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From: Ted Hardie <ted.ietf@gmail.com>
Date: Tue, 9 Apr 2019 10:59:29 -0700
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Subject: [rtcweb] Security-arch IdP determination issue/DISCUSS
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--0000000000007551b605861cb835
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In section 7.5 of the Security-arch draft, the document says:

Authority:  The authority [RFC3986
<https://tools.ietf.org/html/rfc3986>] at which the IdP's service is
      hosted.  Note that this may include a non-default port or a
      userinfo component, but neither will be available in a certificate
      verifying the site.

Benjamin Kaduk raised a DISCUSS on this:

I'm a bit unclear about how the port in the
> IdP URI's Authority (Section 7.5) would get
> discovered. If it can be remotely supplied,
> there may be risks in just trusting blindly
> whatever value is received.
>

Given that we discover this via a .well-known location which is meant to be
deterministic, I went looking for the more general .well-known advice on
this topic.  Turns out that the updated version in 578bis
<https://tools.ietf.org/html/draft-nottingham-rfc5785bis-09#page-2> has
this:

Typically, applications will use the default port
> for the given scheme; if an alternative port is
> used, it MUST be explicitly specified by the
> application in question.
>

Obviously, our doc predates 5785bis, but given the discuss and that advice,
I think the right thing to do here is to drop the ability to have a
non-default port or to specify an alternate port.

Is there anyone currently using this with a non-default port?

Any objections to dropping this or preferences for specifying an alternate
port?

regards,

Ted

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<div dir=3D"ltr"><div><span style=3D"font-family:arial,helvetica,sans-serif=
">I<span style=3D"font-family:arial,helvetica,sans-serif">n</span> section =
7.5 of the Security-arch draft, the document says:</span></div><div><br></d=
iv><div><pre class=3D"gmail-newpage">Authority:  The authority [<a href=3D"=
https://tools.ietf.org/html/rfc3986" title=3D"&quot;Uniform Resource Identi=
fier (URI): Generic Syntax&quot;">RFC3986</a>] at which the IdP&#39;s servi=
ce is
      hosted.  Note that this may include a non-default port or a
      userinfo component, but neither will be available in a certificate
      verifying the site.<br><br></pre><pre class=3D"gmail-newpage"><font f=
ace=3D"arial,helvetica,sans-serif">Benjamin Kaduk raised a DISCUSS on this:=
<br></font></pre><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px =
0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">I&#39;m =
a bit unclear about how the port in the <br>IdP URI&#39;s Authority (Sectio=
n 7.5) would get <br>discovered.  If it can be remotely supplied, <br>there=
 may be risks in just trusting blindly <br>whatever value is received.<br><=
/blockquote></div><div><br></div><div>Given that we discover this via a .we=
ll-known location which is meant to be deterministic, I went looking for th=
e more general .well-known advice on this topic.=C2=A0 Turns out that the u=
pdated version in <a href=3D"https://tools.ietf.org/html/draft-nottingham-r=
fc5785bis-09#page-2" target=3D"_blank">578bis</a> has this:</div><div><br><=
blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-l=
eft:1px solid rgb(204,204,204);padding-left:1ex">Typically, applications wi=
ll use the default port <br>for the given scheme; if an alternative port is=
 <br>used, it MUST be explicitly specified by the <br>application in questi=
on.<br></blockquote><pre class=3D"gmail-m_8568968848071411641gmail-ballot g=
mail-m_8568968848071411641gmail-pasted"><br></pre>Obviously, our doc predat=
es 5785bis, but given the discuss and that advice, I think the right thing =
to do here is to drop the ability to have a non-default port or to specify =
an alternate port.</div><div><br></div><div>Is there anyone currently using=
 this with a non-default port?</div><div><br></div><div>Any objections to d=
ropping this or preferences for specifying an alternate port?</div><div><br=
></div><div>regards,</div><div><br></div><div>Ted<br></div></div>

--0000000000007551b605861cb835--


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> On 9Apr, 2019, at 10:59, Ted Hardie <ted.ietf@gmail.com> wrote:
>=20
> Obviously, our doc predates 5785bis, but given the discuss and that =
advice, I think the right thing to do here is to drop the ability to =
have a non-default port or to specify an alternate port.
>=20
> Is there anyone currently using this with a non-default port?
>=20
> Any objections to dropping this or preferences for specifying an =
alternate port?

I=E2=80=99m aware of small deployments using this. I don=E2=80=99t =
recall if they are using default ports.

But in general no objection against specifying an alternate port.

Best
  Nils Ohlmeier=


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This is a multi-part message in MIME format.
--------------E1AA337D5240D29D152BA485
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I have made a proposal.

Note: This is strictly a *personal* contribution. Do not attach undue
authority to it.


Harald



-------- Forwarded Message --------
Subject: 	New Version Notification for
draft-alvestrand-mmusic-simulcast-ssrc-00.txt
Date: 	Thu, 11 Apr 2019 03:28:17 -0700
From: 	internet-drafts@ietf.org
To: 	Harald Alvestrand <harald@alvestrand.no>




A new version of I-D, draft-alvestrand-mmusic-simulcast-ssrc-00.txt
has been successfully submitted by Harald Alvestrand and posted to the
IETF repository.

Name: draft-alvestrand-mmusic-simulcast-ssrc
Revision: 00
Title: Using SSRC with WebRTC Simulcast
Document date: 2019-04-11
Group: Individual Submission
Pages: 6
URL:
https://www.ietf.org/internet-drafts/draft-alvestrand-mmusic-simulcast-ssrc-00.txt
Status:
https://datatracker.ietf.org/doc/draft-alvestrand-mmusic-simulcast-ssrc/
Htmlized:
https://tools.ietf.org/html/draft-alvestrand-mmusic-simulcast-ssrc-00
Htmlized:
https://datatracker.ietf.org/doc/html/draft-alvestrand-mmusic-simulcast-ssrc


Abstract:
This document describes a convention for sending "a=ssrc" attributes
in SDP together with "a=simulcast" attributes. This allows SFUs that
need SSRC information to have this info easily accessible.

Given that it is intended as an interim measure, it does not aim for
being published as an RFC.



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

The IETF Secretariat


--------------E1AA337D5240D29D152BA485
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<html>
  <head>

    <meta http-equiv="content-type" content="text/html; charset=UTF-8">
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  <body text="#000000" bgcolor="#FFFFFF">
    <p>I have made a proposal.</p>
    <p>Note: This is strictly a *personal* contribution. Do not attach
      undue authority to it.</p>
    <p><br>
    </p>
    <p>Harald<br>
    </p>
    <div class="moz-forward-container"><br>
      <br>
      -------- Forwarded Message --------
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            <th valign="BASELINE" nowrap="nowrap" align="RIGHT">Subject:
            </th>
            <td>New Version Notification for
              draft-alvestrand-mmusic-simulcast-ssrc-00.txt</td>
          </tr>
          <tr>
            <th valign="BASELINE" nowrap="nowrap" align="RIGHT">Date: </th>
            <td>Thu, 11 Apr 2019 03:28:17 -0700</td>
          </tr>
          <tr>
            <th valign="BASELINE" nowrap="nowrap" align="RIGHT">From: </th>
            <td><a class="moz-txt-link-abbreviated" href="mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a></td>
          </tr>
          <tr>
            <th valign="BASELINE" nowrap="nowrap" align="RIGHT">To: </th>
            <td>Harald Alvestrand <a class="moz-txt-link-rfc2396E" href="mailto:harald@alvestrand.no">&lt;harald@alvestrand.no&gt;</a></td>
          </tr>
        </tbody>
      </table>
      <br>
      <br>
      <br>
      A new version of I-D,
      draft-alvestrand-mmusic-simulcast-ssrc-00.txt<br>
      has been successfully submitted by Harald Alvestrand and posted to
      the<br>
      IETF repository.<br>
      <br>
      Name: draft-alvestrand-mmusic-simulcast-ssrc<br>
      Revision: 00<br>
      Title: Using SSRC with WebRTC Simulcast<br>
      Document date: 2019-04-11<br>
      Group: Individual Submission<br>
      Pages: 6<br>
      URL:
<a class="moz-txt-link-freetext" href="https://www.ietf.org/internet-drafts/draft-alvestrand-mmusic-simulcast-ssrc-00.txt">https://www.ietf.org/internet-drafts/draft-alvestrand-mmusic-simulcast-ssrc-00.txt</a><br>
      Status:
      <a class="moz-txt-link-freetext" href="https://datatracker.ietf.org/doc/draft-alvestrand-mmusic-simulcast-ssrc/">https://datatracker.ietf.org/doc/draft-alvestrand-mmusic-simulcast-ssrc/</a><br>
      Htmlized:
      <a class="moz-txt-link-freetext" href="https://tools.ietf.org/html/draft-alvestrand-mmusic-simulcast-ssrc-00">https://tools.ietf.org/html/draft-alvestrand-mmusic-simulcast-ssrc-00</a><br>
      Htmlized:
<a class="moz-txt-link-freetext" href="https://datatracker.ietf.org/doc/html/draft-alvestrand-mmusic-simulcast-ssrc">https://datatracker.ietf.org/doc/html/draft-alvestrand-mmusic-simulcast-ssrc</a><br>
      <br>
      <br>
      Abstract:<br>
      This document describes a convention for sending "a=ssrc"
      attributes<br>
      in SDP together with "a=simulcast" attributes. This allows SFUs
      that<br>
      need SSRC information to have this info easily accessible.<br>
      <br>
      Given that it is intended as an interim measure, it does not aim
      for<br>
      being published as an RFC.<br>
      <br>
      <br>
      <br>
      Please note that it may take a couple of minutes from the time of
      submission<br>
      until the htmlized version and diff are available at
      tools.ietf.org.<br>
      <br>
      The IETF Secretariat<br>
      <br>
    </div>
  </body>
</html>

--------------E1AA337D5240D29D152BA485--


From nobody Thu Apr 11 07:45:09 2019
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From: Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com>
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Subject: Re: [rtcweb] Fwd: New Version Notification for draft-alvestrand-mmusic-simulcast-ssrc-00.txt
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Hi Harald,

Just one quick comment in case you find it useful.

I have been using this format for signaling both rids and ssrcs (and 
association between both):

a=simulcast:send c;b;a a=rid:c send ssrc=2564467881 a=rid:b send ssrc=1 
a=rid:a send ssrc=3 a=ssrc-group:FID 2564467881 32941417 
a=ssrc-group:FID 1 2 a=ssrc-group:FID 3 4 a=ssrc:2564467881 
cname:eRUkt2XxiSTlri77 a=ssrc:32941417 cname:eRUkt2XxiSTlri77 a=ssrc:1 
cname:eRUkt2XxiSTlri77 a=ssrc:2 cname:eRUkt2XxiSTlri77 a=ssrc:3 
cname:eRUkt2XxiSTlri77 a=ssrc:4 cname:eRUkt2XxiSTlri77

Note that the ssrc param (as a rid-param-other) in rid attribute is 
valid per the ABNF in:

https://tools.ietf.org/html/draft-ietf-mmusic-rid-15#page-19 
<https://tools.ietf.org/html/draft-ietf-mmusic-rid-15#page-18>

  rid-param = rid-width-param
                        / rid-height-param
                        / rid-fps-param
                        / rid-fs-param
                        / rid-br-param
                        / rid-pps-param
                        / rid-bpp-param
                        / rid-depend-param
                        / rid-param-other
  rid-param-other   = 1*(alpha-numeric / "-") [ "=" param-val ]
  param-val         = *( %x20-58 / %x60-7E ) ; Any printable character 
except semicolon


IMHO the usage of FID for signaling the "SIM" group is not a good choice 
as it is being used today for signaling the rtx association, and should 
be still present in the SDP if not using rids/rrids (which I understand 
that is the goal of the draft)

Best regards
Sergio

On 11/04/2019 12:30, Harald Alvestrand wrote:
>
> I have made a proposal.
>
> Note: This is strictly a *personal* contribution. Do not attach undue 
> authority to it.
>
>
> Harald
>
>
>
> -------- Forwarded Message --------
> Subject: 	New Version Notification for 
> draft-alvestrand-mmusic-simulcast-ssrc-00.txt
> Date: 	Thu, 11 Apr 2019 03:28:17 -0700
> From: 	internet-drafts@ietf.org
> To: 	Harald Alvestrand <harald@alvestrand.no>
>
>
>
>
> A new version of I-D, draft-alvestrand-mmusic-simulcast-ssrc-00.txt
> has been successfully submitted by Harald Alvestrand and posted to the
> IETF repository.
>
> Name: draft-alvestrand-mmusic-simulcast-ssrc
> Revision: 00
> Title: Using SSRC with WebRTC Simulcast
> Document date: 2019-04-11
> Group: Individual Submission
> Pages: 6
> URL: 
> https://www.ietf.org/internet-drafts/draft-alvestrand-mmusic-simulcast-ssrc-00.txt
> Status: 
> https://datatracker.ietf.org/doc/draft-alvestrand-mmusic-simulcast-ssrc/
> Htmlized: 
> https://tools.ietf.org/html/draft-alvestrand-mmusic-simulcast-ssrc-00
> Htmlized: 
> https://datatracker.ietf.org/doc/html/draft-alvestrand-mmusic-simulcast-ssrc
>
>
> Abstract:
> This document describes a convention for sending "a=ssrc" attributes
> in SDP together with "a=simulcast" attributes. This allows SFUs that
> need SSRC information to have this info easily accessible.
>
> Given that it is intended as an interim measure, it does not aim for
> being published as an RFC.
>
>
>
> Please note that it may take a couple of minutes from the time of 
> submission
> until the htmlized version and diff are available at tools.ietf.org.
>
> The IETF Secretariat
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb



--------------5F67EE26E71BF9CD44BF5712
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<html>
  <head>
    <meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
  </head>
  <body text="#000000" bgcolor="#FFFFFF">
    <div class="moz-cite-prefix">
      <div class="moz-cite-prefix">Hi Harald,</div>
      <div class="moz-cite-prefix"><br>
      </div>
      <div class="moz-cite-prefix">Just one quick comment in case you
        find it useful. <br>
      </div>
      <div class="moz-cite-prefix"><br>
      </div>
      <div class="moz-cite-prefix">I have been using this format for
        signaling both rids and ssrcs (and association between both):<br>
      </div>
      <div class="moz-cite-prefix">
        <pre style="color: rgb(0, 0, 0); font-style: normal; font-variant-ligatures: normal; font-variant-caps: normal; font-weight: 400; letter-spacing: normal; orphans: 2; text-align: start; text-indent: 0px; text-transform: none; widows: 2; word-spacing: 0px; -webkit-text-stroke-width: 0px; text-decoration-style: initial; text-decoration-color: initial;"><font face="Consolas">a=simulcast:send c;b;a
a=rid:c send ssrc=2564467881
a=rid:b send ssrc=1
a=rid:a send ssrc=3
a=ssrc-group:FID 2564467881 32941417
a=ssrc-group:FID 1 2
a=ssrc-group:FID 3 4
a=ssrc:2564467881 cname:eRUkt2XxiSTlri77
a=ssrc:32941417 cname:eRUkt2XxiSTlri77
a=ssrc:1 cname:eRUkt2XxiSTlri77
a=ssrc:2 cname:eRUkt2XxiSTlri77
a=ssrc:3 cname:eRUkt2XxiSTlri77
a=ssrc:4 cname:eRUkt2XxiSTlri77
 </font>
</pre>
      </div>
      <div class="moz-cite-prefix">
        <div class="moz-cite-prefix">Note that the ssrc param (as a
          rid-param-other) in rid attribute is valid per the ABNF in:</div>
        <div class="moz-cite-prefix"><br>
        </div>
      </div>
      <div class="moz-cite-prefix"><a
          href="https://tools.ietf.org/html/draft-ietf-mmusic-rid-15#page-18">https://tools.ietf.org/html/draft-ietf-mmusic-rid-15#page-19</a></div>
      <div class="moz-cite-prefix"><br>
      </div>
      <div class="moz-cite-prefix"><font face="Consolas"> rid-param        
          = rid-width-param<br>
                                 / rid-height-param<br>
                                 / rid-fps-param<br>
                                 / rid-fs-param<br>
                                 / rid-br-param<br>
                                 / rid-pps-param<br>
                                 / rid-bpp-param<br>
                                 / rid-depend-param<br>
                                 / rid-param-other<br>
           rid-param-other   = 1*(alpha-numeric / "-") [ "=" param-val ]<br>
           param-val         = *( %x20-58 / %x60-7E ) ; Any printable
          character except semicolon</font><br>
        <br>
      </div>
      <div class="moz-cite-prefix"><br>
      </div>
      <div class="moz-cite-prefix">IMHO the usage of FID for signaling
        the "SIM" group is not a good choice as it is being used today
        for signaling the rtx association, and should be still present
        in the SDP if not using rids/rrids (which I understand that is
        the goal of the draft)<br>
      </div>
      <div class="moz-cite-prefix"><br>
      </div>
      <div class="moz-cite-prefix">Best regards</div>
      <div class="moz-cite-prefix">Sergio</div>
      <div class="moz-cite-prefix"><br>
      </div>
    </div>
    <div class="moz-cite-prefix">On 11/04/2019 12:30, Harald Alvestrand
      wrote:<br>
    </div>
    <blockquote type="cite"
      cite="mid:a8795c0e-f506-7c6c-7941-2fe847eb472f@alvestrand.no">
      <p>I have made a proposal.</p>
      <p>Note: This is strictly a *personal* contribution. Do not attach
        undue authority to it.</p>
      <p><br>
      </p>
      <p>Harald<br>
      </p>
      <div class="moz-forward-container"><br>
        <br>
        -------- Forwarded Message --------
        <table class="moz-email-headers-table">
          <tbody>
            <tr>
              <th nowrap="nowrap">Subject: </th>
              <td>New Version Notification for
                draft-alvestrand-mmusic-simulcast-ssrc-00.txt</td>
            </tr>
            <tr>
              <th nowrap="nowrap">Date: </th>
              <td>Thu, 11 Apr 2019 03:28:17 -0700</td>
            </tr>
            <tr>
              <th nowrap="nowrap">From: </th>
              <td><a class="moz-txt-link-abbreviated"
                  href="mailto:internet-drafts@ietf.org"
                  moz-do-not-send="true">internet-drafts@ietf.org</a></td>
            </tr>
            <tr>
              <th nowrap="nowrap">To: </th>
              <td>Harald Alvestrand <a class="moz-txt-link-rfc2396E"
                  href="mailto:harald@alvestrand.no"
                  moz-do-not-send="true">&lt;harald@alvestrand.no&gt;</a></td>
            </tr>
          </tbody>
        </table>
        <br>
        <br>
        <br>
        A new version of I-D,
        draft-alvestrand-mmusic-simulcast-ssrc-00.txt<br>
        has been successfully submitted by Harald Alvestrand and posted
        to the<br>
        IETF repository.<br>
        <br>
        Name: draft-alvestrand-mmusic-simulcast-ssrc<br>
        Revision: 00<br>
        Title: Using SSRC with WebRTC Simulcast<br>
        Document date: 2019-04-11<br>
        Group: Individual Submission<br>
        Pages: 6<br>
        URL:
        <a class="moz-txt-link-freetext"
href="https://www.ietf.org/internet-drafts/draft-alvestrand-mmusic-simulcast-ssrc-00.txt"
          moz-do-not-send="true">https://www.ietf.org/internet-drafts/draft-alvestrand-mmusic-simulcast-ssrc-00.txt</a><br>
        Status: <a class="moz-txt-link-freetext"
href="https://datatracker.ietf.org/doc/draft-alvestrand-mmusic-simulcast-ssrc/"
          moz-do-not-send="true">https://datatracker.ietf.org/doc/draft-alvestrand-mmusic-simulcast-ssrc/</a><br>
        Htmlized: <a class="moz-txt-link-freetext"
href="https://tools.ietf.org/html/draft-alvestrand-mmusic-simulcast-ssrc-00"
          moz-do-not-send="true">https://tools.ietf.org/html/draft-alvestrand-mmusic-simulcast-ssrc-00</a><br>
        Htmlized:
        <a class="moz-txt-link-freetext"
href="https://datatracker.ietf.org/doc/html/draft-alvestrand-mmusic-simulcast-ssrc"
          moz-do-not-send="true">https://datatracker.ietf.org/doc/html/draft-alvestrand-mmusic-simulcast-ssrc</a><br>
        <br>
        <br>
        Abstract:<br>
        This document describes a convention for sending "a=ssrc"
        attributes<br>
        in SDP together with "a=simulcast" attributes. This allows SFUs
        that<br>
        need SSRC information to have this info easily accessible.<br>
        <br>
        Given that it is intended as an interim measure, it does not aim
        for<br>
        being published as an RFC.<br>
        <br>
        <br>
        <br>
        Please note that it may take a couple of minutes from the time
        of submission<br>
        until the htmlized version and diff are available at
        tools.ietf.org.<br>
        <br>
        The IETF Secretariat<br>
        <br>
      </div>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <pre class="moz-quote-pre" wrap="">_______________________________________________
rtcweb mailing list
<a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
    </blockquote>
    <p><br>
    </p>
  </body>
</html>

--------------5F67EE26E71BF9CD44BF5712--


From nobody Thu Apr 11 09:20:01 2019
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From: Lorenzo Miniero <lorenzo@meetecho.com>
To: Harald Alvestrand <harald@alvestrand.no>
Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>, "public-webrtc@w3.org" <public-webrtc@w3.org>
Message-ID: <20190411181951.729451c4@lminiero>
In-Reply-To: <a8795c0e-f506-7c6c-7941-2fe847eb472f@alvestrand.no>
References: <155497849785.12672.2988988646899076493.idtracker@ietfa.amsl.com> <a8795c0e-f506-7c6c-7941-2fe847eb472f@alvestrand.no>
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Subject: Re: [rtcweb] New Version Notification for draft-alvestrand-mmusic-simulcast-ssrc-00.txt
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Hi Harald,

why drop SSRC advertisement for rtx for simulcast? While I understand
rtx is still there if you use the repaired-rid extension, Chrome was
able to do both using different ssrc-groups. I realize SIM was not
standard, but it gave the right context, and FID provided the rtx
mappings instead. As such, as an interim solution, that sounded like an
easier approach. Anyway, the rest looks good to me!

Lorenzo


On Thu, 11 Apr 2019 12:30:13 +0200
Harald Alvestrand <harald@alvestrand.no> wrote:

> I have made a proposal.
> 
> Note: This is strictly a *personal* contribution. Do not attach undue
> authority to it.
> 
> 
> Harald
> 
> 
> 
> -------- Forwarded Message --------
> Subject: 	New Version Notification for
> draft-alvestrand-mmusic-simulcast-ssrc-00.txt
> Date: 	Thu, 11 Apr 2019 03:28:17 -0700
> From: 	internet-drafts@ietf.org
> To: 	Harald Alvestrand <harald@alvestrand.no>
> 
> 
> 
> 
> A new version of I-D, draft-alvestrand-mmusic-simulcast-ssrc-00.txt
> has been successfully submitted by Harald Alvestrand and posted to the
> IETF repository.
> 
> Name: draft-alvestrand-mmusic-simulcast-ssrc
> Revision: 00
> Title: Using SSRC with WebRTC Simulcast
> Document date: 2019-04-11
> Group: Individual Submission
> Pages: 6
> URL:
> https://www.ietf.org/internet-drafts/draft-alvestrand-mmusic-simulcast-ssrc-00.txt
> Status:
> https://datatracker.ietf.org/doc/draft-alvestrand-mmusic-simulcast-ssrc/
> Htmlized:
> https://tools.ietf.org/html/draft-alvestrand-mmusic-simulcast-ssrc-00
> Htmlized:
> https://datatracker.ietf.org/doc/html/draft-alvestrand-mmusic-simulcast-ssrc
> 
> 
> Abstract:
> This document describes a convention for sending "a=ssrc" attributes
> in SDP together with "a=simulcast" attributes. This allows SFUs that
> need SSRC information to have this info easily accessible.
> 
> Given that it is intended as an interim measure, it does not aim for
> being published as an RFC.
> 
> 
> 
> Please note that it may take a couple of minutes from the time of
> submission until the htmlized version and diff are available at
> tools.ietf.org.
> 
> The IETF Secretariat
> 



-- 
I'm getting older but, unlike whisky, I'm not getting any better
https://twitter.com/elminiero


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From: Sean Turner <sean@sn3rd.com>
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Date: Sun, 14 Apr 2019 13:02:33 -0400
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Subject: Re: [rtcweb] Benjamin Kaduk's Discuss on draft-ietf-rtcweb-security-11: (with DISCUSS and COMMENT)
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Didn=E2=80=99t get all the way through the comments, but I got a pretty =
start.  If there=E2=80=99s something else we need to discuss I can =
organize a time to get folks on a call.

> On Mar 6, 2019, at 14:08, Datatracker on behalf of Benjamin Kaduk =
<ietf-secretariat-reply@ietf.org> wrote:
>=20
> Benjamin Kaduk has entered the following ballot position for
> draft-ietf-rtcweb-security-11: Discuss
>=20
> When responding, please keep the subject line intact and reply to all
> email addresses included in the To and CC lines. (Feel free to cut =
this
> introductory paragraph, however.)
>=20
>=20
> Please refer to =
https://www.ietf.org/iesg/statement/discuss-criteria.html
> for more information about IESG DISCUSS and COMMENT positions.
>=20
>=20
> The document, along with other ballot positions, can be found here:
> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-security/
>=20
>=20
>=20
> ----------------------------------------------------------------------
> DISCUSS:
> ----------------------------------------------------------------------
>=20
> I'd like to have a brief discussion about a few points, though it's =
not clear that any change
> to the document will be required (details in the COMMENT section for =
all of these):
>=20
> Mutually-verifiable "secure mode" seems to require that the peer's =
browser be included in
> the TCB, which is a bit hard to swallow.  Are we comfortable wrapping =
that in alongside
> "we trust the peer to not be malicious"?
>=20
> It's not clear how much benefit we can get from *optional* third-party =
identity providers;
> won't the calling service have the ability to silently downgrade to =
their non-usage even if
> both calling peers support it?

Looks like you and Adam worked it out and now we need to come up with =
some text.  A PR to address this is here:

https://github.com/rtcweb-wg/security/pull/17

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

I filled a "comments PR=E2=80=9D here:
https://github.com/rtcweb-wg/security/pull/19

> I mostly only have editorial comments, though there are a few that are
> more content-ful.
>=20
> Section 1
>=20
>                                                                     As
>   with any Web application, the Web server can move logic between the
>   server and JavaScript in the browser, but regardless of where the
>   code is executing, it is ultimately under control of the server.
>=20
> The user can observe the javascript running the browser, though maybe
> this distinction is not necessary here.

I re-read this a couple of times and the only thing I could add is what =
is probably very obvious, i.e., the user is in control of the end-point =
so it can see anything.  I am too am not sure that=E2=80=99s really =
worth pointing out here.

> Section 3
>=20
>                                                               Huang et
>   al. [huang-w2sp] summarize the core browser security guarantee as:
>=20
>      Users can safely visit arbitrary web sites and execute scripts
>      provided by those sites.
>=20
> I note that the author of this document is listed as a coauthor on
> huang-w2sp; does the self-cite really add much authority to the
> summary of the guarantee?

Eh ... I can see where maybe not much would be added if there was just =
one author and the paper hadn=E2=80=99t been accepted by a conference, =
but there are five and it was accepted.

> The use of ALL-CAPS to call out new terms feels a bit dated.

Haha - I can fix TCB, WEB ATTACKERS, and NETWORK ATTACKERS :). Fixed in =
this =E2=80=9Ccomments PR=E2=80=9D.

>                                                                   Note
>   that for non-HTTPS traffic, a network attacker is also a Web
>   attacker, since it can inject traffic as if it were any non-HTTPS =
Web
>   site.  Thus, when analyzing HTTP connections, we must assume that
>   traffic is going to the attacker.
>=20
> nit: I know this is a web-centric document, but the privileging of =
https
> as the only "secure" traffic reads a bit oddly to me; something like
> "note that in some cases, a network attacker is also a web attacker,
> since transport protocols that do not provide integrity protection =
allow
> the network to inject traffic as if they were any communications peer.
> TLS, and HTTPS in particular, prevent against these attacks, but when
> analyzing HTTP connections, we must assume that traffic is going to =
the
> attacker."  (A thought experiment might be to consider whether wss://
> traffic counts as "HTTPS traffic=E2=80=9D.)

It=E2=80=99s include the comment PR noted above.

> Section 3.1
>=20
> It might be appropriate to provide some example references in place of
> "extensive research=E2=80=9D.

So there=E2=80=99s lots and I=E2=80=99m not even really sure where to =
start here, but we do reference [cranor-wolf] later so I=E2=80=99ll add =
that here.

> Section 4.1
>=20
>                                                In either case, all the
>   browser is able to do is verify and check authorization for whoever
>   is controlling where the media goes.  [...]
>=20
> nit: the wording here is a bit odd, since in case (1) you're verifying
> you're talking to A, but you still control where the media goes (in
> terms of A or not-A; A can of course then forward on the media =
further).

Maybe =E2=80=9CIn both cases, ..."

> 00000000000000000000000000000000000000000000000000000000000000000000By
>   contrast, consent to send network traffic is about preventing the
>   user's browser from being used to attack its local network.  [...]
>=20
> nit: "local" is perhaps overly restricting, depending on =
interpretation

I can see striking =E2=80=9Clocal=E2=80=9D.

> Section 4.1.1
>=20
> Maybe note that the "result" of the cross-site requests that is leaked
> is in the form of pixels and not structured data, but that does not
> change the information content.

Something like:=20

  A more sophisticated attack would be open up a
  source view window to a site and use the screen sharing result to
  simply view anti cross-site request forgery tokens which would leak =
pixels
  and not structured text.

> Section 4.1.3
>=20
>   Now that we have seen another use case, we can start to reason about
>=20
> nit: I'm confused by "another" here.

I think this should be:

  Now that we have the calling scenarios, we can start to reason about =
..

This links this para back to the previous calling scenarios and user =
expectations.

>                                              While not suitable for =
all
>   cases, this approach may be useful for some.  If we consider the =
case
>   of advertising, it's not particularly convenient to require the
>   advertiser to instantiate an iframe on the hosting site just to get
>   permission; a more convenient approach is to cryptographically tie
>   the advertiser's certificate to the communication directly.  We're
>=20
> This seems to be relying on the reader to have some background =
knowledge
> and make some leaps of reasoning that may not be reasonable to expect.

Okay so -04 included a section that was referred to by this section that =
addressed the =E2=80=9CCalling to an AD Target=E2=80=9D, but that =
section was removed because the WG decided not treat the case =
=E2=80=9Cspecially=E2=80=9D.  So yeah there was some more background =
provided like 6 years ago.  I have to admit I=E2=80=99m not entirely =
sure I know how to address this comment.

>   Another case where media-level cryptographic identity makes sense is
>   when a user really does not trust the calling site.  For instance, I
>   might be worried that the calling service will attempt to bug my
>   computer, but I also want to be able to conveniently call my =
friends.
>=20
> This is especially challenging because if the site (and/or its
> javascript) is in the path for binding a cryptographic identity to a
> real-world identity, then a malicious site can still get whatever keys
> it wants authorized.

yep

> Section 4.1.4
>=20
>   3.  The attacker forges the response apparently http://calling-
>       service.example.com/ to inject JS to initiate a call to himself.
>=20
> seem to be missing a word or two here.

I think r/apparently/from
fixed in comments PR

>   which contain untrusted content.  If a page from a given origin ever
>   loads JavaScript from an attacker, then it is possible for that
>   attacker to infect the browser's notion of that origin semi-
>   permanently.
>=20
> nit: "If any page" is more emphatic, I think.

Yep fixed in comments PR.

> Section 4.2
>=20
> Do we want any discussion of the risks when metered bandwidth (pay per
> byte) is in use?

Do you mean concerns that the connection will be closed if your bill =
isn=E2=80=99t paid up? :)

> Section 4.2.1
>=20
> There's probably some room to tighten up the verbiage here; e.g., "the
> site initiating ICE" is referring to a website that is using a browser
> API to request ICE against some remote peer (right?).

Yes, but that=E2=80=99s the whole point of the intro, this is a browser =
focused world that has a JavaScript API.

>  And "ICE
> keepalives are indications" is using Indication as the technical term
> for a message that doesn't get an ACK response, not in its common
> English usage.

Yes, but those are the actual terms used by ICE, which is referred to in =
the first sentence in the paragraph.

> Section 4.2.2
>=20
> A one- or two-sentence summary of the impact of misinterpretation
> attacks is probably in order, instead of making us follow the =
reference
> (which isn't a section reference).
>=20
>   Where TCP is used the risk is substantial due to the potential
>   presence of transparent proxies and therefore if TCP is to be used,
>   then WebSockets style masking MUST be employed.
>=20
> nit: "employed" to obfuscate what, exactly?
>=20
> Section 4.2.3
>=20
>   refuses to send other traffic until that message has been replied =
to.
>   The message/reply pair must be generated in such a way that an
>   attacker who controls the Web application cannot forge them,
>   generally by having the message contain some secret value that must
>   be incorporated (e.g., echoed, hashed into, etc.).  Non-ICE
>=20
> nit: "incorporated" into what?
>=20
> I think I'm a little confused about which legacy actors we're talking
> about.  Are we still considering the broader situation a
> webserver-mediated interaction between two browsers or brower-adjacent
> applications?  (E.g., a WebRTC client calling some other sort of video
> chat system?)
>=20
>   leaves.  The appropriate technologies here are fairly similar to
>   those for initial consent, though are perhaps weaker since the
>   threats is less severe.
>=20
> nit: "threat is=E2=80=9D

It got switched to =E2=80=9Care=E2=80=9D as a result of some other =
review.

> Section 4.2.4
>=20
>   Note that as soon as the callee sends their ICE candidates, the
>   caller learns the callee's IP addresses.  The callee's server
>   reflexive address reveals a lot of information about the callee's
>   location.  In order to avoid tracking, implementations may wish to
>   suppress the start of ICE negotiation until the callee has answered.
>=20
> Is "answered" supposed to be some interaction with the controlling =
site?
>=20
>   In ordinary operation, the site learns the browser's IP address,
>   though it may be hidden via mechanisms like Tor
>   [http://www.torproject.org] or a VPN.  However, because sites can
>   cause the browser to provide IP addresses, this provides a mechanism
>   for sites to learn about the user's network environment even if the
>   user is behind a VPN that masks their IP address.  [...]
>=20
> Some rewording for clarity is probably in order; "ordinary operation" =
is of
> a website without WebRTC; "sites can cause the browser to provide IP
> addresses" is when the site uses the browser API to request ICE
> initiation; etc.
>=20
> Section 4.3.1
>=20
> [Obligatory note about "Forward Secrecy" vs. "Perfect Forward =
Secrecy"]
>=20
>   to subsequent compromise.  It is this consideration that makes an
>   automatic, public key-based key exchange mechanism imperative for
>   WebRTC (this is a good idea for any communications security system)
>   and this mechanism SHOULD provide perfect forward secrecy (PFS).  =
The
>   signaling channel/calling service can be used to authenticate this
>   mechanism.
>=20
> To be clear, the authentication that the calling service provides is a
> binding between identity and the public keys that are input to the key
> exchange mechanism?
>=20
> Section 4.3.2.1
>=20
>                                                       Even if the user
>   actually checks the other side's name (which all available evidence
>   indicates is unlikely), this would require (a) the browser to =
trusted
>   UI to provide the name and (b) the user to not be fooled by similar
>   appearing names.
>=20
> nit: "browser to use trusted UI=E2=80=9D

Fixed in the comments PR.

> Section 4.3.2.3
>=20
> It's not clear that third-party identity providers actually provide
> downgrade-resistance -- can't the site mediating the calls just =
decline
> to acknowledge that a third-party identity is/was available for the
> peer?
>=20
> Section 4.3.2.4
>=20
>                                    I.e., I must be able to verify that
>   the person I am calling has engaged a secure media mode (see
>   Section 4.3.3).  In order to achieve this it will be necessary to
>   cryptographically bind an indication of the local media access =
policy
>   into the cryptographic authentication procedures detailed in the
>   previous sections.
>=20
> This seems to require extending the TCB from just the local browser to
> the remote browser as well, which is ... a stretch.
> (Also, do we really need the first person?)

Dropped the first person (I also hope the RFC editor) can help with this =
elsewhere.

See the comments PR.

> Section 9.2
>=20
> The coordinates for [OpenID] don't seem quite right.

The title was off.  Fixed in the comments PR.


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From: Sean Turner <sean@sn3rd.com>
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Subject: Re: [rtcweb] Benjamin Kaduk's Discuss on draft-ietf-rtcweb-security-arch-18: (with DISCUSS and COMMENT)
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I didn=E2=80=99t get all the way through the comments, but I got a =
pretty start.  If there=E2=80=99s something else we need to discuss I =
can organize a time to get folks on a call.

> On Mar 6, 2019, at 21:30, Datatracker on behalf of Benjamin Kaduk =
<noreply@ietf.org> wrote:
>=20
> Benjamin Kaduk has entered the following ballot position for
> draft-ietf-rtcweb-security-arch-18: Discuss
>=20
> When responding, please keep the subject line intact and reply to all
> email addresses included in the To and CC lines. (Feel free to cut =
this
> introductory paragraph, however.)
>=20
>=20
> Please refer to =
https://www.ietf.org/iesg/statement/discuss-criteria.html
> for more information about IESG DISCUSS and COMMENT positions.
>=20
>=20
> The document, along with other ballot positions, can be found here:
> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-security-arch/
>=20
>=20
>=20
> ----------------------------------------------------------------------
> DISCUSS:
> ----------------------------------------------------------------------

I filed an issue here for these:
https://github.com/rtcweb-wg/security-arch/issues/89

> The question of when an IdP is trustworthy, whether at all (even for
> "authoritative" IdPs) or for a specific user, is a pretty core topic =
for
> the identity assertion scheme presented here.  These topics do get
> explained in localized sections of the document, but there seem to be
> other portions of the text that do not really acknowledge the risks.
> I've tried to note these in the COMMENT section (though having =
finished
> reading the document, perhaps I am overzealous about determination =
that
> an authoritative IdP is indeed authoritative).

Thanks for the parenthetical ;). We=E2=80=99ll pick through them and see =
what we can do.

> I also think that we need to be more careful about having the IdP know
> the semantics of what it's signing (or otherwise attesting to), so =
that
> it does not turn into a signing oracle, etc..

I am trying to figure out what we could include to address this.  I have =
to tell you my first thought is that the IdP is just doing its job, =
i.e., it signs assertions that an authenticated party wants signed.

> The "Modifying the Session" treatment for the SDP "identity" attribute
> seems incompletely specified.

I am hoping you might expand here a bit on what you think might be =
needed.  We had the SDP directorate review and they (Christer) gave it =
the thumbs up.

> I'm a bit unclear about how the port in the IdP URI's Authority =
(Section
> 7.5) would get discovered.  If it can be remotely supplied, there may =
be
> risks in just trusting blindly whatever value is received.

Ted started a thread on this here:
https://mailarchive.ietf.org/arch/msg/rtcweb/8dtCkSeXWUsb_y2SRyFlbQ5r93w
We=E2=80=99ll see where it lands.

> It seems like there are some unstated privacy considerations in =
allowing
> the IdP proxy to automatically generate an assertion (that reveals the
> user's identity) at the request of javascript from the calling
> application, as described in Section 7.7.

I=E2=80=99m struggling to think what more might be said because the AP =
is asking the IdP to vouch for its identity so the IdP is literally =
doing what the AP asked it to do.

> Section 9.4 talks about how the IdP is attesting to the binding of the
> user identified in the assertion with the key fingerprints, but in
> Sections 7.4 and 7.6 we claim that this assertion is "opaque to the
> IdP"; these statements appear to be in conflict with each other.

I think that all three sections are consistent wrt to the IdP binding =
the user identified in the assertion with the key fingerprints.  Section =
7.4 and 7.6 are just saying that the IdP doesn=E2=80=99t need to know =
what it is signing.

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

I filed an issue for this here:
https://github.com/rtcweb-wg/security-arch/issues/90

> Section 3
>=20
> (My comment about TCB and the other browser from the companion =
document
> is probably relevant here, too.)

I am hoping the text that we put in the -security draft address this =
concern.

> Section 4.1
>=20
>   This message is sent to the signaling server, e.g., by =
XMLHttpRequest
>   [XmlHttpRequest] or by WebSockets [RFC6455], preferably over TLS
>   [RFC5246].  The signaling server processes the message from Alice's
>=20
> This is the optimistic "best security" case, and we already say we're
> talking to the signaling server over HTTPS, so it should be safe to =
just
> say "over TLS" and drop the "preferably".  (Also, s/5246/8446.)

Sure I=E2=80=99ll drop preferably.  Fixed in the =E2=80=9Ccomment" PR:
https://github.com/rtcweb-wg/security-arch/pull/91

>   call and to Alice's identity.  In this case, Alice has provided an
>   identity assertion and so Bob's browser contacts Alice's identity
>   provider (again, this is done in a generic way so the browser has no
>   specific knowledge of the IdP) to verify the assertion.  This allows
>   the browser to display a trusted element in the browser chrome
>   indicating that a call is coming in from Alice.  [...]
>=20
> I think I'm confused.  We're displaying trusted browser chrome based =
on
> an assertion from some IdP that we have no relationship with and no
> reason to trust?

Don=E2=80=99t forget this part from earlier in the Section:

  They are both connected to the
  calling service via HTTPS and so know the origin with some level of
  confidence.  They also have accounts with some identity provider.

> Section 4.3
>=20
>   Once the ICE checks have completed [more specifically, once some ICE
>   checks have completed], [...]
>=20
> nit: that's not really more specific.  Maybe "Once the requisite ICE
> checks have completed=E2=80=9D?

Sounds good.  Fixed in the comments PR.

> Section 5
>=20
> I see that the 4566 <base64> includes the pad characters, though
> sometimes we will mention explicitly whether they are or are not
> included.
>=20
>   Note that long lines in the example are folded to meet the column
>   width constraints of this document; the backslash ("\") at the end =
of
>   a line and the carriage return that follows shall be ignored.
>=20
> leading whitespace, too, right?

Yes :). Fixed in the comments PR.

> Section 5.1
>=20
>   This section defines the SDP Offer/Answer [RFC6454] considerations
>   for the SDP 'identity' attribute.
>=20
> 6454 is "the Web Origin Concept"; presumably this is supposed to be
> 4566 (or 3264?).

good catch.  Fixed in the comments PR

> Section 5.1.3
>=20
> I feel like we need some text here about the (non?)trustworthiness of
> the IdP.
>=20
> Section 5.1.4
>=20
> I'm a bit confused at what's going on here.  Is "MAY send the same"
> supposed to prevent changing it?  If I don't send it, does that =
identity
> continue to apply to the existing DTLS connections but not any new =
ones
> generated by the session modification?  Am I allowed to send a =
different
> one?
>=20
>                  Note that [I-D.ietf-rtcweb-jsep], Section 5.2.1
>   requires that each media section use the same set of fingerprints =
for
>   every media section.
>=20
> nit: is this "each media section"/"every media section" redundant?
>=20
> Section 6.1
>=20
>   Also note that the security architecture depends on the keying
>   material not being available to move between origins.  But, it is
>   assumed that the identity assertion can be passed to anyone that the
>   page cares to.
>=20
> There may be some (weak) privacy considerations if this is literally
> anyone, since it would allow some observers (with weird
> abilities/restrictions) to associate "real" identities with keys in a
> way that they couldn't otherwise do.
>=20
> Section 6.2
>=20
>   Because HTTP origins cannot be securely established against network
>   attackers, implementations MUST NOT allow the setting of permanent
>   access permissions for HTTP origins.  Implementations MUST refuse =
all
>   permissions grants for HTTP origins.
>=20
> Just to check: this last sentence applies for one-time requets, too?
>=20
>                                           The semantics of this =
request
>   are that the media stream from the camera and microphone will only =
be
>   routed through a connection which has been cryptographically =
verified
>   (through the IdP mechanism or an X.509 certificate in the DTLS-SRTP
>   handshake) as being associated with the stated identity.  [...]
>=20
> Does this need to be an exhaustive list or can we leave it open-ended?
> Also, it may be appropriate to mention some concept of "IdP trusted to
> authenticate the stated identity".
>=20
>   API Requirement:  The API MUST provide a mechanism for the =
requesting
>      JS to relinquish the ability to see or modify the media (e.g., =
via
>      MediaStream.record()).  [...]
>=20
> Do we need to say anything about that state transition being visible =
to
> the peer, here?
>=20
>   UI Requirement:  If the UI indication of camera/microphone use are
>      [...]
>      camera and microphone input when the indication is hidden.  =
[Note:
>      this may not be necessary in systems that are non-windows-based
>      but that have good notifications support, such as phones.]
>=20
> nit: s/windows/window/?
>=20
>   Clients MAY permit the formation of data channels without any direct
>   user approval.  Because sites can always tunnel data through the
>   server, further restrictions on the data channel do not provide any
>   additional security.  (though see Section 6.3 for a related issue).
>=20
> Is there anything to say about why clients might not opt to do so (and
> what such approval might look like)?
>=20
> (My comments about "verified user" including the IdP in some way will
> apply here as well.)
>=20
> Section 6.3
>=20
>   While continuing consent is required, the ICE [RFC8445]; Section 10
>   keepalives use STUN Binding Indications which are one-way and
>   therefore not sufficient.  The current WG consensus is to use ICE
>=20
> Is the "the current WG consensus" language going to age well?
>=20
>   Binding Requests for continuing consent freshness.  ICE already
>   requires that implementations respond to such requests, so this
>   approach is maximally compatible.  A separate document will profile
>   the ICE timers to be used; see [RFC7675].
>=20
> Is there a WIP draft for this separate document?
>=20
> Section 6.4
>=20
>   API Requirement:  The API MUST provide a mechanism to allow the JS =
to
>      suppress ICE negotiation (though perhaps to allow candidate
>      gathering) until the user has decided to answer the call [note:
>      determining when the call has been answered is a question for the
>      JS.]  This enables a user to prevent a peer from learning their =
IP
>      address if they elect not to answer a call and also from learning
>      whether the user is online.
>=20
> nit: maybe make it more clear that this only applies for incoming =
calls?
>=20
> Section 6.5
>=20
>                                                           Media =
traffic
>   MUST NOT be sent over plain (unencrypted) RTP or RTCP; that is,
>   implementations MUST NOT negotiate cipher suites with NULL =
encryption
>   modes.  [...]
>=20
> It's not clear to me that the "that is" reflects a strict equivalence;
> would "in particular" be more appropriate?
> (Also, "cipher suite" is a DTLS term, but do we want to disambiguate
> explicitly?)
>=20
> [obligatory "Perfect Forward Secrecy" vs. "Forward Secrecy" note]
>=20
>   Implementations MUST NOT implement DTLS renegotiation and MUST =
reject
>   it with a "no_renegotiation" alert if offered.
>=20
> "MUST NOT implement" isn't really something that 2119 language can
> enforce; "MUST NOT use" is the best we can get.
>=20
>   Endpoints MUST NOT implement TLS False Start [RFC7918].
>=20
> (7918 doesn't claim to be applicable to DTLS anyway)
>=20
>=20
>   API Requirement:  Unless the user specifically configures an =
external
>      key pair, different key pairs MUST be used for each origin.  =
(This
>      avoids creating a super-cookie.)
>=20
> nit: might be appropriate to note why we care about a super-cookie =
(and
> what it is)
>=20
>      *  The "security characteristics" MUST indicate the cryptographic
>         algorithms in use (For example: "AES-CBC".)  However, if Null
>         ciphers are used, that MUST be presented to the user at the
>         top-level UI.
>=20
> I'm not sure I see anywhere that we allow the usage of null ciphers.
>=20
> Section 7
>=20
>   Recently, a number of Web-based identity technologies (OAuth,
>   Facebook Connect etc.) have been developed.  While the details vary,
>   what these technologies share is that they have a Web-based (i.e.,
>   HTTP/HTTPS) identity provider which attests to your identity.  For
>   instance, if I have an account at example.org, I could use the
>   example.org identity provider to prove to others that I was
>   alice@example.org.  [...]
>=20
> I agree with Alissa that the first person is not needed here.
>=20
> Section 7.1
>=20
>   Third-Party:   IdPs which don't have control of their section of the
>      [...]
>      identity space.  Probably the best-known example of a third-party
>      identity provider is SSL/TLS certificates, where there are a =
large
>      number of CAs all of whom can attest to any domain name.
>=20
> This probably needs some qualifier, given recent developments with CAA
> and similar mechanisms.
>=20
>   If an AP is authenticating via an authoritative IdP, then the RP =
does
>   not need to explicitly configure trust in the IdP at all.  The
>=20
> The RP still needs to establish somehow that the IdP in use is in fact
> an authoritative IdP, though!
>=20
> Section 7.2
>=20
>   In order to provide security without trusting the calling site, the
>   PeerConnection component of the browser must interact directly with
>   the IdP.  The details of the mechanism are described in the W3C API
>   specification, but the general idea is that the PeerConnection
>=20
> A reference to that W3C API spec might be handy.
>=20
> Section 7.3
>=20
>   There are two parts to this work:
>=20
>   o  The precise information from the signaling message that must be
>      cryptographically bound to the user's identity and a mechanism =
for
>      carrying assertions in JSEP messages.  This is specified in
>      Section 7.4.
>=20
> nit: the grammar is a bit weird here, as the "information from the
> signaling message" isn't really a part of this work, but rather the
> specification for what information that is.
>=20
> Section 7.4
>=20
> The indentation of the line with "}, {" is a bit confusing.

I tweaked it a bit in the comments PR.

>   This object is encoded in a JSON [RFC8259] string for passing to the
>   IdP.  The identity assertion returned by the IdP, which is encoded =
in
>=20
> I'm a little confused what this "encoded in a JSON string" is supposed
> to mean.
>=20
>   This structure does not need to be interpreted by the IdP or the IdP
>   proxy.  It is consumed solely by the RP's browser.  The IdP merely
>   treats it as an opaque value to be attested to.  Thus, new =
parameters
>   can be added to the assertion without modifying the IdP.
>=20
> The IdP probably wants to know enough about its structure to not turn
> into a signing oracle for other protocols, though.

It=E2=80=99s as specified in the webrtc-api :)

> Section 7.4.1
>=20
> (RFC 8259 JSON inherently is UTF-8, so maybe we don't need to mention
> that.)

True but it doesn=E2=80=99t really hurt to leave it.

> It's a little surprising to see sha-1 fingerprint in use (since
> "examples are recommendations"), though I didn't find anything that
> would actually formally deprecate such usage yet.

Can somebody please generate a SHA-256 example for me?

>   Note that long lines in the example are folded to meet the column
>   width constraints of this document; the backslash ("\") at the end =
of
>   a line and the carriage return that follows shall be ignored.
>=20
> leading whitespace, too, right?

Yes :) Added in the comments PR.

> Section 7.5.2
>=20
> (Still need to say how it's know than authoritative assertions are in
> fact authoritative for what they claim.)
>=20
> Section 7.6
>=20
>   The input to identity assertion is the JSON-encoded object described
>   in Section 7.4 that contains the set of certificate fingerprints the
>   browser intends to use.  This string is treated as opaque from the
>   perspective of the IdP.
>=20
> (IdP still doesn't want to become a signing oracle.)
>=20
>   For use in signaling, the assertion is serialized into JSON,
>   Base64-encoded [RFC4648], and used as the value of the "identity"
>   attribute.
>=20
> nit: it's unclear that "serialized into JSON" adds any value, since =
the
> thing is defined to be a JSON object.
>=20
> Section 7.7
>=20
> I think that the framing of HTTP Basic (7617) here is not great.
> RFC 7235 might be a better link for HTTP Authentication in general, =
and
> of course there are mechanisms that don't include sending the password
> in plaintext, like SCRAM (RFC7804).
>=20
> Section 8
>=20
>   The IdP proxy verifies the assertion.  Depending on the identity
>   protocol, the proxy might contact the IdP server or other servers.
>   For instance, an OAuth-based protocol will likely require using the
>   IdP as an oracle, whereas with a signature-based scheme might be =
able
>   to verify the assertion without contacting the IdP, provided that it
>   has cached the relevant public key.
>=20
> IMPORTANT: Do we need a freshness property for the assertion?  Some of
> these schemes do not provide freshness.
>=20
>   Figure 6 shows an example response formatted as JSON for =
illustrative
>   purposes.
>=20
> (Doesn't the W3C API spec need to say how the response is formatted?  =
Is
> the JSON formatting actually "illustrative" then, or is this just an
> example output?)

he webrtc-api spec says its JSON so:

 Figure 6 shows an example response, which is JSON-formatted.

> Section 8.1
>=20
>   2.  If the domain portion of the string is not equal to the domain
>       name of the IdP proxy, then the PeerConnection object MUST =
reject
>       the assertion unless:
>=20
> Reading closely, I think this is supposed to be "unless either", but
> it's easy to assume it should be read as "unless both", so I think
> clarification is in order.

since it=E2=80=99s =E2=80=9Cand=E2=80=9D after the first sub-bullet =
I=E2=80=99ll add =E2=80=9Cboth=E2=80=9D.

>   Any "@" or "%" characters in the "user" portion of the identity MUST
>   be escaped according to the "Percent-Encoding" rules defined in
>=20
> We just said in the first paragraph that "user" has "any character
> except '@'", so this is a bit redundant.

I dropped =E2=80=9Cexcept =E2=80=98@=E2=80=98 from the 1st para.  =
Somebody will tell me if I did this wrong, but there is some kind of x1

> Section 9.1
>=20
>            Users who wish to assure themselves of security against a
>   malicious identity provider can only do so by verifying peer
>   credentials directly, e.g., by checking the peer's fingerprint
>   against a value delivered out of band.
>=20
> I suppose an "untrustworthy" IdP is basically a malicious one, though
> there are perhaps some subtleties that could be distinguished here.
>=20
>   In order to protect against malicious content JavaScript, that
>   JavaScript MUST NOT be allowed to have direct access to---or perform
>   computations with---DTLS keys.  For instance, if content JS were =
able
>   to compute digital signatures, then it would be possible for content
>   JS to get an identity assertion for a browser's generated key and
>   then use that assertion plus a signature by the key to authenticate =
a
>   call protected under an ephemeral Diffie-Hellman (DH) key controlled
>   by the content JS, thus violating the security guarantees otherwise
>   provided by the IdP mechanism.
>=20
> I don't think I fully understand the scenario described in this last
> sentence.  Is "compute digital signatures" supposed to be with some
> specific secret key, and/or is "a browser's generated key" one that is
> covered under the fingerprint in the IdP assertion?
>=20
> Section 9.2
>=20
>   Otherwise, the other side will learn linkable information.
>=20
> nit: "linkable information that would allow them to correlate the
> browser across multiple calls=E2=80=9D.

Added in the comments PR.

> Section 9.3
>=20
>   Consider the case of a call center which accepts calls via WebRTC.
>   An attacker proxies the call center's front-end and arranges for
>   multiple clients to initiate calls to the call center.  Note that
>   this requires user consent in many cases but because the data =
channel
>   does not need consent, he can use that directly.
>=20
> I think I'm missing a step here.  How is the attacker using the data
> channel directly when the point is to get the multiple browsers to =
send
> the data on the data channel?
>=20
>              Muxing multiple media flows over a single transport makes
>   it harder to individually suppress a single flow by denying ICE
>   keepalives.  Either media-level (RTCP) mechanisms must be used or =
the
>   implementation must deny responses entirely, thus terminating the
>   call.
>=20
> nit: "must be used to suppress the misbehaving flow", I think.
>=20
> Section 9.4.3
>=20
>   The "origin" field of the signature request can be used to check =
that
>   the user has agreed to disclose their identity to the calling site;
>   because it is supplied by the PeerConnection it can be trusted to be
>   correct.
>=20
> I don't see an "origin" field in the signature request; is this =
supposed
> to be the "domain=E2=80=9D?

Changed it to =E2=80=9Cdomain=E2=80=9D to match the structure.  Also, =
instead of "signature request=E2=80=9D I change it to =E2=80=9Cassertion =
request=E2=80=9D to match the earlier heading title.

> Section 9.4.5.1
>=20
> nit: it might be friendlier to the reader to prefix this with "When
> popup blocking is in use, =E2=80=9C.

Added in comments PR.

> Section 13.2
>=20
> It's perhaps debatable that JSEP is only an informative reference.

Moved it to normative in the comments PR.


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Date: Thu, 18 Apr 2019 18:38:14 -0600
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I agree this change makes sense. It is so easy to get domain names and =
certs and SNI is so widely adopted that I do not see many arguments =
against this.=20

> On Apr 9, 2019, at 11:59 AM, Ted Hardie <ted.ietf@gmail.com> wrote:
>=20
> In section 7.5 of the Security-arch draft, the document says:
>=20
> Authority:  The authority [RFC3986 =
<https://tools.ietf.org/html/rfc3986>] at which the IdP's service is
>       hosted.  Note that this may include a non-default port or a
>       userinfo component, but neither will be available in a =
certificate
>       verifying the site.
>=20
> Benjamin Kaduk raised a DISCUSS on this:
> I'm a bit unclear about how the port in the=20
> IdP URI's Authority (Section 7.5) would get=20
> discovered. If it can be remotely supplied,=20
> there may be risks in just trusting blindly=20
> whatever value is received.
>=20
> Given that we discover this via a .well-known location which is meant =
to be deterministic, I went looking for the more general .well-known =
advice on this topic.  Turns out that the updated version in 578bis =
<https://tools.ietf.org/html/draft-nottingham-rfc5785bis-09#page-2> has =
this:
>=20
> Typically, applications will use the default port=20
> for the given scheme; if an alternative port is=20
> used, it MUST be explicitly specified by the=20
> application in question.
>=20
> Obviously, our doc predates 5785bis, but given the discuss and that =
advice, I think the right thing to do here is to drop the ability to =
have a non-default port or to specify an alternate port.
>=20
> Is there anyone currently using this with a non-default port?
>=20
> Any objections to dropping this or preferences for specifying an =
alternate port?
>=20
> regards,
>=20
> Ted
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


--Apple-Mail=_FE14113F-78A8-4296-92F0-9F4BB8809E13
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	charset=us-ascii

<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; line-break: after-white-space;" class=3D""><div =
class=3D""><br class=3D""></div>I agree this change makes sense. It is =
so easy to get domain names and certs and SNI is so widely adopted that =
I do not see many arguments against this.&nbsp;<br class=3D""><div><br =
class=3D""><blockquote type=3D"cite" class=3D""><div class=3D"">On Apr =
9, 2019, at 11:59 AM, Ted Hardie &lt;<a href=3D"mailto:ted.ietf@gmail.com"=
 class=3D"">ted.ietf@gmail.com</a>&gt; wrote:</div><br =
class=3D"Apple-interchange-newline"><div class=3D""><div dir=3D"ltr" =
class=3D""><div class=3D""><span =
style=3D"font-family:arial,helvetica,sans-serif" class=3D"">I<span =
style=3D"font-family:arial,helvetica,sans-serif" class=3D"">n</span> =
section 7.5 of the Security-arch draft, the document =
says:</span></div><div class=3D""><br class=3D""></div><div =
class=3D""><pre class=3D"gmail-newpage">Authority:  The authority [<a =
href=3D"https://tools.ietf.org/html/rfc3986" title=3D"&quot;Uniform =
Resource Identifier (URI): Generic Syntax&quot;" class=3D"">RFC3986</a>] =
at which the IdP's service is
      hosted.  Note that this may include a non-default port or a
      userinfo component, but neither will be available in a certificate
      verifying the site.<br class=3D""><br class=3D""></pre><pre =
class=3D"gmail-newpage"><font face=3D"arial,helvetica,sans-serif" =
class=3D"">Benjamin Kaduk raised a DISCUSS on this:<br =
class=3D""></font></pre><blockquote class=3D"gmail_quote" =
style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid =
rgb(204,204,204);padding-left:1ex">I'm a bit unclear about how the port =
in the <br class=3D"">IdP URI's Authority (Section 7.5) would get <br =
class=3D"">discovered.  If it can be remotely supplied, <br =
class=3D"">there may be risks in just trusting blindly <br =
class=3D"">whatever value is received.<br =
class=3D""></blockquote></div><div class=3D""><br class=3D""></div><div =
class=3D"">Given that we discover this via a .well-known location which =
is meant to be deterministic, I went looking for the more general =
.well-known advice on this topic.&nbsp; Turns out that the updated =
version in <a =
href=3D"https://tools.ietf.org/html/draft-nottingham-rfc5785bis-09#page-2"=
 target=3D"_blank" class=3D"">578bis</a> has this:</div><div =
class=3D""><br class=3D""><blockquote class=3D"gmail_quote" =
style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid =
rgb(204,204,204);padding-left:1ex">Typically, applications will use the =
default port <br class=3D"">for the given scheme; if an alternative port =
is <br class=3D"">used, it MUST be explicitly specified by the <br =
class=3D"">application in question.<br class=3D""></blockquote><pre =
class=3D"gmail-m_8568968848071411641gmail-ballot =
gmail-m_8568968848071411641gmail-pasted"><br class=3D""></pre>Obviously, =
our doc predates 5785bis, but given the discuss and that advice, I think =
the right thing to do here is to drop the ability to have a non-default =
port or to specify an alternate port.</div><div class=3D""><br =
class=3D""></div><div class=3D"">Is there anyone currently using this =
with a non-default port?</div><div class=3D""><br class=3D""></div><div =
class=3D"">Any objections to dropping this or preferences for specifying =
an alternate port?</div><div class=3D""><br class=3D""></div><div =
class=3D"">regards,</div><div class=3D""><br class=3D""></div><div =
class=3D"">Ted<br class=3D""></div></div>
_______________________________________________<br class=3D"">rtcweb =
mailing list<br class=3D""><a href=3D"mailto:rtcweb@ietf.org" =
class=3D"">rtcweb@ietf.org</a><br =
class=3D"">https://www.ietf.org/mailman/listinfo/rtcweb<br =
class=3D""></div></blockquote></div><br class=3D""></body></html>=

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Folks

Please take a look at the proposed update below and let the authors and 
MMUSIC chairs know if you have any concerns with this change (by April 29).

Thanks

-- Flemming (as MMUSIC co-chair)

On 4/21/19 7:12 PM, Christer Holmberg wrote:
>
> Hi,
>
> I am cross-posting this on multiple list, to make people aware of the 
> issue. However, I ask that any comments and discussions take place on 
> the MMUSIC list only.
>
> I found the following statement in section 4.5 of 
> draft-ietf-mmusic-sctp-sdp:
>
>    “NOTE: 'webrtc-datachannel' indicates the WebRTC Data Channel
>
>    Establishment Protocol defined in [I-D.ietf-rtcweb-data-protocol].”
>
> In addition, in the IANA registration of ‘webrtc-datachannel’ (section 
> 15.3), draft-ietf-rtcweb-data-protocol is referenced.
>
> To refresh everyone, draft-ietf-rtcweb-data-protocol defines DCEP (the 
> inband protocol for handling WebRTC Data Channels), and 
> ‘webrtc-datachannel’ is used in the SDP m= line when establishing an 
> SCTP association for such channel.
>
> However, both draft-ietf-mmusic-datachannel-sdp-neg and 
> draft-ietf-clue-data-channel use ‘webrtc-datachannel’ in the “m=” 
> line, and those drafts do not use DCEP.
>
> My suggestion to fix this is to modify the text in 
> draft-ietf-mmusic-sctp-sdp, to always use ‘webrtc-datachannel’ for 
> WebRTC Data Channels – no matter whether DCEP is used or not.
>
> I have created a pull request:
>
> https://github.com/cdh4u/draft-sctp-sdp/pull/15
>
> Regards,
>
> Christer
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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    Folks<br>
    <br>
    Please take a look at the proposed update below and let the authors
    and MMUSIC chairs know if you have any concerns with this change (by
    April 29). <br>
    <br>
    Thanks <br>
    <br>
    -- Flemming (as MMUSIC co-chair)<br>
    <br>
    <div class="moz-cite-prefix">On 4/21/19 7:12 PM, Christer Holmberg
      wrote:<br>
    </div>
    <blockquote type="cite"
      cite="mid:D591F3E7-EBC8-4844-A0FA-6ED64DD2B5F6@ericsson.com">
      <meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
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        <p class="MsoNormal"><span style="font-size:11.0pt">Hi,<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size:11.0pt"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span style="font-size:11.0pt" lang="EN-US">I
            am cross-posting this on multiple list, to make people aware
            of the issue. However, I ask that any comments and
            discussions take place on the MMUSIC list only.<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size:11.0pt" lang="EN-US"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span style="font-size:11.0pt" lang="EN-US">I
            found the following statement in section 4.5 of
            draft-ietf-mmusic-sctp-sdp:<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size:11.0pt" lang="EN-US"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;Courier
            New&quot;;color:black" lang="EN-US">   “NOTE:
            'webrtc-datachannel' indicates the WebRTC Data Channel<o:p></o:p></span></p>
        <p class="MsoNormal"><span
            style="font-size:10.0pt;font-family:&quot;Courier
            New&quot;;color:black" lang="EN-US">   Establishment
            Protocol defined in [I-D.ietf-rtcweb-data-protocol].”<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size:11.0pt" lang="EN-US"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span style="font-size:11.0pt" lang="EN-US">In
            addition, in the IANA registration of ‘webrtc-datachannel’
            (section 15.3), draft-ietf-rtcweb-data-protocol is
            referenced.<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size:11.0pt" lang="EN-US"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span style="font-size:11.0pt" lang="EN-US">To
            refresh everyone, draft-ietf-rtcweb-data-protocol defines
            DCEP (the inband protocol for handling WebRTC Data
            Channels), and ‘webrtc-datachannel’ is used in the SDP m=
            line when establishing an SCTP association for such channel.<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size:11.0pt" lang="EN-US"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span style="font-size:11.0pt" lang="EN-US">However,
            both draft-ietf-mmusic-datachannel-sdp-neg and
            draft-ietf-clue-data-channel use ‘webrtc-datachannel’ in the
            “m=” line, and those drafts do not use DCEP.<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size:11.0pt" lang="EN-US"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span style="font-size:11.0pt" lang="EN-US">My
            suggestion to fix this is to modify the text in
            draft-ietf-mmusic-sctp-sdp, to always use
            ‘webrtc-datachannel’ for WebRTC Data Channels – no matter
            whether DCEP is used or not.<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size:11.0pt" lang="EN-US"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span style="font-size:11.0pt" lang="EN-US">I
            have created a pull request:<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size:11.0pt" lang="EN-US"><o:p> </o:p></span></p>
        <p class="MsoNormal"><a
            href="https://github.com/cdh4u/draft-sctp-sdp/pull/15"
            moz-do-not-send="true"><span lang="EN-US">https://github.com/cdh4u/draft-sctp-sdp/pull/15</span></a><span
            lang="EN-US"><o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size:11.0pt" lang="EN-US"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span style="font-size:11.0pt" lang="EN-US">Regards,<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size:11.0pt" lang="EN-US"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span style="font-size:11.0pt" lang="EN-US">Christer<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size:11.0pt" lang="EN-US"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span style="font-size:11.0pt" lang="EN-US"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span style="font-size:11.0pt" lang="EN-US"><o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size:11.0pt" lang="EN-US"><o:p> </o:p></span></p>
      </div>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <pre class="moz-quote-pre" wrap="">_______________________________________________
rtcweb mailing list
<a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
    </blockquote>
    <br>
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From: "Martin Thomson" <mt@lowentropy.net>
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Subject: Re: [rtcweb] Security-arch IdP determination issue/DISCUSS
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On Wed, Apr 10, 2019, at 04:00, Ted Hardie wrote:
> In section 7.5 of the Security-arch draft, the document says:
> 
> Authority:  The authority [RFC3986 
> <https://tools.ietf.org/html/rfc3986>] at which the IdP's service is
>       hosted.  Note that this may include a non-default port or a
>       userinfo component, but neither will be available in a certificate
>       verifying the site.
> 
> Benjamin Kaduk raised a DISCUSS on this:
> > I'm a bit unclear about how the port in the 
> > IdP URI's Authority (Section 7.5) would get 
> > discovered. If it can be remotely supplied, 
> > there may be risks in just trusting blindly 
> > whatever value is received.
> 
> Given that we discover this via a .well-known location which is meant 
> to be deterministic, I went looking for the more general .well-known 
> advice on this topic. Turns out that the updated version in 578bis 
> <https://tools.ietf.org/html/draft-nottingham-rfc5785bis-09#page-2> has 
> this:
> 
> > Typically, applications will use the default port 
> > for the given scheme; if an alternative port is 
> > used, it MUST be explicitly specified by the 
> > application in question.
> 
> Obviously, our doc predates 5785bis, but given the discuss and that 
> advice, I think the right thing to do here is to drop the ability to 
> have a non-default port or to specify an alternate port.
> 
> Is there anyone currently using this with a non-default port?
> 
> Any objections to dropping this or preferences for specifying an alternate port?

I hope that this also includes dropping the userinfo part, so that this is just the host portion (https://tools.ietf.org/html/rfc3986#section-3.2.2).

Firefox does support the port number, but I'm not aware of anyone using it.  As Cullen points out, getting a unique name is not the biggest barrier to deployment.  You might even argue that there is a security benefit in that arbitrary ports can be used to assert identities.


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Reviewer: Nancy Cam-Winget
Review result: Has Nits

SECDIR review of draft-ietf-rtcweb-security-11

Reviewer: Nancy Cam-Winget
Review result: Ready with Nits

I have reviewed this document as part of the security directorate'sÊ
ongoing effort to review all IETF documents being processed by theÊ
IESG.ÊÊThese comments were written primarily for the benefit of theÊ
security area directors.ÊÊDocument editors and WG chairs should treatÊ
these comments just like any other last call comments.

This document defines the threat model and security analysis of WebRTC.
There are a few normative clauses to suggest requirements, I think the document
could benefit of having more normative requirements to make it a stronger
standards track document. Otherwise, it reads more as an informational document.

The following are general comments and suggestions (and editorial nits at the
end):

Section 3. Should it also be noted that as it (browser) has no purview to the
actual application running, attacks from the application layer can still occur
but is not in scope for WebRTC?

Section 4.1
- The conceptual model is a bit confusing, as I think Entity can refer to both
the webRTC server as well as the receiving client application?  The notion is
that the User is trusting the webRTC (client) application (Entity A) to send
the media to another user, but really to the receiving application (Entity B). 
Perhaps model 1 has a typo where the Òtalk toÓ should be Entity B?

- The paragraph following the conceptual models is a bit awkward.  If I
understand correctly, the intent is to state that the user believes Entity A
when attempting to call Entity B, Entity A (the client app) could in fact also
send it to Entity C?  Which is valid, but the writing is awkward as the AÕs,
BÕs and CÕs are not explicitly stated as such or at least from the end userÕs
viewpoint.

- ÒIn either case, all the browser is able to do is verify and check
authorization for whoever is controlling where the media goesÓ Is ÒwhoeverÓ
meant to be the webRTC application controlling media paths?  That should be
clarified

- ÒÉconsent to access local devices is largely orthogonal to consent to
transmitÉÓ It should be both consent to local devices and the deviceÕs
resources, unless you mean Òlocal devicesÓ as in camera and microphone; but I
think processes and stored or cached files (e.g. other resources) need to be
considered too. § - Òconsent to send network traffic is about preventing the
        userÕs browser from being used to attack its local networkÓ
This is true, though, I think you are inferring that the network traffic is
enforced to be encrypted, which IÕm not sure is always the case; so I would
think it is up to the browser to ensure that this is the case unless the
browserÕs policy has made an exception

Section 4.1.2.1  ÒÉbug my computerÓ should be clarified.  I think there are at
least two dimensions to the ÒbugÓ, being it has ÒfreeÓ access to my resources
and it can actually potentially ÒlistenÓ to my calls; it could also potentially
override the use of my resources. The last sentence on the paragraph ÒNote that
question of consentÉ.Ó, the note makes sense, but I am not sure how the last
clause ÒÉ.the site is not listening inÓ, can you clarify this?

Section 4.1.2.2 ÒÉthe need for a second consentÓ, perhaps it is the need for a
ÒdistinctÓ consent as there may not be a previous consent?  By ÒdistinctÓ I
mean that it is a different type of consent than what may have been granted to
the car manuf (in your example) from getting my geolocationÉ.or perhaps I
missed the ÒfirstÓ consent type.

The last sentence of the paragraph is difficult to parse.  I think it is
asserting a requirement that the GUI used to launch the call must show the call
status (active, continuing, stopped)?

This section eludes to granting ÒcallÓ access only for the duration of the call
and the access should be limited (Òjust because I want some information on a
car doesnÕt meanÉ.Ó); the attack vector should be better highlighted.  As I
also believe that the browser can only grant the full client application
access, so for the duration of the call, it can very well be that the app can
get access to my resources beyond just the call (audio only vs. audio + videoÉ.)

Section 4.1.3
- The countermeasures can also be combined (e.g. consent is only given to a
given user for each call, as well as also having the appropriate keying
material). It is subtly eluded to in the 2nd to last paragraph but doesnÕt
consider that it can be done for all calls.

Section 4.1.4
- It should also be noted that weaknesses in the HTTPS stack can also be
exploited (weak authentication or key establishment use) by an attackerÉ.and
perhaps should be a MUST enforce strong mutual authentication and key
management.

Section 4.2.3
- As IÕm unfamiliar to ICE/STUN, IÕm not sure what checks are referred to in
ÒÉunsafe to completely remove the requirement for some checkÓ, this should be
clarified. Not sure (in the succeeding sentence) if there is a forward
reference to proposed checks or if they are listed elsewhere?

Section 4.3.1:  since the draft is listing requirements, ÒÉ.exchange mechanism
imperative forÉÓ
 The ÒimperativeÓ should be normative ÒMUSTÓ?

Section 4.3.2: who is the Òremote endpointÓ?

Editorial nits:
Section 1. It may be useful to describe why it is Òimmediately apparentÓ that
new security challenges resultÉ..suggest remove ÒimmediatelyÓ.

Section 4.1.1 (last sentence in the 2nd to last paragraph) is missing a toÕ:
Òsophisticated attack would be open up aÓ

Section 4.1.3 : ÒNow that we have seen another use caseÓ  Seems odd or
superfluous.  Suggest to just remove that clause or ÒWith the aforementioned
use cases, we can startÉ.Ó

Section 4.2.1: as it is a first reference, title should call out ÒInteractive
Connectivity Establishment (ICE)Ó, same for STUN (and STUNÕs reference, RFC
5389) should be called out.

Section 4.2.2 SRTP as a first reference should have its full reference to match
the acronym and TURN TCP should have reference too

Section 4.2.3 RTCP needs a reference?

Section 4.3: is it Òa problem from the SIP worldÓ or Òa problem familiar to the
SIP worldÓ? - Extra is Òcalling service is is non-maliciousÓ

Section 4.3.1: ÒinÓ should be removed ÒÉ.if end-to-end keying is in usedÓ

Section 4.3.2.1: ÒOne natural approach is to use É.Ó, natural approach to what?
 I think it is to mitigate the during-call attack only?


