=
 is my view that WebRTC should be further designed and developed with an eye=
 to mitigating this serious problem, so that users of services which might r=
ely partially or wholly upon WebRTC would not have to be concerned about suc=
h information leakage.<br></p><p>Respectfully,</p>Colin Gallagher<br><br></d=
iv>
<br></div></div>______________________________<wbr>_________________<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" t=
arget=3D"_blank">https://www.ietf.org/mailman/l<wbr>istinfo/rtcweb</a><br>
<br></blockquote></div><br></div>
</blockquote></div><br></div>
</div></div></blockquote></div></div></div><br></div></div>
</blockquote></div><br></div>
</div></blockquote><blockquote type=3D"cite"><div><span>____________________=
___________________________</span><br><span>rtcweb mailing list</span><br><s=
pan><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>=

--Apple-Mail-94BC0043-9133-4009-BD1E-EED78C1D5BFA--

