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Subject: Re: [rtcweb] Sending of zero-length messages over data channels
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On 07/24/2014 10:25 PM, Makaraju, Maridi Raju (Raju) wrote:
>
> */In conclusion, I think websocket API supports zero length because 
> websocket protocol supports zero-length data frames. So, it does not 
> have the issue of 'bufferedAmount' discrepancy. /*
>
> *//*
>
> Yes, that is what I meant.
>
> */<Raju>/*
>
> */Ok, good. Websocket is not sending any dummy bytes, so 
> /**/bufferedAmount is always accurate./*
>
> */I am still not convinced that to keep data channel API consistent 
> with websocket API we need to introduce of dummy byte!?/*
>
> */How about keeping consistency of what goes on wire vs. what user 
> sent? websocket preserves that, where as data channel won't. /*
>

The WebSocket protocol is not on-the-wire compatible with the 
DataChannel protocol.
This has not been a design goal, and I don't see any reason to make it one.

> *//*
>
> */IMHO opinion, sending such a dummy byte may have other consequences:/*
>
> */1./**/App is not aware of what's going on? send() probably meant to 
> be called with non-empty data, but some error leg could have caused it 
> become empty, but still a byte will be sent out!!/*
>
> */2./**/Debugging becomes bit more complex and not intuitive. WebRTC 
> is already complex, why add more complexity?/*
>
> */3./**/Interop issues. The more the exceptions the more the interop 
> issues you see as some implementations may overlook these or implement 
> them later./*
>
> *//*
>
> */It's usually a bad idea to send a dummy byte on wire, without the 
> application direct involvement./*
>

I recommend spending some time with Wireshark looking at actual traces 
before making any such statement. Many protocols have dummy bytes, for 
many different reasons.

> *//*
>
> */IMHO, if doing PPID_OOB is more effort and can't fit in webrtc 1.0 
> then it would be better to delay and do it in more proper way than 
> taking a shortcut with PPID_EMPTY. With PPID_OOB approach underlying 
> signaling protocol can negotiate OOB before using it; though such 
> negotiation is not mandatory, but this approach allows such possibility./*
>

/*PPID_OOB would make the compatibility problem worse, not better.*/
>
> *//*
>
> *//*
>
> */I then, rest my case! /**/J/**//*
>
> *//*
>
> */</Raju>/*
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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    <div class="moz-cite-prefix">On 07/24/2014 10:25 PM, Makaraju,
      Maridi Raju (Raju) wrote:<br>
    </div>
    <blockquote
cite="mid:E1FE4C082A89A246A11D7F32A95A17828E4C61A5@US70UWXCHMBA02.zam.alcatel-lucent.com"
      type="cite">
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                      1.5pt;padding:0in 0in 0in 4.0pt">
                      <div>
                        <div>
                          <div>
                            <div>
                              <p class="MsoNormal"
                                style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><b><i><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">In
                                      conclusion, I think websocket API
                                      supports zero length because
                                      websocket protocol supports
                                      zero-length data frames. So, it
                                      does not have the issue of
                                      &#8216;bufferedAmount&#8217; discrepancy.
                                    </span></i></b><o:p></o:p></p>
                              <p class="MsoNormal"
                                style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><b><i><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</span></i></b><o:p></o:p></p>
                            </div>
                          </div>
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                    </div>
                  </div>
                </div>
                <div>
                  <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
                </div>
                <div>
                  <p class="MsoNormal">Yes, that is what I meant.&nbsp;<o:p></o:p></p>
                  <p class="MsoNormal"><b><i><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&lt;Raju&gt;<o:p></o:p></span></i></b></p>
                  <p class="MsoNormal"><b><i><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Ok,
                          good. Websocket is not sending any dummy
                          bytes, so
                        </span></i></b><b><i><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">bufferedAmount
                          is always accurate.<o:p></o:p></span></i></b></p>
                  <p class="MsoNormal"><b><i><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">I
                          am still not convinced that to keep data
                          channel API consistent with websocket API we
                          need to introduce of dummy byte!?<o:p></o:p></span></i></b></p>
                  <p class="MsoNormal"><b><i><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">How
                          about keeping consistency of what goes on wire
                          vs. what user sent? websocket preserves that,
                          where as data channel won&#8217;t.
                        </span></i></b></p>
                </div>
              </div>
            </div>
          </div>
        </div>
      </div>
    </blockquote>
    <br>
    The WebSocket protocol is not on-the-wire compatible with the
    DataChannel protocol.<br>
    This has not been a design goal, and I don't see any reason to make
    it one.<br>
    <br>
    <blockquote
cite="mid:E1FE4C082A89A246A11D7F32A95A17828E4C61A5@US70UWXCHMBA02.zam.alcatel-lucent.com"
      type="cite">
      <div class="WordSection1">
        <div style="border:none;border-left:solid blue 1.5pt;padding:0in
          0in 0in 4.0pt">
          <div>
            <div>
              <div>
                <div>
                  <p class="MsoNormal"><b><i><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p></o:p></span></i></b></p>
                  <p class="MsoNormal"><b><i><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">IMHO
                          opinion, sending such a dummy byte may have
                          other consequences:<o:p></o:p></span></i></b></p>
                  <p class="MsoListParagraph"
                    style="text-indent:-.25in;mso-list:l0 level1 lfo1"><!--[if !supportLists]--><b><i><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><span
                            style="mso-list:Ignore">1.<span
                              style="font:7.0pt &quot;Times New
                              Roman&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
                            </span></span></span></i></b><!--[endif]--><b><i><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">App
                          is not aware of what&#8217;s going on? send()
                          probably meant to be called with non-empty
                          data, but some error leg could have caused it
                          become empty, but still a byte will be sent
                          out!!<o:p></o:p></span></i></b></p>
                  <p class="MsoListParagraph"
                    style="text-indent:-.25in;mso-list:l0 level1 lfo1"><!--[if !supportLists]--><b><i><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><span
                            style="mso-list:Ignore">2.<span
                              style="font:7.0pt &quot;Times New
                              Roman&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
                            </span></span></span></i></b><!--[endif]--><b><i><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Debugging
                          becomes bit more complex and not intuitive.
                          WebRTC is already complex, why add more
                          complexity?<o:p></o:p></span></i></b></p>
                  <p class="MsoListParagraph"
                    style="text-indent:-.25in;mso-list:l0 level1 lfo1"><!--[if !supportLists]--><b><i><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><span
                            style="mso-list:Ignore">3.<span
                              style="font:7.0pt &quot;Times New
                              Roman&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
                            </span></span></span></i></b><!--[endif]--><b><i><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Interop
                          issues. The more the exceptions the more the
                          interop issues you see as some implementations
                          may overlook these or implement them later.<o:p></o:p></span></i></b></p>
                  <p class="MsoNormal"><b><i><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span></i></b></p>
                  <p class="MsoNormal"><b><i><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">It&#8217;s
                          usually a bad idea to send a dummy byte on
                          wire, without the application direct
                          involvement.</span></i></b></p>
                </div>
              </div>
            </div>
          </div>
        </div>
      </div>
    </blockquote>
    <br>
    I recommend spending some time with Wireshark looking at actual
    traces before making any such statement. Many protocols have dummy
    bytes, for many different reasons.<br>
    <br>
    <blockquote
cite="mid:E1FE4C082A89A246A11D7F32A95A17828E4C61A5@US70UWXCHMBA02.zam.alcatel-lucent.com"
      type="cite">
      <div class="WordSection1">
        <div style="border:none;border-left:solid blue 1.5pt;padding:0in
          0in 0in 4.0pt">
          <div>
            <div>
              <div>
                <div>
                  <p class="MsoNormal"><b><i><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p></o:p></span></i></b></p>
                  <p class="MsoNormal"><b><i><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">IMHO,
                          if doing PPID_OOB is more effort and can&#8217;t fit
                          in webrtc 1.0 then it would be better to delay
                          and do it in more proper way than taking a
                          shortcut with PPID_EMPTY. With PPID_OOB
                          approach underlying signaling protocol can
                          negotiate OOB before using it; though such
                          negotiation is not mandatory, but this
                          approach allows such possibility.</span></i></b></p>
                </div>
              </div>
            </div>
          </div>
        </div>
      </div>
    </blockquote>
    <br>
    <i><b>PPID_OOB would make the compatibility problem worse, not
        better.</b></i><br>
    <blockquote
cite="mid:E1FE4C082A89A246A11D7F32A95A17828E4C61A5@US70UWXCHMBA02.zam.alcatel-lucent.com"
      type="cite">
      <div class="WordSection1">
        <div style="border:none;border-left:solid blue 1.5pt;padding:0in
          0in 0in 4.0pt">
          <div>
            <div>
              <div>
                <div>
                  <p class="MsoNormal"><b><i><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p></o:p></span></i></b></p>
                  <p class="MsoNormal"><b><i><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span></i></b></p>
                  <p class="MsoNormal"><b><i><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">I
                          then, rest my case!
                        </span></i></b><b><i><span
                          style="font-size:11.0pt;font-family:Wingdings;color:#1F497D">J</span></i></b><b><i><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p></o:p></span></i></b></p>
                  <p class="MsoNormal"><b><i><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span></i></b></p>
                  <p class="MsoNormal"><b><i><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&lt;/Raju&gt;</span></i></b><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p></o:p></span></p>
                </div>
              </div>
            </div>
          </div>
        </div>
      </div>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_______________________________________________
rtcweb mailing list
<a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
    </blockquote>
    <br>
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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Sending of zero-length messages over data channels
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Date: Mon, 4 Aug 2014 18:08:55 +0000
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Subject: Re: [rtcweb] Sending of zero-length messages over data channels
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>IMHO opinion, sending such a dummy byte may have other consequences:

1.      >App is not aware of what's going on? send() probably meant to be c=
alled >with non-empty data, but some error leg could have caused it become =
>empty, but still a byte will be sent out!!

2.      >Debugging becomes bit more complex and not intuitive. WebRTC is >a=
lready complex, why add more complexity?

3.      >Interop issues. The more the exceptions the more the interop issue=
s you >see as some implementations may overlook these or implement them >la=
ter.

>It's usually a bad idea to send a dummy byte on wire, without the applicat=
ion >direct involvement.

>I recommend spending some time with Wireshark looking at actual traces >be=
fore making any such statement. Many protocols have dummy bytes, for >many =
different reasons.

<Raju2>
I am afraid there is some confusion about my comment and as a result we may=
 be talking about dummy bytes at 2 different levels: app level and underlyi=
ng protocol level.
I am aware of many protocols using reserved bits, bytes, padded bits/bytes,=
 protocol-level heartbeats/PING-PONGs (SCTP heartbeats use of opaque data).=
 In all these cases the mechanism is built into the protocol below app-leve=
l protocol itself with no direct involvement of application-level protocol.=
 My comment was in the context of application-level protocol.
If app calls send() 3 times with data "abc", "" (empty/null data with size =
0) and then "xyz" then on the wire in the application-level payload you see=
 "abc" <dummy byte> "xyz".
So, can you please point to me a protocol where this kind of dummy byte(s) =
are inserted in between application payload?

Btw, If you leave the app choice to send the byte (using OOB_PPID) then the=
re is a possibility for the app to send 256 distinct values instead of brow=
ser hard-coding the byte to zero always, which just conveys a single value.

</Raju2>


>PPID_OOB would make the compatibility problem worse, not better.
<Raju2>
I would appreciate if you can provide reasons on why PPID_OOB making compat=
ibility worse?
With PPID_OOB, apps have a choice to negotiate (this negotiation is like an=
y other negotiation for webrtc.) and use this option. PPID_EMPTY option see=
ms to be enabled always and no way to turn-off.
Without an option to turn off PPID_EMPTY, you will get into more compatibil=
ity issues as some webrtc implementations may not implement it on day one; =
a native client could be using an interim version of a popular webrtc lib; =
or it might be using its own webrtc lib, in which case we can't expect it t=
o implement all these later-adders in a timely manner.

</Raju2>

BR
Raju


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<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></=
span></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></=
span></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&gt;IMHO opinion, s=
ending such a dummy byte may have other consequences:</span></i></b><o:p></=
o:p></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-.25in;mso-list:l0 level=
1 lfo2"><![if !supportLists]><span style=3D"mso-list:Ignore">1.<span style=
=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]><b><i><span style=3D"font-size:11.0pt;font-family:&=
quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&gt;App is not awa=
re of what&#8217;s going on? send() probably meant to be called &gt;with no=
n-empty data, but some error leg could have caused it become &gt;empty,
 but still a byte will be sent out!!</span></i></b><o:p></o:p></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-.25in;mso-list:l0 level=
1 lfo2"><![if !supportLists]><span style=3D"mso-list:Ignore">2.<span style=
=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]><b><i><span style=3D"font-size:11.0pt;font-family:&=
quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&gt;Debugging beco=
mes bit more complex and not intuitive. WebRTC is &gt;already complex, why =
add more complexity?</span></i></b><o:p></o:p></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-.25in;mso-list:l0 level=
1 lfo2"><![if !supportLists]><span style=3D"mso-list:Ignore">3.<span style=
=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]><b><i><span style=3D"font-size:11.0pt;font-family:&=
quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&gt;Interop issues=
. The more the exceptions the more the interop issues you &gt;see as some i=
mplementations may overlook these or implement them &gt;later.</span></i></=
b><o:p></o:p></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</span></i></=
b><o:p></o:p></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&gt;It&#8217;s usua=
lly a bad idea to send a dummy byte on wire, without the application &gt;di=
rect involvement.</span></i></b><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"margin-left:5.25pt"><br>
&gt;I recommend spending some time with Wireshark looking at actual traces =
&gt;before making any such statement. Many protocols have dummy bytes, for =
&gt;many different reasons.<br>
<br>
<o:p></o:p></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&lt;Raju2&gt;
<o:p></o:p></span></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">I am afraid there i=
s some confusion about my comment and as a result we may be talking about d=
ummy bytes at 2 different levels: app level and underlying
 protocol level.<o:p></o:p></span></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">I am aware of many =
protocols using reserved bits, bytes, padded bits/bytes, protocol-level hea=
rtbeats/PING-PONGs (SCTP heartbeats use of opaque data).
 In all these cases the mechanism is built into the protocol below app-leve=
l protocol itself with no direct involvement of application-level protocol.=
 My comment was in the context of application-level protocol.
<o:p></o:p></span></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">If app calls send()=
 3 times with data &#8220;abc&#8221;, &#8220;&#8221; (empty/null data with =
size 0) and then &#8220;xyz&#8221; then on the wire in the application-leve=
l payload you see
 &#8220;abc&#8221; &lt;dummy byte&gt; &#8220;xyz&#8221;.<o:p></o:p></span><=
/i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">So, can you please =
point to me a protocol where this kind of dummy byte(s) are inserted in bet=
ween application payload?<o:p></o:p></span></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></=
span></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Btw, If you leave t=
he app choice to send the byte (using OOB_PPID) then there is a possibility=
 for the app to send 256 distinct values instead of browser
 hard-coding the byte to zero always, which just conveys a single value.<o:=
p></o:p></span></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></=
span></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&lt;/Raju2&gt;</spa=
n></i></b><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&=
quot;sans-serif&quot;;color:#1F497D"><o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><br>
<b><i>&gt;PPID_OOB would make the compatibility problem worse, not better.<=
/i></b><br>
<b><i><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot=
;sans-serif&quot;;color:#1F497D">&lt;Raju2&gt;<o:p></o:p></span></i></b></p=
>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">I would appreciate =
if you can provide reasons on why PPID_OOB making compatibility worse?<o:p>=
</o:p></span></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">With PPID_OOB, apps=
 have a choice to negotiate (this negotiation is like any other negotiation=
 for webrtc.) and use this option. PPID_EMPTY option seems
 to be enabled always and no way to turn-off. <o:p></o:p></span></i></b></p=
>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Without an option t=
o turn off PPID_EMPTY, you will get into more compatibility issues as some =
webrtc implementations may not implement it on day one;
 a native client could be using an interim version of a popular webrtc lib;=
 or it might be using its own webrtc lib, in which case we can&#8217;t expe=
ct it to implement all these later-adders in a timely manner.<o:p></o:p></s=
pan></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></=
span></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&lt;/Raju2&gt;<o:p>=
</o:p></span></i></b></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">BR<o:p></o:p></span=
></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Raju</span></i></b>=
<span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-=
serif&quot;;color:#1F497D"><o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
</div>
</div>
</body>
</html>

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On 08/04/2014 08:08 PM, Makaraju, Maridi Raju (Raju) wrote:
>
> */>PPID_OOB would make the compatibility problem worse, not better./*
> */<Raju2>/*
>
> */I would appreciate if you can provide reasons on why PPID_OOB making 
> compatibility worse?/*
>

Because WebSockets doesn't have any functionality corresponding to PPID_OOB.

Adding more functions that WebSockets doesn't have makes compatibility 
with WebSockets worse.



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    <div class="moz-cite-prefix">Not responding to rest of message.<br>
      <br>
      On 08/04/2014 08:08 PM, Makaraju, Maridi Raju (Raju) wrote:<br>
    </div>
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            <b><i>&gt;PPID_OOB would make the compatibility problem
                worse, not better.</i></b><br>
            <b><i><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&lt;Raju2&gt;<o:p></o:p></span></i></b></p>
          <p class="MsoNormal"><b><i><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">I
                  would appreciate if you can provide reasons on why
                  PPID_OOB making compatibility worse?</span></i></b></p>
        </div>
      </div>
    </blockquote>
    <br>
    Because WebSockets doesn't have any functionality corresponding to
    PPID_OOB.<br>
    <br>
    Adding more functions that WebSockets doesn't have makes
    compatibility with WebSockets worse.<br>
    <br>
    <br>
  </body>
</html>

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From nobody Mon Aug  4 13:49:21 2014
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On 4 August 2014 13:44, Harald Alvestrand <harald@alvestrand.no> wrote:
> Adding more functions that WebSockets doesn't have makes compatibility with
> WebSockets worse.

Not that this is a good example, but I really wish this wasn't the
yardstick we keep on trotting out.


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>Because WebSockets doesn't have any functionality corresponding to >PPID_O=
OB.

>Adding more functions that WebSockets doesn't have makes compatibility >wi=
th WebSockets worse.
<Raju>
Per the discussions I thought only backward compatibility with Websocket AP=
I is what's desired with data channel API. Does it also mean data channel A=
PI can't add new APIs while maintaining compatibility to existing ones? I d=
on't see a reason why new APIs can't be added and be restricted by websocke=
t API though! Such restriction will limit the data channel API and won't al=
low leveraging on the flexibility provided by data channel's SCTP transport=
 only.

</Raju>

BR
Raju

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4.0pt">
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"color:=
#1F497D">&gt;</span>Because WebSockets doesn't have any functionality corre=
sponding to
<span style=3D"color:#1F497D">&gt;</span>PPID_OOB.<br>
<br>
<span style=3D"color:#1F497D">&gt;</span>Adding more functions that WebSock=
ets doesn't have makes compatibility
<span style=3D"color:#1F497D">&gt;</span>with WebSockets worse.<span style=
=3D"color:#1F497D"><o:p></o:p></span></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&lt;Raju&gt;
<o:p></o:p></span></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Per the discussions=
 I thought only backward compatibility with Websocket API is what&#8217;s d=
esired with data channel API. Does it also mean data channel API
 can&#8217;t add new APIs while maintaining compatibility to existing ones?=
 I don&#8217;t see a reason why new APIs can&#8217;t be added and be restri=
cted by websocket API though! Such restriction will limit the data channel =
API and won&#8217;t allow leveraging on the flexibility
 provided by data channel&#8217;s SCTP transport only.<o:p></o:p></span></i=
></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></=
span></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&lt;/Raju&gt;<o:p><=
/o:p></span></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></=
span></i></b></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">BR</span><o:p></o:p></=
p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Raju</span></i></b>=
<span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-=
serif&quot;;color:#1F497D"><o:p></o:p></span></p>
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From: Justin Uberti <juberti@google.com>
Date: Mon, 4 Aug 2014 17:18:51 -0700
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To: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
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Subject: Re: [rtcweb] Sending of zero-length messages over data channels
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--089e0117772106ba2504ffd6cefc
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

On Mon, Aug 4, 2014 at 11:08 AM, Makaraju, Maridi Raju (Raju) <
Raju.Makaraju@alcatel-lucent.com> wrote:

>
>
>
>
> *>IMHO opinion, sending such a dummy byte may have other consequences:*
>
> 1.      *>App is not aware of what=E2=80=99s going on? send() probably me=
ant to
> be called >with non-empty data, but some error leg could have caused it
> become >empty, but still a byte will be sent out!!*
>
> 2.      *>Debugging becomes bit more complex and not intuitive. WebRTC is
> >already complex, why add more complexity?*
>
> 3.      *>Interop issues. The more the exceptions the more the interop
> issues you >see as some implementations may overlook these or implement
> them >later.*
>
>
>
> *>It=E2=80=99s usually a bad idea to send a dummy byte on wire, without t=
he
> application >direct involvement.*
>
>
> >I recommend spending some time with Wireshark looking at actual traces
> >before making any such statement. Many protocols have dummy bytes, for
> >many different reasons.
>
>  *<Raju2> *
>
> *I am afraid there is some confusion about my comment and as a result we
> may be talking about dummy bytes at 2 different levels: app level and
> underlying protocol level.*
>
> *I am aware of many protocols using reserved bits, bytes, padded
> bits/bytes, protocol-level heartbeats/PING-PONGs (SCTP heartbeats use of
> opaque data). In all these cases the mechanism is built into the protocol
> below app-level protocol itself with no direct involvement of
> application-level protocol. My comment was in the context of
> application-level protocol. *
>
> *If app calls send() 3 times with data =E2=80=9Cabc=E2=80=9D, =E2=80=9C=
=E2=80=9D (empty/null data with
> size 0) and then =E2=80=9Cxyz=E2=80=9D then on the wire in the applicatio=
n-level payload
> you see =E2=80=9Cabc=E2=80=9D <dummy byte> =E2=80=9Cxyz=E2=80=9D.*
>
> *So, can you please point to me a protocol where this kind of dummy
> byte(s) are inserted in between application payload?*
>
>
>
> *Btw, If you leave the app choice to send the byte (using OOB_PPID) then
> there is a possibility for the app to send 256 distinct values instead of
> browser hard-coding the byte to zero always, which just conveys a single
> value.*
>
>
>
> *</Raju2>*
>
>
>

What value would this provide to justify the increased complexity?


>
> *>PPID_OOB would make the compatibility problem worse, not better.*
> *<Raju2>*
>
> *I would appreciate if you can provide reasons on why PPID_OOB making
> compatibility worse?*
>
> *With PPID_OOB, apps have a choice to negotiate (this negotiation is like
> any other negotiation for webrtc.) and use this option. PPID_EMPTY option
> seems to be enabled always and no way to turn-off. *
>
> *Without an option to turn off PPID_EMPTY, you will get into more
> compatibility issues as some webrtc implementations may not implement it =
on
> day one; a native client could be using an interim version of a popular
> webrtc lib; or it might be using its own webrtc lib, in which case we can=
=E2=80=99t
> expect it to implement all these later-adders in a timely manner.*
>
>
> I doubt this will be a real-world problem, given that the protocol is
still evolving and all implementations need to keep up, but if this turns
out to be an issue we can negotiate it in SDP.

--089e0117772106ba2504ffd6cefc
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Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Mon, Aug 4, 2014 at 11:08 AM, Makaraju, Maridi Raju (Raju) <span=
 dir=3D"ltr">&lt;<a href=3D"mailto:Raju.Makaraju@alcatel-lucent.com" target=
=3D"_blank">Raju.Makaraju@alcatel-lucent.com</a>&gt;</span> wrote:<br>

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





<div bgcolor=3D"white" lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt"><div class=3D"">
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=C2=A0<u></u=
></span></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=C2=A0<u></u=
></span></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">&gt;IMHO opinion, s=
ending such a dummy byte may have other consequences:</span></i></b><u></u>=
<u></u></p>


<p><u></u><span>1.<span style=3D"font:7.0pt &quot;Times New Roman&quot;">=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0
</span></span><u></u><b><i><span style=3D"font-size:11.0pt;font-family:&quo=
t;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">&gt;App is not aware =
of what=E2=80=99s going on? send() probably meant to be called &gt;with non=
-empty data, but some error leg could have caused it become &gt;empty,
 but still a byte will be sent out!!</span></i></b><u></u><u></u></p>
<p><u></u><span>2.<span style=3D"font:7.0pt &quot;Times New Roman&quot;">=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0
</span></span><u></u><b><i><span style=3D"font-size:11.0pt;font-family:&quo=
t;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">&gt;Debugging becomes=
 bit more complex and not intuitive. WebRTC is &gt;already complex, why add=
 more complexity?</span></i></b><u></u><u></u></p>


<p><u></u><span>3.<span style=3D"font:7.0pt &quot;Times New Roman&quot;">=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0
</span></span><u></u><b><i><span style=3D"font-size:11.0pt;font-family:&quo=
t;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">&gt;Interop issues. T=
he more the exceptions the more the interop issues you &gt;see as some impl=
ementations may overlook these or implement them &gt;later.</span></i></b><=
u></u><u></u></p>


<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">=C2=A0</span></i></=
b><u></u><u></u></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">&gt;It=E2=80=99s us=
ually a bad idea to send a dummy byte on wire, without the application &gt;=
direct involvement.</span></i></b><u></u><u></u></p>


<p class=3D"MsoNormal" style=3D"margin-left:5.25pt"><br>
&gt;I recommend spending some time with Wireshark looking at actual traces =
&gt;before making any such statement. Many protocols have dummy bytes, for =
&gt;many different reasons.<br>
<br>
<u></u><u></u></p>
</div><p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-fam=
ily:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">&lt;Raju2&gt;
<u></u><u></u></span></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">I am afraid there i=
s some confusion about my comment and as a result we may be talking about d=
ummy bytes at 2 different levels: app level and underlying
 protocol level.<u></u><u></u></span></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">I am aware of many =
protocols using reserved bits, bytes, padded bits/bytes, protocol-level hea=
rtbeats/PING-PONGs (SCTP heartbeats use of opaque data).
 In all these cases the mechanism is built into the protocol below app-leve=
l protocol itself with no direct involvement of application-level protocol.=
 My comment was in the context of application-level protocol.
<u></u><u></u></span></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">If app calls send()=
 3 times with data =E2=80=9Cabc=E2=80=9D, =E2=80=9C=E2=80=9D (empty/null da=
ta with size 0) and then =E2=80=9Cxyz=E2=80=9D then on the wire in the appl=
ication-level payload you see
 =E2=80=9Cabc=E2=80=9D &lt;dummy byte&gt; =E2=80=9Cxyz=E2=80=9D.<u></u><u><=
/u></span></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">So, can you please =
point to me a protocol where this kind of dummy byte(s) are inserted in bet=
ween application payload?<u></u><u></u></span></i></b></p>


<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=C2=A0<u></u=
></span></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">Btw, If you leave t=
he app choice to send the byte (using OOB_PPID) then there is a possibility=
 for the app to send 256 distinct values instead of browser
 hard-coding the byte to zero always, which just conveys a single value.<u>=
</u><u></u></span></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=C2=A0<u></u=
></span></i></b></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">&lt;/Raju2&gt;</spa=
n></i></b><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&=
quot;sans-serif&quot;;color:#1f497d"><u></u><u></u></span></p>


<p class=3D"MsoNormal"><span style=3D"color:#1f497d"><u></u>=C2=A0</span></=
p></div></div></div></blockquote><div><br></div><div>What value would this =
provide to justify the increased complexity?</div><div>=C2=A0</div><blockqu=
ote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc s=
olid;padding-left:1ex">

<div bgcolor=3D"white" lang=3D"EN-US" link=3D"blue" vlink=3D"purple"><div><=
div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in 4=
.0pt"><p class=3D"MsoNormal"><span style=3D"color:#1f497d"><u></u></span></=
p>
<p class=3D"MsoNormal"></p><div class=3D""><br>
<b><i>&gt;PPID_OOB would make the compatibility problem worse, not better.<=
/i></b><br>
</div><b><i><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;=
,&quot;sans-serif&quot;;color:#1f497d">&lt;Raju2&gt;<u></u><u></u></span></=
i></b><p></p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">I would appreciate =
if you can provide reasons on why PPID_OOB making compatibility worse?<u></=
u><u></u></span></i></b></p>


<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">With PPID_OOB, apps=
 have a choice to negotiate (this negotiation is like any other negotiation=
 for webrtc.) and use this option. PPID_EMPTY option seems
 to be enabled always and no way to turn-off. <u></u><u></u></span></i></b>=
</p>
<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">Without an option t=
o turn off PPID_EMPTY, you will get into more compatibility issues as some =
webrtc implementations may not implement it on day one;
 a native client could be using an interim version of a popular webrtc lib;=
 or it might be using its own webrtc lib, in which case we can=E2=80=99t ex=
pect it to implement all these later-adders in a timely manner.<u></u><u></=
u></span></i></b></p>


<p class=3D"MsoNormal"><br></p></div></div></div></blockquote><div>I doubt =
this will be a real-world problem, given that the protocol is still evolvin=
g and all implementations need to keep up, but if this turns out to be an i=
ssue we can negotiate it in SDP.</div>

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

--089e0117772106ba2504ffd6cefc--


From nobody Tue Aug  5 05:25:04 2014
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Date: Tue, 05 Aug 2014 05:24:53 -0700
From: Harald Alvestrand <harald@alvestrand.no>
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Subject: [rtcweb] ALPN question - optional confidentiality
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In draft-ietf-rtcweb-alpn-00, the following text occurs:

   Only one of these labels can be used for any given session.  A peer
   acting in the client role MUST NOT offer both identifiers.  A peer in
   the server role that receives a ClientHello containing both labels
   MUST reject the session, though it MAY accept the confidential option
   and protect content accordingly.

I can't make this match up in my head; the MAY seems to contradict the MUST.

If a client sends a ClientHello(c-webrtc, webrtc), the MUST seems to say
that the server MUST reject the session, while the MAY seems to say that
the server MAY do Accept(c-webrtc).

Suggested reformulation:

If a client offers both c-webrtc and webrtc, the server MAY accept
c-webrtc, but MUST NOT accept webrtc (possibly add: because this makes a
downgrade attack possible???)

-- 
Surveillance is pervasive. Go Dark.


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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Justin Uberti <juberti@google.com>
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Date: Tue, 05 Aug 2014 14:35:27 -0400
From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: [rtcweb] draft-ietf-rtcweb-alpn-00 - in what way is WebRTC an application?
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Why is WebRTC considered an application? It provides a 
platform/layer/transport in support of applications.

Applications ride on top of it. I guess Meetecho or Webex++ or Skype++ 
would be plausible applications.

Of course this is a constantly changing game. We keep adding layers to 
the stack, and each time we do what used to be an application becomes 
just a transport for the next layer.

But AFAIK WebRTC is *never* an application in its own right.

ISTM that it would be more appropriate if the application identifier 
identified the complete protocol stack that is riding on top of the 
layer (in this case DTLS) that is carrying the identifier.

E.g. "meetecho/webex"

	Thanks,
	Paul


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From: Ted Hardie <ted.ietf@gmail.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [rtcweb] draft-ietf-rtcweb-alpn-00 - in what way is WebRTC an application?
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--90e6ba6149aa2d904304ffe74cfb
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Hi Paul,

Think of it this way:  ALPN tokens identify application layer protocols.
WebRTC may support a raft of applications, but it is a common application
layer.

Attempting to negotiate each one of the potential applications which might
use the layer would be thoroughly unworkable, given the ability of a
developer to mint one at will.  Thus, it is more important to signal the
common aspect--that this application is built on WebRTC protocols and
expectations.

regards,

Ted Hardie


On Tue, Aug 5, 2014 at 11:35 AM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:

> Why is WebRTC considered an application? It provides a
> platform/layer/transport in support of applications.
>
> Applications ride on top of it. I guess Meetecho or Webex++ or Skype++
> would be plausible applications.
>
> Of course this is a constantly changing game. We keep adding layers to the
> stack, and each time we do what used to be an application becomes just a
> transport for the next layer.
>
> But AFAIK WebRTC is *never* an application in its own right.
>
> ISTM that it would be more appropriate if the application identifier
> identified the complete protocol stack that is riding on top of the layer
> (in this case DTLS) that is carrying the identifier.
>
> E.g. "meetecho/webex"
>
>         Thanks,
>         Paul
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--90e6ba6149aa2d904304ffe74cfb
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<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:garamond=
,serif;font-size:small">Hi Paul,<br><br></div><div class=3D"gmail_default" =
style=3D"font-family:garamond,serif;font-size:small">Think of it this way:=
=C2=A0 ALPN tokens identify application layer protocols.=C2=A0 WebRTC may s=
upport a raft of applications, but it is a common application layer.<br>
<br></div><div class=3D"gmail_default" style=3D"font-family:garamond,serif;=
font-size:small">Attempting to negotiate each one of the potential applicat=
ions which might use the layer would be thoroughly unworkable, given the ab=
ility of a developer to mint one at will.=C2=A0 Thus, it is more important =
to signal the common aspect--that this application is built on WebRTC proto=
cols and expectations.<br>
<br></div><div class=3D"gmail_default" style=3D"font-family:garamond,serif;=
font-size:small">regards,<br><br>Ted Hardie<br></div></div><div class=3D"gm=
ail_extra"><br><br><div class=3D"gmail_quote">On Tue, Aug 5, 2014 at 11:35 =
AM, Paul Kyzivat <span dir=3D"ltr">&lt;<a href=3D"mailto:pkyzivat@alum.mit.=
edu" target=3D"_blank">pkyzivat@alum.mit.edu</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Why is WebRTC considered an application? It =
provides a platform/layer/transport in support of applications.<br>
<br>
Applications ride on top of it. I guess Meetecho or Webex++ or Skype++ woul=
d be plausible applications.<br>
<br>
Of course this is a constantly changing game. We keep adding layers to the =
stack, and each time we do what used to be an application becomes just a tr=
ansport for the next layer.<br>
<br>
But AFAIK WebRTC is *never* an application in its own right.<br>
<br>
ISTM that it would be more appropriate if the application identifier identi=
fied the complete protocol stack that is riding on top of the layer (in thi=
s case DTLS) that is carrying the identifier.<br>
<br>
E.g. &quot;meetecho/webex&quot;<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Thanks,<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Paul<br>
<br>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</blockquote></div><br></div>

--90e6ba6149aa2d904304ffe74cfb--


From nobody Tue Aug  5 13:15:35 2014
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On 5 August 2014 12:59, Ted Hardie <ted.ietf@gmail.com> wrote:
> Think of it this way:  ALPN tokens identify application layer protocols.
> WebRTC may support a raft of applications, but it is a common application
> layer.

We commonly say the same thing about HTTP as an application layer
protocol.  I don't think that anyone is going to suggest that nothing
else goes on above HTTP either.


From nobody Tue Aug  5 18:27:39 2014
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-alpn-00.txt
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The draft defines "webrtc" mixed media and data communications using 
SRTP and data channels. Does there need to be a value that describes the 
use of "webrtc data channel only"? E.g. An endpoint may establish a data 
channel for multiple applications (CLUE, BFCP etc.) without using WebRTC 
media.

Regards, Christian

On 23/07/2014 11:32 PM, internet-drafts@ietf.org wrote:
> A New Internet-Draft is available from the on-line Internet-Drafts directories.
>   This draft is a work item of the Real-Time Communication in WEB-browsers Working Group of the IETF.
>
>          Title           : Application Layer Protocol Negotiation for Web Real-Time Communications (WebRTC)
>          Author          : Martin Thomson
> 	Filename        : draft-ietf-rtcweb-alpn-00.txt
> 	Pages           : 7
> 	Date            : 2014-07-23
>
> Abstract:
>     Application Layer Protocol Negotiation (ALPN) labels are defined for
>     use in identifying Web Real-Time Communications (WebRTC) usages of
>     Datagram Transport Layer Security (DTLS).  Labels are provided for
>     identifying a session that uses a combination of WebRTC compatible
>     media and data, and for identifying a session requiring
>     confidentiality protection.
>
>
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-alpn/
>
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-rtcweb-alpn-00
>
>
> Please note that it may take a couple of minutes from the time of submission
> until the htmlized version and diff are available at tools.ietf.org.
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


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Subject: [rtcweb] Raw notes from IETF91
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You can find the raw notes at (and you can comment on them) at

=
https://docs.google.com/document/d/1Yl-qToN5KDdGSIhcOfsI5vbqGrDAbyzln-_ktO=
OYKM0/edit?usp=3Dsharing

As we finalize these, we will get proper ones submitted to list.=20



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From: Justin Uberti <juberti@google.com>
Date: Thu, 7 Aug 2014 12:31:45 -0700
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To: Victoria Kirst <vrk@google.com>, Benjamin Schwartz <bemasc@google.com>,  "rtcweb@ietf.org" <rtcweb@ietf.org>
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Subject: Re: [rtcweb] Handling system suspend during an active call
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--20cf307cffd6cb916e05000f245f
Content-Type: text/plain; charset=UTF-8

When screen-sharing in a video conference, it's quite common for the
screen-sharing user to stop sharing by closing their laptop. However, when
this happens, the browser doesn't give the application or MCU any
indication, leading to zombie participants in the conference. (Of course,
the MCU will stop receiving packets, but as that could also happen due to
transient connectivity issues, we don't want to eject the user upon a brief
interruption of RTP packets).

One way to deal with this would be for the browser, upon receipt of a
suspend notification from the OS, to send an indication that would let the
MCU know that the stream is officially terminating. This has to be done by
the browser, since the application doesn't get any such notification. For
example, the browser could send RTCP BYE for all active RTP streams and
terminate any outstanding SCTP associations. While this is different than
how the browser treats HTTP connections on suspend, it makes sense in this
case because the session really can't be resumed after suspend, as ICE
consent will have disappeared within a minute. Alternatively, if we wanted
to allow resumption, the browser could send RTCP PAUSED
<http://tools.ietf.org/html/draft-ietf-avtext-rtp-stream-pause-02#section-8.2>
indication
(via TMMBN 0) instead.

Receipt of BYE/PAUSED message prior to RTP packets ceasing would tell the
remote side that the sender intentionally stopped the flow, instead of
falling off the Internet. While not perfect, this seems like a simple
solution that avoids the complexities of trying to expose power events
directly to the application.

--20cf307cffd6cb916e05000f245f
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_extra">When screen-sharing in a video =
conference, it&#39;s quite common for the screen-sharing user to stop shari=
ng by closing their laptop. However, when this happens, the browser doesn&#=
39;t give the application or MCU any indication, leading to zombie particip=
ants in the conference. (Of course, the MCU will stop receiving packets, bu=
t as that could also happen due to transient connectivity issues, we don&#3=
9;t want to eject the user upon a brief interruption of RTP packets).<div>

<br></div><div>One way to deal with this would be for the browser, upon rec=
eipt of a suspend notification from the OS, to send an indication that woul=
d let the MCU know that the stream is officially terminating. This has to b=
e done by the browser, since the application doesn&#39;t get any such notif=
ication. For example, the browser could send RTCP BYE for all active RTP st=
reams and terminate any outstanding SCTP associations. While this is differ=
ent than how the browser treats HTTP connections on suspend, it makes sense=
 in this case because the session really can&#39;t be resumed after suspend=
, as ICE consent will have disappeared within a minute. Alternatively, if w=
e wanted to allow resumption, the browser could send=C2=A0<a href=3D"http:/=
/tools.ietf.org/html/draft-ietf-avtext-rtp-stream-pause-02#section-8.2" tar=
get=3D"_blank">RTCP PAUSED</a>=C2=A0indication (via TMMBN 0) instead.=C2=A0=
</div>

<div><br></div><div>Receipt of BYE/PAUSED message prior to RTP packets ceas=
ing would tell the remote side that the sender intentionally stopped the fl=
ow, instead of falling off the Internet. While not perfect, this seems like=
 a simple solution that avoids the complexities of trying to expose power e=
vents directly to the application.</div>

</div></div>

--20cf307cffd6cb916e05000f245f--


From nobody Thu Aug  7 14:24:36 2014
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Subject: [rtcweb] ALPN question - other labels?
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Section 2 says:

    The following identifiers are defined for use in ALPN:
...
    Only one of these labels can be used for any given session.  A peer
    acting in the client role MUST NOT offer both identifiers.  A peer in
    the server role that receives a ClientHello containing both labels
    MUST reject the session, though it MAY accept the confidential option
    and protect content accordingly.

This does not quite say that a peer in a server role must reject a 
session that contains some *other* label. It doesn't even explicitly 
forbid a peer in a client role from offering some other label. Does it 
intend to do so?

This could be an issue in interoperation. For instance, if CLUE 
endpoints want to be capable of interoperating with WebRTC devices must 
they always use one of the webrtc ALPN values?

This could be handled saying that any other ALPN value is to be treated 
by browsers as equivalent to "webrtc".

(Note: we haven't worked out in detail how to interoperate CLUE and 
WebRTC, but we are trying to ensure we don't do anything to preclude it.)

	Thanks,
	Paul


From nobody Thu Aug  7 16:26:03 2014
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Subject: Re: [rtcweb] ALPN question - other labels?
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On 7 August 2014 14:24, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
>    Only one of these labels can be used for any given session.  A peer
>    acting in the client role MUST NOT offer both identifiers.  A peer in
>    the server role that receives a ClientHello containing both labels
>    MUST reject the session, though it MAY accept the confidential option
>    and protect content accordingly.
>
> This does not quite say that a peer in a server role must reject a session
> that contains some *other* label. It doesn't even explicitly forbid a peer
> in a client role from offering some other label. Does it intend to do so?

That's actually old, incorrect text.  I need to update the draft, but
don't want to do every time that I make a change (I'm not looking to
top Ian Hickson's record here).

Here's what my copy says:

"""A peer that is not aware of whether it needs to request
confidentiality can use either form. A peer in the client role MUST
offer both identifiers if it is not aware of a need for
confidentiality. A peer in the server role SHOULD select webrtc if it
does not prefer either.""
-- https://martinthomson.github.io/drafts/draft-ietf-rtcweb-alpn.html#rfc.section.2

As to whether a completely different protocol is acceptable, the
intent was to only cover the interaction of the two labels that are
defined in the document.  If you want to do webrtc OR some other
protocol, I see no reason not to permit that.


From nobody Thu Aug  7 19:25:39 2014
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Date: Thu, 07 Aug 2014 22:25:33 -0400
From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [rtcweb] ALPN question - other labels?
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On 8/7/14 7:25 PM, Martin Thomson wrote:
> On 7 August 2014 14:24, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
>>     Only one of these labels can be used for any given session.  A peer
>>     acting in the client role MUST NOT offer both identifiers.  A peer in
>>     the server role that receives a ClientHello containing both labels
>>     MUST reject the session, though it MAY accept the confidential option
>>     and protect content accordingly.
>>
>> This does not quite say that a peer in a server role must reject a session
>> that contains some *other* label. It doesn't even explicitly forbid a peer
>> in a client role from offering some other label. Does it intend to do so?
>
> That's actually old, incorrect text.  I need to update the draft, but
> don't want to do every time that I make a change (I'm not looking to
> top Ian Hickson's record here).
>
> Here's what my copy says:
>
> """A peer that is not aware of whether it needs to request
> confidentiality can use either form. A peer in the client role MUST
> offer both identifiers if it is not aware of a need for
> confidentiality. A peer in the server role SHOULD select webrtc if it
> does not prefer either.""
> -- https://martinthomson.github.io/drafts/draft-ietf-rtcweb-alpn.html#rfc.section.2
>
> As to whether a completely different protocol is acceptable, the
> intent was to only cover the interaction of the two labels that are
> defined in the document.  If you want to do webrtc OR some other
> protocol, I see no reason not to permit that.

My primary question is how is a browser expected to behave if some other 
label is offered? Will it still support use of the JSEP APIs?

And, for interop reasons, will there be a way to get a browser to offer 
another label and still support the JSEP APIs if the other end accepts it?

	Thanks,
	Paul


From nobody Thu Aug  7 19:41:47 2014
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Subject: Re: [rtcweb] ALPN question - other labels?
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--f46d043c7faa4530380500152599
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On Aug 7, 2014 7:25 PM, "Paul Kyzivat" <pkyzivat@alum.mit.edu> wrote:
> My primary question is how is a browser expected to behave if some other
label is offered? Will it still support use of the JSEP APIs?

ALPN requires that a peer in the server role reject the connection if it
can't select a protocol. That is what Firefox will do, though it will
tolerate no ALPN extension as long as it doesn't have confidential media.

> And, for interop reasons, will there be a way to get a browser to offer
another label and still support the JSEP APIs if the other end accepts it?

That is an interesting question. You would need to teach the browser that
other protocol, I guess.

--f46d043c7faa4530380500152599
Content-Type: text/html; charset=UTF-8

<p dir="ltr"><br>
On Aug 7, 2014 7:25 PM, &quot;Paul Kyzivat&quot; &lt;<a href="mailto:pkyzivat@alum.mit.edu">pkyzivat@alum.mit.edu</a>&gt; wrote:<br>
&gt; My primary question is how is a browser expected to behave if some other label is offered? Will it still support use of the JSEP APIs?</p>
<p dir="ltr">ALPN requires that a peer in the server role reject the connection if it can&#39;t select a protocol. That is what Firefox will do, though it will tolerate no ALPN extension as long as it doesn&#39;t have confidential media.</p>

<p dir="ltr">&gt; And, for interop reasons, will there be a way to get a browser to offer another label and still support the JSEP APIs if the other end accepts it?</p>
<p dir="ltr">That is an interesting question. You would need to teach the browser that other protocol, I guess.<br>
</p>

--f46d043c7faa4530380500152599--


From nobody Fri Aug  8 03:01:30 2014
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Fri, 8 Aug 2014 12:01:00 +0200
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Subject: [rtcweb] How to reset DTLS over TCP if the connection is closed
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Hi, let's assume the following scenario:

- Browser and server (ICE-Lite) with just TCP candidates.

- Browser sends the DTLS handshake over the TCP nominated (and single)
connection.

- Media begins.

- Later the TCP connection is abruptly closed. The client want to
"reset" the communication.


So at this point I expect that:

1) Browser MUST open a new TCP connection (obviously).

2) When done it must send Binding Request for the server to allow media fro=
m it.

3) Browser MUST NOT perform again the DTLS handshake.

The rationale on bullet 3) is that ICE is supposed to be a "virtual
socket" and thus a single DTLS handshake must be done regardless which
UDP/TCP/SCTP/X.25 transport is used to carried the DTLS data between
peers.


4) Said that, and assuming that there is also a UDP ICE valid pair, I
also expect (and this is a bit exotic but IMHO it SHOULD work) that
the browser should be able to send a DTLS Close Alert over the UDP
valid pair (this is, even when the DTLS handshake and RTP is being
sent over the TCP nominated pair).


Well, I'm pretty sure of all the above, but I consider it a bit exotic
and would like to hear opinions :)

Regards.


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


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On 8/7/14 10:41 PM, Martin Thomson wrote:
>
> On Aug 7, 2014 7:25 PM, "Paul Kyzivat" <pkyzivat@alum.mit.edu
> <mailto:pkyzivat@alum.mit.edu>> wrote:
>  > My primary question is how is a browser expected to behave if some
> other label is offered? Will it still support use of the JSEP APIs?
>
> ALPN requires that a peer in the server role reject the connection if it
> can't select a protocol. That is what Firefox will do, though it will
> tolerate no ALPN extension as long as it doesn't have confidential media.

Are you saying that the offering and acceptance of ALPN values is 
controlled totally by the browser, without any influence by the JS app?

>  > And, for interop reasons, will there be a way to get a browser to
> offer another label and still support the JSEP APIs if the other end
> accepts it?
>
> That is an interesting question. You would need to teach the browser
> that other protocol, I guess.

IMO this would only be interesting if other values can also be used in 
the server role.

For now I think it is more important that you be very clear what you 
intend.

I think it means that CLUE implementations will always need to use 
"webrtc" as the ALPN for the CLUE data channel, so that it will be right 
when interoperating with WebRTC. That is possible, though it seems weird 
when not interoperating.

(I'm struggling to figure out what c-webrtc means for a native clue 
device, such as an MCU.)

	Thanks,
	Paul


From nobody Fri Aug  8 09:43:32 2014
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Subject: Re: [rtcweb] ALPN question - other labels?
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On 8 August 2014 09:12, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
> Are you saying that the offering and acceptance of ALPN values is controlled
> totally by the browser, without any influence by the JS app?

Well, for the confidential media use case, it has to be.

>>  > And, for interop reasons, will there be a way to get a browser to
>> offer another label and still support the JSEP APIs if the other end
>> accepts it?
>>
>> That is an interesting question. You would need to teach the browser
>> that other protocol, I guess.
>
> IMO this would only be interesting if other values can also be used in the
> server role.

Do you mean the server can pick something that the client didn't offer?

> For now I think it is more important that you be very clear what you intend.
>
> I think it means that CLUE implementations will always need to use "webrtc"
> as the ALPN for the CLUE data channel, so that it will be right when
> interoperating with WebRTC. That is possible, though it seems weird when not
> interoperating.

If the intent is to interoperate with WebRTC, by effectively sitting
on top of the WebRTC stack, then I don't find that to be particularly
strange.  That doesn't mean that you can't define a "clue" token,
which may or may not be identical or at least very similar to
"webrtc".


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--089e01182c0adfd46d050024082b
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I have reviewed this document in preparation for IETF LC.  In general, it
looks good, and I have requested LC.  There are some comments below that
should be addressed along with IETF LC comments.
--Richard


MAJOR:

Section 4. "To avoid glare..."
The term "glare" is undefined in this document, and will probably be
unfamiliar to many readers.  Please add it to the Terminology section.

Section 4. "When using SCTP over DTLS..."
This qualification is unnecessary, since data channels always use
SCTP/DTLS/UDP:
"""
   The DTLS encapsulation of SCTP packets as described in
   [I-D.ietf-tsvwg-sctp-dtls-encaps] MUST be used.
"""
So this should be "Since data channels use the DTLS encapsulation of
SCTP..."

Section 6. "... all parameters in the DATA_CHANNEL_OPEN message are valid"
The receiver MUST fail if the parameters are invalid, but you never define
what validation needs to be performed.  For example, is the DCEP layer
supposed to verify that the Protocol and Label fields are valid UTF-8?

Section 10.2. [I-D.ietf-rtcweb-data-channel]
This reference needs to be normative, since, after all, this is a protocol
for negotiating data channels.



MINOR / EDITORIAL:

Section 5.1. [packet diagram]
The diagram seems to imply that the protocol field starts on a 4-octet
boundary, even though this is not true.  It might be good if you could find
a way to reformat the diagram to make this clearer.

Section 5.1. "The suggested value of this field for IANA is 0x03"
You're defining the initial contents of this registry.  Just say it.
Likewise for the same text in Section 5.2.

Section 6. "an Stream identifier" -> "a stream identifier"

--089e01182c0adfd46d050024082b
Content-Type: text/html; charset=UTF-8
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<div dir=3D"ltr"><div><div><div>I have reviewed this document in preparatio=
n for IETF LC.=C2=A0 In general, it looks good, and I have requested LC.=C2=
=A0 There are some comments below that should be addressed along with IETF =
LC comments.<br>
</div>--Richard<br><br><br></div><div>MAJOR:<br></div><div><br></div></div>=
<div>Section 4. &quot;To avoid glare...&quot;<br></div><div>The term &quot;=
glare&quot; is undefined in this document, and will probably be unfamiliar =
to many readers.=C2=A0 Please add it to the Terminology section.<br>
<br></div><div>Section 4. &quot;When using SCTP over DTLS...&quot;<br></div=
><div>This qualification is unnecessary, since data channels always use SCT=
P/DTLS/UDP:<br>&quot;&quot;&quot;<br>=C2=A0=C2=A0 The DTLS encapsulation of=
 SCTP packets as described in<br>
=C2=A0=C2=A0 [I-D.ietf-tsvwg-sctp-dtls-encaps] MUST be used.<br>&quot;&quot=
;&quot;<br></div><div>So this should be &quot;Since data channels use the D=
TLS encapsulation of SCTP...&quot;<br></div><div><br>Section 6. &quot;... a=
ll parameters in the DATA_CHANNEL_OPEN message are valid&quot;<br>
<div>The
 receiver MUST fail if the parameters are invalid, but you never define=20
what validation needs to be performed.=C2=A0 For example, is the DCEP layer=
=20
supposed to verify that the Protocol and Label fields are valid UTF-8?<br><=
br></div><div>Section 10.2. [I-D.ietf-rtcweb-data-channel]<br></div><div>Th=
is reference needs to be normative, since, after all, this is a protocol fo=
r negotiating data channels.<br>
</div><br><br><br></div><div>MINOR / EDITORIAL:<br></div><div><br></div><di=
v>Section 5.1. [packet diagram]<br></div><div>The diagram seems to imply th=
at the protocol field starts on a 4-octet boundary, even though this is not=
 true.=C2=A0 It might be good if you could find a way to reformat the diagr=
am to make this clearer.<br>
<br>Section 5.1. &quot;The suggested value of this field for IANA is 0x03&q=
uot;<br></div><div>You&#39;re defining the initial contents of this registr=
y.=C2=A0 Just say it.=C2=A0 Likewise for the same text in Section 5.2.<br><=
br></div>
<div>Section 6. &quot;an Stream identifier&quot; -&gt; &quot;a stream ident=
ifier&quot;<br><br></div></div>

--089e01182c0adfd46d050024082b--


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--047d7b47217aff677f050024080a
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I have reviewed this document in preparation for IETF Last Call.  I have
gone ahead and requested last call, but there are some comments below that
I would like to be addressed before the document goes to the IESG.
--Richard


MAJOR:

Section 6.2.
I'm very concerned about interoperability of data channels given that,
e.g., (1) DCEP [draft-ietf-rtcweb-data-protocol] is not required, and (2)
DCEP only works if it is used by both endpoints (it resets any streams that
are used without DCEP negotiation).  In order to avoid interoperability
failure of this type, whatever mechanism is used to establish an SCTP
association that is used for data channels must also negotiate the
mechanism(s) that will be used to establish data channels over that
association.  That could be directly (by negotiating some data channels at
the same time as the association), or indirectly (by indicating that DCEP
or some other protocol will be used), but either way, the endpoints of the
association need to know how data channels will be negotiated before they
can create any data channels.  This section needs to state this requirement
so that applications using data channels can know that they need to meet it.

Section 6.4.
This section lists in passing several properties of a data channel, but
doesn't give a full definition of all the things that need to be defined in
order to define a data channel.  Fortunately, such a list is given in
draft-ietf-rtcweb-data-protocol (Section 4); I would suggest moving that
here.



MINOR / EDITORIAL:

Figures 1 and 2 list ICE as a separate layer.  That seems incorrect, since
ICE isn't actually involved once the connection is set up.

Section 6.4. "... in-sequence, out-of-sequence, reliable and unreliable as
properties of channels"
"... (as opposed to SCTP, which applies these concepts to messages)"

Section 6.4. "... and waiting for onopen to fire ... meaning of the streams"
This sentence doesn't really parse, and it's not clear why these API-level
considerations are relevant here.

Section 6.4. "stream SCTP identifier" -> "SCTP stream identifier" ?

Section 6.6. "The sender SHOULD disable the Nagle algorithm..."
A reference to RFC 1122 would be helpful here.

--047d7b47217aff677f050024080a
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div><div><div><div><div><div>I have reviewed this documen=
t in preparation for IETF Last Call.=C2=A0 I have gone ahead and requested =
last call, but there are some comments below that I would like to be addres=
sed before the document goes to the IESG.<br>
</div><div>--Richard<br><br></div><div><br></div>MAJOR:<br><br></div><div>S=
ection 6.2. <br>I&#39;m very concerned about interoperability of data chann=
els given that, e.g., (1) DCEP [draft-ietf-rtcweb-data-protocol] is not req=
uired, and (2) DCEP only works if it is used by both endpoints (it resets a=
ny streams that are used without DCEP negotiation).=C2=A0 In order to avoid=
 interoperability failure of this type, whatever mechanism is used to estab=
lish an SCTP association that is used for data channels must also negotiate=
 the mechanism(s) that will be used to establish data channels over that as=
sociation.=C2=A0 That could be directly (by negotiating some data channels =
at the same time as the association), or indirectly (by indicating that DCE=
P or some other protocol will be used), but either way, the endpoints of th=
e association need to know how data channels will be negotiated before they=
 can create any data channels.=C2=A0 This section needs to state this requi=
rement so that applications using data channels can know that they need to =
meet it.<br>
</div><div><br></div><div>Section 6.4. <br></div><div>This section lists in=
 passing several properties of a data channel, but doesn&#39;t give a full =
definition of all the things that need to be defined in order to define a d=
ata channel.=C2=A0 Fortunately, such a list is given in draft-ietf-rtcweb-d=
ata-protocol (Section 4); I would suggest moving that here.<br>
</div><br><div><br><br></div>MINOR / EDITORIAL:<br><br></div>Figures 1 and =
2 list ICE as a separate layer.=C2=A0 That seems incorrect, since ICE isn&#=
39;t actually involved once the connection is set up.<br><br></div>Section =
6.4. &quot;... in-sequence, out-of-sequence, reliable and unreliable as pro=
perties of channels&quot;<br>
</div>&quot;... (as opposed to SCTP, which applies these concepts to messag=
es)&quot;<br><br></div><div>Section 6.4. &quot;... and waiting for onopen t=
o fire ... meaning of the streams&quot;<br></div><div>This sentence doesn&#=
39;t really parse, and it&#39;s not clear why these API-level consideration=
s are relevant here.<br>
</div><div><br></div><div>Section 6.4. &quot;stream SCTP identifier&quot; -=
&gt; &quot;SCTP stream identifier&quot; ?<br><div><br></div><div>Section 6.=
6. &quot;The sender SHOULD disable the Nagle algorithm...&quot;<br></div>
A reference to RFC 1122 would be helpful here.<br></div></div>

--047d7b47217aff677f050024080a--


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The IESG has received a request from the Real-Time Communication in
WEB-browsers WG (rtcweb) to consider the following document:
- 'WebRTC Data Channels'
  <draft-ietf-rtcweb-data-channel-11.txt> as Proposed Standard

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action. Please send substantive comments to the
ietf@ietf.org mailing lists by 2014-08-22. Exceptionally, comments may be
sent to iesg@ietf.org instead. In either case, please retain the
beginning of the Subject line to allow automated sorting.

Abstract


   The WebRTC framework specifies protocol support for direct
   interactive rich communication using audio, video, and data between
   two peers' web-browsers.  This document specifies the non-media data
   transport aspects of the WebRTC framework.  It provides an
   architectural overview of how the Stream Control Transmission
   Protocol (SCTP) is used in the WebRTC context as a generic transport
   service allowing WEB-browsers to exchange generic data from peer to
   peer.




The file can be obtained via
http://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-channel/

IESG discussion can be tracked via
http://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-channel/ballot/


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



From nobody Fri Aug  8 14:24:40 2014
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The IESG has received a request from the Real-Time Communication in
WEB-browsers WG (rtcweb) to consider the following document:
- 'WebRTC Data Channel Establishment Protocol'
  <draft-ietf-rtcweb-data-protocol-07.txt> as Proposed Standard

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action. Please send substantive comments to the
ietf@ietf.org mailing lists by 2014-08-22. Exceptionally, comments may be
sent to iesg@ietf.org instead. In either case, please retain the
beginning of the Subject line to allow automated sorting.

Abstract


   The WebRTC framework specifies protocol support for direct
   interactive rich communication using audio, video, and data between
   two peers' web-browsers.  This document specifies a simple protocol
   for establishing symmetric Data Channels between the peers.  It uses
   a two way handshake and allows sending of user data without waiting
   for the handshake to complete.




The file can be obtained via
http://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-protocol/

IESG discussion can be tracked via
http://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-protocol/ballot/


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



From nobody Fri Aug  8 14:57:35 2014
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From: Richard Barnes <rlb@ipv.sx>
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Subject: Re: [rtcweb] Last Call: <draft-ietf-rtcweb-data-channel-11.txt> (WebRTC Data Channels) to Proposed Standard
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--089e0158bab485e15b0500254a83
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Dear IETF,

I clicked the "Last Call" button a little too quickly on this one, and have
had to pull the draft back into the AD Evaluation state.  So feel free to
comment in this thread, but know that there will be another (real) Last
Call soon.  I will make sure that comments from both threads are addressed.

Thanks,
--Richard


On Fri, Aug 8, 2014 at 5:21 PM, The IESG <iesg-secretary@ietf.org> wrote:

>
> The IESG has received a request from the Real-Time Communication in
> WEB-browsers WG (rtcweb) to consider the following document:
> - 'WebRTC Data Channels'
>   <draft-ietf-rtcweb-data-channel-11.txt> as Proposed Standard
>
> The IESG plans to make a decision in the next few weeks, and solicits
> final comments on this action. Please send substantive comments to the
> ietf@ietf.org mailing lists by 2014-08-22. Exceptionally, comments may be
> sent to iesg@ietf.org instead. In either case, please retain the
> beginning of the Subject line to allow automated sorting.
>
> Abstract
>
>
>    The WebRTC framework specifies protocol support for direct
>    interactive rich communication using audio, video, and data between
>    two peers' web-browsers.  This document specifies the non-media data
>    transport aspects of the WebRTC framework.  It provides an
>    architectural overview of how the Stream Control Transmission
>    Protocol (SCTP) is used in the WebRTC context as a generic transport
>    service allowing WEB-browsers to exchange generic data from peer to
>    peer.
>
>
>
>
> The file can be obtained via
> http://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-channel/
>
> IESG discussion can be tracked via
> http://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-channel/ballot/
>
>
> No IPR declarations have been submitted directly on this I-D.
>
>
>

--089e0158bab485e15b0500254a83
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div><div>Dear IETF,<br><br></div>I clicked the &quot;Last=
 Call&quot; button a little too quickly on this one, and have had to pull t=
he draft back into the AD Evaluation state.=C2=A0 So feel free to comment i=
n this thread, but know that there will be another (real) Last Call soon.=
=C2=A0 I will make sure that comments from both threads are addressed.<br>
<br>Thanks,<br></div>--Richard<br></div><div class=3D"gmail_extra"><br><br>=
<div class=3D"gmail_quote">On Fri, Aug 8, 2014 at 5:21 PM, The IESG <span d=
ir=3D"ltr">&lt;<a href=3D"mailto:iesg-secretary@ietf.org" target=3D"_blank"=
>iesg-secretary@ietf.org</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><br>
The IESG has received a request from the Real-Time Communication in<br>
WEB-browsers WG (rtcweb) to consider the following document:<br>
- &#39;WebRTC Data Channels&#39;<br>
=C2=A0 &lt;draft-ietf-rtcweb-data-channel-11.txt&gt; as Proposed Standard<b=
r>
<br>
The IESG plans to make a decision in the next few weeks, and solicits<br>
final comments on this action. Please send substantive comments to the<br>
<a href=3D"mailto:ietf@ietf.org">ietf@ietf.org</a> mailing lists by 2014-08=
-22. Exceptionally, comments may be<br>
sent to <a href=3D"mailto:iesg@ietf.org">iesg@ietf.org</a> instead. In eith=
er case, please retain the<br>
beginning of the Subject line to allow automated sorting.<br>
<br>
Abstract<br>
<br>
<br>
=C2=A0 =C2=A0The WebRTC framework specifies protocol support for direct<br>
=C2=A0 =C2=A0interactive rich communication using audio, video, and data be=
tween<br>
=C2=A0 =C2=A0two peers&#39; web-browsers. =C2=A0This document specifies the=
 non-media data<br>
=C2=A0 =C2=A0transport aspects of the WebRTC framework. =C2=A0It provides a=
n<br>
=C2=A0 =C2=A0architectural overview of how the Stream Control Transmission<=
br>
=C2=A0 =C2=A0Protocol (SCTP) is used in the WebRTC context as a generic tra=
nsport<br>
=C2=A0 =C2=A0service allowing WEB-browsers to exchange generic data from pe=
er to<br>
=C2=A0 =C2=A0peer.<br>
<br>
<br>
<br>
<br>
The file can be obtained via<br>
<a href=3D"http://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-channel/"=
 target=3D"_blank">http://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-c=
hannel/</a><br>
<br>
IESG discussion can be tracked via<br>
<a href=3D"http://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-channel/b=
allot/" target=3D"_blank">http://datatracker.ietf.org/doc/draft-ietf-rtcweb=
-data-channel/ballot/</a><br>
<br>
<br>
No IPR declarations have been submitted directly on this I-D.<br>
<br>
<br>
</blockquote></div><br></div>

--089e0158bab485e15b0500254a83--


From nobody Fri Aug  8 14:58:00 2014
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From: Richard Barnes <rlb@ipv.sx>
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--001a11c336ca9575480500254bb3
Content-Type: text/plain; charset=UTF-8

Dear IETF,

I clicked the "Last Call" button a little too quickly on this one, and have
had to pull the draft back into the AD Evaluation state.  So feel free to
comment in this thread, but know that there will be another (real) Last
Call soon.  I will make sure that comments from both threads are addressed.

Thanks,
--Richard


On Fri, Aug 8, 2014 at 5:24 PM, The IESG <iesg-secretary@ietf.org> wrote:

>
> The IESG has received a request from the Real-Time Communication in
> WEB-browsers WG (rtcweb) to consider the following document:
> - 'WebRTC Data Channel Establishment Protocol'
>   <draft-ietf-rtcweb-data-protocol-07.txt> as Proposed Standard
>
> The IESG plans to make a decision in the next few weeks, and solicits
> final comments on this action. Please send substantive comments to the
> ietf@ietf.org mailing lists by 2014-08-22. Exceptionally, comments may be
> sent to iesg@ietf.org instead. In either case, please retain the
> beginning of the Subject line to allow automated sorting.
>
> Abstract
>
>
>    The WebRTC framework specifies protocol support for direct
>    interactive rich communication using audio, video, and data between
>    two peers' web-browsers.  This document specifies a simple protocol
>    for establishing symmetric Data Channels between the peers.  It uses
>    a two way handshake and allows sending of user data without waiting
>    for the handshake to complete.
>
>
>
>
> The file can be obtained via
> http://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-protocol/
>
> IESG discussion can be tracked via
> http://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-protocol/ballot/
>
>
> No IPR declarations have been submitted directly on this I-D.
>
>
>

--001a11c336ca9575480500254bb3
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div><div>Dear IETF,<br><br></div>I clicked the &quot;Last=
 Call&quot; button a=20
little too quickly on this one, and have had to pull the draft back into
 the AD Evaluation state.=C2=A0 So feel free to comment in this thread, but=
=20
know that there will be another (real) Last Call soon.=C2=A0 I will make su=
re
 that comments from both threads are addressed.<br><br>Thanks,<br></div>--R=
ichard</div><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">O=
n Fri, Aug 8, 2014 at 5:24 PM, The IESG <span dir=3D"ltr">&lt;<a href=3D"ma=
ilto:iesg-secretary@ietf.org" target=3D"_blank">iesg-secretary@ietf.org</a>=
&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><br>
The IESG has received a request from the Real-Time Communication in<br>
WEB-browsers WG (rtcweb) to consider the following document:<br>
- &#39;WebRTC Data Channel Establishment Protocol&#39;<br>
=C2=A0 &lt;draft-ietf-rtcweb-data-protocol-07.txt&gt; as Proposed Standard<=
br>
<br>
The IESG plans to make a decision in the next few weeks, and solicits<br>
final comments on this action. Please send substantive comments to the<br>
<a href=3D"mailto:ietf@ietf.org">ietf@ietf.org</a> mailing lists by 2014-08=
-22. Exceptionally, comments may be<br>
sent to <a href=3D"mailto:iesg@ietf.org">iesg@ietf.org</a> instead. In eith=
er case, please retain the<br>
beginning of the Subject line to allow automated sorting.<br>
<br>
Abstract<br>
<br>
<br>
=C2=A0 =C2=A0The WebRTC framework specifies protocol support for direct<br>
=C2=A0 =C2=A0interactive rich communication using audio, video, and data be=
tween<br>
=C2=A0 =C2=A0two peers&#39; web-browsers. =C2=A0This document specifies a s=
imple protocol<br>
=C2=A0 =C2=A0for establishing symmetric Data Channels between the peers. =
=C2=A0It uses<br>
=C2=A0 =C2=A0a two way handshake and allows sending of user data without wa=
iting<br>
=C2=A0 =C2=A0for the handshake to complete.<br>
<br>
<br>
<br>
<br>
The file can be obtained via<br>
<a href=3D"http://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-protocol/=
" target=3D"_blank">http://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-=
protocol/</a><br>
<br>
IESG discussion can be tracked via<br>
<a href=3D"http://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-protocol/=
ballot/" target=3D"_blank">http://datatracker.ietf.org/doc/draft-ietf-rtcwe=
b-data-protocol/ballot/</a><br>
<br>
<br>
No IPR declarations have been submitted directly on this I-D.<br>
<br>
<br>
</blockquote></div><br></div>

--001a11c336ca9575480500254bb3--


From nobody Sat Aug  9 11:49:41 2014
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Subject: [rtcweb] Fwd: WGLC: draft-ietf-dart-dscp-rtp-02
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Hi,

The DART working group has started a last call for =
draft-ietf-dart-dscp-rtp-02, ending on August 22. This informational =
draft describes considerations for the use of DiffServ code points in =
situations where one multiplexes multiple RTP packet streams, and =
potentially other protocol streams, that share the same 5-tuple.=20

Since this draft is likely of interest to several working groups, I =
would like to solicit additional reviews from participants of TSVWG, =
AVTCORE, MMUSIC, CLUE, RTCWEB, and RMCAT. Please send any feedback =
(including feedback to the effect of "this is ready to progress") to the =
DART working group mailing list.

Thanks!

Ben.

Begin forwarded message:

> Resent-From: draft-alias-bounces@tools.ietf.org
> From: Ben Campbell <ben@nostrum.com>
> Subject: WGLC: draft-ietf-dart-dscp-rtp-02
> Date: August 9, 2014 at 12:50:16 PM CDT
> Resent-To: ben@nostrum.com, david.black@emc.com, paulej@packetizer.com
> To: dart@ietf.org
> Cc: draft-ietf-dart-dscp-rtp.all@tools.ietf.org, mls.ietf@gmail.com, =
Richard Barnes <rlb@ipv.sx>
>=20
> This is a DART Working Group Last Call of draft-ietf-dart-dscp-rtp-02. =
It's available on the data tracker at the following URL:
>=20
> https://datatracker.ietf.org/doc/draft-ietf-dart-dscp-rtp/
>=20
> The WGLC will conclude on 22 August, 2014. Please send your comments =
to the authors and the DART mailing list. If you've reviewed it and =
think it's good to go, please say so.
>=20
> Thanks!
>=20
> Ben.
>=20


From nobody Mon Aug 11 03:53:23 2014
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Subject: [rtcweb] Negotiating the opening protocol (Re: AD review of draft-ietf-rtcweb-data-channel)
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On 08/08/2014 10:27 PM, Richard Barnes wrote:
> I have reviewed this document in preparation for IETF Last Call.  I 
> have gone ahead and requested last call, but there are some comments 
> below that I would like to be addressed before the document goes to 
> the IESG.
> --Richard
>
>
> MAJOR:
>
> Section 6.2.
> I'm very concerned about interoperability of data channels given that, 
> e.g., (1) DCEP [draft-ietf-rtcweb-data-protocol] is not required, and 
> (2) DCEP only works if it is used by both endpoints (it resets any 
> streams that are used without DCEP negotiation).  In order to avoid 
> interoperability failure of this type, whatever mechanism is used to 
> establish an SCTP association that is used for data channels must also 
> negotiate the mechanism(s) that will be used to establish data 
> channels over that association.  That could be directly (by 
> negotiating some data channels at the same time as the association), 
> or indirectly (by indicating that DCEP or some other protocol will be 
> used), but either way, the endpoints of the association need to know 
> how data channels will be negotiated before they can create any data 
> channels.  This section needs to state this requirement so that 
> applications using data channels can know that they need to meet it.

I *think* this is a concern we have discussed before, and decided that 
it's not relevant.

There are two states a receiver can be in:

1) It has external knowledge of what is expected on a particular data 
channel
2) It has no such external knowledge.

There are two actions a sender can do:

A) Send a data-protocol open packet
B) Send data without a data-protocol packet

This produces four outcomes:

1A: The sender sends an open packet that the receiver was not expecting.
1B: The sender sends data, and the receiver knows what to expect. All is OK.
2A: The sender sends an open packet, and the receiver knows from the 
packet what to do. All is OK.
2B: The sender sends data, and the receiver has no configuration to match.

1A and 2B are the problematic cases, and both can occur only as a result 
of misconfiguration, where the two sides have different opinions.

The WG decided to leave out the actual mechanism for this configuration 
from the spec, and most emphatically did not design an SDP-based 
mechanism to negotiate some channels.

The WG also decided that this decision is done for each channel 
independently - there is no need for an overall decision for all 
channels up front. As long as a channel number is free, either mechanism 
can start using it.

Summary: I disagree with the concern. I think this issue has been 
discussed, and the WG has found consensus that it is a problem that does 
not need to be solved.

Harald


From nobody Mon Aug 11 08:08:02 2014
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Subject: [rtcweb] EarlyBird registration deadline for SIPit 31 is Monday August 18
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The Early Bird registration deadline for SIPit 31 is Monday August 18.

If you haven't already registered, please do so as soon as you can.

RjS
> SIPit 31 will be held in Nice, France Sep 28 - Oct 3, 2014.
> The event will be hosted by ETSI.
>
> More information, and the registration page,  is available at
> <http://www.etsi.org/news-events/events/750-sipit-31>
> Please take advantage of the opportunity to register early.
>
> In addition to the normal team-to-team testing at the event,
> we will be concentrating on several specific interoperability areas:
> * Nat and Firewall Traversal using Outbound, TURN, and STUN
> * RTCWeb
> * Advanced Video scenarios
> * Use of TLS and DTLS
>
> If you're planning to attend and have other areas you would like
> have a multi-team focus session around, please let me know.
>
> RjS 


From nobody Mon Aug 11 11:14:02 2014
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Subject: [rtcweb] RETRACTION: Last Call: <draft-ietf-rtcweb-data-channel-11.txt> (WebRTC Data Channels) to Proposed Standard
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The IESG would like to retract the current Last Call on the following 
document:

- 'WebRTC Data Channels'
  <draft-ietf-rtcweb-data-channel-11.txt> as Proposed Standard

The document's state has been set back to "AD Evaluation."


From nobody Mon Aug 11 11:14:38 2014
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The IESG would like to retract the current Last Call on the following 
document:

- 'WebRTC Data Channel Establishment Protocol'
  <draft-ietf-rtcweb-data-protocol-07.txt> as Proposed Standard

The document's state has been set back to "AD Evaluation."


From nobody Mon Aug 11 11:42:37 2014
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Date: Mon, 11 Aug 2014 14:42:34 -0400
From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: [rtcweb] Some final comments on draft-ietf-rtcweb-data-channel-11
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I just did another read-through of this draft, and identified a few 
things that I think need to be cleared up:

Association Usage:

draft-ietf-mmusic-sctp-sdp specifies new SDP notation for describing 
SCTP associations. It is not specific to rtcweb. For SCTP m-lines, it is 
necessary to name an "association-usage". Support for rtcweb data 
channels is one possible usage.

Currently draft-ietf-mmusic-sctp-sdp gives all of its examples using 
"webrtc-datachannel" as the association-usage. It does *not* currently 
define an IANA registry for association-usage, but it should. But it 
really is the wrong place to define this specific usage. Certainly 
draft-ietf-rtcweb-data-channel provides the *definition* of the usage. 
IMO it also ought to define and do the IANA registration of the name. 
(Obviously some coordination is required between this draft and the 
sctp-sdp draft, so that they are aligned on this.)

Also, can we PLEASE use a name that not specific to webrtc? This *will* 
be used in contexts where webrtc is not being used. (At least by CLUE.)

I propose that the name be "data-channel".

SCTP port:

*Something* needs to talk about SCTP ports. Currently they aren't 
mentioned either here or in JSEP. Since this document makes the point 
that this will be a user layer stack, and talks about the layering over 
DTLS, I think they ought to be mentioned here. They also probably need 
to be mentioned in JSEP, as part of aligning to the most recent version 
of draft-ietf-mmusic-sctp-sdp.

Only a little bit needs to be stated: that SCTP supports multiple SCTP 
ports, but that a particular SCTP association uses a single sctp port 
pair. (One port at each end of the SCTP association.) JSEP needs to 
specify how these ports are negotiated. (Via reference to 
draft-ietf-mmusic-sctp-sdp.)

Normative requirement for DCEP support:

While this draft references ietf-rtcweb-data-protocol normatively, and 
talks about it, I can find no explicit statement that implementations of 
this draft MUST also implement DCEP. AFAIK negotiation of the sctp 
association-usage defined by this draft in intended to mean that DCEP is 
also supported.

	Thanks,
	Paul


From nobody Mon Aug 11 11:42:48 2014
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I just did another read-through of this draft, and identified a few 
things that I think need to be cleared up:

IIUC the data channel APIs expect that all channels, even those defined 
out of band, have a well defined set of properties. But those properties 
are defined in *this* draft that applies only to channels created using 
DCEP. OTOH, draft-ietf-rtcweb-data-channel-11 is much more vague about 
channel properties.

ISTM that most of the channel property specification should be moved to, 
or replicated in, draft-ietf-rtcweb-data-channel.

And this extends to the definitions for Channel Type, Priority, and 
Reliability since those also are intended to apply for the data channel 
API for channels defined out of band.

Alternatively, the data channel document could more extensively cross 
reference this document, or the two documents could be merged into one.

One more thing - section 4 says:

    Please note that
    there is also no attempt to resolve protocol glare.

I cannot figure out what this means. It is unambiguous which end is 
creating the channel, and that end is *declaring* the protocol to be 
used on it. There is no possibility of glare.

	Thanks,
	Paul


From nobody Mon Aug 11 14:26:26 2014
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Mon, 11 Aug 2014 23:26:01 +0200
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Subject: [rtcweb] "A DTLS-SRTP handshake establishes one or more SRTP crypto contexts"
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Hi, may somebody clarify what the following sentence in RFC 5764 (DTLS
"use_srtp" extension) Section 4.1 means?

"A DTLS-SRTP handshake establishes one or more SRTP crypto contexts"

AFAIU a DTLS-SRTP handshake generates keys for inbound and outbound
RTP or RTCP or RTP+RTCP media flows, this is, one or two. Is that what
the above sentence means?

Thanks a lot.

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


From nobody Mon Aug 11 14:37:31 2014
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Subject: Re: [rtcweb] "A DTLS-SRTP handshake establishes one or more SRTP crypto contexts"
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See:

http://tools.ietf.org/html/rfc3711#section-3.2.3

For the definition of a crypto context.

-Ekr



On Mon, Aug 11, 2014 at 2:26 PM, I=C3=B1aki Baz Castillo <ibc@aliax.net> wr=
ote:

> Hi, may somebody clarify what the following sentence in RFC 5764 (DTLS
> "use_srtp" extension) Section 4.1 means?
>
> "A DTLS-SRTP handshake establishes one or more SRTP crypto contexts"
>
> AFAIU a DTLS-SRTP handshake generates keys for inbound and outbound
> RTP or RTCP or RTP+RTCP media flows, this is, one or two. Is that what
> the above sentence means?
>
> Thanks a lot.
>
> --
> I=C3=B1aki Baz Castillo
> <ibc@aliax.net>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<div dir=3D"ltr">See:<div><br><div><a href=3D"http://tools.ietf.org/html/rf=
c3711#section-3.2.3">http://tools.ietf.org/html/rfc3711#section-3.2.3</a></=
div></div><div><br></div><div>For the definition of a crypto context.</div>=
<div>

<br></div><div>-Ekr</div><div><br></div></div><div class=3D"gmail_extra"><b=
r><br><div class=3D"gmail_quote">On Mon, Aug 11, 2014 at 2:26 PM, I=C3=B1ak=
i Baz Castillo <span dir=3D"ltr">&lt;<a href=3D"mailto:ibc@aliax.net" targe=
t=3D"_blank">ibc@aliax.net</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Hi, may somebody clarify what the following =
sentence in RFC 5764 (DTLS<br>
&quot;use_srtp&quot; extension) Section 4.1 means?<br>
<br>
&quot;A DTLS-SRTP handshake establishes one or more SRTP crypto contexts&qu=
ot;<br>
<br>
AFAIU a DTLS-SRTP handshake generates keys for inbound and outbound<br>
RTP or RTCP or RTP+RTCP media flows, this is, one or two. Is that what<br>
the above sentence means?<br>
<br>
Thanks a lot.<br>
<span class=3D"HOEnZb"><font color=3D"#888888"><br>
--<br>
I=C3=B1aki Baz Castillo<br>
&lt;<a href=3D"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt;<br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</font></span></blockquote></div><br></div>

--089e0102e6e687c3ad0500615c84--


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Subject: Re: [rtcweb] "A DTLS-SRTP handshake establishes one or more SRTP crypto contexts"
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2014-08-11 23:36 GMT+02:00 Eric Rescorla <ekr@rtfm.com>:
> See:
>
> http://tools.ietf.org/html/rfc3711#section-3.2.3
>
> For the definition of a crypto context.


Clear. Thanks a lot.

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


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--bcaec51f8f9d524f4505006212ea
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On Fri, Aug 8, 2014 at 3:01 AM, I=C3=B1aki Baz Castillo <ibc@aliax.net> wro=
te:

> Hi, let's assume the following scenario:
>
> - Browser and server (ICE-Lite) with just TCP candidates.
>
> - Browser sends the DTLS handshake over the TCP nominated (and single)
> connection.
>
> - Media begins.
>
> - Later the TCP connection is abruptly closed. The client want to
> "reset" the communication.
>

> So at this point I expect that:
>
> 1) Browser MUST open a new TCP connection (obviously).
>

Yes, per http://tools.ietf.org/html/rfc6544#section-11.1

>
> 2) When done it must send Binding Request for the server to allow media
> from it.
>

Yes, as indicated in that same section.

>
> 3) Browser MUST NOT perform again the DTLS handshake.
>
> The rationale on bullet 3) is that ICE is supposed to be a "virtual
> socket" and thus a single DTLS handshake must be done regardless which
> UDP/TCP/SCTP/X.25 transport is used to carried the DTLS data between
> peers.
>

Yes, the DTLS layer is not affected.

>
>
> 4) Said that, and assuming that there is also a UDP ICE valid pair, I
> also expect (and this is a bit exotic but IMHO it SHOULD work) that
> the browser should be able to send a DTLS Close Alert over the UDP
> valid pair (this is, even when the DTLS handshake and RTP is being
> sent over the TCP nominated pair).
>

Any DTLS close alert would nuke the entire DTLS connection, which is
currently fatal in Chrome.

--bcaec51f8f9d524f4505006212ea
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Fri, Aug 8, 2014 at 3:01 AM, I=C3=B1aki Baz Castillo <span dir=
=3D"ltr">&lt;<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.n=
et</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">Hi, let&#39;s assume the following scenario:<br>
<br>
- Browser and server (ICE-Lite) with just TCP candidates.<br>
<br>
- Browser sends the DTLS handshake over the TCP nominated (and single)<br>
connection.<br>
<br>
- Media begins.<br>
<br>
- Later the TCP connection is abruptly closed. The client want to<br>
&quot;reset&quot; the communication.=C2=A0<br></blockquote><blockquote clas=
s=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;b=
order-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"=
>
<br>
So at this point I expect that:<br>
<br>
1) Browser MUST open a new TCP connection (obviously).<br></blockquote><div=
><br></div><div>Yes, per <a href=3D"http://tools.ietf.org/html/rfc6544#sect=
ion-11.1">http://tools.ietf.org/html/rfc6544#section-11.1</a>=C2=A0</div><b=
lockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-le=
ft-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;pad=
ding-left:1ex">


<br>
2) When done it must send Binding Request for the server to allow media fro=
m it.<br></blockquote><div><br></div><div>Yes, as indicated in that same se=
ction.</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0=
.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-s=
tyle:solid;padding-left:1ex">


<br>
3) Browser MUST NOT perform again the DTLS handshake.<br>
<br>
The rationale on bullet 3) is that ICE is supposed to be a &quot;virtual<br=
>
socket&quot; and thus a single DTLS handshake must be done regardless which=
<br>
UDP/TCP/SCTP/X.25 transport is used to carried the DTLS data between<br>
peers.<br></blockquote><div><br></div><div>Yes, the DTLS layer is not affec=
ted.</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8=
ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-sty=
le:solid;padding-left:1ex">


<br>
<br>
4) Said that, and assuming that there is also a UDP ICE valid pair, I<br>
also expect (and this is a bit exotic but IMHO it SHOULD work) that<br>
the browser should be able to send a DTLS Close Alert over the UDP<br>
valid pair (this is, even when the DTLS handshake and RTP is being<br>
sent over the TCP nominated pair).<br></blockquote><div><br></div><div>Any =
DTLS close alert would nuke the entire DTLS connection, which is currently =
fatal in Chrome.</div></div></div></div>

--bcaec51f8f9d524f4505006212ea--


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Subject: Re: [rtcweb] ALPN question - other labels?
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>> For now I think it is more important that you be very clear what you intend.
>>
>> I think it means that CLUE implementations will always need to use "webrtc"
>> as the ALPN for the CLUE data channel, so that it will be right when
>> interoperating with WebRTC. That is possible, though it seems weird when not
>> interoperating.
> If the intent is to interoperate with WebRTC, by effectively sitting
> on top of the WebRTC stack, then I don't find that to be particularly
> strange.  That doesn't mean that you can't define a "clue" token,
> which may or may not be identical or at least very similar to
> "webrtc".

I think it would be simpler to offer a "webrtc-datachannel" ALPN in 
addition to the ones defined that allows an endpoint to indicate that it 
is only using datachannel .

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


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From: Kiran Kumar Guduru <kiran.guduru@samsung.com>
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Subject: Re: [rtcweb] New Version Notification for draft-guduru-rtcweb-codec-preferences-01.txt
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From nobody Tue Aug 12 02:00:40 2014
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From: "Schwarz, Albrecht (Albrecht)" <albrecht.schwarz@alcatel-lucent.com>
To: Justin Uberti <juberti@google.com>
Thread-Topic: [rtcweb] How to reset DTLS over TCP if the connection is closed
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Date: Tue, 12 Aug 2014 09:00:15 +0000
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Subject: Re: [rtcweb] How to reset DTLS over TCP if the connection is closed
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Subject: Re: [rtcweb] How to reset DTLS over TCP if the connection is closed
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2014-08-12 11:00 GMT+02:00 Schwarz, Albrecht (Albrecht)
<albrecht.schwarz@alcatel-lucent.com>:
> if finally TCP would be selected as L4 protocol (instead of UDP), then TL=
S
> would be used instead of DTLS, right?

No. It is STUN, DTLS and SRTP/SRTCP over TCP.

RFC 6544 (ICE TCP) allows both DTLS and TLS:

                       +----------+
                       |          |
                       |    App   |
            +----------+----------+     +----------+----------+
            |          |          |     |          |          |
            |   STUN   |  (D)TLS  |     |   STUN   |    App   |
            +----------+----------+     +----------+----------+
            |                     |     |                     |
            |      RFC 4571       |     |      RFC 4571       |
            +---------------------+     +---------------------+
            |                     |     |                     |
            |         TCP         |     |         TCP         |
            +---------------------+     +---------------------+
            |                     |     |                     |
            |         IP          |     |         IP          |
            +---------------------+     +---------------------+

              Figure 1: ICE TCP Stack with and without (D)TLS

but AFAIK WebRTC just uses DTLS and not TLS (note that we are not
talking about TURN in which TLS is supported).




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2014-08-12 0:27 GMT+02:00 Justin Uberti <juberti@google.com>:
>> 4) Said that, and assuming that there is also a UDP ICE valid pair, I
>> also expect (and this is a bit exotic but IMHO it SHOULD work) that
>> the browser should be able to send a DTLS Close Alert over the UDP
>> valid pair (this is, even when the DTLS handshake and RTP is being
>> sent over the TCP nominated pair).
>
>
> Any DTLS close alert would nuke the entire DTLS connection, which is
> currently fatal in Chrome.

Sure, I just meant that, theoretically, the DTLS alert can be sent
over any valid pair (even if it is not the selected one).

Thanks a lot. This is now clear for me, but IMHO we have a serious
problem given that RFC 5764 insists on "A single DTLS-SRTP session
only protects data carried over a single UDP source and destination
port pair".



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On Tue, Aug 12, 2014 at 2:33 AM, I=C3=B1aki Baz Castillo <ibc@aliax.net> wr=
ote:

> 2014-08-12 11:00 GMT+02:00 Schwarz, Albrecht (Albrecht)
> <albrecht.schwarz@alcatel-lucent.com>:
> > if finally TCP would be selected as L4 protocol (instead of UDP), then
> TLS
> > would be used instead of DTLS, right?
>
> No. It is STUN, DTLS and SRTP/SRTCP over TCP.
>
> RFC 6544 (ICE TCP) allows both DTLS and TLS:
>
>                        +----------+
>                        |          |
>                        |    App   |
>             +----------+----------+     +----------+----------+
>             |          |          |     |          |          |
>             |   STUN   |  (D)TLS  |     |   STUN   |    App   |
>             +----------+----------+     +----------+----------+
>             |                     |     |                     |
>             |      RFC 4571       |     |      RFC 4571       |
>             +---------------------+     +---------------------+
>             |                     |     |                     |
>             |         TCP         |     |         TCP         |
>             +---------------------+     +---------------------+
>             |                     |     |                     |
>             |         IP          |     |         IP          |
>             +---------------------+     +---------------------+
>
>               Figure 1: ICE TCP Stack with and without (D)TLS
>
> but AFAIK WebRTC just uses DTLS and not TLS (note that we are not
> talking about TURN in which TLS is supported).
>
>
Exactly. Even when TCP is used as a L4 protocol, there can still be packet
losses, as we will never buffer data when entering a flow control state.

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From: Justin Uberti <juberti@google.com>
Date: Tue, 12 Aug 2014 09:27:53 -0700
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Subject: Re: [rtcweb] How to reset DTLS over TCP if the connection is closed
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On Tue, Aug 12, 2014 at 3:19 AM, I=C3=B1aki Baz Castillo <ibc@aliax.net> wr=
ote:

> 2014-08-12 0:27 GMT+02:00 Justin Uberti <juberti@google.com>:
> >> 4) Said that, and assuming that there is also a UDP ICE valid pair, I
> >> also expect (and this is a bit exotic but IMHO it SHOULD work) that
> >> the browser should be able to send a DTLS Close Alert over the UDP
> >> valid pair (this is, even when the DTLS handshake and RTP is being
> >> sent over the TCP nominated pair).
> >
> >
> > Any DTLS close alert would nuke the entire DTLS connection, which is
> > currently fatal in Chrome.
>
> Sure, I just meant that, theoretically, the DTLS alert can be sent
> over any valid pair (even if it is not the selected one).
>

Yes.


>
> Thanks a lot. This is now clear for me, but IMHO we have a serious
> problem given that RFC 5764 insists on "A single DTLS-SRTP session
> only protects data carried over a single UDP source and destination
> port pair".


Agreed, this was written before the DTLS-ICE interaction had been fully
understood. Errata should be filed against 5764 so that it can be made more
consistent with ICE.

--001a1133901c7094150500712898
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Tue, Aug 12, 2014 at 3:19 AM, I=C3=B1aki Baz Castillo <span dir=
=3D"ltr">&lt;<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.n=
et</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">2014-08-12 0:27 GMT+02:00 Justin Uberti &lt;=
<a href=3D"mailto:juberti@google.com">juberti@google.com</a>&gt;:<br>
<div class=3D"">&gt;&gt; 4) Said that, and assuming that there is also a UD=
P ICE valid pair, I<br>
&gt;&gt; also expect (and this is a bit exotic but IMHO it SHOULD work) tha=
t<br>
&gt;&gt; the browser should be able to send a DTLS Close Alert over the UDP=
<br>
&gt;&gt; valid pair (this is, even when the DTLS handshake and RTP is being=
<br>
&gt;&gt; sent over the TCP nominated pair).<br>
&gt;<br>
&gt;<br>
&gt; Any DTLS close alert would nuke the entire DTLS connection, which is<b=
r>
&gt; currently fatal in Chrome.<br>
<br>
</div>Sure, I just meant that, theoretically, the DTLS alert can be sent<br=
>
over any valid pair (even if it is not the selected one).<br></blockquote><=
div><br></div><div>Yes.</div><div>=C2=A0</div><blockquote class=3D"gmail_qu=
ote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex=
">
<br>
Thanks a lot. This is now clear for me, but IMHO we have a serious<br>
problem given that RFC 5764 insists on &quot;A single DTLS-SRTP session<br>
only protects data carried over a single UDP source and destination<br>
port pair&quot;.</blockquote><div><br></div><div>Agreed, this was written b=
efore the DTLS-ICE interaction had been fully understood. Errata should be =
filed against 5764 so that it can be made more consistent with ICE.</div>

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

--001a1133901c7094150500712898--


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On Tue, Aug 12, 2014 at 9:27 AM, Justin Uberti <juberti@google.com> wrote:

>
>
>
> On Tue, Aug 12, 2014 at 3:19 AM, I=C3=B1aki Baz Castillo <ibc@aliax.net> =
wrote:
>
>> 2014-08-12 0:27 GMT+02:00 Justin Uberti <juberti@google.com>:
>> >> 4) Said that, and assuming that there is also a UDP ICE valid pair, I
>> >> also expect (and this is a bit exotic but IMHO it SHOULD work) that
>> >> the browser should be able to send a DTLS Close Alert over the UDP
>> >> valid pair (this is, even when the DTLS handshake and RTP is being
>> >> sent over the TCP nominated pair).
>> >
>> >
>> > Any DTLS close alert would nuke the entire DTLS connection, which is
>> > currently fatal in Chrome.
>>
>> Sure, I just meant that, theoretically, the DTLS alert can be sent
>> over any valid pair (even if it is not the selected one).
>>
>
> Yes.
>
>
>>
>> Thanks a lot. This is now clear for me, but IMHO we have a serious
>> problem given that RFC 5764 insists on "A single DTLS-SRTP session
>> only protects data carried over a single UDP source and destination
>> port pair".
>
>
> Agreed, this was written before the DTLS-ICE interaction had been fully
> understood. Errata should be filed against 5764 so that it can be made mo=
re
> consistent with ICE.
>

Agreed. I think this is just a textual problem, though. It's pretty clear
what
has to happen.

-Ekr

--f46d04389533b16b360500713a85
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Tue, Aug 12, 2014 at 9:27 AM, Justin Uberti <span dir=3D"ltr">&l=
t;<a href=3D"mailto:juberti@google.com" target=3D"_blank">juberti@google.co=
m</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div dir=3D"ltr"><br><div class=3D"gmail_ext=
ra"><br><br><div class=3D"gmail_quote"><div class=3D"">On Tue, Aug 12, 2014=
 at 3:19 AM, I=C3=B1aki Baz Castillo <span dir=3D"ltr">&lt;<a href=3D"mailt=
o:ibc@aliax.net" target=3D"_blank">ibc@aliax.net</a>&gt;</span> wrote:<br>



<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">2014-08-12 0:27 GMT+02:00 Justin Uberti &lt;=
<a href=3D"mailto:juberti@google.com" target=3D"_blank">juberti@google.com<=
/a>&gt;:<br>


<div>&gt;&gt; 4) Said that, and assuming that there is also a UDP ICE valid=
 pair, I<br>
&gt;&gt; also expect (and this is a bit exotic but IMHO it SHOULD work) tha=
t<br>
&gt;&gt; the browser should be able to send a DTLS Close Alert over the UDP=
<br>
&gt;&gt; valid pair (this is, even when the DTLS handshake and RTP is being=
<br>
&gt;&gt; sent over the TCP nominated pair).<br>
&gt;<br>
&gt;<br>
&gt; Any DTLS close alert would nuke the entire DTLS connection, which is<b=
r>
&gt; currently fatal in Chrome.<br>
<br>
</div>Sure, I just meant that, theoretically, the DTLS alert can be sent<br=
>
over any valid pair (even if it is not the selected one).<br></blockquote><=
div><br></div></div><div>Yes.</div><div class=3D""><div>=C2=A0</div><blockq=
uote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc =
solid;padding-left:1ex">


<br>
Thanks a lot. This is now clear for me, but IMHO we have a serious<br>
problem given that RFC 5764 insists on &quot;A single DTLS-SRTP session<br>
only protects data carried over a single UDP source and destination<br>
port pair&quot;.</blockquote><div><br></div></div><div>Agreed, this was wri=
tten before the DTLS-ICE interaction had been fully understood. Errata shou=
ld be filed against 5764 so that it can be made more consistent with ICE.</=
div>

</div></div></div></blockquote><div><br></div><div>Agreed. I think this is =
just a textual problem, though. It&#39;s pretty clear what</div><div>has to=
 happen.</div><div><br></div><div>-Ekr</div><div>=C2=A0</div></div><br></di=
v>

</div>

--f46d04389533b16b360500713a85--


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From: Dave Taht <dave.taht@gmail.com>
To: Ben Campbell <ben@nostrum.com>
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Subject: Re: [rtcweb] Fwd: WGLC: draft-ietf-dart-dscp-rtp-02
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This is also of some interest to me at least, and perhaps some members
of the AQM working group who are working out weighted fair queuing and
wred like approaches to these needs.

Anyway, I do try to track thinking in this area in the ongoing
development of cerowrt's sqm system, which has pluggable modules for
all the known aqm and fq schedulers, and a three tier system modeled
on wondershaper for managing traffic,
which uses what diffserv markings that I have observed make sense...

Some relevant links:

http://www.bufferbloat.net/projects/cerowrt/wiki/Wondershaper_Must_Die

http://www.bufferbloat.net/projects/cerowrt/wiki/Setting_up_SQM_for_CeroWrt=
_310

http://www.bufferbloat.net/projects/cerowrt/wiki/Smart_Queue_Management


Missing from this discussion so far is what is actually deployed out
there... I am only familiar with stuff like the above as an
egress/ingress queue management system...

AND: in the case of linux based wireless routers, none of the AF
codepoints are respected explicitly, and how the codepoints are mapped
are described by 802.11e - and that is far more about scheduling
priority for airtime rather than drop probability.

It is kind of hard to wrap your head around how 802.11e actually
works, but, this is a start:

http://en.wikipedia.org/wiki/IEEE_802.11e-2005#Enhanced_distributed_channel=
_access_.28EDCA.29

In linux wifi at least:

CS0, CS3 are mapped into the BE hw queue
CS1, CS2 are mapped into the BK hw queue
CS4, CS5 are mapped into the VI hw queue
CS6, CS7 are mapped into the VO  hw queue

Bit patterns for the other classes to whatever extent they map into
CSX are mapped into the underlying hardware queue.

I am not saying this is an optimal state of affairs, (as notably VO
should be as completely disabled as possible in the 802.11n and ac
age), and am very interested in how other OSes (ios, osx, windows),
and router/switch OSes do it, but the wireless scheduling aspect of
all these codepoints does not seem to be well touched upon.

On a personal note I tend to think that having 17+ ways to slice up
traffic is about 13 or 14 too many, particularly those targetting a
drop probability. When you have drop rates well below a few percentage
points, adding another percentage point of granularity seems futile.

Lastly, there is also an implementation issue in linux presently in
that doing extensive diffserv classification basically requires
one iptables or tc rule per class, which is inefficient.  Someone (not
me!) doing some code for doing diffserv saner there would go a long
way...




On Sat, Aug 9, 2014 at 11:48 AM, Ben Campbell <ben@nostrum.com> wrote:
> Hi,
>
> The DART working group has started a last call for draft-ietf-dart-dscp-r=
tp-02, ending on August 22. This informational draft describes consideratio=
ns for the use of DiffServ code points in situations where one multiplexes =
multiple RTP packet streams, and potentially other protocol streams, that s=
hare the same 5-tuple.
>
> Since this draft is likely of interest to several working groups, I would=
 like to solicit additional reviews from participants of TSVWG, AVTCORE, MM=
USIC, CLUE, RTCWEB, and RMCAT. Please send any feedback (including feedback=
 to the effect of "this is ready to progress") to the DART working group ma=
iling list.
>
> Thanks!
>
> Ben.
>
> Begin forwarded message:
>
>> Resent-From: draft-alias-bounces@tools.ietf.org
>> From: Ben Campbell <ben@nostrum.com>
>> Subject: WGLC: draft-ietf-dart-dscp-rtp-02
>> Date: August 9, 2014 at 12:50:16 PM CDT
>> Resent-To: ben@nostrum.com, david.black@emc.com, paulej@packetizer.com
>> To: dart@ietf.org
>> Cc: draft-ietf-dart-dscp-rtp.all@tools.ietf.org, mls.ietf@gmail.com, Ric=
hard Barnes <rlb@ipv.sx>
>>
>> This is a DART Working Group Last Call of draft-ietf-dart-dscp-rtp-02. I=
t's available on the data tracker at the following URL:
>>
>> https://datatracker.ietf.org/doc/draft-ietf-dart-dscp-rtp/
>>
>> The WGLC will conclude on 22 August, 2014. Please send your comments to =
the authors and the DART mailing list. If you've reviewed it and think it's=
 good to go, please say so.
>>
>> Thanks!
>>
>> Ben.
>>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb



--=20
Dave T=C3=A4ht

NSFW: https://w2.eff.org/Censorship/Internet_censorship_bills/russell_0296_=
indecent.article


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From nobody Wed Aug 13 01:35:26 2014
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the Real-Time Communication in WEB-browsers Working Group of the IETF.

        Title           : Transports for WebRTC
        Author          : Harald Alvestrand
	Filename        : draft-ietf-rtcweb-transports-06.txt
	Pages           : 14
	Date            : 2014-08-13

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


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-transports-06

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-transports-06


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 Aug 13 05:50:42 2014
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Finally got to this. Sorry it's taken so long.

On 07/18/2014 03:09 AM, Paul Kyzivat wrote:
> I haven't read the overview carefully in a long time. This time I did.
> I came away with the following thoughts:
>
> Terminology:
>
>    Media path:  The path that media data follows from one browser to
>       another.
>
> I don't think this is quite right by restricting to browsers. I think 
> you mean:
>
>    Media path:  The path that media data follows from one WebRTC device
>       to another.
Yep. Thanks!
>
> The intent is clear (e.g. from figure 2) but the definition doesn't 
> say it.
>
> Also, why aren't "WebRTC browser" and "WebRTC device" included in the 
> Terminology section? Seems like they should be.
Fixed now.
>
> Section 5:
>
>    Considerations for the transfer of data that is not in RTP format is
>    described in [I-D.ietf-rtcweb-data-channel], and a supporting
>    protocol is described in [I-D.ietf-rtcweb-data-protocol]. Webrtc
>    devices MUST implement these two specifications.
>
> Actually ietf-rtcweb-data-protocol only describes a protocol used for 
> data channel *establishment*. As such, I think it is technically a 
> Connection management protocol. This becomes more evident when you 
> consider all the discussion on how to establish channels via SDP.
>
> In principle I think ietf-rtcweb-data-protocol should be referenced 
> from section 7. If that seems heretical, then how about at least saying:
>
>    Considerations for the transfer of data that is not in RTP format is
>    described in [I-D.ietf-rtcweb-data-channel], and a supporting
>    protocol for establishing individual data channels is described in
>    [I-D.ietf-rtcweb-data-protocol].  Webrtc devices MUST implement
>    these two specifications.

Fix applied. I think it fits better here than in section 7, since 
section 7 deals with management of the overall connection.
>
> Otherwise this document is remarkably clear.

Thank you!

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


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Catching up on mail.....

On 07/23/2014 04:37 PM, Emil Ivov wrote:
> Hey all,
>
> Lately, we have been working on SFU-based conferencing. SFUs and 
> WebRTC go together quite well: SFUs scale beautifully and are very 
> latency friendly which makes them great for cloud deployment.
>
> But!
>
> There is a glass ceiling currently that I was hoping we could discuss.
>
> Performance.
>
> Currently the only way an SFU can operate is by terminating DTLS/SRTP 
> for every participant. This means decrypting and (worse) re-encrypting 
> all traffic that traverses the SFU. This easily accounts for 90% of 
> the CPU use on the SFU.

Do you have an actual measurement giving this figure? Over what 
configuration and AES implementation?

Last time I was evaluating this, I think profiling of the conferencing 
engine showed that less than 5% of the time was spent in the 
decrypt/encrypt functions - of course, we might just have been massively 
inefficient in the rest of the things that implementation was doing.

>
> This is not a difficult problem to solve and AVTCORE even has a WG 
> document that addresses it. It removes the need for decrypting and 
> allows SFUs to just forward packets.
>
> However, while chatting with many others in Toronto it appeared that 
> most of us are focused on solving a slightly different problem:
>
> Privacy.
>
> Or in other words: making it impossible (as opposed to unnecessary) 
> for SFUs to decrypt.
>
> While this is a notable effort, it is also much more complex and parts 
> of it (i.e. not requiring end users to trust their web provider) would 
> be arguably very hard to solve.
>
> Therefore, my question to the working group is: wouldn't it make sense 
> to try and solve the efficiency issue as a first step and then worry 
> about the privacy one on a later stage?
>
> Emil
>


From nobody Wed Aug 13 08:51:22 2014
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On Wed, Aug 13, 2014 at 9:36 AM, Harald Alvestrand <harald@alvestrand.no>
wrote:

> On 07/23/2014 04:37 PM, Emil Ivov wrote:
>
>> Currently the only way an SFU can operate is by terminating DTLS/SRTP for
>> every participant. This means decrypting and (worse) re-encrypting all
>> traffic that traverses the SFU. This easily accounts for 90% of the CPU use
>> on the SFU.
>>
>
> Do you have an actual measurement giving this figure? Over what
> configuration and AES implementation?
>
> Last time I was evaluating this, I think profiling of the conferencing
> engine showed that less than 5% of the time was spent in the
> decrypt/encrypt functions - of course, we might just have been massively
> inefficient in the rest of the things that implementation was doing.
>

I think we are talking about different conferencing servers here. In case
of audio conferencing server where all the audio streams are received,
de-jittered, decoded, mixed, re-encoded, and then re-encrypted,
decryption/encryption procedures account for about 5-10% of the total CPU
load. In case of packet forwarding unit, where streams from one participant
is forwarded to several other participants as is (i.e, each conference
participant receives audio and video stream from each other participant and
then mixes audio on the client side and displays each video stream
independently), the forwarding unit receives packets from the network
decrypts/encrypts and forwards the data. A typical modern dual CPU Xeon
server can do about 1-2GB of traffic in cases where it needs to do
decryption/encryption and forward traffic. In cases where the server can
simply forward packets, it can easily do around 10GB of traffic. In other
words, encryption accounts for 90% of the CPU load on such server.
_____________
Roman Shpount

--089e01493c541bf9ee050084c2a4
Content-Type: text/html; charset=UTF-8
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div>On Wed, Aug 13, 2014 at 9:=
36 AM, Harald Alvestrand <span dir=3D"ltr">&lt;<a href=3D"mailto:harald@alv=
estrand.no" target=3D"_blank">harald@alvestrand.no</a>&gt;</span> wrote:<br=
></div>
<div class=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"margi=
n:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204=
);border-left-style:solid;padding-left:1ex">On 07/23/2014 04:37 PM, Emil Iv=
ov wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">Currently the only way an SFU can operate is by terminatin=
g DTLS/SRTP for every participant. This means decrypting and (worse) re-enc=
rypting all traffic that traverses the SFU. This easily accounts for 90% of=
 the CPU use on the SFU.<br>

</blockquote>
<br>
Do you have an actual measurement giving this figure? Over what configurati=
on and AES implementation?<br>
<br>
Last time I was evaluating this, I think profiling of the conferencing engi=
ne showed that less than 5% of the time was spent in the decrypt/encrypt fu=
nctions - of course, we might just have been massively inefficient in the r=
est of the things that implementation was doing.<br>
</blockquote><div><br></div><div>I think we are talking about different con=
ferencing servers here. In case of audio conferencing server where all the =
audio streams are received, de-jittered, decoded, mixed, re-encoded, and th=
en re-encrypted, decryption/encryption procedures account for about 5-10% o=
f the total CPU load. In case of packet forwarding unit, where streams from=
 one participant is forwarded to several other participants as is (i.e, eac=
h conference participant receives audio and video stream from each other pa=
rticipant and then mixes audio on the client side and displays each video s=
tream independently), the forwarding unit receives packets from the network=
 decrypts/encrypts and forwards the data. A typical modern dual CPU Xeon se=
rver can do about 1-2GB of traffic in cases where it needs to do decryption=
/encryption and forward traffic. In cases where the server can simply forwa=
rd packets, it can easily do around 10GB of traffic. In other words, encryp=
tion accounts for 90% of the CPU load on such server.</div>
<div>_____________<br>Roman Shpount</div><div>=C2=A0</div></div></div></div=
>

--089e01493c541bf9ee050084c2a4--


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For a while now the draft has said:

   For UDP, this specification assumes the ability to set the DSCP code
   point of the sockets opened on a per-packet basis, in order to
   achieve the prioritizations described in [I-D.ietf-tsvwg-rtcweb-qos]
   (see Section 4.1) when multiple media types are multiplexed.  It does
   not assume that the DSCP codepoints will be honored, and does assume
   that they may be zeroed or changed, since this is a local
   configuration issue.

   Platforms that do not give access to these interfaces will not be
   able to support a conforming WebRTC implementation.

If a platform does not have this capability or does not honor the request
to use it (since it may be limited to privileged users), it seems possible
that an alternate approach is for the WebRTC implementation to suspend
cross-media type multiplexing when it is not available.  There's a cost to
that, especially for IPv4, but that still seems to support a conforming use
of WebRTC.

Would the working group support describing that alternative as conforming?

My concern here is that access to the ability to set different DSCP
markings on a per packet basis for a flow may not be available generally on
all platforms.  Even if we want to make a forward looking push toward that
ability, naming what we think is the appropriate alternative may make sense.

regards,

Ted (as an individual)

--90e6ba6149aabde6f705008506e3
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<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:garamond=
,serif;font-size:small">For a while now the draft has said:<br><pre>   For =
UDP, this specification assumes the ability to set the DSCP code
   point of the sockets opened on a per-packet basis, in order to
   achieve the prioritizations described in [I-D.ietf-tsvwg-rtcweb-qos]
   (see Section 4.1) when multiple media types are multiplexed.  It does
   not assume that the DSCP codepoints will be honored, and does assume
   that they may be zeroed or changed, since this is a local
   configuration issue.

   Platforms that do not give access to these interfaces will not be
   able to support a conforming WebRTC implementation.</pre>If a=20
platform does not have this capability or does not honor the request to=20
use it (since it may be limited to privileged users), it seems possible=20
that an alternate approach is for the WebRTC implementation to suspend=20
cross-media type multiplexing when it is not available.=C2=A0 There&#39;s a=
 cost to that, especially for=20
IPv4, but that still seems to support a conforming use of WebRTC.<br>
<br></div><div class=3D"gmail_default" style=3D"font-family:garamond,serif;=
font-size:small">Would the working group support describing that alternativ=
e as conforming?<br><br></div><div class=3D"gmail_default" style=3D"font-fa=
mily:garamond,serif;font-size:small">
My concern here is that access to the ability to set different DSCP marking=
s on a per packet basis for a flow may not be available generally on all pl=
atforms.=C2=A0 Even if we want to make a forward looking push toward that a=
bility, naming what we think is the appropriate alternative may make sense.=
<br>
<br>regards,<br><br>Ted (as an individual)<br></div><div class=3D"gmail_def=
ault" style=3D"font-family:garamond,serif;font-size:small"><br></div></div>

--90e6ba6149aabde6f705008506e3--


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On 13 August 2014 09:10, Ted Hardie <ted.ietf@gmail.com> wrote:
> My concern here is that access to the ability to set different DSCP markings
> on a per packet basis for a flow may not be available generally on all
> platforms.  Even if we want to make a forward looking push toward that
> ability, naming what we think is the appropriate alternative may make sense.

Yeah, requiring per-packet DSCP seems overly enthusiastic.


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Subject: Re: [rtcweb] TLS or DTLS?; RE:  How to reset DTLS over TCP if the connection is closed
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On Wed, Aug 13, 2014 at 12:34 AM, Schwarz, Albrecht (Albrecht) <
albrecht.schwarz@alcatel-lucent.com> wrote:

> Right, this is roughly described in the overview section, which provides
> only informative statements, but not any normative behaviour.
> The later sections talk about a "TLS or DTLS-SRTP session" (and not a TLS=
,
> DTLS or DTLS-SRTP session), which would be interpreted that the key
> exchange for SRTP uses DTLS-SRTP, but the "application" itself would use =
a
> TLS/RFC 4571/TCP transport.
>

If DTLS-SRTP/4571/TCP is permitted, then clearly application/DTLS/4571/TCP
must be permitted, especially in the context of BUNDLE.

You could of course use application/TLS/4571/TCP as well, but for real-time
applications I think that is generally inadvisable, since such applications
will typically not make any reliability guarantees regarding the ICE layer
(e.g. may discard outgoing packets when flow control is encountered)


> If RFC 6544 would allow an "application/DTLS/RFC 4571/TCP" transport, the=
n
> I would expect some information how the interactions between the DTLS and
> TCP layers are resolved in case of packet loss ("given the two separate
> functions how packet loss is handled by TCP and DTLS"), right?


> For instance, RFC 6544 could state that a "RFC 6544 DTLS session" would
> need to be profiled for emulating TLS behaviour by a correspondent
> alignment of TCP retransmission timer and DTLS retransmission timer
> settings ... (might be feasible, but looks odd, why not TLS instead?).
>

TLS requires a reliable transport, but in the 6544 case, TCP is not being
used to guarantee reliability, but rather, provide connectivity in
situations where use of UDP is not possible. TCP optimizations for
real-time apps, such as Minion, will only exacerbate this. Therefore use of
TLS is contraindicated in these situations.

It would probably be worthwhile for DTLS and any higher-level protocols
(e.g. SCTP) to adjust their timers when running over TCP.

>
> Do I miss sth.?
>
> Albrecht
>
> -----Original Message-----
> From: I=C3=B1aki Baz Castillo [mailto:ibc@aliax.net]
> Sent: Dienstag, 12. August 2014 11:34
> To: Schwarz, Albrecht (Albrecht)
> Cc: Justin Uberti; rtcweb@ietf.org
> Subject: Re: [rtcweb] How to reset DTLS over TCP if the connection is
> closed
>
> 2014-08-12 11:00 GMT+02:00 Schwarz, Albrecht (Albrecht)
> <albrecht.schwarz@alcatel-lucent.com>:
> > if finally TCP would be selected as L4 protocol (instead of UDP), then
> > TLS would be used instead of DTLS, right?
>
> No. It is STUN, DTLS and SRTP/SRTCP over TCP.
>
> RFC 6544 (ICE TCP) allows both DTLS and TLS:
>
>                        +----------+
>                        |          |
>                        |    App   |
>             +----------+----------+     +----------+----------+
>             |          |          |     |          |          |
>             |   STUN   |  (D)TLS  |     |   STUN   |    App   |
>             +----------+----------+     +----------+----------+
>             |                     |     |                     |
>             |      RFC 4571       |     |      RFC 4571       |
>             +---------------------+     +---------------------+
>             |                     |     |                     |
>             |         TCP         |     |         TCP         |
>             +---------------------+     +---------------------+
>             |                     |     |                     |
>             |         IP          |     |         IP          |
>             +---------------------+     +---------------------+
>
>               Figure 1: ICE TCP Stack with and without (D)TLS
>
> but AFAIK WebRTC just uses DTLS and not TLS (note that we are not talking
> about TURN in which TLS is supported).
>
>
>
>
> --
> I=C3=B1aki Baz Castillo
> <ibc@aliax.net>
>
>
> From: Schwarz, Albrecht (Albrecht)
> Sent: Dienstag, 12. August 2014 11:00
> To: 'Justin Uberti'
> Cc: rtcweb@ietf.org
> Subject: RE: [rtcweb] How to reset DTLS over TCP if the connection is
> closed
>
> Justin,
> question for clarification:
>
> if finally TCP would be selected as L4 protocol (instead of UDP), then TL=
S
> would be used instead of DTLS, right?
> (If yes, then discussion would be about =E2=80=9Creset TLS over TCP =E2=
=80=A6=E2=80=9D in
> understanding).
>
> -Albrecht
>
>

--bcaec548a399907fa8050088ebd1
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Wed, Aug 13, 2014 at 12:34 AM, Schwarz, Albrecht (Albrecht) <spa=
n dir=3D"ltr">&lt;<a href=3D"mailto:albrecht.schwarz@alcatel-lucent.com" ta=
rget=3D"_blank">albrecht.schwarz@alcatel-lucent.com</a>&gt;</span> wrote:<b=
r>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Right, this is roughly described in the over=
view section, which provides only informative statements, but not any norma=
tive behaviour.<br>


The later sections talk about a &quot;TLS or DTLS-SRTP session&quot; (and n=
ot a TLS, DTLS or DTLS-SRTP session), which would be interpreted that the k=
ey exchange for SRTP uses DTLS-SRTP, but the &quot;application&quot; itself=
 would use a TLS/RFC 4571/TCP transport.<br>

</blockquote><div><br></div><div>If DTLS-SRTP/4571/TCP is permitted, then c=
learly application/DTLS/4571/TCP must be permitted, especially in the conte=
xt of BUNDLE.</div><div><br></div><div>You could of course use application/=
TLS/4571/TCP as well, but for real-time applications I think that is genera=
lly inadvisable, since such applications will typically not make any reliab=
ility guarantees regarding the ICE layer (e.g. may discard outgoing packets=
 when flow control is encountered)</div>

<div><br></div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex=
;border-left:1px #ccc solid;padding-left:1ex">
<br>
If RFC 6544 would allow an &quot;application/DTLS/RFC 4571/TCP&quot; transp=
ort, then I would expect some information how the interactions between the =
DTLS and TCP layers are resolved in case of packet loss (&quot;given the tw=
o separate functions how packet loss is handled by TCP and DTLS&quot;), rig=
ht?=C2=A0</blockquote>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<br>
For instance, RFC 6544 could state that a &quot;RFC 6544 DTLS session&quot;=
 would need to be profiled for emulating TLS behaviour by a correspondent a=
lignment of TCP retransmission timer and DTLS retransmission timer settings=
 ... (might be feasible, but looks odd, why not TLS instead?).<br>

</blockquote><div><br></div><div>TLS requires a reliable transport, but in =
the 6544 case, TCP is not being used to guarantee reliability, but rather, =
provide connectivity in situations where use of UDP is not possible. TCP op=
timizations for real-time apps, such as Minion, will only exacerbate this. =
Therefore use of TLS is contraindicated in these situations.</div>

<div><br></div><div>It would probably be worthwhile for DTLS and any higher=
-level protocols (e.g. SCTP) to adjust their timers when running over TCP.<=
/div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-le=
ft:1px #ccc solid;padding-left:1ex">


<br>
Do I miss sth.?<br>
<br>
Albrecht<br>
<br>
-----Original Message-----<br>
From: I=C3=B1aki Baz Castillo [mailto:<a href=3D"mailto:ibc@aliax.net">ibc@=
aliax.net</a>]<br>
Sent: Dienstag, 12. August 2014 11:34<br>
To: Schwarz, Albrecht (Albrecht)<br>
Cc: Justin Uberti; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><b=
r>
Subject: Re: [rtcweb] How to reset DTLS over TCP if the connection is close=
d<br>
<br>
2014-08-12 11:00 GMT+02:00 Schwarz, Albrecht (Albrecht)<br>
&lt;<a href=3D"mailto:albrecht.schwarz@alcatel-lucent.com">albrecht.schwarz=
@alcatel-lucent.com</a>&gt;:<br>
&gt; if finally TCP would be selected as L4 protocol (instead of UDP), then=
<br>
&gt; TLS would be used instead of DTLS, right?<br>
<br>
No. It is STUN, DTLS and SRTP/SRTCP over TCP.<br>
<br>
RFC 6544 (ICE TCP) allows both DTLS and TLS:<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0+----------+<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0| =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0|<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0| =C2=A0 =C2=A0App =C2=A0 |<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 +----------+----------+ =C2=A0 =
=C2=A0 +----------+----------+<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 | =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0| =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0| =C2=A0 =C2=A0 | =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0| =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0|<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 | =C2=A0 STUN =C2=A0 | =C2=A0(D)T=
LS =C2=A0| =C2=A0 =C2=A0 | =C2=A0 STUN =C2=A0 | =C2=A0 =C2=A0App =C2=A0 |<b=
r>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 +----------+----------+ =C2=A0 =
=C2=A0 +----------+----------+<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 | =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 | =C2=A0 =C2=A0 | =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 |<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 | =C2=A0 =C2=A0 =C2=A0RFC 4571 =
=C2=A0 =C2=A0 =C2=A0 | =C2=A0 =C2=A0 | =C2=A0 =C2=A0 =C2=A0RFC 4571 =C2=A0 =
=C2=A0 =C2=A0 |<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 +---------------------+ =C2=A0 =
=C2=A0 +---------------------+<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 | =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 | =C2=A0 =C2=A0 | =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 |<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 | =C2=A0 =C2=A0 =C2=A0 =C2=A0 TCP=
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 | =C2=A0 =C2=A0 | =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
TCP =C2=A0 =C2=A0 =C2=A0 =C2=A0 |<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 +---------------------+ =C2=A0 =
=C2=A0 +---------------------+<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 | =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 | =C2=A0 =C2=A0 | =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 |<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 | =C2=A0 =C2=A0 =C2=A0 =C2=A0 IP =
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0| =C2=A0 =C2=A0 | =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 IP =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0|<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 +---------------------+ =C2=A0 =
=C2=A0 +---------------------+<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 Figure 1: ICE TCP Stack wi=
th and without (D)TLS<br>
<br>
but AFAIK WebRTC just uses DTLS and not TLS (note that we are not talking a=
bout TURN in which TLS is supported).<br>
<br>
<br>
<br>
<br>
--<br>
I=C3=B1aki Baz Castillo<br>
&lt;<a href=3D"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt;<br>
<br>
<br>
From: Schwarz, Albrecht (Albrecht)<br>
Sent: Dienstag, 12. August 2014 11:00<br>
To: &#39;Justin Uberti&#39;<br>
Cc: <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
Subject: RE: [rtcweb] How to reset DTLS over TCP if the connection is close=
d<br>
<br>
Justin,<br>
question for clarification:<br>
<br>
if finally TCP would be selected as L4 protocol (instead of UDP), then TLS =
would be used instead of DTLS, right?<br>
(If yes, then discussion would be about =E2=80=9Creset TLS over TCP =E2=80=
=A6=E2=80=9D in understanding).<br>
<span class=3D"HOEnZb"><font color=3D"#888888"><br>
-Albrecht<br>
<br>
</font></span></blockquote></div><br></div></div>

--bcaec548a399907fa8050088ebd1--


From nobody Wed Aug 13 18:46:46 2014
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Subject: [rtcweb] I-D Action: draft-ietf-rtcweb-stun-consent-freshness-06.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 Working Group of the IETF.

        Title           : STUN Usage for Consent Freshness
        Authors         : Muthu Arul Mozhi Perumal
                          Dan Wing
                          Ram Mohan Ravindranath
                          Tirumaleswar Reddy
                          Martin Thomson
	Filename        : draft-ietf-rtcweb-stun-consent-freshness-06.txt
	Pages           : 9
	Date            : 2014-08-13

Abstract:
   To prevent sending excessive traffic to an endpoint, periodic consent
   needs to be obtained from that remote endpoint.

   This document describes a consent mechanism using a new STUN usage.
   This same mechanism can also determine connection loss ("liveness")
   with a remote peer.


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-stun-consent-freshness-06

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-stun-consent-freshness-06


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/


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On 08/13/2014 09:07 PM, Martin Thomson wrote:
> On 13 August 2014 09:10, Ted Hardie <ted.ietf@gmail.com> wrote:
>> My concern here is that access to the ability to set different DSCP markings
>> on a per packet basis for a flow may not be available generally on all
>> platforms.  Even if we want to make a forward looking push toward that
>> ability, naming what we think is the appropriate alternative may make sense.
> Yeah, requiring per-packet DSCP seems overly enthusiastic.

I know it's possible to do this on Linux (for UDP, of course). I think 
it still requires an extra sysctl before sending each packet, which is 
reasonably high overhead if your app is sysctl-limited. But I don't know 
of any case where you would be allowed to set per-flow DSCP markings and 
would not be allowed to set per-packet DSCP markings; all the reasons to 
not do marking seem to be reasons to not do marking.

I'd like to know if the APIs exist on other platforms (whether they're 
overridden by default policy or not).

The two alternatives if we don't get per-packet setting are:

- Deaggregate, so that per-port-flow classifiers can do their thing
- Don't ask the network to differentiate, keep precedence of flow a 
sender-stack thing only

I don't know which is the more (or less) distasteful / useful of those 
two paths.


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


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From: Ted Hardie <ted.ietf@gmail.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>, Sean Turner <turners@ieca.com>,  Cullen Jennings <fluffy@cisco.com>
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Subject: [rtcweb] WG Last Call for draft-ietf-rtcweb-stun-consent-freshness
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This starts a WG Last Call for draft-ietf-rtcweb-stun-consent-freshness
(available at
http://datatracker.ietf.org/doc/draft-ietf-rtcweb-stun-consent-freshness/).
Please review the document and send comments to the list by September 10,
2014.

thanks,

Ted Hardie

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<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:garamond=
,serif;font-size:small">This starts a WG Last Call for<font> <span style=3D=
"font-weight:normal">draft-ietf-rtcweb-stun-consent-freshness (available at=
 <a href=3D"http://datatracker.ietf.org/doc/draft-ietf-rtcweb-stun-consent-=
freshness/">http://datatracker.ietf.org/doc/draft-ietf-rtcweb-stun-consent-=
freshness/</a>).=C2=A0 Please review the document and send comments to the =
list by September 10, 2014.<br>
<br></span></font></div><div class=3D"gmail_default" style=3D"font-family:g=
aramond,serif;font-size:small"><font><span style=3D"font-weight:normal">tha=
nks,<br><br>Ted Hardie<br></span></font></div></div>

--90e6ba6149aa9b21aa0500984c9d--


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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
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They have now been uploaded to IETF site. If any additional problems are fo=
und, we can still correct them.=20

Thanks, the chairs =85 =


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From: Justin Uberti <juberti@google.com>
Date: Thu, 14 Aug 2014 11:17:36 -0700
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Subject: Re: [rtcweb] -transports- discussion of DSCP
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On Thu, Aug 14, 2014 at 1:18 AM, Harald Alvestrand <harald@alvestrand.no>
wrote:

> On 08/13/2014 09:07 PM, Martin Thomson wrote:
>
>> On 13 August 2014 09:10, Ted Hardie <ted.ietf@gmail.com> wrote:
>>
>>> My concern here is that access to the ability to set different DSCP
>>> markings
>>> on a per packet basis for a flow may not be available generally on all
>>> platforms.  Even if we want to make a forward looking push toward that
>>> ability, naming what we think is the appropriate alternative may make
>>> sense.
>>>
>> Yeah, requiring per-packet DSCP seems overly enthusiastic.
>>
>
> I know it's possible to do this on Linux (for UDP, of course). I think it
> still requires an extra sysctl before sending each packet, which is
> reasonably high overhead if your app is sysctl-limited. But I don't know of
> any case where you would be allowed to set per-flow DSCP markings and would
> not be allowed to set per-packet DSCP markings; all the reasons to not do
> marking seem to be reasons to not do marking.
>
> I'd like to know if the APIs exist on other platforms (whether they're
> overridden by default policy or not).
>
> The two alternatives if we don't get per-packet setting are:
>
> - Deaggregate, so that per-port-flow classifiers can do their thing
> - Don't ask the network to differentiate, keep precedence of flow a
> sender-stack thing only
>
> I don't know which is the more (or less) distasteful / useful of those two
> paths.


The other option, which I believe works on MacOS, is to setsockopt before
each sendto:

    setsockopt(socket_, IPPROTO_IP, IP_TOS, &dscp, sizeof(dscp));

--bcaec51f8f9d87d88e05009aec44
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<div dir=3D"ltr"><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quo=
te">On Thu, Aug 14, 2014 at 1:18 AM, Harald Alvestrand <span dir=3D"ltr">&l=
t;<a href=3D"mailto:harald@alvestrand.no" target=3D"_blank">harald@alvestra=
nd.no</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><div class=3D"">On 08/13/2014 09:07 PM, Martin Thomson wro=
te:<br>


<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
On 13 August 2014 09:10, Ted Hardie &lt;<a href=3D"mailto:ted.ietf@gmail.co=
m" target=3D"_blank">ted.ietf@gmail.com</a>&gt; wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
My concern here is that access to the ability to set different DSCP marking=
s<br>
on a per packet basis for a flow may not be available generally on all<br>
platforms.=C2=A0 Even if we want to make a forward looking push toward that=
<br>
ability, naming what we think is the appropriate alternative may make sense=
.<br>
</blockquote>
Yeah, requiring per-packet DSCP seems overly enthusiastic.<br>
</blockquote>
<br></div>
I know it&#39;s possible to do this on Linux (for UDP, of course). I think =
it still requires an extra sysctl before sending each packet, which is reas=
onably high overhead if your app is sysctl-limited. But I don&#39;t know of=
 any case where you would be allowed to set per-flow DSCP markings and woul=
d not be allowed to set per-packet DSCP markings; all the reasons to not do=
 marking seem to be reasons to not do marking.<br>


<br>
I&#39;d like to know if the APIs exist on other platforms (whether they&#39=
;re overridden by default policy or not).<br>
<br>
The two alternatives if we don&#39;t get per-packet setting are:<br>
<br>
- Deaggregate, so that per-port-flow classifiers can do their thing<br>
- Don&#39;t ask the network to differentiate, keep precedence of flow a sen=
der-stack thing only<br>
<br>
I don&#39;t know which is the more (or less) distasteful / useful of those =
two paths.</blockquote><div><br></div><div>The other option, which I believ=
e works on MacOS, is to setsockopt before each sendto:</div><div>=C2=A0 =C2=
=A0=C2=A0</div>

<div>=C2=A0 =C2=A0 setsockopt(socket_, IPPROTO_IP, IP_TOS, &amp;dscp, sizeo=
f(dscp));=C2=A0</div></div></div></div>

--bcaec51f8f9d87d88e05009aec44--


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From: Martin Thomson <martin.thomson@gmail.com>
To: Justin Uberti <juberti@google.com>
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Subject: Re: [rtcweb] -transports- discussion of DSCP
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On 14 August 2014 11:17, Justin Uberti <juberti@google.com> wrote:
> The other option, which I believe works on MacOS, is to setsockopt before
> each sendto:
>
>     setsockopt(socket_, IPPROTO_IP, IP_TOS, &dscp, sizeof(dscp));

That is a solution to the problem, not a fallback position :)

I don't mind whether we require the fallback to be no markings, or we
specify a whole-flow marking that can be used (which is functionally
the same has having no markings, though much harder to define
consistent behaviour as things change), or whether we require
disaggregation.


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From: Roman Shpount <roman@telurix.com>
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Subject: Re: [rtcweb] -transports- discussion of DSCP
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--e89a8f647859ac7c1d05009cfc6e
Content-Type: text/plain; charset=UTF-8

On Thu, Aug 14, 2014 at 2:17 PM, Justin Uberti <juberti@google.com> wrote:

>
>
> On Thu, Aug 14, 2014 at 1:18 AM, Harald Alvestrand <harald@alvestrand.no>
> wrote:
>
>> On 08/13/2014 09:07 PM, Martin Thomson wrote:
>>
>>> On 13 August 2014 09:10, Ted Hardie <ted.ietf@gmail.com> wrote:
>>>
>>>> My concern here is that access to the ability to set different DSCP
>>>> markings
>>>> on a per packet basis for a flow may not be available generally on all
>>>> platforms.  Even if we want to make a forward looking push toward that
>>>> ability, naming what we think is the appropriate alternative may make
>>>> sense.
>>>>
>>> Yeah, requiring per-packet DSCP seems overly enthusiastic.
>>>
>>
>> I know it's possible to do this on Linux (for UDP, of course). I think it
>> still requires an extra sysctl before sending each packet, which is
>> reasonably high overhead if your app is sysctl-limited. But I don't know of
>> any case where you would be allowed to set per-flow DSCP markings and would
>> not be allowed to set per-packet DSCP markings; all the reasons to not do
>> marking seem to be reasons to not do marking.
>>
>> I'd like to know if the APIs exist on other platforms (whether they're
>> overridden by default policy or not).
>>
>> The two alternatives if we don't get per-packet setting are:
>>
>> - Deaggregate, so that per-port-flow classifiers can do their thing
>> - Don't ask the network to differentiate, keep precedence of flow a
>> sender-stack thing only
>>
>> I don't know which is the more (or less) distasteful / useful of those
>> two paths.
>
>
> The other option, which I believe works on MacOS, is to setsockopt before
> each sendto:
>
>     setsockopt(socket_, IPPROTO_IP, IP_TOS, &dscp, sizeof(dscp));
>
>
And on windows you can do the same by calling the following before sending
each UDP packet:
setsockopt(d_socket, IPPROTO_IP, IP_TOS, (char*)&tos, sizeof(tos));

I believe this will work on any BSD based network stack.

The problem with this approach is that it requires two system calls for
each sent packet. It also has some implications on threading, since without
setting TOS you can send data via the same socket without locking. Now,
because each send is two system calls, lock will be required to ensure
right TOS value, so the overhead can be even higher. There was a discussion
about updating Linux kernel to add support for setting IP_TOS via the
control structure in sendmsg, but I do not think this patch was ever
accepted.

It was also recommended that ICE packets are sent with the same TOS as main
data. Technically, you can end up with the situation where some TOS values
go through the network and some do not. If data is sent with a different
TOS value then ICE, ICE connectivity can succeed but sending the data will
fail. This means an ICE connectivity check for each TOS value, which might
produce different ICE candidates for different TOS values.

Taking all of this into account I would suggest that a single TOS value
should be used for each flow and data with different TOS values should be
dis-aggregated across different flows.
_____________
Roman Shpount

--e89a8f647859ac7c1d05009cfc6e
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On T=
hu, Aug 14, 2014 at 2:17 PM, Justin Uberti <span dir=3D"ltr">&lt;<a href=3D=
"mailto:juberti@google.com" target=3D"_blank">juberti@google.com</a>&gt;</s=
pan> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><div dir=3D"ltr"><div class=3D"gmail_extra"><br><br><div c=
lass=3D"gmail_quote">
On Thu, Aug 14, 2014 at 1:18 AM, Harald Alvestrand <span dir=3D"ltr">&lt;<a=
 href=3D"mailto:harald@alvestrand.no" target=3D"_blank">harald@alvestrand.n=
o</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><div>On 08/13/2014 09:07 PM, Martin Thomson wrote:<br>


<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
On 13 August 2014 09:10, Ted Hardie &lt;<a href=3D"mailto:ted.ietf@gmail.co=
m" target=3D"_blank">ted.ietf@gmail.com</a>&gt; wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
My concern here is that access to the ability to set different DSCP marking=
s<br>
on a per packet basis for a flow may not be available generally on all<br>
platforms.=C2=A0 Even if we want to make a forward looking push toward that=
<br>
ability, naming what we think is the appropriate alternative may make sense=
.<br>
</blockquote>
Yeah, requiring per-packet DSCP seems overly enthusiastic.<br>
</blockquote>
<br></div>
I know it&#39;s possible to do this on Linux (for UDP, of course). I think =
it still requires an extra sysctl before sending each packet, which is reas=
onably high overhead if your app is sysctl-limited. But I don&#39;t know of=
 any case where you would be allowed to set per-flow DSCP markings and woul=
d not be allowed to set per-packet DSCP markings; all the reasons to not do=
 marking seem to be reasons to not do marking.<br>



<br>
I&#39;d like to know if the APIs exist on other platforms (whether they&#39=
;re overridden by default policy or not).<br>
<br>
The two alternatives if we don&#39;t get per-packet setting are:<br>
<br>
- Deaggregate, so that per-port-flow classifiers can do their thing<br>
- Don&#39;t ask the network to differentiate, keep precedence of flow a sen=
der-stack thing only<br>
<br>
I don&#39;t know which is the more (or less) distasteful / useful of those =
two paths.</blockquote><div><br></div><div>The other option, which I believ=
e works on MacOS, is to setsockopt before each sendto:</div><div>=C2=A0 =C2=
=A0=C2=A0</div>


<div>=C2=A0 =C2=A0 setsockopt(socket_, IPPROTO_IP, IP_TOS, &amp;dscp, sizeo=
f(dscp));=C2=A0</div></div></div></div>
<br></blockquote><div><br></div><div>And on windows you can do the same by =
calling the following before sending each UDP packet:</div>setsockopt(d_soc=
ket, IPPROTO_IP, IP_TOS, (char*)&amp;tos, sizeof(tos));</div><div class=3D"=
gmail_quote">
<br></div><div class=3D"gmail_quote">I believe this will work on any BSD ba=
sed network stack.</div><div class=3D"gmail_quote"><br></div><div class=3D"=
gmail_quote">The problem with this approach is that it requires two system =
calls for each sent packet. It also has some implications on threading, sin=
ce without setting TOS you can send data via the same socket without lockin=
g. Now, because each send is two system calls, lock will be required to ens=
ure right TOS value, so the overhead can be even higher. There was a discus=
sion about updating Linux kernel to add support for setting IP_TOS via the =
control structure in sendmsg, but I do not think this patch was ever accept=
ed.<br clear=3D"all">
<div><br></div><div>It was also recommended that ICE packets are sent with =
the same TOS as main data. Technically, you can end up with the situation w=
here some TOS values go through the network and some do not. If data is sen=
t with a different TOS value then ICE, ICE connectivity can succeed but sen=
ding the data will fail. This means an ICE connectivity check for each TOS =
value, which might produce different ICE candidates for different TOS value=
s.</div>
<div><br></div><div>Taking all of this into account I would suggest that a =
single TOS value should be used for each flow and data with different TOS v=
alues should be dis-aggregated across different flows.</div><div>__________=
___<br>
Roman Shpount</div><br><div>=C2=A0</div></div></div></div>

--e89a8f647859ac7c1d05009cfc6e--


From nobody Thu Aug 14 14:52:31 2014
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Subject: Re: [rtcweb] -transports- discussion of DSCP
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On 14 Aug 2014, at 22:45, Roman Shpount <roman@telurix.com> wrote:

> On Thu, Aug 14, 2014 at 2:17 PM, Justin Uberti <juberti@google.com> =
wrote:
>=20
>=20
> On Thu, Aug 14, 2014 at 1:18 AM, Harald Alvestrand =
<harald@alvestrand.no> wrote:
> On 08/13/2014 09:07 PM, Martin Thomson wrote:
> On 13 August 2014 09:10, Ted Hardie <ted.ietf@gmail.com> wrote:
> My concern here is that access to the ability to set different DSCP =
markings
> on a per packet basis for a flow may not be available generally on all
> platforms.  Even if we want to make a forward looking push toward that
> ability, naming what we think is the appropriate alternative may make =
sense.
> Yeah, requiring per-packet DSCP seems overly enthusiastic.
>=20
> I know it's possible to do this on Linux (for UDP, of course). I think =
it still requires an extra sysctl before sending each packet, which is =
reasonably high overhead if your app is sysctl-limited. But I don't know =
of any case where you would be allowed to set per-flow DSCP markings and =
would not be allowed to set per-packet DSCP markings; all the reasons to =
not do marking seem to be reasons to not do marking.
>=20
> I'd like to know if the APIs exist on other platforms (whether they're =
overridden by default policy or not).
>=20
> The two alternatives if we don't get per-packet setting are:
>=20
> - Deaggregate, so that per-port-flow classifiers can do their thing
> - Don't ask the network to differentiate, keep precedence of flow a =
sender-stack thing only
>=20
> I don't know which is the more (or less) distasteful / useful of those =
two paths.
>=20
> The other option, which I believe works on MacOS, is to setsockopt =
before each sendto:
>    =20
>     setsockopt(socket_, IPPROTO_IP, IP_TOS, &dscp, sizeof(dscp));=20
>=20
>=20
> And on windows you can do the same by calling the following before =
sending each UDP packet:
> setsockopt(d_socket, IPPROTO_IP, IP_TOS, (char*)&tos, sizeof(tos));
>=20
> I believe this will work on any BSD based network stack.
>=20
> The problem with this approach is that it requires two system calls =
for each sent packet. It also has some implications on threading, since =
without setting TOS you can send data via the same socket without =
locking. Now, because each send is two system calls, lock will be =
required to ensure right TOS value, so the overhead can be even higher. =
There was a discussion about updating Linux kernel to add support for =
setting IP_TOS via the control structure in sendmsg, but I do not think =
this patch was ever accepted.
FreeBSD supports using a cmsg. So there is only one system call: =
sendmsg().

Best regards
Michael
>=20
> It was also recommended that ICE packets are sent with the same TOS as =
main data. Technically, you can end up with the situation where some TOS =
values go through the network and some do not. If data is sent with a =
different TOS value then ICE, ICE connectivity can succeed but sending =
the data will fail. This means an ICE connectivity check for each TOS =
value, which might produce different ICE candidates for different TOS =
values.
>=20
> Taking all of this into account I would suggest that a single TOS =
value should be used for each flow and data with different TOS values =
should be dis-aggregated across different flows.
> _____________
> Roman Shpount
>=20
> =20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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From: Roman Shpount <roman@telurix.com>
To: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
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Subject: Re: [rtcweb] -transports- discussion of DSCP
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_____________
Roman Shpount


On Thu, Aug 14, 2014 at 5:52 PM, Michael Tuexen <
Michael.Tuexen@lurchi.franken.de> wrote:

> On 14 Aug 2014, at 22:45, Roman Shpount <roman@telurix.com> wrote:
>
> > On Thu, Aug 14, 2014 at 2:17 PM, Justin Uberti <juberti@google.com>
> wrote:
> >
> >
> > On Thu, Aug 14, 2014 at 1:18 AM, Harald Alvestrand <harald@alvestrand.no>
> wrote:
> > On 08/13/2014 09:07 PM, Martin Thomson wrote:
> > On 13 August 2014 09:10, Ted Hardie <ted.ietf@gmail.com> wrote:
> > My concern here is that access to the ability to set different DSCP
> markings
> > on a per packet basis for a flow may not be available generally on all
> > platforms.  Even if we want to make a forward looking push toward that
> > ability, naming what we think is the appropriate alternative may make
> sense.
> > Yeah, requiring per-packet DSCP seems overly enthusiastic.
> >
> > I know it's possible to do this on Linux (for UDP, of course). I think
> it still requires an extra sysctl before sending each packet, which is
> reasonably high overhead if your app is sysctl-limited. But I don't know of
> any case where you would be allowed to set per-flow DSCP markings and would
> not be allowed to set per-packet DSCP markings; all the reasons to not do
> marking seem to be reasons to not do marking.
> >
> > I'd like to know if the APIs exist on other platforms (whether they're
> overridden by default policy or not).
> >
> > The two alternatives if we don't get per-packet setting are:
> >
> > - Deaggregate, so that per-port-flow classifiers can do their thing
> > - Don't ask the network to differentiate, keep precedence of flow a
> sender-stack thing only
> >
> > I don't know which is the more (or less) distasteful / useful of those
> two paths.
> >
> > The other option, which I believe works on MacOS, is to setsockopt
> before each sendto:
> >
> >     setsockopt(socket_, IPPROTO_IP, IP_TOS, &dscp, sizeof(dscp));
> >
> >
> > And on windows you can do the same by calling the following before
> sending each UDP packet:
> > setsockopt(d_socket, IPPROTO_IP, IP_TOS, (char*)&tos, sizeof(tos));
> >
> > I believe this will work on any BSD based network stack.
> >
> > The problem with this approach is that it requires two system calls for
> each sent packet. It also has some implications on threading, since without
> setting TOS you can send data via the same socket without locking. Now,
> because each send is two system calls, lock will be required to ensure
> right TOS value, so the overhead can be even higher. There was a discussion
> about updating Linux kernel to add support for setting IP_TOS via the
> control structure in sendmsg, but I do not think this patch was ever
> accepted.
> FreeBSD supports using a cmsg. So there is only one system call: sendmsg().
>
> Best regards
> Michael
> >
> > It was also recommended that ICE packets are sent with the same TOS as
> main data. Technically, you can end up with the situation where some TOS
> values go through the network and some do not. If data is sent with a
> different TOS value then ICE, ICE connectivity can succeed but sending the
> data will fail. This means an ICE connectivity check for each TOS value,
> which might produce different ICE candidates for different TOS values.
> >
> > Taking all of this into account I would suggest that a single TOS value
> should be used for each flow and data with different TOS values should be
> dis-aggregated across different flows.
> > _____________
> > Roman Shpount
> >
> >
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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<div dir=3D"ltr"><br></div><div class=3D"gmail_extra"><br clear=3D"all"><di=
v>_____________<br>Roman Shpount</div>
<br><br><div class=3D"gmail_quote">On Thu, Aug 14, 2014 at 5:52 PM, Michael=
 Tuexen <span dir=3D"ltr">&lt;<a href=3D"mailto:Michael.Tuexen@lurchi.frank=
en.de" target=3D"_blank">Michael.Tuexen@lurchi.franken.de</a>&gt;</span> wr=
ote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"">On 14 Aug 2014, at 22:45, Ro=
man Shpount &lt;<a href=3D"mailto:roman@telurix.com">roman@telurix.com</a>&=
gt; wrote:<br>

<br>
&gt; On Thu, Aug 14, 2014 at 2:17 PM, Justin Uberti &lt;<a href=3D"mailto:j=
uberti@google.com">juberti@google.com</a>&gt; wrote:<br>
&gt;<br>
&gt;<br>
&gt; On Thu, Aug 14, 2014 at 1:18 AM, Harald Alvestrand &lt;<a href=3D"mail=
to:harald@alvestrand.no">harald@alvestrand.no</a>&gt; wrote:<br>
&gt; On 08/13/2014 09:07 PM, Martin Thomson wrote:<br>
&gt; On 13 August 2014 09:10, Ted Hardie &lt;<a href=3D"mailto:ted.ietf@gma=
il.com">ted.ietf@gmail.com</a>&gt; wrote:<br>
&gt; My concern here is that access to the ability to set different DSCP ma=
rkings<br>
&gt; on a per packet basis for a flow may not be available generally on all=
<br>
&gt; platforms. =C2=A0Even if we want to make a forward looking push toward=
 that<br>
&gt; ability, naming what we think is the appropriate alternative may make =
sense.<br>
&gt; Yeah, requiring per-packet DSCP seems overly enthusiastic.<br>
&gt;<br>
&gt; I know it&#39;s possible to do this on Linux (for UDP, of course). I t=
hink it still requires an extra sysctl before sending each packet, which is=
 reasonably high overhead if your app is sysctl-limited. But I don&#39;t kn=
ow of any case where you would be allowed to set per-flow DSCP markings and=
 would not be allowed to set per-packet DSCP markings; all the reasons to n=
ot do marking seem to be reasons to not do marking.<br>

&gt;<br>
&gt; I&#39;d like to know if the APIs exist on other platforms (whether the=
y&#39;re overridden by default policy or not).<br>
&gt;<br>
&gt; The two alternatives if we don&#39;t get per-packet setting are:<br>
&gt;<br>
&gt; - Deaggregate, so that per-port-flow classifiers can do their thing<br=
>
&gt; - Don&#39;t ask the network to differentiate, keep precedence of flow =
a sender-stack thing only<br>
&gt;<br>
&gt; I don&#39;t know which is the more (or less) distasteful / useful of t=
hose two paths.<br>
&gt;<br>
&gt; The other option, which I believe works on MacOS, is to setsockopt bef=
ore each sendto:<br>
&gt;<br>
&gt; =C2=A0 =C2=A0 setsockopt(socket_, IPPROTO_IP, IP_TOS, &amp;dscp, sizeo=
f(dscp));<br>
&gt;<br>
&gt;<br>
&gt; And on windows you can do the same by calling the following before sen=
ding each UDP packet:<br>
&gt; setsockopt(d_socket, IPPROTO_IP, IP_TOS, (char*)&amp;tos, sizeof(tos))=
;<br>
&gt;<br>
&gt; I believe this will work on any BSD based network stack.<br>
&gt;<br>
&gt; The problem with this approach is that it requires two system calls fo=
r each sent packet. It also has some implications on threading, since witho=
ut setting TOS you can send data via the same socket without locking. Now, =
because each send is two system calls, lock will be required to ensure righ=
t TOS value, so the overhead can be even higher. There was a discussion abo=
ut updating Linux kernel to add support for setting IP_TOS via the control =
structure in sendmsg, but I do not think this patch was ever accepted.<br>

</div>FreeBSD supports using a cmsg. So there is only one system call: send=
msg().<br>
<br>
Best regards<br>
<span class=3D"HOEnZb"><font color=3D"#888888">Michael<br>
</font></span><div class=3D"im HOEnZb">&gt;<br>
&gt; It was also recommended that ICE packets are sent with the same TOS as=
 main data. Technically, you can end up with the situation where some TOS v=
alues go through the network and some do not. If data is sent with a differ=
ent TOS value then ICE, ICE connectivity can succeed but sending the data w=
ill fail. This means an ICE connectivity check for each TOS value, which mi=
ght produce different ICE candidates for different TOS values.<br>

&gt;<br>
&gt; Taking all of this into account I would suggest that a single TOS valu=
e should be used for each flow and data with different TOS values should be=
 dis-aggregated across different flows.<br>
&gt; _____________<br>
&gt; Roman Shpount<br>
&gt;<br>
&gt;<br>
</div><div class=3D"HOEnZb"><div class=3D"h5">&gt; ________________________=
_______________________<br>
&gt; rtcweb mailing list<br>
&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br>
</div></div></blockquote></div><br></div>

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From: Roman Shpount <roman@telurix.com>
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On Thu, Aug 14, 2014 at 5:52 PM, Michael Tuexen <
Michael.Tuexen@lurchi.franken.de> wrote:

> FreeBSD supports using a cmsg. So there is only one system call: sendmsg().
>
>
Do you know if setting TOS via cmsg in sendmsg works for OSX as well?

Thank You,
_____________
Roman Shpount

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<div dir=3D"ltr"><div class=3D"gmail_extra"><div>On Thu, Aug 14, 2014 at 5:=
52 PM, Michael Tuexen <span dir=3D"ltr">&lt;<a href=3D"mailto:Michael.Tuexe=
n@lurchi.franken.de" target=3D"_blank">Michael.Tuexen@lurchi.franken.de</a>=
&gt;</span> wrote:<br>
</div><div class=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D=
"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,2=
04,204);border-left-style:solid;padding-left:1ex"><div class=3D"">FreeBSD s=
upports using a cmsg. So there is only one system call: sendmsg().<br>
</div>
<br></blockquote><div>=C2=A0</div><div>Do you know if setting TOS via cmsg =
in sendmsg works for OSX as well?</div><div><br></div><div>Thank You,</div>=
<div>_____________<br>Roman Shpount</div><div>=C2=A0</div></div></div></div=
>

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On 15 Aug 2014, at 00:16, Roman Shpount <roman@telurix.com> wrote:

> On Thu, Aug 14, 2014 at 5:52 PM, Michael Tuexen =
<Michael.Tuexen@lurchi.franken.de> wrote:
> FreeBSD supports using a cmsg. So there is only one system call: =
sendmsg().
>=20
> =20
> Do you know if setting TOS via cmsg in sendmsg works for OSX as well?
=46rom reading the code: They don't provide a direct way to access it.
They have a proprietary cmsg type SO_TRAFFIC_CLASS, which their own =
values
priority levels which they map to DSCPs.

Best regards
Michael
>=20
> Thank You,
> _____________
> Roman Shpount
> =20


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--------------090401000506040409060206
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As per discussions in Toronto, I have submitted an initial draft of a 
gateways draft.

Uwe Rauschenbach and Christer Holmberg, who signed up to be co-authors, 
have contributed text and/or comments, but since both of them are on 
vacation at this time, and I couldn't get a signoff from them, I'm 
taking sole responsibility for the errors myself this time around.

Comments welcome!

               Harald


-------- Forwarded Message --------
Subject: 	New Version Notification for 
draft-alvestrand-rtcweb-gateways-00.txt
Date: 	Fri, 15 Aug 2014 06:33:24 -0700
From: 	internet-drafts@ietf.org
To: 	Harald T. Alvestrand <harald@alvestrand.no>, Harald Alvestrand 
<harald@alvestrand.no>



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

Name:		draft-alvestrand-rtcweb-gateways
Revision:	00
Title:		WebRTC Gateways
Document date:	2014-08-15
Group:		Individual Submission
Pages:		5
URL:            http://www.ietf.org/internet-drafts/draft-alvestrand-rtcweb-gateways-00.txt
Status:         https://datatracker.ietf.org/doc/draft-alvestrand-rtcweb-gateways/
Htmlized:       http://tools.ietf.org/html/draft-alvestrand-rtcweb-gateways-00


Abstract:
    This document specifies conformance requirements for a class of
    WebRTC devices called "WebRTC gateways".

    This type of device forms interconnects between WebRTC browsers and
    devices that are not WebRTC browsers.


                                                                                   


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




--------------090401000506040409060206
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<html>
  <head>

    <meta http-equiv="content-type" content="text/html; charset=utf-8">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    As per discussions in Toronto, I have submitted an initial draft of
    a gateways draft.<br>
    <br>
    Uwe Rauschenbach and Christer Holmberg, who signed up to be
    co-authors, have contributed text and/or comments, but since both of
    them are on vacation at this time, and I couldn't get a signoff from
    them, I'm taking sole responsibility for the errors myself this time
    around.<br>
    <br>
    Comments welcome!<br>
    <br>
    Â Â Â Â Â Â Â Â Â Â Â Â Â  Harald<br>
    <div class="moz-forward-container"><br>
      <br>
      -------- Forwarded Message --------
      <table class="moz-email-headers-table" border="0" cellpadding="0"
        cellspacing="0">
        <tbody>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Subject:
            </th>
            <td>New Version Notification for
              draft-alvestrand-rtcweb-gateways-00.txt</td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Date: </th>
            <td>Fri, 15 Aug 2014 06:33:24 -0700</td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">From: </th>
            <td><a class="moz-txt-link-abbreviated" href="mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a></td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">To: </th>
            <td>Harald T. Alvestrand <a class="moz-txt-link-rfc2396E" href="mailto:harald@alvestrand.no">&lt;harald@alvestrand.no&gt;</a>,
              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>
      <pre>A new version of I-D, draft-alvestrand-rtcweb-gateways-00.txt
has been successfully submitted by Harald Alvestrand and posted to the
IETF repository.

Name:		draft-alvestrand-rtcweb-gateways
Revision:	00
Title:		WebRTC Gateways
Document date:	2014-08-15
Group:		Individual Submission
Pages:		5
URL:            <a class="moz-txt-link-freetext" href="http://www.ietf.org/internet-drafts/draft-alvestrand-rtcweb-gateways-00.txt">http://www.ietf.org/internet-drafts/draft-alvestrand-rtcweb-gateways-00.txt</a>
Status:         <a class="moz-txt-link-freetext" href="https://datatracker.ietf.org/doc/draft-alvestrand-rtcweb-gateways/">https://datatracker.ietf.org/doc/draft-alvestrand-rtcweb-gateways/</a>
Htmlized:       <a class="moz-txt-link-freetext" href="http://tools.ietf.org/html/draft-alvestrand-rtcweb-gateways-00">http://tools.ietf.org/html/draft-alvestrand-rtcweb-gateways-00</a>


Abstract:
   This document specifies conformance requirements for a class of
   WebRTC devices called "WebRTC gateways".

   This type of device forms interconnects between WebRTC browsers and
   devices that are not WebRTC browsers.


                                                                                  


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

</pre>
      <br>
    </div>
    <br>
  </body>
</html>

--------------090401000506040409060206--


From nobody Fri Aug 15 07:20:25 2014
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From: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
To: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>, Roman Shpount <roman@telurix.com>
Thread-Topic: [rtcweb] -transports- discussion of DSCP
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I thought SO_TRAFFIC_CLASS was only for IPv6. Does it also work for IPv4?

Also, I thought these cmsg types were only for the recvmsg direction (when
SO_RECV_TRAFFIC_CLASS is set). Do they also work in the sendmsg direction?
If so, do they override the traffic class only for the specific sendmsg or
all subsequent sends on the socket? This last question would also apply to
FreeBSD, so that answer would be useful even if you have no idea what OSX
does.

Mo

-----Original Message-----
From: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
Date: Thursday, August 14, 2014 at 6:30 PM
To: Roman Shpount <roman@telurix.com>
Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] -transports- discussion of DSCP

On 15 Aug 2014, at 00:16, Roman Shpount <roman@telurix.com> wrote:

> On Thu, Aug 14, 2014 at 5:52 PM, Michael Tuexen
><Michael.Tuexen@lurchi.franken.de> wrote:
> FreeBSD supports using a cmsg. So there is only one system call:
>sendmsg().
>=20
> =20
> Do you know if setting TOS via cmsg in sendmsg works for OSX as well?
>From reading the code: They don't provide a direct way to access it.
They have a proprietary cmsg type SO_TRAFFIC_CLASS, which their own values
priority levels which they map to DSCPs.

Best regards
Michael
>=20
> Thank You,
> _____________
> Roman Shpount
> =20

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


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From: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
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Date: Fri, 15 Aug 2014 14:51:35 +0000
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References: <CA+9kkMBBfPyZdq0uzo96sxYaWK75bAHu8madAd5P4OTXLtb=Wg@mail.gmail.com> <CABkgnnVBtyb2wNZfo7Fto_X0Rhr_e2wAHWUxqcgBksj6YMLgFA@mail.gmail.com> <53EC70E5.7070105@alvestrand.no> <CAOJ7v-1sa0iq0jJ+=PwnDSdjf3=HJAVxDY-8QXLLbHXVHp-TuQ@mail.gmail.com> <CABkgnnX3_3hXY=Kaa+RU2xYstaWqRWdtF-gDzzE3uJ9MyUw31g@mail.gmail.com>
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Archived-At: http://mailarchive.ietf.org/arch/msg/rtcweb/_iPBKsieN4Usvp4Pd4R4zyRtjX0
Subject: Re: [rtcweb] -transports- discussion of DSCP
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> setsockopt(socket_, IPPROTO_IP, IP_TOS, &dscp, sizeof(dscp));

This works if your UDP transmit queue is always empty. If it ever backs
up, it fails, because this option clears the transmit queue. So while the
extra syscall overhead is trivial, it is unreliable for general use.

Also note that DART is likely to recommend a UDP flow only multiplex
different drop precedences within the same class (e.g. AF41, AF42), but
not different classes (e.g. AF41, EF). So even if there was a reliable way
to set DSCP per packet on every OS, we must still deal with the higher
level issue of whether to unbundle or unmark in cases DART does not
recommend. Should -transports- give guidance on this? Or leave it up to
implementations?

Mo


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From: "Parthasarathi R" <partha@parthasarathi.co.in>
To: "'Harald Alvestrand'" <harald@alvestrand.no>, <rtcweb@ietf.org>
References: <20140815133324.23199.55890.idtracker@ietfa.amsl.com> <53EE0CAD.7090301@alvestrand.no>
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Archived-At: http://mailarchive.ietf.org/arch/msg/rtcweb/CzUV9iyzzMamEwbm4QvqFvQ65BE
Subject: Re: [rtcweb] Gateway draft submitted
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------=_NextPart_000_0011_01CFB91A.6147D6E0
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Hi Harald,

=20

The draft looks as a good starting for WebRTC gateway. My initial review =
comment are as follows:

=20

1)      Abstract & Sec  1 Para 2:  WebRTC gateways are the type device =
which shall be between WebRTC browsers=20

2)      Sec 1: Figure shall be added to show the entities involved in =
case WebRTC gateways

3)      Sec 1.2  Bullet 1 & bullet 2:  JSEP offer/answer has to be =
mentioned instead of WebRTC Browser API

4)      Sec 2:  SDP extensions like MSID & AppID shall be relaxed

5)      Sec 2: IDP shall be relaxed

6)      The reference has to be added=20

=20

Thanks

Partha

=20

From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Harald =
Alvestrand
Sent: Friday, August 15, 2014 7:06 PM
To: rtcweb@ietf.org
Subject: [rtcweb] Gateway draft submitted

=20

As per discussions in Toronto, I have submitted an initial draft of a =
gateways draft.

Uwe Rauschenbach and Christer Holmberg, who signed up to be co-authors, =
have contributed text and/or comments, but since both of them are on =
vacation at this time, and I couldn't get a signoff from them, I'm =
taking sole responsibility for the errors myself this time around.

Comments welcome!

              Harald



-------- Forwarded Message --------=20


Subject:=20

New Version Notification for draft-alvestrand-rtcweb-gateways-00.txt


Date:=20

Fri, 15 Aug 2014 06:33:24 -0700


From:=20

internet-drafts@ietf.org


To:=20

Harald T. Alvestrand  <mailto:harald@alvestrand.no> =
<harald@alvestrand.no>, Harald Alvestrand  <mailto:harald@alvestrand.no> =
<harald@alvestrand.no>

=20

A new version of I-D, draft-alvestrand-rtcweb-gateways-00.txt
has been successfully submitted by Harald Alvestrand and posted to the
IETF repository.
=20
Name:         draft-alvestrand-rtcweb-gateways
Revision:     00
Title:        WebRTC Gateways
Document date: 2014-08-15
Group:        Individual Submission
Pages:        5
URL:            =
http://www.ietf.org/internet-drafts/draft-alvestrand-rtcweb-gateways-00.t=
xt
Status:         =
https://datatracker.ietf.org/doc/draft-alvestrand-rtcweb-gateways/
Htmlized:       =
http://tools.ietf.org/html/draft-alvestrand-rtcweb-gateways-00
=20
=20
Abstract:
   This document specifies conformance requirements for a class of
   WebRTC devices called "WebRTC gateways".
=20
   This type of device forms interconnects between WebRTC browsers and
   devices that are not WebRTC browsers.
=20
=20
                                                                         =
        =20
=20
=20
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.
=20
The IETF Secretariat
=20

=20

=20


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<div class=3DSection1>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Hi Harald,<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>The draft looks as a good starting for WebRTC gateway. My
initial review comment are as follows:<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo2'><![if !supportLists]><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><span
style=3D'mso-list:Ignore'>1)<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Abstract &amp; Sec=C2=A0 1 Para 2:=C2=A0 WebRTC gateways =
are the type
device which shall be between WebRTC browsers <o:p></o:p></span></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo2'><![if !supportLists]><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><span
style=3D'mso-list:Ignore'>2)<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Sec 1: Figure shall be added to show the entities =
involved in
case WebRTC gateways<o:p></o:p></span></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo2'><![if !supportLists]><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><span
style=3D'mso-list:Ignore'>3)<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Sec 1.2=C2=A0 Bullet 1 &amp; bullet 2:=C2=A0 JSEP =
offer/answer has to be
mentioned instead of WebRTC Browser API<o:p></o:p></span></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo2'><![if !supportLists]><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><span
style=3D'mso-list:Ignore'>4)<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Sec 2: =C2=A0SDP extensions like MSID &amp; AppID shall =
be relaxed<o:p></o:p></span></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo2'><![if !supportLists]><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><span
style=3D'mso-list:Ignore'>5)<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Sec 2: IDP shall be relaxed<o:p></o:p></span></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo2'><![if !supportLists]><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><span
style=3D'mso-list:Ignore'>6)<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>The reference has to be added <o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Thanks<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Partha<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<div style=3D'border:none;border-left:solid blue 1.5pt;padding:0in 0in =
0in 4.0pt'>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt =
0in 0in 0in'>

<p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";
color:windowtext'>From:</span></b><span =
style=3D'font-size:10.0pt;font-family:
"Tahoma","sans-serif";color:windowtext'> rtcweb
[mailto:rtcweb-bounces@ietf.org] <b>On Behalf Of </b>Harald =
Alvestrand<br>
<b>Sent:</b> Friday, August 15, 2014 7:06 PM<br>
<b>To:</b> rtcweb@ietf.org<br>
<b>Subject:</b> [rtcweb] Gateway draft submitted<o:p></o:p></span></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>As per discussions in Toronto, I have submitted an =
initial
draft of a gateways draft.<br>
<br>
Uwe Rauschenbach and Christer Holmberg, who signed up to be co-authors, =
have
contributed text and/or comments, but since both of them are on vacation =
at
this time, and I couldn't get a signoff from them, I'm taking sole
responsibility for the errors myself this time around.<br>
<br>
Comments welcome!<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;
Harald<o:p></o:p></p>

<div>

<p class=3DMsoNormal><br>
<br>
-------- Forwarded Message -------- <o:p></o:p></p>

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  <p class=3DMsoNormal align=3Dright =
style=3D'text-align:right'><b>Subject: <o:p></o:p></b></p>
  </td>
  <td style=3D'padding:0in 0in 0in 0in'>
  <p class=3DMsoNormal>New Version Notification for =
draft-alvestrand-rtcweb-gateways-00.txt<o:p></o:p></p>
  </td>
 </tr>
 <tr>
  <td nowrap valign=3Dtop style=3D'padding:0in 0in 0in 0in'>
  <p class=3DMsoNormal align=3Dright style=3D'text-align:right'><b>Date: =
<o:p></o:p></b></p>
  </td>
  <td style=3D'padding:0in 0in 0in 0in'>
  <p class=3DMsoNormal>Fri, 15 Aug 2014 06:33:24 -0700<o:p></o:p></p>
  </td>
 </tr>
 <tr>
  <td nowrap valign=3Dtop style=3D'padding:0in 0in 0in 0in'>
  <p class=3DMsoNormal align=3Dright style=3D'text-align:right'><b>From: =
<o:p></o:p></b></p>
  </td>
  <td style=3D'padding:0in 0in 0in 0in'>
  <p class=3DMsoNormal><a =
href=3D"mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a><o:p=
></o:p></p>
  </td>
 </tr>
 <tr>
  <td nowrap valign=3Dtop style=3D'padding:0in 0in 0in 0in'>
  <p class=3DMsoNormal align=3Dright style=3D'text-align:right'><b>To: =
<o:p></o:p></b></p>
  </td>
  <td style=3D'padding:0in 0in 0in 0in'>
  <p class=3DMsoNormal>Harald T. Alvestrand <a =
href=3D"mailto:harald@alvestrand.no">&lt;harald@alvestrand.no&gt;</a>,
  Harald Alvestrand <a =
href=3D"mailto:harald@alvestrand.no">&lt;harald@alvestrand.no&gt;</a><o:p=
></o:p></p>
  </td>
 </tr>
</table>

<p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'><o:p>&nbsp;</o:p></p>

<pre>A new version of I-D, =
draft-alvestrand-rtcweb-gateways-00.txt<o:p></o:p></pre><pre>has been =
successfully submitted by Harald Alvestrand and posted to =
the<o:p></o:p></pre><pre>IETF =
repository.<o:p></o:p></pre><pre><o:p>&nbsp;</o:p></pre><pre>Name:=C2=A0=C2=
=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 =
draft-alvestrand-rtcweb-gateways<o:p></o:p></pre><pre>Revision:=C2=A0=C2=A0=
=C2=A0=C2=A0 =
00<o:p></o:p></pre><pre>Title:=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 =
WebRTC Gateways<o:p></o:p></pre><pre>Document date: =
2014-08-15<o:p></o:p></pre><pre>Group:=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0 Individual =
Submission<o:p></o:p></pre><pre>Pages:=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0 =
5<o:p></o:p></pre><pre>URL:=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0 <a
href=3D"http://www.ietf.org/internet-drafts/draft-alvestrand-rtcweb-gatew=
ays-00.txt">http://www.ietf.org/internet-drafts/draft-alvestrand-rtcweb-g=
ateways-00.txt</a><o:p></o:p></pre><pre>Status:=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0=C2=A0 <a
href=3D"https://datatracker.ietf.org/doc/draft-alvestrand-rtcweb-gateways=
/">https://datatracker.ietf.org/doc/draft-alvestrand-rtcweb-gateways/</a>=
<o:p></o:p></pre><pre>Htmlized:=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 <a
href=3D"http://tools.ietf.org/html/draft-alvestrand-rtcweb-gateways-00">h=
ttp://tools.ietf.org/html/draft-alvestrand-rtcweb-gateways-00</a><o:p></o=
:p></pre><pre><o:p>&nbsp;</o:p></pre><pre><o:p>&nbsp;</o:p></pre><pre>Abs=
tract:<o:p></o:p></pre><pre>=C2=A0=C2=A0 This document specifies =
conformance requirements for a class =
of<o:p></o:p></pre><pre>=C2=A0=C2=A0 WebRTC devices called &quot;WebRTC =
gateways&quot;.<o:p></o:p></pre><pre><o:p>&nbsp;</o:p></pre><pre>=C2=A0=C2=
=A0 This type of device forms interconnects between WebRTC browsers =
and<o:p></o:p></pre><pre>=C2=A0=C2=A0 devices that are not WebRTC =
browsers.<o:p></o:p></pre><pre><o:p>&nbsp;</o:p></pre><pre><o:p>&nbsp;</o=
:p></pre><pre>=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=
=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 =
<o:p></o:p></pre><pre><o:p>&nbsp;</o:p></pre><pre><o:p>&nbsp;</o:p></pre>=
<pre>Please note that it may take a couple of minutes from the time of =
submission<o:p></o:p></pre><pre>until the htmlized version and diff are =
available at =
tools.ietf.org.<o:p></o:p></pre><pre><o:p>&nbsp;</o:p></pre><pre>The =
IETF Secretariat<o:p></o:p></pre><pre><o:p>&nbsp;</o:p></pre>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

</div>

</body>

</html>

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From nobody Fri Aug 15 23:44:45 2014
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Date: Sat, 16 Aug 2014 08:44:35 +0200
From: Gunnar Hellstrom <gunnar.hellstrom@omnitor.se>
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Subject: [rtcweb] draft-ietf-rtcweb-data-channel failure handling description
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We have discussed the lack of description of failure handling in 
draft-ietf-rtcweb-data-channel a couple of times.

Explanations of mechanisms have been provided in mail answers that I 
have not seen documented elsewhere, but they have not been introduced in 
the draft.

I think that for specification and implementation of the WebRTC API, 
there is a need for clear specification how the channels behave in 
failure situations.

I suggest to insert a new chapter 6.7 in the draft with the following 
contents.

-------------------------------------------------------------------Proposed 
new section 
6.7-------------------------------------------------------------------------------------------
6.7 Transmission failure and error handling

Failures can occur in an open channel by transmission error detection 
and by SCTP watchdog error indications.
If transmission in a reliable channel fails, the association where the 
channel belongs will be aborted. If all retries for a watchdog expires, 
the corresponding association will be aborted.  All channels in an 
association that is aborted because of errors need to be closed by the 
party detecting the failure. Network error indications shall be provided 
to upper layers on all closed channels.

If retransmissions or transmission time is exceeded in a partially 
reliable channel, the transmission SHALL be allowed to continue with 
other data items. A transmission error indication SHALL be provided to 
upper layers on that channel.

If reception out-of order is detected from an SCTP channel for which 
ordered transmission is requested, the receiver SHALL close the data 
channel, and provide an transmission error indication to upper layers.

If a message with an unsupported PPID is received or some logical error 
is detected by the receiver, the receiver SHALL close the corresponding 
data channel.
-----------------------------------------------------------------------end 
of 
proposal---------------------------------------------------------------------------------------------------------------

I am not sure about what the two last kinds of errors should result in 
and hope that the experts can adjust the wording.

Regards

Gunnar
------------------------------------------------------------------------
Gunnar HellstrÃ¶m
Omnitor
gunnar.hellstrom@omnitor.se

On 2014-08-11 20:13, IESG Secretary wrote:
> The IESG would like to retract the current Last Call on the following
> document:
>
> - 'WebRTC Data Channels'
>    <draft-ietf-rtcweb-data-channel-11.txt> as Proposed Standard
>
> The document's state has been set back to "AD Evaluation."
>


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<html>
  <head>
    <meta content="text/html; charset=UTF-8" http-equiv="Content-Type">
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  <body bgcolor="#FFFFFF" text="#000000">
    <div class="moz-cite-prefix">We have discussed the lack of
      description of failure handling in draft-ietf-rtcweb-data-channel
      a couple of times.<br>
      <br>
      Explanations of mechanisms have been provided in mail answers that
      I have not seen documented elsewhere, but they have not been
      introduced in the draft.<br>
      <br>
      I think that for specification and implementation of the WebRTC
      API, there is a need for clear specification how the channels
      behave in failure situations.<br>
      <br>
      I suggest to insert a new chapter 6.7 in the draft with the
      following contents.Â  <br>
      <br>
      -------------------------------------------------------------------Proposed
      new section
6.7-------------------------------------------------------------------------------------------<br>
      6.7 Transmission failure and error handling
      <br>
      <br>
      Failures can occur in an open channel by transmission error
      detection and by SCTP watchdog error indications.<br>
      If transmission in a reliable channel fails, the association where
      the channel belongs will be aborted. If all retries for a watchdog
      expires, the corresponding association will be aborted.Â  All
      channels in an association that is aborted because of errors need
      to be closed by the party detecting the failure. Network error
      indications shall be provided to upper layers on all closed
      channels.<br>
      <br>
      If retransmissions or transmission time is exceeded in a partially
      reliable channel, the transmission SHALL be allowed to continue
      with other data items. A transmission error indication SHALL be
      provided to upper layers on that channel.
      <br>
      <br>
      If reception out-of order is detected from an SCTP channel for
      which ordered transmission is requested, the receiver SHALL close
      the data channel, and provide an transmission error indication to
      upper layers.Â 
      <br>
      <br>
      If a message with an unsupported PPID is received or some logical
      error is detected by the receiver, the receiver SHALL close the
      corresponding data channel.
      <br>
      -----------------------------------------------------------------------end
      of
proposal---------------------------------------------------------------------------------------------------------------<br>
      <br>
      I am not sure about what the two last kinds of errors should
      result in and hope that the experts can adjust the wording.<br>
      <br>
      Regards<br>
      <br>
      Gunnar<br>
      <div class="moz-signature">
        <hr>
        Gunnar HellstrÃ¶m<br>
        Omnitor<br>
        <a class="moz-txt-link-abbreviated" href="mailto:gunnar.hellstrom@omnitor.se">gunnar.hellstrom@omnitor.se</a><br>
        <br>
      </div>
      On 2014-08-11 20:13, IESG Secretary wrote:<br>
    </div>
    <blockquote
      cite="mid:20140811181357.613.72705.idtracker@ietfa.amsl.com"
      type="cite">
      <pre wrap="">The IESG would like to retract the current Last Call on the following 
document:

- 'WebRTC Data Channels'
  &lt;draft-ietf-rtcweb-data-channel-11.txt&gt; as Proposed Standard

The document's state has been set back to "AD Evaluation."

</pre>
    </blockquote>
    <br>
  </body>
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--------------090802080707010108050401--


From nobody Sat Aug 16 05:01:29 2014
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From: Barry Dingle <btdingle@gmail.com>
Date: Sat, 16 Aug 2014 22:01:00 +1000
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Subject: Re: [rtcweb] draft-ietf-rtcweb-data-channel failure handling description
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--089e01493c945c3e830500bde583
Content-Type: text/plain; charset=UTF-8
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I agree that error handling is missing from draft-ietf-rtcweb-data-channel
.

Should the 'error' indications' in the proposed text be more specific? That
is, should the error indications indicate the type of error. (The same
error indication for all errors could be sent and be compliant with the
currently proposed text.)

Note - I assume that you are proposing adding the new Section 6.7 to
draft-ietf-rtcweb-data-channel-11 . It already has a Section 6.7 titled
"Closing a channel". I assume you mean a new Section 6.8.

Cheers,
/Barry Dingle




On Sat, Aug 16, 2014 at 4:44 PM, Gunnar Hellstrom <
gunnar.hellstrom@omnitor.se> wrote:

>  We have discussed the lack of description of failure handling in
> draft-ietf-rtcweb-data-channel a couple of times.
>
> Explanations of mechanisms have been provided in mail answers that I have
> not seen documented elsewhere, but they have not been introduced in the
> draft.
>
> I think that for specification and implementation of the WebRTC API, ther=
e
> is a need for clear specification how the channels behave in failure
> situations.
>
> I suggest to insert a new chapter 6.7 in the draft with the following
> contents.
>
> -------------------------------------------------------------------Propos=
ed
> new section
> 6.7----------------------------------------------------------------------=
---------------------
> 6.7 Transmission failure and error handling
>
> Failures can occur in an open channel by transmission error detection and
> by SCTP watchdog error indications.
> If transmission in a reliable channel fails, the association where the
> channel belongs will be aborted. If all retries for a watchdog expires, t=
he
> corresponding association will be aborted.  All channels in an associatio=
n
> that is aborted because of errors need to be closed by the party detectin=
g
> the failure. Network error indications shall be provided to upper layers =
on
> all closed channels.
>
> If retransmissions or transmission time is exceeded in a partially
> reliable channel, the transmission SHALL be allowed to continue with othe=
r
> data items. A transmission error indication SHALL be provided to upper
> layers on that channel.
>
> If reception out-of order is detected from an SCTP channel for which
> ordered transmission is requested, the receiver SHALL close the data
> channel, and provide an transmission error indication to upper layers.
>
> If a message with an unsupported PPID is received or some logical error i=
s
> detected by the receiver, the receiver SHALL close the corresponding data
> channel.
> -----------------------------------------------------------------------en=
d
> of
> proposal-----------------------------------------------------------------=
----------------------------------------------
>
> I am not sure about what the two last kinds of errors should result in an=
d
> hope that the experts can adjust the wording.
>
> Regards
>
> Gunnar
>  ------------------------------
> Gunnar Hellstr=C3=B6m
> Omnitor
> gunnar.hellstrom@omnitor.se
>
>  On 2014-08-11 20:13, IESG Secretary wrote:
>
> The IESG would like to retract the current Last Call on the following
> document:
>
> - 'WebRTC Data Channels'
>   <draft-ietf-rtcweb-data-channel-11.txt> as Proposed Standard
>
> The document's state has been set back to "AD Evaluation."
>
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--089e01493c945c3e830500bde583
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<div dir=3D"ltr"><div>I agree that error handling is missing from draft-iet=
f-rtcweb-data-channel . <br><br></div><div>Should the &#39;error&#39; indic=
ations&#39; in the proposed text be more specific? That is, should the erro=
r indications indicate the type of error. (The same error indication for al=
l errors could be sent and be compliant with the currently proposed text.) =
<br>

</div><div><br></div>Note - I assume that you are proposing adding the new =
Section 6.7 to=C2=A0 draft-ietf-rtcweb-data-channel-11 . It already has a S=
ection 6.7 titled &quot;Closing a channel&quot;. I assume you mean a new Se=
ction 6.8. <br>

<div class=3D"gmail_extra"><br clear=3D"all"><div><div dir=3D"ltr">Cheers,<=
br>/Barry Dingle<br><br><br></div></div>
<br><br><div class=3D"gmail_quote">On Sat, Aug 16, 2014 at 4:44 PM, Gunnar =
Hellstrom <span dir=3D"ltr">&lt;<a href=3D"mailto:gunnar.hellstrom@omnitor.=
se" target=3D"_blank">gunnar.hellstrom@omnitor.se</a>&gt;</span> wrote:<br>=
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">


 =20
   =20
 =20
  <div bgcolor=3D"#FFFFFF" text=3D"#000000">
    <div>We have discussed the lack of
      description of failure handling in draft-ietf-rtcweb-data-channel
      a couple of times.<br>
      <br>
      Explanations of mechanisms have been provided in mail answers that
      I have not seen documented elsewhere, but they have not been
      introduced in the draft.<br>
      <br>
      I think that for specification and implementation of the WebRTC
      API, there is a need for clear specification how the channels
      behave in failure situations.<br>
      <br>
      I suggest to insert a new chapter 6.7 in the draft with the
      following contents.=C2=A0 <br>
      <br>
      -------------------------------------------------------------------Pr=
oposed
      new section
6.7------------------------------------------------------------------------=
-------------------<br>
      6.7 Transmission failure and error handling
      <br>
      <br>
      Failures can occur in an open channel by transmission error
      detection and by SCTP watchdog error indications.<br>
      If transmission in a reliable channel fails, the association where
      the channel belongs will be aborted. If all retries for a watchdog
      expires, the corresponding association will be aborted.=C2=A0 All
      channels in an association that is aborted because of errors need
      to be closed by the party detecting the failure. Network error
      indications shall be provided to upper layers on all closed
      channels.<br>
      <br>
      If retransmissions or transmission time is exceeded in a partially
      reliable channel, the transmission SHALL be allowed to continue
      with other data items. A transmission error indication SHALL be
      provided to upper layers on that channel.
      <br>
      <br>
      If reception out-of order is detected from an SCTP channel for
      which ordered transmission is requested, the receiver SHALL close
      the data channel, and provide an transmission error indication to
      upper layers.=C2=A0
      <br>
      <br>
      If a message with an unsupported PPID is received or some logical
      error is detected by the receiver, the receiver SHALL close the
      corresponding data channel.
      <br>
      ---------------------------------------------------------------------=
--end
      of
proposal-------------------------------------------------------------------=
--------------------------------------------<br>
      <br>
      I am not sure about what the two last kinds of errors should
      result in and hope that the experts can adjust the wording.<br>
      <br>
      Regards<br>
      <br>
      Gunnar<br>
      <div>
        <hr>
        Gunnar Hellstr=C3=B6m<br>
        Omnitor<br>
        <a href=3D"mailto:gunnar.hellstrom@omnitor.se" target=3D"_blank">gu=
nnar.hellstrom@omnitor.se</a><br>
        <br>
      </div>
      On 2014-08-11 20:13, IESG Secretary wrote:<br>
    </div>
    <blockquote type=3D"cite">
      <pre>The IESG would like to retract the current Last Call on the foll=
owing=20
document:

- &#39;WebRTC Data Channels&#39;
  &lt;draft-ietf-rtcweb-data-channel-11.txt&gt; as Proposed Standard

The document&#39;s state has been set back to &quot;AD Evaluation.&quot;

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

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

--089e01493c945c3e830500bde583--


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Date: Sat, 16 Aug 2014 15:25:28 +0200
From: Gunnar Hellstrom <gunnar.hellstrom@omnitor.se>
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On 2014-08-16 14:01, Barry Dingle wrote:
> I agree that error handling is missing from 
> draft-ietf-rtcweb-data-channel .
>
> Should the 'error' indications' in the proposed text be more specific? 
> That is, should the error indications indicate the type of error. (The 
> same error indication for all errors could be sent and be compliant 
> with the currently proposed text.)
Yes, possibly more detailed failure cause information is beneficial. But 
this needs to be matched against the ambitions of the WebRTC API to be 
as similar as possible to the Websocket API.
And the Websocket API groups many error reasons into one error 
indication because of security risks.
See http://dev.w3.org/html5/websockets/

Maintenance and debugging might be hard without further refinement of 
error causes, while security is better with less specific information. A 
balanced proposal would be needed.

A specific improvement would be to indicate if it is the sending side or 
the receiving side or both that will get an error indication.

I have not seen the association concept in the WebRTC API. So, it seems 
that the association concept will be hidden in the implementation 
between the WebRTC API and the SCTP, because in interaction with SCTP, 
it is important to know when a new association needs to be opened, and 
when just a new channel can be opened in an existing association. The 
effect of the abortion of the association also needs to be understood by 
that layer and translated into separate rtcDataChannel API actions.

One further note: I have assumed that ICE connectivity failures is 
handled separately by the WebRTC API implementation and will cause both 
indications to API users and to the RTCweb data channel implementations. 
Is that right?


>
> Note - I assume that you are proposing adding the new Section 6.7 to  
> draft-ietf-rtcweb-data-channel-11 . It already has a Section 6.7 
> titled "Closing a channel". I assume you mean a new Section 6.8.
Right, thanks.
>
> Cheers,
> /Barry Dingle
>
>
>
>
> On Sat, Aug 16, 2014 at 4:44 PM, Gunnar Hellstrom 
> <gunnar.hellstrom@omnitor.se <mailto:gunnar.hellstrom@omnitor.se>> wrote:
>
>     We have discussed the lack of description of failure handling in
>     draft-ietf-rtcweb-data-channel a couple of times.
>
>     Explanations of mechanisms have been provided in mail answers that
>     I have not seen documented elsewhere, but they have not been
>     introduced in the draft.
>
>     I think that for specification and implementation of the WebRTC
>     API, there is a need for clear specification how the channels
>     behave in failure situations.
>
>     I suggest to insert a new chapter 6.7 in the draft with the
>     following contents.
>
>     -------------------------------------------------------------------Proposed
>     new section
>     6.7-------------------------------------------------------------------------------------------
>     6.7 Transmission failure and error handling
>
>     Failures can occur in an open channel by transmission error
>     detection and by SCTP watchdog error indications.
>     If transmission in a reliable channel fails, the association where
>     the channel belongs will be aborted. If all retries for a watchdog
>     expires, the corresponding association will be aborted.  All
>     channels in an association that is aborted because of errors need
>     to be closed by the party detecting the failure. Network error
>     indications shall be provided to upper layers on all closed channels.
>
>     If retransmissions or transmission time is exceeded in a partially
>     reliable channel, the transmission SHALL be allowed to continue
>     with other data items. A transmission error indication SHALL be
>     provided to upper layers on that channel.
>
>     If reception out-of order is detected from an SCTP channel for
>     which ordered transmission is requested, the receiver SHALL close
>     the data channel, and provide an transmission error indication to
>     upper layers.
>
>     If a message with an unsupported PPID is received or some logical
>     error is detected by the receiver, the receiver SHALL close the
>     corresponding data channel.
>     -----------------------------------------------------------------------end
>     of
>     proposal---------------------------------------------------------------------------------------------------------------
>
>     I am not sure about what the two last kinds of errors should
>     result in and hope that the experts can adjust the wording.
>
>     Regards
>
>     Gunnar
>     ------------------------------------------------------------------------
>     Gunnar HellstrÃ¶m
>     Omnitor
>     gunnar.hellstrom@omnitor.se <mailto:gunnar.hellstrom@omnitor.se>
>
>     On 2014-08-11 20:13, IESG Secretary wrote:
>>     The IESG would like to retract the current Last Call on the following
>>     document:
>>
>>     - 'WebRTC Data Channels'
>>        <draft-ietf-rtcweb-data-channel-11.txt> as Proposed Standard
>>
>>     The document's state has been set back to "AD Evaluation."
>>
>
>
>     _______________________________________________
>     rtcweb mailing list
>     rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>     https://www.ietf.org/mailman/listinfo/rtcweb
>
>


--------------070703010100050303070609
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<html>
  <head>
    <meta content="text/html; charset=UTF-8" http-equiv="Content-Type">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    <div class="moz-cite-prefix">On 2014-08-16 14:01, Barry Dingle
      wrote:<br>
    </div>
    <blockquote
cite="mid:CAN=GVAuc-DEzEh3qUVugcRWm85tUJggmA+-zw3hLwT+v6m1srw@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div>I agree that error handling is missing from
          draft-ietf-rtcweb-data-channel . <br>
          <br>
        </div>
        <div>Should the 'error' indications' in the proposed text be
          more specific? That is, should the error indications indicate
          the type of error. (The same error indication for all errors
          could be sent and be compliant with the currently proposed
          text.) <br>
        </div>
      </div>
    </blockquote>
    Yes, possibly more detailed failure cause information is beneficial.
    But this needs to be matched against the ambitions of the WebRTC API
    to be as similar as possible to the Websocket API.<br>
    And the Websocket API groups many error reasons into one error
    indication because of security risks.<br>
    See <a class="moz-txt-link-freetext" href="http://dev.w3.org/html5/websockets/">http://dev.w3.org/html5/websockets/</a><br>
    <br>
    Maintenance and debugging might be hard without further refinement
    of error causes, while security is better with less specific
    information. A balanced proposal would be needed. <br>
    <br>
    A specific improvement would be to indicate if it is the sending
    side or the receiving side or both that will get an error
    indication. <br>
    <br>
    I have not seen the association concept in the WebRTC API. So, it
    seems that the association concept will be hidden in the
    implementation between the WebRTC API and the SCTP, because in
    interaction with SCTP, it is important to know when a new
    association needs to be opened, and when just a new channel can be
    opened in an existing association. The effect of the abortion of the
    association also needs to be understood by that layer and translated
    into separate rtcDataChannel API actions.<br>
    <br>
    One further note: I have assumed that ICE connectivity failures is
    handled separately by the WebRTC API implementation and will cause
    both indications to API users and to the RTCweb data channel
    implementations. Is that right?<br>
    <br>
    Â  <br>
    <blockquote
cite="mid:CAN=GVAuc-DEzEh3qUVugcRWm85tUJggmA+-zw3hLwT+v6m1srw@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div>
        </div>
        <div><br>
        </div>
        Note - I assume that you are proposing adding the new Section
        6.7 toÂ  draft-ietf-rtcweb-data-channel-11 . It already has a
        Section 6.7 titled "Closing a channel". I assume you mean a new
        Section 6.8. <br>
      </div>
    </blockquote>
    Right, thanks.<br>
    <blockquote
cite="mid:CAN=GVAuc-DEzEh3qUVugcRWm85tUJggmA+-zw3hLwT+v6m1srw@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div class="gmail_extra"><br clear="all">
          <div>
            <div dir="ltr">Cheers,<br>
              /Barry Dingle<br>
              <br>
              <br>
            </div>
          </div>
          <br>
          <br>
          <div class="gmail_quote">On Sat, Aug 16, 2014 at 4:44 PM,
            Gunnar Hellstrom <span dir="ltr">&lt;<a
                moz-do-not-send="true"
                href="mailto:gunnar.hellstrom@omnitor.se"
                target="_blank">gunnar.hellstrom@omnitor.se</a>&gt;</span>
            wrote:<br>
            <blockquote class="gmail_quote" style="margin:0 0 0
              .8ex;border-left:1px #ccc solid;padding-left:1ex">
              <div bgcolor="#FFFFFF" text="#000000">
                <div>We have discussed the lack of description of
                  failure handling in draft-ietf-rtcweb-data-channel a
                  couple of times.<br>
                  <br>
                  Explanations of mechanisms have been provided in mail
                  answers that I have not seen documented elsewhere, but
                  they have not been introduced in the draft.<br>
                  <br>
                  I think that for specification and implementation of
                  the WebRTC API, there is a need for clear
                  specification how the channels behave in failure
                  situations.<br>
                  <br>
                  I suggest to insert a new chapter 6.7 in the draft
                  with the following contents.Â  <br>
                  <br>
                  -------------------------------------------------------------------Proposed

                  new section
6.7-------------------------------------------------------------------------------------------<br>
                  6.7 Transmission failure and error handling <br>
                  <br>
                  Failures can occur in an open channel by transmission
                  error detection and by SCTP watchdog error
                  indications.<br>
                  If transmission in a reliable channel fails, the
                  association where the channel belongs will be aborted.
                  If all retries for a watchdog expires, the
                  corresponding association will be aborted.Â  All
                  channels in an association that is aborted because of
                  errors need to be closed by the party detecting the
                  failure. Network error indications shall be provided
                  to upper layers on all closed channels.<br>
                  <br>
                  If retransmissions or transmission time is exceeded in
                  a partially reliable channel, the transmission SHALL
                  be allowed to continue with other data items. A
                  transmission error indication SHALL be provided to
                  upper layers on that channel. <br>
                  <br>
                  If reception out-of order is detected from an SCTP
                  channel for which ordered transmission is requested,
                  the receiver SHALL close the data channel, and provide
                  an transmission error indication to upper layers.Â  <br>
                  <br>
                  If a message with an unsupported PPID is received or
                  some logical error is detected by the receiver, the
                  receiver SHALL close the corresponding data channel. <br>
                  -----------------------------------------------------------------------end

                  of
proposal---------------------------------------------------------------------------------------------------------------<br>
                  <br>
                  I am not sure about what the two last kinds of errors
                  should result in and hope that the experts can adjust
                  the wording.<br>
                  <br>
                  Regards<br>
                  <br>
                  Gunnar<br>
                  <div>
                    <hr> Gunnar HellstrÃ¶m<br>
                    Omnitor<br>
                    <a moz-do-not-send="true"
                      href="mailto:gunnar.hellstrom@omnitor.se"
                      target="_blank">gunnar.hellstrom@omnitor.se</a><br>
                    <br>
                  </div>
                  On 2014-08-11 20:13, IESG Secretary wrote:<br>
                </div>
                <blockquote type="cite">
                  <pre>The IESG would like to retract the current Last Call on the following 
document:

- 'WebRTC Data Channels'
  &lt;draft-ietf-rtcweb-data-channel-11.txt&gt; as Proposed Standard

The document's state has been set back to "AD Evaluation."

</pre>
                </blockquote>
                <br>
              </div>
              <br>
              _______________________________________________<br>
              rtcweb mailing list<br>
              <a moz-do-not-send="true" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
              <a moz-do-not-send="true"
                href="https://www.ietf.org/mailman/listinfo/rtcweb"
                target="_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
              <br>
            </blockquote>
          </div>
          <br>
        </div>
      </div>
    </blockquote>
    <br>
  </body>
</html>

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--------------020007050400010202000802
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Hi Harald,

Lorenzo Miniero, Victor Pascual and me wrote the "Guidelines to support 
RTCP end-to-end in Back-to-Back User Agents  (B2BUAs)":

http://tools.ietf.org/html/draft-miniero-straw-b2bua-rtcp-00

that I think could be related to this one. Maybe it could be worthy to 
reference it?

Best regards
Sergio
On 15/08/2014 15:35, Harald Alvestrand wrote:
> As per discussions in Toronto, I have submitted an initial draft of a 
> gateways draft.
>
> Uwe Rauschenbach and Christer Holmberg, who signed up to be 
> co-authors, have contributed text and/or comments, but since both of 
> them are on vacation at this time, and I couldn't get a signoff from 
> them, I'm taking sole responsibility for the errors myself this time 
> around.
>
> Comments welcome!
>
>               Harald
>
>
> -------- Forwarded Message --------
> Subject: 	New Version Notification for 
> draft-alvestrand-rtcweb-gateways-00.txt
> Date: 	Fri, 15 Aug 2014 06:33:24 -0700
> From: 	internet-drafts@ietf.org
> To: 	Harald T. Alvestrand <harald@alvestrand.no>, Harald Alvestrand 
> <harald@alvestrand.no>
>
>
>
> A new version of I-D, draft-alvestrand-rtcweb-gateways-00.txt
> has been successfully submitted by Harald Alvestrand and posted to the
> IETF repository.
>
> Name:		draft-alvestrand-rtcweb-gateways
> Revision:	00
> Title:		WebRTC Gateways
> Document date:	2014-08-15
> Group:		Individual Submission
> Pages:		5
> URL:http://www.ietf.org/internet-drafts/draft-alvestrand-rtcweb-gateways-00.txt
> Status:https://datatracker.ietf.org/doc/draft-alvestrand-rtcweb-gateways/
> Htmlized:http://tools.ietf.org/html/draft-alvestrand-rtcweb-gateways-00
>
>
> Abstract:
>     This document specifies conformance requirements for a class of
>     WebRTC devices called "WebRTC gateways".
>
>     This type of device forms interconnects between WebRTC browsers and
>     devices that are not WebRTC browsers.
>
>
>                                                                                    
>
>
> 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


--------------020007050400010202000802
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<html>
  <head>
    <meta content="text/html; charset=ISO-8859-1"
      http-equiv="Content-Type">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    <div class="moz-cite-prefix">Hi Harald,<br>
      <br>
      Lorenzo Miniero, Victor Pascual and me wrote the "Guidelines to
      support RTCP end-to-end in Back-to-Back User Agents&nbsp; (B2BUAs)":<br>
      <br>
      <a class="moz-txt-link-freetext" href="http://tools.ietf.org/html/draft-miniero-straw-b2bua-rtcp-00">http://tools.ietf.org/html/draft-miniero-straw-b2bua-rtcp-00</a><br>
      <br>
      that I think could be related to this one. Maybe it could be
      worthy to reference it?<br>
      <br>
      Best regards<br>
      Sergio<br>
      On 15/08/2014 15:35, Harald Alvestrand wrote:<br>
    </div>
    <blockquote cite="mid:53EE0CAD.7090301@alvestrand.no" type="cite">
      <meta http-equiv="content-type" content="text/html;
        charset=ISO-8859-1">
      As per discussions in Toronto, I have submitted an initial draft
      of a gateways draft.<br>
      <br>
      Uwe Rauschenbach and Christer Holmberg, who signed up to be
      co-authors, have contributed text and/or comments, but since both
      of them are on vacation at this time, and I couldn't get a signoff
      from them, I'm taking sole responsibility for the errors myself
      this time around.<br>
      <br>
      Comments welcome!<br>
      <br>
      &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Harald<br>
      <div class="moz-forward-container"><br>
        <br>
        -------- Forwarded Message --------
        <table class="moz-email-headers-table" cellpadding="0"
          cellspacing="0" border="0">
          <tbody>
            <tr>
              <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Subject:

              </th>
              <td>New Version Notification for
                draft-alvestrand-rtcweb-gateways-00.txt</td>
            </tr>
            <tr>
              <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Date:
              </th>
              <td>Fri, 15 Aug 2014 06:33:24 -0700</td>
            </tr>
            <tr>
              <th align="RIGHT" nowrap="nowrap" valign="BASELINE">From:
              </th>
              <td><a moz-do-not-send="true"
                  class="moz-txt-link-abbreviated"
                  href="mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a></td>
            </tr>
            <tr>
              <th align="RIGHT" nowrap="nowrap" valign="BASELINE">To: </th>
              <td>Harald T. Alvestrand <a moz-do-not-send="true"
                  class="moz-txt-link-rfc2396E"
                  href="mailto:harald@alvestrand.no">&lt;harald@alvestrand.no&gt;</a>,
                Harald Alvestrand <a moz-do-not-send="true"
                  class="moz-txt-link-rfc2396E"
                  href="mailto:harald@alvestrand.no">&lt;harald@alvestrand.no&gt;</a></td>
            </tr>
          </tbody>
        </table>
        <br>
        <br>
        <pre>A new version of I-D, draft-alvestrand-rtcweb-gateways-00.txt
has been successfully submitted by Harald Alvestrand and posted to the
IETF repository.

Name:		draft-alvestrand-rtcweb-gateways
Revision:	00
Title:		WebRTC Gateways
Document date:	2014-08-15
Group:		Individual Submission
Pages:		5
URL:            <a moz-do-not-send="true" class="moz-txt-link-freetext" href="http://www.ietf.org/internet-drafts/draft-alvestrand-rtcweb-gateways-00.txt">http://www.ietf.org/internet-drafts/draft-alvestrand-rtcweb-gateways-00.txt</a>
Status:         <a moz-do-not-send="true" class="moz-txt-link-freetext" href="https://datatracker.ietf.org/doc/draft-alvestrand-rtcweb-gateways/">https://datatracker.ietf.org/doc/draft-alvestrand-rtcweb-gateways/</a>
Htmlized:       <a moz-do-not-send="true" class="moz-txt-link-freetext" href="http://tools.ietf.org/html/draft-alvestrand-rtcweb-gateways-00">http://tools.ietf.org/html/draft-alvestrand-rtcweb-gateways-00</a>


Abstract:
   This document specifies conformance requirements for a class of
   WebRTC devices called "WebRTC gateways".

   This type of device forms interconnects between WebRTC browsers and
   devices that are not WebRTC browsers.


                                                                                  


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

</pre>
        <br>
      </div>
      <br>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_______________________________________________
rtcweb mailing list
<a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
    </blockquote>
    <br>
  </body>
</html>

--------------020007050400010202000802--


From nobody Sun Aug 17 12:36:31 2014
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From: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
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To: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
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Subject: Re: [rtcweb] -transports- discussion of DSCP
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On 15 Aug 2014, at 16:20, Mo Zanaty (mzanaty) <mzanaty@cisco.com> wrote:

> I thought SO_TRAFFIC_CLASS was only for IPv6. Does it also work for =
IPv4?
I guess so.
=
http://fxr.watson.org/fxr/source/bsd/netinet/udp_usrreq.c?v=3Dxnu-2050.18.=
24#L1330
should handle IPv4 and IPv6 sockets.
>=20
> Also, I thought these cmsg types were only for the recvmsg direction =
(when
> SO_RECV_TRAFFIC_CLASS is set). Do they also work in the sendmsg =
direction?
Yepp. The above is in udp_output(), a routine called when sending UDP =
packets.
> If so, do they override the traffic class only for the specific =
sendmsg or
> all subsequent sends on the socket? This last question would also =
apply to
Only on that packet. The value is not stored at the socket layer / =
endpoint.
> FreeBSD, so that answer would be useful even if you have no idea what =
OSX
> does.
Same for FreeBSD.

At least on FreeBSD you have a socket option to set the default value =
for all
future packets. Using the cmsg overwrites the the default on a =
per-packet
base.

Best regards
Michael
>=20
> Mo
>=20
> -----Original Message-----
> From: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
> Date: Thursday, August 14, 2014 at 6:30 PM
> To: Roman Shpount <roman@telurix.com>
> Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
> Subject: Re: [rtcweb] -transports- discussion of DSCP
>=20
> On 15 Aug 2014, at 00:16, Roman Shpount <roman@telurix.com> wrote:
>=20
>> On Thu, Aug 14, 2014 at 5:52 PM, Michael Tuexen
>> <Michael.Tuexen@lurchi.franken.de> wrote:
>> FreeBSD supports using a cmsg. So there is only one system call:
>> sendmsg().
>>=20
>>=20
>> Do you know if setting TOS via cmsg in sendmsg works for OSX as well?
> =46rom reading the code: They don't provide a direct way to access it.
> They have a proprietary cmsg type SO_TRAFFIC_CLASS, which their own =
values
> priority levels which they map to DSCPs.
>=20
> Best regards
> Michael
>>=20
>> Thank You,
>> _____________
>> Roman Shpount
>>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
>=20


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On 15 Aug 2014, at 16:51, Mo Zanaty (mzanaty) <mzanaty@cisco.com> wrote:

>> setsockopt(socket_, IPPROTO_IP, IP_TOS, &dscp, sizeof(dscp));
> 
> This works if your UDP transmit queue is always empty. If it ever backs
> up, it fails, because this option clears the transmit queue. So while the
> extra syscall overhead is trivial, it is unreliable for general use.
Which OS has a UDP send queue?

Best regards
Michael
> 
> Also note that DART is likely to recommend a UDP flow only multiplex
> different drop precedences within the same class (e.g. AF41, AF42), but
> not different classes (e.g. AF41, EF). So even if there was a reliable way
> to set DSCP per packet on every OS, we must still deal with the higher
> level issue of whether to unbundle or unmark in cases DART does not
> recommend. Should -transports- give guidance on this? Or leave it up to
> implementations?
> 
> Mo
> 
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> 


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Subject: Re: [rtcweb] -transports- discussion of DSCP
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DSCP can be set by using setsockopt() system call.

Regards
Ranjit

-----Original Message-----
From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of ext Michael Tuex=
en
Sent: Friday, August 15, 2014 4:01 AM
To: Roman Shpount
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] -transports- discussion of DSCP

On 15 Aug 2014, at 00:16, Roman Shpount <roman@telurix.com> wrote:

> On Thu, Aug 14, 2014 at 5:52 PM, Michael Tuexen <Michael.Tuexen@lurchi.fr=
anken.de> wrote:
> FreeBSD supports using a cmsg. So there is only one system call: sendmsg(=
).
>=20
> =20
> Do you know if setting TOS via cmsg in sendmsg works for OSX as well?
>From reading the code: They don't provide a direct way to access it.
They have a proprietary cmsg type SO_TRAFFIC_CLASS, which their own values
priority levels which they map to DSCPs.

Best regards
Michael
>=20
> Thank You,
> _____________
> Roman Shpount
> =20

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


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Subject: Re: [rtcweb] -transports- discussion of DSCP
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On 08/15/2014 04:51 PM, Mo Zanaty (mzanaty) wrote:
>> setsockopt(socket_, IPPROTO_IP, IP_TOS, &dscp, sizeof(dscp));
> This works if your UDP transmit queue is always empty. If it ever backs
> up, it fails, because this option clears the transmit queue. So while the
> extra syscall overhead is trivial, it is unreliable for general use.

I'd like to see a demo program that shows there's an issue.

A natural implementation strategy would be to mark the DSCP when 
creating a buffer in sendmsg(), before enqueueing it; there would only 
be an issue if some implementation had a strategy of marking the DSCP 
with the socket's currently selected DSCP value at dequeueing time (and 
one could argue that such an implementation has a bug that needs fixing).

>
> Also note that DART is likely to recommend a UDP flow only multiplex
> different drop precedences within the same class (e.g. AF41, AF42), but
> not different classes (e.g. AF41, EF). So even if there was a reliable way
> to set DSCP per packet on every OS, we must still deal with the higher
> level issue of whether to unbundle or unmark in cases DART does not
> recommend. Should -transports- give guidance on this? Or leave it up to
> implementations?

Either rtcweb-transports- says what to do, or it refers to something 
normative that says what to do. The only higher level target in the 
chain is -overview, and that's clearly the wrong place.

Since DART's target is informative, by Hobson's choice it seems that 
rtcweb-transports- is the place.

I see the implementation choices to be unbundling, marking at the lowest 
class (?) recommended, or not marking at all - I assume that BE is a 
different class than AF4*.

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


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On 08/16/2014 03:25 PM, Gunnar Hellstrom wrote:
> On 2014-08-16 14:01, Barry Dingle wrote:
>> I agree that error handling is missing from 
>> draft-ietf-rtcweb-data-channel .
>>
>> Should the 'error' indications' in the proposed text be more 
>> specific? That is, should the error indications indicate the type of 
>> error. (The same error indication for all errors could be sent and be 
>> compliant with the currently proposed text.)
> Yes, possibly more detailed failure cause information is beneficial. 
> But this needs to be matched against the ambitions of the WebRTC API 
> to be as similar as possible to the Websocket API.
> And the Websocket API groups many error reasons into one error 
> indication because of security risks.
> See http://dev.w3.org/html5/websockets/
>
> Maintenance and debugging might be hard without further refinement of 
> error causes, while security is better with less specific information. 
> A balanced proposal would be needed.

The need for debugging is handled by advice on what information needs to 
be in the "message" part of the error indication.

If there is a need for consistent action to be taken by applications, I 
think that needs to be in the "name" part - that is, different errors - 
or in additional information passed in the error.

>
> A specific improvement would be to indicate if it is the sending side 
> or the receiving side or both that will get an error indication.
>
> I have not seen the association concept in the WebRTC API. So, it 
> seems that the association concept will be hidden in the 
> implementation between the WebRTC API and the SCTP, because in 
> interaction with SCTP, it is important to know when a new association 
> needs to be opened, and when just a new channel can be opened in an 
> existing association. The effect of the abortion of the association 
> also needs to be understood by that layer and translated into separate 
> rtcDataChannel API actions.
>
> One further note: I have assumed that ICE connectivity failures is 
> handled separately by the WebRTC API implementation and will cause 
> both indications to API users and to the RTCweb data channel 
> implementations. Is that right?

I can't parse this in terms that fit my implementation - there is no 
distinction between the "RTCweb data channel implementation" and "the 
WebRTC API implementation" - it's all the WebRTC implementation, and 
there is no distinction that can be surfaced.

What's the change that needs to be there?


>
>
>>
>> Note - I assume that you are proposing adding the new Section 6.7 to  
>> draft-ietf-rtcweb-data-channel-11 . It already has a Section 6.7 
>> titled "Closing a channel". I assume you mean a new Section 6.8.
> Right, thanks.
>>
>> Cheers,
>> /Barry Dingle
>>
>>
>>
>>
>> On Sat, Aug 16, 2014 at 4:44 PM, Gunnar Hellstrom 
>> <gunnar.hellstrom@omnitor.se <mailto:gunnar.hellstrom@omnitor.se>> wrote:
>>
>>     We have discussed the lack of description of failure handling in
>>     draft-ietf-rtcweb-data-channel a couple of times.
>>
>>     Explanations of mechanisms have been provided in mail answers
>>     that I have not seen documented elsewhere, but they have not been
>>     introduced in the draft.
>>
>>     I think that for specification and implementation of the WebRTC
>>     API, there is a need for clear specification how the channels
>>     behave in failure situations.
>>
>>     I suggest to insert a new chapter 6.7 in the draft with the
>>     following contents.
>>
>>     -------------------------------------------------------------------Proposed
>>     new section
>>     6.7-------------------------------------------------------------------------------------------
>>     6.7 Transmission failure and error handling
>>
>>     Failures can occur in an open channel by transmission error
>>     detection and by SCTP watchdog error indications.
>>     If transmission in a reliable channel fails, the association
>>     where the channel belongs will be aborted. If all retries for a
>>     watchdog expires, the corresponding association will be aborted. 
>>     All channels in an association that is aborted because of errors
>>     need to be closed by the party detecting the failure. Network
>>     error indications shall be provided to upper layers on all closed
>>     channels.
>>
>>     If retransmissions or transmission time is exceeded in a
>>     partially reliable channel, the transmission SHALL be allowed to
>>     continue with other data items. A transmission error indication
>>     SHALL be provided to upper layers on that channel.
>>
>>     If reception out-of order is detected from an SCTP channel for
>>     which ordered transmission is requested, the receiver SHALL close
>>     the data channel, and provide an transmission error indication to
>>     upper layers.
>>
>>     If a message with an unsupported PPID is received or some logical
>>     error is detected by the receiver, the receiver SHALL close the
>>     corresponding data channel.
>>     -----------------------------------------------------------------------end
>>     of
>>     proposal---------------------------------------------------------------------------------------------------------------
>>
>>     I am not sure about what the two last kinds of errors should
>>     result in and hope that the experts can adjust the wording.
>>
>>     Regards
>>
>>     Gunnar
>>     ------------------------------------------------------------------------
>>     Gunnar Hellström
>>     Omnitor
>>     gunnar.hellstrom@omnitor.se <mailto:gunnar.hellstrom@omnitor.se>
>>
>>     On 2014-08-11 20:13, IESG Secretary wrote:
>>>     The IESG would like to retract the current Last Call on the following
>>>     document:
>>>
>>>     - 'WebRTC Data Channels'
>>>        <draft-ietf-rtcweb-data-channel-11.txt> as Proposed Standard
>>>
>>>     The document's state has been set back to "AD Evaluation."
>>>
>>
>>
>>     _______________________________________________
>>     rtcweb mailing list
>>     rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>>     https://www.ietf.org/mailman/listinfo/rtcweb
>>
>>
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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<html>
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      http-equiv="Content-Type">
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  <body bgcolor="#FFFFFF" text="#000000">
    <div class="moz-cite-prefix">On 08/16/2014 03:25 PM, Gunnar
      Hellstrom wrote:<br>
    </div>
    <blockquote cite="mid:53EF5BC8.7060803@omnitor.se" type="cite">
      <meta content="text/html; charset=windows-1252"
        http-equiv="Content-Type">
      <div class="moz-cite-prefix">On 2014-08-16 14:01, Barry Dingle
        wrote:<br>
      </div>
      <blockquote
cite="mid:CAN=GVAuc-DEzEh3qUVugcRWm85tUJggmA+-zw3hLwT+v6m1srw@mail.gmail.com"
        type="cite">
        <div dir="ltr">
          <div>I agree that error handling is missing from
            draft-ietf-rtcweb-data-channel . <br>
            <br>
          </div>
          <div>Should the 'error' indications' in the proposed text be
            more specific? That is, should the error indications
            indicate the type of error. (The same error indication for
            all errors could be sent and be compliant with the currently
            proposed text.) <br>
          </div>
        </div>
      </blockquote>
      Yes, possibly more detailed failure cause information is
      beneficial. But this needs to be matched against the ambitions of
      the WebRTC API to be as similar as possible to the Websocket API.<br>
      And the Websocket API groups many error reasons into one error
      indication because of security risks.<br>
      See <a moz-do-not-send="true" class="moz-txt-link-freetext"
        href="http://dev.w3.org/html5/websockets/">http://dev.w3.org/html5/websockets/</a><br>
      <br>
      Maintenance and debugging might be hard without further refinement
      of error causes, while security is better with less specific
      information. A balanced proposal would be needed. <br>
    </blockquote>
    <br>
    The need for debugging is handled by advice on what information
    needs to be in the "message" part of the error indication.<br>
    <br>
    If there is a need for consistent action to be taken by
    applications, I think that needs to be in the "name" part - that is,
    different errors - or in additional information passed in the error.<br>
    <br>
    <blockquote cite="mid:53EF5BC8.7060803@omnitor.se" type="cite"> <br>
      A specific improvement would be to indicate if it is the sending
      side or the receiving side or both that will get an error
      indication. <br>
      <br>
      I have not seen the association concept in the WebRTC API. So, it
      seems that the association concept will be hidden in the
      implementation between the WebRTC API and the SCTP, because in
      interaction with SCTP, it is important to know when a new
      association needs to be opened, and when just a new channel can be
      opened in an existing association. The effect of the abortion of
      the association also needs to be understood by that layer and
      translated into separate rtcDataChannel API actions.<br>
      <br>
      One further note: I have assumed that ICE connectivity failures is
      handled separately by the WebRTC API implementation and will cause
      both indications to API users and to the RTCweb data channel
      implementations. Is that right?<br>
    </blockquote>
    <br>
    I can't parse this in terms that fit my implementation - there is no
    distinction between the "RTCweb data channel implementation" and
    "the WebRTC API implementation" - it's all the WebRTC
    implementation, and there is no distinction that can be surfaced.<br>
    <br>
    What's the change that needs to be there?<br>
    <br>
    <br>
    <blockquote cite="mid:53EF5BC8.7060803@omnitor.se" type="cite"> <br>
        <br>
      <blockquote
cite="mid:CAN=GVAuc-DEzEh3qUVugcRWm85tUJggmA+-zw3hLwT+v6m1srw@mail.gmail.com"
        type="cite">
        <div dir="ltr">
          <div> </div>
          <div><br>
          </div>
          Note - I assume that you are proposing adding the new Section
          6.7 to  draft-ietf-rtcweb-data-channel-11 . It already has a
          Section 6.7 titled "Closing a channel". I assume you mean a
          new Section 6.8. <br>
        </div>
      </blockquote>
      Right, thanks.<br>
      <blockquote
cite="mid:CAN=GVAuc-DEzEh3qUVugcRWm85tUJggmA+-zw3hLwT+v6m1srw@mail.gmail.com"
        type="cite">
        <div dir="ltr">
          <div class="gmail_extra"><br clear="all">
            <div>
              <div dir="ltr">Cheers,<br>
                /Barry Dingle<br>
                <br>
                <br>
              </div>
            </div>
            <br>
            <br>
            <div class="gmail_quote">On Sat, Aug 16, 2014 at 4:44 PM,
              Gunnar Hellstrom <span dir="ltr">&lt;<a
                  moz-do-not-send="true"
                  href="mailto:gunnar.hellstrom@omnitor.se"
                  target="_blank">gunnar.hellstrom@omnitor.se</a>&gt;</span>
              wrote:<br>
              <blockquote class="gmail_quote" style="margin:0 0 0
                .8ex;border-left:1px #ccc solid;padding-left:1ex">
                <div bgcolor="#FFFFFF" text="#000000">
                  <div>We have discussed the lack of description of
                    failure handling in draft-ietf-rtcweb-data-channel a
                    couple of times.<br>
                    <br>
                    Explanations of mechanisms have been provided in
                    mail answers that I have not seen documented
                    elsewhere, but they have not been introduced in the
                    draft.<br>
                    <br>
                    I think that for specification and implementation of
                    the WebRTC API, there is a need for clear
                    specification how the channels behave in failure
                    situations.<br>
                    <br>
                    I suggest to insert a new chapter 6.7 in the draft
                    with the following contents.  <br>
                    <br>
                    -------------------------------------------------------------------Proposed


                    new section
6.7-------------------------------------------------------------------------------------------<br>
                    6.7 Transmission failure and error handling <br>
                    <br>
                    Failures can occur in an open channel by
                    transmission error detection and by SCTP watchdog
                    error indications.<br>
                    If transmission in a reliable channel fails, the
                    association where the channel belongs will be
                    aborted. If all retries for a watchdog expires, the
                    corresponding association will be aborted.  All
                    channels in an association that is aborted because
                    of errors need to be closed by the party detecting
                    the failure. Network error indications shall be
                    provided to upper layers on all closed channels.<br>
                    <br>
                    If retransmissions or transmission time is exceeded
                    in a partially reliable channel, the transmission
                    SHALL be allowed to continue with other data items.
                    A transmission error indication SHALL be provided to
                    upper layers on that channel. <br>
                    <br>
                    If reception out-of order is detected from an SCTP
                    channel for which ordered transmission is requested,
                    the receiver SHALL close the data channel, and
                    provide an transmission error indication to upper
                    layers.  <br>
                    <br>
                    If a message with an unsupported PPID is received or
                    some logical error is detected by the receiver, the
                    receiver SHALL close the corresponding data channel.
                    <br>
                    -----------------------------------------------------------------------end


                    of
proposal---------------------------------------------------------------------------------------------------------------<br>
                    <br>
                    I am not sure about what the two last kinds of
                    errors should result in and hope that the experts
                    can adjust the wording.<br>
                    <br>
                    Regards<br>
                    <br>
                    Gunnar<br>
                    <div>
                      <hr> Gunnar Hellström<br>
                      Omnitor<br>
                      <a moz-do-not-send="true"
                        href="mailto:gunnar.hellstrom@omnitor.se"
                        target="_blank">gunnar.hellstrom@omnitor.se</a><br>
                      <br>
                    </div>
                    On 2014-08-11 20:13, IESG Secretary wrote:<br>
                  </div>
                  <blockquote type="cite">
                    <pre>The IESG would like to retract the current Last Call on the following 
document:

- 'WebRTC Data Channels'
  &lt;draft-ietf-rtcweb-data-channel-11.txt&gt; as Proposed Standard

The document's state has been set back to "AD Evaluation."

</pre>
                  </blockquote>
                  <br>
                </div>
                <br>
                _______________________________________________<br>
                rtcweb mailing list<br>
                <a moz-do-not-send="true" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
                <a moz-do-not-send="true"
                  href="https://www.ietf.org/mailman/listinfo/rtcweb"
                  target="_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
                <br>
              </blockquote>
            </div>
            <br>
          </div>
        </div>
      </blockquote>
      <br>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_______________________________________________
rtcweb mailing list
<a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
    </blockquote>
    <br>
  </body>
</html>

--------------040102020100000204000709--


From nobody Mon Aug 18 04:11:46 2014
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Date: Mon, 18 Aug 2014 13:11:26 +0200
From: Harald Alvestrand <harald@alvestrand.no>
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Subject: [rtcweb] WebRTC Device: Lowering the bar, or inventing a new term?
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When writing the gateway document, I found myself grappling with the 
"device" description.

It is somewhat uncomfortable, in that as first defined, it requires full 
WebRTC functionality except for the API part - this means that it should 
support data channels, full ICE, bundling and so on.

But for gateways, it makes sense to let go of a lot of these 
requirements. So a gateway is not a device. But what is the name for the 
thing that a gateway is, a device is, and a browser is, that names "the 
set of things that can successfully negotiate connectivity"? And what 
are the requirements on *that*?

I see a couple of options:

- Redefine "Device" to mean "that which can successfully communicate 
with a browser". It's free to negotiate away features if it can, but 
needs to interoperate with a browser that implements only the absolute 
minimum of what a browser has to implement.

- Define a new class of "thing" which is "the thing that can talk to 
browsers" - of which a gateway has to be an exemplar.

This is a kind of word game in some ways, but when we make definitions, 
we also define what's easy to talk about. And that can influence what we 
end up building.

What do people think?

    Harald


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Subject: [rtcweb] Unsupported PPID (Re: draft-ietf-rtcweb-data-channel failure handling description)
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Picking up a single point....

On 08/16/2014 08:44 AM, Gunnar Hellstrom wrote:
>
> If a message with an unsupported PPID is received or some logical 
> error is detected by the receiver, the receiver SHALL close the 
> corresponding data channel.

This struct me when thinking about the zero-length-data proposal: If we 
ever want to add another such feature *without* explicitly negotiating 
the use of the feature at setup time, we can't do this.

If we want the possibility of un-negotiated extensions, we HAVE to say 
(somewhere)

   If a message with an unsupported PPID is received, it is silently 
ignored.

OTOH, if we're convinced we never need that kind of extensibility, any 
way we do this is fine.

              Harald


--------------050405090503000108090502
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    <meta content="text/html; charset=windows-1252"
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    <div class="moz-cite-prefix">Picking up a single point....<br>
      <br>
      On 08/16/2014 08:44 AM, Gunnar Hellstrom wrote:<br>
    </div>
    <blockquote cite="mid:53EEFDD3.6040607@omnitor.se" type="cite">
      <meta content="text/html; charset=windows-1252"
        http-equiv="Content-Type">
      <div class="moz-cite-prefix"><br>
        If a message with an unsupported PPID is received or some
        logical error is detected by the receiver, the receiver SHALL
        close the corresponding data channel. </div>
    </blockquote>
    <br>
    This struct me when thinking about the zero-length-data proposal: If
    we ever want to add another such feature *without* explicitly
    negotiating the use of the feature at setup time, we can't do this.<br>
    <br>
    If we want the possibility of un-negotiated extensions, we HAVE to
    say (somewhere)<br>
    <br>
      If a message with an unsupported PPID is received, it is silently
    ignored.<br>
    <br>
    OTOH, if we're convinced we never need that kind of extensibility,
    any way we do this is fine.<br>
    <br>
                 Harald<br>
    <br>
  </body>
</html>

--------------050405090503000108090502--


From nobody Mon Aug 18 04:19:23 2014
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the Real-Time Communication in WEB-browsers Working Group of the IETF.

        Title           : Overview: Real Time Protocols for Browser-based Applications
        Author          : Harald T. Alvestrand
	Filename        : draft-ietf-rtcweb-overview-11.txt
	Pages           : 22
	Date            : 2014-08-18

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.

   This document is an Applicability Statement - it does not itself
   specify any protocol, but specifies which other specifications RTCWEB
   compliant implementations are supposed to follow.

   This document is a work item of the RTCWEB working group.


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

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

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


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

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


From nobody Mon Aug 18 04:23:54 2014
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On 2014-08-18 13:15, Harald Alvestrand wrote:
> Picking up a single point....
>
> On 08/16/2014 08:44 AM, Gunnar Hellstrom wrote:
>>
>> If a message with an unsupported PPID is received or some logical 
>> error is detected by the receiver, the receiver SHALL close the 
>> corresponding data channel.
>
> This struct me when thinking about the zero-length-data proposal: If 
> we ever want to add another such feature *without* explicitly 
> negotiating the use of the feature at setup time, we can't do this.
>
> If we want the possibility of un-negotiated extensions, we HAVE to say 
> (somewhere)
>
>   If a message with an unsupported PPID is received, it is silently 
> ignored.
[GH] That sounds wise, and also in line with the security aspects that 
make Websocket be very conservative about signaling errors and their 
reasons. WebRTC data channels have the ambition to be similar.
/Gunnar
>
> OTOH, if we're convinced we never need that kind of extensibility, any 
> way we do this is fine.
>
>              Harald
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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    <div class="moz-cite-prefix">On 2014-08-18 13:15, Harald Alvestrand
      wrote:<br>
    </div>
    <blockquote cite="mid:53F1E05D.9040804@alvestrand.no" type="cite">
      <meta content="text/html; charset=ISO-8859-1"
        http-equiv="Content-Type">
      <div class="moz-cite-prefix">Picking up a single point....<br>
        <br>
        On 08/16/2014 08:44 AM, Gunnar Hellstrom wrote:<br>
      </div>
      <blockquote cite="mid:53EEFDD3.6040607@omnitor.se" type="cite">
        <meta content="text/html; charset=ISO-8859-1"
          http-equiv="Content-Type">
        <div class="moz-cite-prefix"><br>
          If a message with an unsupported PPID is received or some
          logical error is detected by the receiver, the receiver SHALL
          close the corresponding data channel.&nbsp;</div>
      </blockquote>
      <br>
      This struct me when thinking about the zero-length-data proposal:
      If we ever want to add another such feature *without* explicitly
      negotiating the use of the feature at setup time, we can't do
      this.<br>
      <br>
      If we want the possibility of un-negotiated extensions, we HAVE to
      say (somewhere)<br>
      <br>
      &nbsp; If a message with an unsupported PPID is received, it is
      silently ignored.<br>
    </blockquote>
    [GH] That sounds wise, and also in line with the security aspects
    that make Websocket be very conservative about signaling errors and
    their reasons. WebRTC data channels have the ambition to be similar.<br>
    /Gunnar<br>
    <blockquote cite="mid:53F1E05D.9040804@alvestrand.no" type="cite"> <br>
      OTOH, if we're convinced we never need that kind of extensibility,
      any way we do this is fine.<br>
      <br>
      &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Harald<br>
      <br>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_______________________________________________
rtcweb mailing list
<a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
    </blockquote>
    <br>
  </body>
</html>

--------------060406050801020407090607--


From nobody Mon Aug 18 04:24:49 2014
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From: Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com>
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Subject: Re: [rtcweb] WebRTC Device: Lowering the bar, or inventing a new term?
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Hi Harald,

I always liked "endpoint" much more than "device".

We could have different types of endpoints, and define a browser like: a 
WebRTC "full" endpoint exposing the W3C WebRTC APIs.

Just my 2 cents.

Best regards
Sergio
On 18/08/2014 13:11, Harald Alvestrand wrote:
> When writing the gateway document, I found myself grappling with the 
> "device" description.
>
> It is somewhat uncomfortable, in that as first defined, it requires 
> full WebRTC functionality except for the API part - this means that it 
> should support data channels, full ICE, bundling and so on.
>
> But for gateways, it makes sense to let go of a lot of these 
> requirements. So a gateway is not a device. But what is the name for 
> the thing that a gateway is, a device is, and a browser is, that names 
> "the set of things that can successfully negotiate connectivity"? And 
> what are the requirements on *that*?
>
> I see a couple of options:
>
> - Redefine "Device" to mean "that which can successfully communicate 
> with a browser". It's free to negotiate away features if it can, but 
> needs to interoperate with a browser that implements only the absolute 
> minimum of what a browser has to implement.
>
> - Define a new class of "thing" which is "the thing that can talk to 
> browsers" - of which a gateway has to be an exemplar.
>
> This is a kind of word game in some ways, but when we make 
> definitions, we also define what's easy to talk about. And that can 
> influence what we end up building.
>
> What do people think?
>
>    Harald
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>, Harald Alvestrand <harald@alvestrand.no>
Thread-Topic: [rtcweb] WebRTC Device: Lowering the bar,	or inventing a new term?
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Subject: Re: [rtcweb] WebRTC Device: Lowering the bar, or inventing a new term?
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Hi Harald,

How about "gw-endpoint"? in the sense that it behaves like an "endpoint" bu=
t has different requirements than an "endpoint".

BR
Raju


-----Original Message-----
From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Sergio Garcia Mu=
rillo
Sent: Monday, August 18, 2014 6:25 AM
To: rtcweb@ietf.org
Subject: Re: [rtcweb] WebRTC Device: Lowering the bar, or inventing a new t=
erm?

Hi Harald,

I always liked "endpoint" much more than "device".

We could have different types of endpoints, and define a browser like: a=20
WebRTC "full" endpoint exposing the W3C WebRTC APIs.

Just my 2 cents.

Best regards
Sergio
On 18/08/2014 13:11, Harald Alvestrand wrote:
> When writing the gateway document, I found myself grappling with the=20
> "device" description.
>
> It is somewhat uncomfortable, in that as first defined, it requires=20
> full WebRTC functionality except for the API part - this means that it=20
> should support data channels, full ICE, bundling and so on.
>
> But for gateways, it makes sense to let go of a lot of these=20
> requirements. So a gateway is not a device. But what is the name for=20
> the thing that a gateway is, a device is, and a browser is, that names=20
> "the set of things that can successfully negotiate connectivity"? And=20
> what are the requirements on *that*?
>
> I see a couple of options:
>
> - Redefine "Device" to mean "that which can successfully communicate=20
> with a browser". It's free to negotiate away features if it can, but=20
> needs to interoperate with a browser that implements only the absolute=20
> minimum of what a browser has to implement.
>
> - Define a new class of "thing" which is "the thing that can talk to=20
> browsers" - of which a gateway has to be an exemplar.
>
> This is a kind of word game in some ways, but when we make=20
> definitions, we also define what's easy to talk about. And that can=20
> influence what we end up building.
>
> What do people think?
>
>    Harald
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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


From nobody Mon Aug 18 04:56:54 2014
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On 2014-08-18 10:30, Harald Alvestrand wrote:
> On 08/16/2014 03:25 PM, Gunnar Hellstrom wrote:
>> On 2014-08-16 14:01, Barry Dingle wrote:
>>> I agree that error handling is missing from 
>>> draft-ietf-rtcweb-data-channel .
>>>
>>> Should the 'error' indications' in the proposed text be more 
>>> specific? That is, should the error indications indicate the type of 
>>> error. (The same error indication for all errors could be sent and 
>>> be compliant with the currently proposed text.)
>> Yes, possibly more detailed failure cause information is beneficial. 
>> But this needs to be matched against the ambitions of the WebRTC API 
>> to be as similar as possible to the Websocket API.
>> And the Websocket API groups many error reasons into one error 
>> indication because of security risks.
>> See http://dev.w3.org/html5/websockets/
>>
>> Maintenance and debugging might be hard without further refinement of 
>> error causes, while security is better with less specific 
>> information. A balanced proposal would be needed.
>
> The need for debugging is handled by advice on what information needs 
> to be in the "message" part of the error indication.
>
> If there is a need for consistent action to be taken by applications, 
> I think that needs to be in the "name" part - that is, different 
> errors - or in additional information passed in the error.
There are at least the following reasons:

1. Sending seems to have failed ( you do not know, it might be the 
feedback that failed and the receiver got the message ).

2. Transport aborted because of network errors. (transmission retry 
limit exceeded on a reliable channel in the association, or a watchdog 
expired )

( further  error causes appear in the WebRTC API, such as buffer full 
when requesting SEND )


>
>>
>> A specific improvement would be to indicate if it is the sending side 
>> or the receiving side or both that will get an error indication.
>>
>> I have not seen the association concept in the WebRTC API. So, it 
>> seems that the association concept will be hidden in the 
>> implementation between the WebRTC API and the SCTP, because in 
>> interaction with SCTP, it is important to know when a new association 
>> needs to be opened, and when just a new channel can be opened in an 
>> existing association. The effect of the abortion of the association 
>> also needs to be understood by that layer and translated into 
>> separate rtcDataChannel API actions.
>>
>> One further note: I have assumed that ICE connectivity failures is 
>> handled separately by the WebRTC API implementation and will cause 
>> both indications to API users and to the RTCweb data channel 
>> implementations. Is that right?
>
> I can't parse this in terms that fit my implementation - there is no 
> distinction between the "RTCweb data channel implementation" and "the 
> WebRTC API implementation" - it's all the WebRTC implementation, and 
> there is no distinction that can be surfaced.
>
> What's the change that needs to be there?
Probably none for the lack of the Association concept on the application 
level.
Below the API, the implementation need to keep track of the association 
and all channels using that association, so that a new association is 
requested only when the old has been aborted.

For the ICE connectivity failure, I want to know if my assumption is 
right that the ICE connectivity and its indications of failing 
connectivity are handled by separate API elements, so that it is the 
application that needs to request closing of channels when ICE 
connectivity fails,   while for retransmission excess and connection 
watchdog failures the association will be aborted by underlaying levels, 
and closing with error will be signaled on all channels in that 
association from lower layer.

Therefore, I have not mentioned failing ICE connectivity in the comments 
to rtcweb-data-channel. If the assumption is wrong, failing ICE 
connectivity should be added to the reasons for the association to be 
aborted in the rtcweb-data-channel draft.



/Gunnar

>
>
>>
>>
>>>
>>> Note - I assume that you are proposing adding the new Section 6.7 
>>> to  draft-ietf-rtcweb-data-channel-11 . It already has a Section 6.7 
>>> titled "Closing a channel". I assume you mean a new Section 6.8.
>> Right, thanks.
>>>
>>> Cheers,
>>> /Barry Dingle
>>>
>>>
>>>
>>>
>>> On Sat, Aug 16, 2014 at 4:44 PM, Gunnar Hellstrom 
>>> <gunnar.hellstrom@omnitor.se <mailto:gunnar.hellstrom@omnitor.se>> 
>>> wrote:
>>>
>>>     We have discussed the lack of description of failure handling in
>>>     draft-ietf-rtcweb-data-channel a couple of times.
>>>
>>>     Explanations of mechanisms have been provided in mail answers
>>>     that I have not seen documented elsewhere, but they have not
>>>     been introduced in the draft.
>>>
>>>     I think that for specification and implementation of the WebRTC
>>>     API, there is a need for clear specification how the channels
>>>     behave in failure situations.
>>>
>>>     I suggest to insert a new chapter 6.7 in the draft with the
>>>     following contents.
>>>
>>>     -------------------------------------------------------------------Proposed
>>>     new section
>>>     6.7-------------------------------------------------------------------------------------------
>>>     6.7 Transmission failure and error handling
>>>
>>>     Failures can occur in an open channel by transmission error
>>>     detection and by SCTP watchdog error indications.
>>>     If transmission in a reliable channel fails, the association
>>>     where the channel belongs will be aborted. If all retries for a
>>>     watchdog expires, the corresponding association will be aborted.
>>>     All channels in an association that is aborted because of errors
>>>     need to be closed by the party detecting the failure. Network
>>>     error indications shall be provided to upper layers on all
>>>     closed channels.
>>>
>>>     If retransmissions or transmission time is exceeded in a
>>>     partially reliable channel, the transmission SHALL be allowed to
>>>     continue with other data items. A transmission error indication
>>>     SHALL be provided to upper layers on that channel.
>>>
>>>     If reception out-of order is detected from an SCTP channel for
>>>     which ordered transmission is requested, the receiver SHALL
>>>     close the data channel, and provide an transmission error
>>>     indication to upper layers.
>>>
>>>     If a message with an unsupported PPID is received or some
>>>     logical error is detected by the receiver, the receiver SHALL
>>>     close the corresponding data channel.
>>>     -----------------------------------------------------------------------end
>>>     of
>>>     proposal---------------------------------------------------------------------------------------------------------------
>>>
>>>     I am not sure about what the two last kinds of errors should
>>>     result in and hope that the experts can adjust the wording.
>>>
>>>     Regards
>>>
>>>     Gunnar
>>>     ------------------------------------------------------------------------
>>>     Gunnar Hellström
>>>     Omnitor
>>>     gunnar.hellstrom@omnitor.se <mailto:gunnar.hellstrom@omnitor.se>
>>>
>>>     On 2014-08-11 20:13, IESG Secretary wrote:
>>>>     The IESG would like to retract the current Last Call on the following
>>>>     document:
>>>>
>>>>     - 'WebRTC Data Channels'
>>>>        <draft-ietf-rtcweb-data-channel-11.txt> as Proposed Standard
>>>>
>>>>     The document's state has been set back to "AD Evaluation."
>>>>
>>>
>>>
>>>     _______________________________________________
>>>     rtcweb mailing list
>>>     rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>>>     https://www.ietf.org/mailman/listinfo/rtcweb
>>>
>>>
>>
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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    <div class="moz-cite-prefix"><br>
      On 2014-08-18 10:30, Harald Alvestrand wrote:<br>
    </div>
    <blockquote cite="mid:53F1B99A.1010404@alvestrand.no" type="cite">
      <meta content="text/html; charset=ISO-8859-1"
        http-equiv="Content-Type">
      <div class="moz-cite-prefix">On 08/16/2014 03:25 PM, Gunnar
        Hellstrom wrote:<br>
      </div>
      <blockquote cite="mid:53EF5BC8.7060803@omnitor.se" type="cite">
        <meta content="text/html; charset=ISO-8859-1"
          http-equiv="Content-Type">
        <div class="moz-cite-prefix">On 2014-08-16 14:01, Barry Dingle
          wrote:<br>
        </div>
        <blockquote
cite="mid:CAN=GVAuc-DEzEh3qUVugcRWm85tUJggmA+-zw3hLwT+v6m1srw@mail.gmail.com"
          type="cite">
          <div dir="ltr">
            <div>I agree that error handling is missing from
              draft-ietf-rtcweb-data-channel . <br>
              <br>
            </div>
            <div>Should the 'error' indications' in the proposed text be
              more specific? That is, should the error indications
              indicate the type of error. (The same error indication for
              all errors could be sent and be compliant with the
              currently proposed text.) <br>
            </div>
          </div>
        </blockquote>
        Yes, possibly more detailed failure cause information is
        beneficial. But this needs to be matched against the ambitions
        of the WebRTC API to be as similar as possible to the Websocket
        API.<br>
        And the Websocket API groups many error reasons into one error
        indication because of security risks.<br>
        See <a moz-do-not-send="true" class="moz-txt-link-freetext"
          href="http://dev.w3.org/html5/websockets/">http://dev.w3.org/html5/websockets/</a><br>
        <br>
        Maintenance and debugging might be hard without further
        refinement of error causes, while security is better with less
        specific information. A balanced proposal would be needed. <br>
      </blockquote>
      <br>
      The need for debugging is handled by advice on what information
      needs to be in the "message" part of the error indication.<br>
      <br>
      If there is a need for consistent action to be taken by
      applications, I think that needs to be in the "name" part - that
      is, different errors - or in additional information passed in the
      error.<br>
    </blockquote>
    There are at least the following reasons:<br>
    <br>
    1. Sending seems to have failed ( you do not know, it might be the
    feedback that failed and the receiver got the message ).<br>
    <br>
    2. Transport aborted because of network errors. (transmission retry
    limit exceeded on a reliable channel in the association, or a
    watchdog expired )<br>
    <br>
    ( further&nbsp; error causes appear in the WebRTC API, such as buffer
    full when requesting SEND ) <br>
    <br>
    <br>
    <blockquote cite="mid:53F1B99A.1010404@alvestrand.no" type="cite"> <br>
      <blockquote cite="mid:53EF5BC8.7060803@omnitor.se" type="cite"> <br>
        A specific improvement would be to indicate if it is the sending
        side or the receiving side or both that will get an error
        indication. <br>
        <br>
        I have not seen the association concept in the WebRTC API. So,
        it seems that the association concept will be hidden in the
        implementation between the WebRTC API and the SCTP, because in
        interaction with SCTP, it is important to know when a new
        association needs to be opened, and when just a new channel can
        be opened in an existing association. The effect of the abortion
        of the association also needs to be understood by that layer and
        translated into separate rtcDataChannel API actions.<br>
        <br>
        One further note: I have assumed that ICE connectivity failures
        is handled separately by the WebRTC API implementation and will
        cause both indications to API users and to the RTCweb data
        channel implementations. Is that right?<br>
      </blockquote>
      <br>
      I can't parse this in terms that fit my implementation - there is
      no distinction between the "RTCweb data channel implementation"
      and "the WebRTC API implementation" - it's all the WebRTC
      implementation, and there is no distinction that can be surfaced.<br>
      <br>
      What's the change that needs to be there?<br>
    </blockquote>
    Probably none for the lack of the Association concept on the
    application level. <br>
    Below the API, the implementation need to keep track of the
    association and all channels using that association, so that a new
    association is requested only when the old has been aborted.<br>
    <br>
    For the ICE connectivity failure, I want to know if my assumption is
    right that the ICE connectivity and its indications of failing
    connectivity are handled by separate API elements, so that it is the
    application that needs to request closing of channels when ICE
    connectivity fails,&nbsp;&nbsp; while for retransmission excess and connection
    watchdog failures the association will be aborted by underlaying
    levels, and closing with error will be signaled on all channels in
    that association from lower layer.<br>
    <br>
    Therefore, I have not mentioned failing ICE connectivity in the
    comments to rtcweb-data-channel. If the assumption is wrong, failing
    ICE connectivity should be added to the reasons for the association
    to be aborted in the rtcweb-data-channel draft.<br>
    <br>
    <br>
    <br>
    /Gunnar<br>
    <br>
    <blockquote cite="mid:53F1B99A.1010404@alvestrand.no" type="cite"> <br>
      <br>
      <blockquote cite="mid:53EF5BC8.7060803@omnitor.se" type="cite"> <br>
        &nbsp; <br>
        <blockquote
cite="mid:CAN=GVAuc-DEzEh3qUVugcRWm85tUJggmA+-zw3hLwT+v6m1srw@mail.gmail.com"
          type="cite">
          <div dir="ltr">
            <div> </div>
            <div><br>
            </div>
            Note - I assume that you are proposing adding the new
            Section 6.7 to&nbsp; draft-ietf-rtcweb-data-channel-11 . It
            already has a Section 6.7 titled "Closing a channel". I
            assume you mean a new Section 6.8. <br>
          </div>
        </blockquote>
        Right, thanks.<br>
        <blockquote
cite="mid:CAN=GVAuc-DEzEh3qUVugcRWm85tUJggmA+-zw3hLwT+v6m1srw@mail.gmail.com"
          type="cite">
          <div dir="ltr">
            <div class="gmail_extra"><br clear="all">
              <div>
                <div dir="ltr">Cheers,<br>
                  /Barry Dingle<br>
                  <br>
                  <br>
                </div>
              </div>
              <br>
              <br>
              <div class="gmail_quote">On Sat, Aug 16, 2014 at 4:44 PM,
                Gunnar Hellstrom <span dir="ltr">&lt;<a
                    moz-do-not-send="true"
                    href="mailto:gunnar.hellstrom@omnitor.se"
                    target="_blank">gunnar.hellstrom@omnitor.se</a>&gt;</span>
                wrote:<br>
                <blockquote class="gmail_quote" style="margin:0 0 0
                  .8ex;border-left:1px #ccc solid;padding-left:1ex">
                  <div bgcolor="#FFFFFF" text="#000000">
                    <div>We have discussed the lack of description of
                      failure handling in draft-ietf-rtcweb-data-channel
                      a couple of times.<br>
                      <br>
                      Explanations of mechanisms have been provided in
                      mail answers that I have not seen documented
                      elsewhere, but they have not been introduced in
                      the draft.<br>
                      <br>
                      I think that for specification and implementation
                      of the WebRTC API, there is a need for clear
                      specification how the channels behave in failure
                      situations.<br>
                      <br>
                      I suggest to insert a new chapter 6.7 in the draft
                      with the following contents.&nbsp; <br>
                      <br>
                      -------------------------------------------------------------------Proposed



                      new section
6.7-------------------------------------------------------------------------------------------<br>
                      6.7 Transmission failure and error handling <br>
                      <br>
                      Failures can occur in an open channel by
                      transmission error detection and by SCTP watchdog
                      error indications.<br>
                      If transmission in a reliable channel fails, the
                      association where the channel belongs will be
                      aborted. If all retries for a watchdog expires,
                      the corresponding association will be aborted.&nbsp;
                      All channels in an association that is aborted
                      because of errors need to be closed by the party
                      detecting the failure. Network error indications
                      shall be provided to upper layers on all closed
                      channels.<br>
                      <br>
                      If retransmissions or transmission time is
                      exceeded in a partially reliable channel, the
                      transmission SHALL be allowed to continue with
                      other data items. A transmission error indication
                      SHALL be provided to upper layers on that channel.
                      <br>
                      <br>
                      If reception out-of order is detected from an SCTP
                      channel for which ordered transmission is
                      requested, the receiver SHALL close the data
                      channel, and provide an transmission error
                      indication to upper layers.&nbsp; <br>
                      <br>
                      If a message with an unsupported PPID is received
                      or some logical error is detected by the receiver,
                      the receiver SHALL close the corresponding data
                      channel. <br>
                      -----------------------------------------------------------------------end



                      of
proposal---------------------------------------------------------------------------------------------------------------<br>
                      <br>
                      I am not sure about what the two last kinds of
                      errors should result in and hope that the experts
                      can adjust the wording.<br>
                      <br>
                      Regards<br>
                      <br>
                      Gunnar<br>
                      <div>
                        <hr> Gunnar Hellstr&ouml;m<br>
                        Omnitor<br>
                        <a moz-do-not-send="true"
                          href="mailto:gunnar.hellstrom@omnitor.se"
                          target="_blank">gunnar.hellstrom@omnitor.se</a><br>
                        <br>
                      </div>
                      On 2014-08-11 20:13, IESG Secretary wrote:<br>
                    </div>
                    <blockquote type="cite">
                      <pre>The IESG would like to retract the current Last Call on the following 
document:

- 'WebRTC Data Channels'
  &lt;draft-ietf-rtcweb-data-channel-11.txt&gt; as Proposed Standard

The document's state has been set back to "AD Evaluation."

</pre>
                    </blockquote>
                    <br>
                  </div>
                  <br>
                  _______________________________________________<br>
                  rtcweb mailing list<br>
                  <a moz-do-not-send="true"
                    href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
                  <a moz-do-not-send="true"
                    href="https://www.ietf.org/mailman/listinfo/rtcweb"
                    target="_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
                  <br>
                </blockquote>
              </div>
              <br>
            </div>
          </div>
        </blockquote>
        <br>
        <br>
        <fieldset class="mimeAttachmentHeader"></fieldset>
        <br>
        <pre wrap="">_______________________________________________
rtcweb mailing list
<a moz-do-not-send="true" class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a moz-do-not-send="true" class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
      </blockquote>
      <br>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_______________________________________________
rtcweb mailing list
<a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
    </blockquote>
    <br>
  </body>
</html>

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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Unsupported PPID (Re: draft-ietf-rtcweb-data-channel failure handling description)
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Picking up a single point....

On 08/16/2014 08:44 AM, Gunnar Hellstrom wrote:

If a message with an unsupported PPID is received or some logical error is =
detected by the receiver, the receiver SHALL close the corresponding data c=
hannel.

<Raju/> This is one of the interop concern I expressed on using EMPTY PPID =
without negotiating it. </Raju>

This struct me when thinking about the zero-length-data proposal: If we eve=
r want to add another such feature *without* explicitly negotiating the use=
 of the feature at setup time, we can't do this.
<Raju> Right. I am in favor of explicitly negotiating such new PPIDs. </Raj=
u>


If we want the possibility of un-negotiated extensions, we HAVE to say (som=
ewhere)

  If a message with an unsupported PPID is received, it is silently ignored=
.

OTOH, if we're convinced we never need that kind of extensibility, any way =
we do this is fine.
<Raju> Silently dropping these "unknown" PPIDs may have issues like the sen=
der may "assume" the msg is "accepted" by the receiver, especially for the =
EMPTY_PPID kind of use case where there is no response required. Also, at t=
he receiving side an "unknown" PPID might  have been received due to an err=
or at the sending side (rather than transit). In such error cases it is bet=
ter to drop the call rather than "limp along". Having an explicit negotiati=
on allows the implementation to take an informed decision.
Sorry to keep going back, but the earlier proposal I made about OOB PPID ma=
y solve this issue by allowing user to negotiate an additional OOB PPID.
The same mechanism can be extended to allow users to define and use arbitra=
ry PPIDs. I am aware that this complicates the spec/API further, but listin=
g it here for completeness and further discussion.
</Raju>
BR
Raju

             Harald

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<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">Picking up a single point....<br>
<br>
On 08/16/2014 08:44 AM, Gunnar Hellstrom wrote:<o:p></o:p></p>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<p class=3D"MsoNormal"><br>
If a message with an unsupported PPID is received or some logical error is =
detected by the receiver, the receiver SHALL close the corresponding data c=
hannel.&nbsp;<span style=3D"color:#1F497D"><o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:0in;margin-right:.5in;ma=
rgin-bottom:5.0pt;margin-left:0in">
<span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-=
serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span></p>
</div>
</blockquote>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"color:=
#1F497D">&lt;Raju/&gt; This is one of the interop concern I expressed on us=
ing EMPTY PPID without negotiating it. &lt;/Raju&gt;<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
This struct me when thinking about the zero-length-data proposal: If we eve=
r want to add another such feature *without* explicitly negotiating the use=
 of the feature at setup time, we can't do this.<span style=3D"color:#1F497=
D"><o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F=
497D">&lt;Raju&gt; Right. I am in favor of explicitly negotiating such new =
PPIDs. &lt;/Raju&gt;<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
<br>
If we want the possibility of un-negotiated extensions, we HAVE to say (som=
ewhere)<br>
<br>
&nbsp; If a message with an unsupported PPID is received, it is silently ig=
nored.<br>
<br>
OTOH, if we're convinced we never need that kind of extensibility, any way =
we do this is fine.<span style=3D"color:#1F497D"><o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F=
497D">&lt;Raju&gt; Silently dropping these &#8220;unknown&#8221; PPIDs may =
have issues like the sender may &#8220;assume&#8221; the msg is &#8220;acce=
pted&#8221; by the receiver,
 especially for the EMPTY_PPID kind of use case where there is no response =
required. Also, at the receiving side an &#8220;unknown&#8221; PPID might &=
nbsp;have been received due to an error at the sending side (rather than tr=
ansit). In such error cases it is better to drop
 the call rather than &#8220;limp along&#8221;. Having an explicit negotiat=
ion allows the implementation to take an informed decision.<o:p></o:p></spa=
n></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F=
497D">Sorry to keep going back, but the earlier proposal I made about OOB P=
PID may solve this issue by allowing user to negotiate an
 additional OOB PPID. <o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F=
497D">The same mechanism can be extended to allow users to define and use a=
rbitrary PPIDs. I am aware that this complicates the spec/API
 further, but listing it here for completeness and further discussion.<o:p>=
</o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F=
497D">&lt;/Raju&gt;<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">BR<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Raju<br>
</span><br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Ha=
rald<o:p></o:p></p>
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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Gunnar Hellstrom <gunnar.hellstrom@omnitor.se>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Unsupported PPID (Re: draft-ietf-rtcweb-data-channel failure handling description)
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Subject: Re: [rtcweb] Unsupported PPID (Re: draft-ietf-rtcweb-data-channel failure handling description)
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From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Gunnar Hellstrom
Sent: Monday, August 18, 2014 6:24 AM

On 2014-08-18 13:15, Harald Alvestrand wrote:
Picking up a single point....

On 08/16/2014 08:44 AM, Gunnar Hellstrom wrote:

If a message with an unsupported PPID is received or some logical error is =
detected by the receiver, the receiver SHALL close the corresponding data c=
hannel.

This struct me when thinking about the zero-length-data proposal: If we eve=
r want to add another such feature *without* explicitly negotiating the use=
 of the feature at setup time, we can't do this.

If we want the possibility of un-negotiated extensions, we HAVE to say (som=
ewhere)

  If a message with an unsupported PPID is received, it is silently ignored=
.
[GH] That sounds wise, and also in line with the security aspects that make=
 Websocket be very conservative about signaling errors and their reasons. W=
ebRTC data channels have the ambition to be similar.
/Gunnar
<Raju>
Per the following excerpts from websocket RFC, if an "unknown" opcode is re=
ceived then the connection fails. I think PPID is same level as Opcode (web=
socket binary and text data are represented as 2 opcodes). Did I miss somet=
hing?
5.2<http://tools.ietf.org/html/rfc6455#section-5.2>.  Base Framing Protocol

Opcode:  4 bits

      Defines the interpretation of the "Payload data".  If an unknown
      opcode is received, the receiving endpoint MUST _Fail the
      WebSocket Connection_.  The following values are defined.

</Raju>

BR
Raju


OTOH, if we're convinced we never need that kind of extensibility, any way =
we do this is fine.

             Harald





_______________________________________________

rtcweb mailing list

rtcweb@ietf.org<mailto:rtcweb@ietf.org>

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


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<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
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<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext">From:</span></b><spa=
n style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif=
&quot;;color:windowtext"> rtcweb [mailto:rtcweb-bounces@ietf.org]
<b>On Behalf Of </b>Gunnar Hellstrom<br>
<b>Sent:</b> Monday, August 18, 2014 6:24 AM<br>
<br>
</span><o:p></o:p></p>
<div>
<p class=3D"MsoNormal">On 2014-08-18 13:15, Harald Alvestrand wrote:<o:p></=
o:p></p>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<p class=3D"MsoNormal">Picking up a single point....<br>
<br>
On 08/16/2014 08:44 AM, Gunnar Hellstrom wrote:<o:p></o:p></p>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<p class=3D"MsoNormal"><br>
If a message with an unsupported PPID is received or some logical error is =
detected by the receiver, the receiver SHALL close the corresponding data c=
hannel.&nbsp;<o:p></o:p></p>
</div>
</blockquote>
<p class=3D"MsoNormal"><br>
This struct me when thinking about the zero-length-data proposal: If we eve=
r want to add another such feature *without* explicitly negotiating the use=
 of the feature at setup time, we can't do this.<br>
<br>
If we want the possibility of un-negotiated extensions, we HAVE to say (som=
ewhere)<br>
<br>
&nbsp; If a message with an unsupported PPID is received, it is silently ig=
nored.<o:p></o:p></p>
</blockquote>
<p class=3D"MsoNormal">[GH] That sounds wise, and also in line with the sec=
urity aspects that make Websocket be very conservative about signaling erro=
rs and their reasons. WebRTC data channels have the ambition to be similar.=
<br>
/Gunnar<span style=3D"color:#1F497D"><o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Calibri&quot;,&quot=
;sans-serif&quot;;color:#1F497D">&lt;Raju&gt;<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Calibri&quot;,&quot=
;sans-serif&quot;;color:#1F497D">Per the following excerpts from websocket =
RFC, if an &#8220;unknown&#8221; opcode is received then the connection fai=
ls. I think PPID is same level as Opcode (websocket binary and text
 data are represented as 2 opcodes). Did I miss something?<o:p></o:p></span=
></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;mso-line-height-alt:0pt;page-break-before:always">
<a name=3D"section-5.2"></a><a href=3D"http://tools.ietf.org/html/rfc6455#s=
ection-5.2"><b><span style=3D"font-family:&quot;Courier New&quot;,&quot;ser=
if&quot;;color:black">5.2</span></b></a><b><span style=3D"font-family:&quot=
;Courier New&quot;,&quot;serif&quot;">.&nbsp; Base Framing Protocol<o:p></o=
:p></span></b></p>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Calibri&quot;,&quot=
;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal" style=3D"page-break-before:always"><span style=3D"fo=
nt-family:&quot;Courier New&quot;,&quot;serif&quot;">Opcode:&nbsp; 4 bits<o=
:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"page-break-before:always"><span style=3D"fo=
nt-family:&quot;Courier New&quot;,&quot;serif&quot;"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal" style=3D"page-break-before:always"><span style=3D"fo=
nt-family:&quot;Courier New&quot;,&quot;serif&quot;">&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp; Defines the interpretation of the &quot;Payload data&quot;.&nbsp; =
If an unknown<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"page-break-before:always"><span style=3D"fo=
nt-family:&quot;Courier New&quot;,&quot;serif&quot;">&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp; opcode is received, the receiving endpoint MUST _Fail the<o:p></o:=
p></span></p>
<p class=3D"MsoNormal" style=3D"page-break-before:always"><span style=3D"fo=
nt-family:&quot;Courier New&quot;,&quot;serif&quot;">&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp; WebSocket Connection_.&nbsp; The following values are defined.<o:p=
></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Calibri&quot;,&quot=
;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Calibri&quot;,&quot=
;sans-serif&quot;;color:#1F497D">&lt;/Raju&gt;<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Calibri&quot;,&quot=
;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Calibri&quot;,&quot=
;sans-serif&quot;;color:#1F497D">BR<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Calibri&quot;,&quot=
;sans-serif&quot;;color:#1F497D">Raju<br>
<br>
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><br>
OTOH, if we're convinced we never need that kind of extensibility, any way =
we do this is fine.<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Ha=
rald<br>
<br>
<br>
<br>
<br>
<o:p></o:p></p>
<pre>_______________________________________________<o:p></o:p></pre>
<pre>rtcweb mailing list<o:p></o:p></pre>
<pre><a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><o:p></o:p></pre=
>
<pre><a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.i=
etf.org/mailman/listinfo/rtcweb</a><o:p></o:p></pre>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
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Archived-At: http://mailarchive.ietf.org/arch/msg/rtcweb/DJDq4R6sedTj0tINvk2agjeEUhE
Subject: [rtcweb]  WebRTC Device: Lowering the bar, or inventing a new term?
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--=_NamoWEC-4v4yag1e2n--



From nobody Mon Aug 18 09:49:58 2014
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Date: Mon, 18 Aug 2014 09:49:54 -0700
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From: Martin Thomson <martin.thomson@gmail.com>
To: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] Unsupported PPID (Re: draft-ietf-rtcweb-data-channel failure handling description)
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On 18 August 2014 04:15, Harald Alvestrand <harald@alvestrand.no> wrote:
> OTOH, if we're convinced we never need that kind of extensibility, any way
> we do this is fine.

I think that I want that kind of extensibility.  People who care about
secure text and files might also.


From nobody Mon Aug 18 10:34:22 2014
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From: "Parthasarathi R" <partha@parthasarathi.co.in>
To: "'Makaraju, Maridi Raju \(Raju\)'" <Raju.Makaraju@alcatel-lucent.com>, <rtcweb@ietf.org>, "'Harald Alvestrand'" <harald@alvestrand.no>
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Date: Mon, 18 Aug 2014 23:04:09 +0530
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Subject: Re: [rtcweb] WebRTC Device: Lowering the bar, or inventing a new term?
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Hi Harald & all,

I'm seeing that we are discussing about four WebRTC entities in three term
which is confusing. I propose the following four entities to discuss:

1) WebRTC browser (full compliant Protocol + API) - as in
draft-ietf-rtcweb-overview-10

2) WebRTC device (Full complaint protocol) - as in
draft-ietf-rtcweb-overview-10

3) Invent new WebRTC term ("WebRTC endpoint") by lowering the bar. WebRTC
endpoint mean "that the entity which can successfully communicate with a
WebRTC browser by implementing only the absolute minimum of what a WebRTC
browser has to implement". Ex: WebRTC IVR application which is hosted in the
public internet.

4) "WebRTC gateway" is entity which has the WebRTC endpoint in one side and
interwork with other entities (like SIP entities, Jingle, ISDN or WebRTC
entities) in the another side. Here, we need additional requirements in
WebRTC gateway which has to be documented in Sec 3 of
draft-alvestrand-rtcweb-gateways-00.txt. Ex additional requirements: ICE-to
Non-ICE handling, DTLS-SRTP to RTP handling.

Please let me know your opinion on the same.

Thanks
Partha


> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Makaraju,
> Maridi Raju (Raju)
> Sent: Monday, August 18, 2014 5:11 PM
> To: rtcweb@ietf.org; Harald Alvestrand
> Subject: Re: [rtcweb] WebRTC Device: Lowering the bar, or inventing a
> new term?
> 
> Hi Harald,
> 
> How about "gw-endpoint"? in the sense that it behaves like an
> "endpoint" but has different requirements than an "endpoint".
> 
> BR
> Raju
> 
> 
> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Sergio
> Garcia Murillo
> Sent: Monday, August 18, 2014 6:25 AM
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] WebRTC Device: Lowering the bar, or inventing a
> new term?
> 
> Hi Harald,
> 
> I always liked "endpoint" much more than "device".
> 
> We could have different types of endpoints, and define a browser like:
> a
> WebRTC "full" endpoint exposing the W3C WebRTC APIs.
> 
> Just my 2 cents.
> 
> Best regards
> Sergio
> On 18/08/2014 13:11, Harald Alvestrand wrote:
> > When writing the gateway document, I found myself grappling with the
> > "device" description.
> >
> > It is somewhat uncomfortable, in that as first defined, it requires
> > full WebRTC functionality except for the API part - this means that
> it
> > should support data channels, full ICE, bundling and so on.
> >
> > But for gateways, it makes sense to let go of a lot of these
> > requirements. So a gateway is not a device. But what is the name for
> > the thing that a gateway is, a device is, and a browser is, that
> names
> > "the set of things that can successfully negotiate connectivity"? And
> > what are the requirements on *that*?
> >
> > I see a couple of options:
> >
> > - Redefine "Device" to mean "that which can successfully communicate
> > with a browser". It's free to negotiate away features if it can, but
> > needs to interoperate with a browser that implements only the
> absolute
> > minimum of what a browser has to implement.
> >
> > - Define a new class of "thing" which is "the thing that can talk to
> > browsers" - of which a gateway has to be an exemplar.
> >
> > This is a kind of word game in some ways, but when we make
> > definitions, we also define what's easy to talk about. And that can
> > influence what we end up building.
> >
> > What do people think?
> >
> >    Harald
> >
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
> 
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> 
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From nobody Mon Aug 18 15:10:16 2014
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From: Silvia Pfeiffer <silviapfeiffer1@gmail.com>
Date: Tue, 19 Aug 2014 08:09:52 +1000
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To: Parthasarathi R <partha@parthasarathi.co.in>
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Subject: Re: [rtcweb] WebRTC Device: Lowering the bar, or inventing a new term?
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On Tue, Aug 19, 2014 at 3:34 AM, Parthasarathi R
<partha@parthasarathi.co.in> wrote:
> Hi Harald & all,
>
> I'm seeing that we are discussing about four WebRTC entities in three term
> which is confusing. I propose the following four entities to discuss:
>
> 1) WebRTC browser (full compliant Protocol + API) - as in
> draft-ietf-rtcweb-overview-10

To me, this is a WebRTC endpoint.


> 2) WebRTC device (Full complaint protocol) - as in
> draft-ietf-rtcweb-overview-10

To me, this is a RTCWeb endpoint (it doesn't care about the browser
API, but only the stuff that is defined in the rtcweb group of IEEE).

Silvia.

> 3) Invent new WebRTC term ("WebRTC endpoint") by lowering the bar. WebRTC
> endpoint mean "that the entity which can successfully communicate with a
> WebRTC browser by implementing only the absolute minimum of what a WebRTC
> browser has to implement". Ex: WebRTC IVR application which is hosted in the
> public internet.
>
> 4) "WebRTC gateway" is entity which has the WebRTC endpoint in one side and
> interwork with other entities (like SIP entities, Jingle, ISDN or WebRTC
> entities) in the another side. Here, we need additional requirements in
> WebRTC gateway which has to be documented in Sec 3 of
> draft-alvestrand-rtcweb-gateways-00.txt. Ex additional requirements: ICE-to
> Non-ICE handling, DTLS-SRTP to RTP handling.
>
> Please let me know your opinion on the same.
>
> Thanks
> Partha
>
>
>> -----Original Message-----
>> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Makaraju,
>> Maridi Raju (Raju)
>> Sent: Monday, August 18, 2014 5:11 PM
>> To: rtcweb@ietf.org; Harald Alvestrand
>> Subject: Re: [rtcweb] WebRTC Device: Lowering the bar, or inventing a
>> new term?
>>
>> Hi Harald,
>>
>> How about "gw-endpoint"? in the sense that it behaves like an
>> "endpoint" but has different requirements than an "endpoint".
>>
>> BR
>> Raju
>>
>>
>> -----Original Message-----
>> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Sergio
>> Garcia Murillo
>> Sent: Monday, August 18, 2014 6:25 AM
>> To: rtcweb@ietf.org
>> Subject: Re: [rtcweb] WebRTC Device: Lowering the bar, or inventing a
>> new term?
>>
>> Hi Harald,
>>
>> I always liked "endpoint" much more than "device".
>>
>> We could have different types of endpoints, and define a browser like:
>> a
>> WebRTC "full" endpoint exposing the W3C WebRTC APIs.
>>
>> Just my 2 cents.
>>
>> Best regards
>> Sergio
>> On 18/08/2014 13:11, Harald Alvestrand wrote:
>> > When writing the gateway document, I found myself grappling with the
>> > "device" description.
>> >
>> > It is somewhat uncomfortable, in that as first defined, it requires
>> > full WebRTC functionality except for the API part - this means that
>> it
>> > should support data channels, full ICE, bundling and so on.
>> >
>> > But for gateways, it makes sense to let go of a lot of these
>> > requirements. So a gateway is not a device. But what is the name for
>> > the thing that a gateway is, a device is, and a browser is, that
>> names
>> > "the set of things that can successfully negotiate connectivity"? And
>> > what are the requirements on *that*?
>> >
>> > I see a couple of options:
>> >
>> > - Redefine "Device" to mean "that which can successfully communicate
>> > with a browser". It's free to negotiate away features if it can, but
>> > needs to interoperate with a browser that implements only the
>> absolute
>> > minimum of what a browser has to implement.
>> >
>> > - Define a new class of "thing" which is "the thing that can talk to
>> > browsers" - of which a gateway has to be an exemplar.
>> >
>> > This is a kind of word game in some ways, but when we make
>> > definitions, we also define what's easy to talk about. And that can
>> > influence what we end up building.
>> >
>> > What do people think?
>> >
>> >    Harald
>> >
>> > _______________________________________________
>> > rtcweb mailing list
>> > rtcweb@ietf.org
>> > https://www.ietf.org/mailman/listinfo/rtcweb
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From nobody Mon Aug 18 15:32:26 2014
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From: Silvia Pfeiffer <silviapfeiffer1@gmail.com>
Date: Tue, 19 Aug 2014 08:32:00 +1000
Message-ID: <CAHp8n2kD_TaDKVbvd5qkW--FHsG2vtj7mK7-bzmLzgAsWj45qA@mail.gmail.com>
To: Parthasarathi R <partha@parthasarathi.co.in>
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Subject: Re: [rtcweb] WebRTC Device: Lowering the bar, or inventing a new term?
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On Tue, Aug 19, 2014 at 8:09 AM, Silvia Pfeiffer
<silviapfeiffer1@gmail.com> wrote:
> On Tue, Aug 19, 2014 at 3:34 AM, Parthasarathi R
> <partha@parthasarathi.co.in> wrote:
>> Hi Harald & all,
>>
>> I'm seeing that we are discussing about four WebRTC entities in three term
>> which is confusing. I propose the following four entities to discuss:
>>
>> 1) WebRTC browser (full compliant Protocol + API) - as in
>> draft-ietf-rtcweb-overview-10
>
> To me, this is a WebRTC endpoint.
>
>
>> 2) WebRTC device (Full complaint protocol) - as in
>> draft-ietf-rtcweb-overview-10
>
> To me, this is a RTCWeb endpoint (it doesn't care about the browser
> API, but only the stuff that is defined in the rtcweb group of IEEE).


BTW: I don't actually mind calling this a "WebRTC device" either. Just
thought I should add this for discussion.

Cheers,
Silvia.

> Silvia.
>
>> 3) Invent new WebRTC term ("WebRTC endpoint") by lowering the bar. WebRTC
>> endpoint mean "that the entity which can successfully communicate with a
>> WebRTC browser by implementing only the absolute minimum of what a WebRTC
>> browser has to implement". Ex: WebRTC IVR application which is hosted in the
>> public internet.
>>
>> 4) "WebRTC gateway" is entity which has the WebRTC endpoint in one side and
>> interwork with other entities (like SIP entities, Jingle, ISDN or WebRTC
>> entities) in the another side. Here, we need additional requirements in
>> WebRTC gateway which has to be documented in Sec 3 of
>> draft-alvestrand-rtcweb-gateways-00.txt. Ex additional requirements: ICE-to
>> Non-ICE handling, DTLS-SRTP to RTP handling.
>>
>> Please let me know your opinion on the same.
>>
>> Thanks
>> Partha
>>
>>
>>> -----Original Message-----
>>> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Makaraju,
>>> Maridi Raju (Raju)
>>> Sent: Monday, August 18, 2014 5:11 PM
>>> To: rtcweb@ietf.org; Harald Alvestrand
>>> Subject: Re: [rtcweb] WebRTC Device: Lowering the bar, or inventing a
>>> new term?
>>>
>>> Hi Harald,
>>>
>>> How about "gw-endpoint"? in the sense that it behaves like an
>>> "endpoint" but has different requirements than an "endpoint".
>>>
>>> BR
>>> Raju
>>>
>>>
>>> -----Original Message-----
>>> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Sergio
>>> Garcia Murillo
>>> Sent: Monday, August 18, 2014 6:25 AM
>>> To: rtcweb@ietf.org
>>> Subject: Re: [rtcweb] WebRTC Device: Lowering the bar, or inventing a
>>> new term?
>>>
>>> Hi Harald,
>>>
>>> I always liked "endpoint" much more than "device".
>>>
>>> We could have different types of endpoints, and define a browser like:
>>> a
>>> WebRTC "full" endpoint exposing the W3C WebRTC APIs.
>>>
>>> Just my 2 cents.
>>>
>>> Best regards
>>> Sergio
>>> On 18/08/2014 13:11, Harald Alvestrand wrote:
>>> > When writing the gateway document, I found myself grappling with the
>>> > "device" description.
>>> >
>>> > It is somewhat uncomfortable, in that as first defined, it requires
>>> > full WebRTC functionality except for the API part - this means that
>>> it
>>> > should support data channels, full ICE, bundling and so on.
>>> >
>>> > But for gateways, it makes sense to let go of a lot of these
>>> > requirements. So a gateway is not a device. But what is the name for
>>> > the thing that a gateway is, a device is, and a browser is, that
>>> names
>>> > "the set of things that can successfully negotiate connectivity"? And
>>> > what are the requirements on *that*?
>>> >
>>> > I see a couple of options:
>>> >
>>> > - Redefine "Device" to mean "that which can successfully communicate
>>> > with a browser". It's free to negotiate away features if it can, but
>>> > needs to interoperate with a browser that implements only the
>>> absolute
>>> > minimum of what a browser has to implement.
>>> >
>>> > - Define a new class of "thing" which is "the thing that can talk to
>>> > browsers" - of which a gateway has to be an exemplar.
>>> >
>>> > This is a kind of word game in some ways, but when we make
>>> > definitions, we also define what's easy to talk about. And that can
>>> > influence what we end up building.
>>> >
>>> > What do people think?
>>> >
>>> >    Harald
>>> >
>>> > _______________________________________________
>>> > rtcweb mailing list
>>> > rtcweb@ietf.org
>>> > https://www.ietf.org/mailman/listinfo/rtcweb
>>>
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb


From nobody Tue Aug 19 03:19:20 2014
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From: "Hutton, Andrew" <andrew.hutton@unify.com>
To: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] WebRTC Device: Lowering the bar, or inventing a new term?
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My thinking is that there are only two entities we need to be concerned wit=
h.

1. WebRTC Browsers that implement both the IETF protocol requirements and t=
he W3C API in full.

2. Things that can interoperate directly with WebRTC browsers but are not b=
rowsers. These have to implement at least a minimum set of the IETF protoco=
l and it is irrelevant as to whether they have anything link a PeerConnecti=
on API internally. I was happy with calling these WebRTC Devices and think =
it is useful to define the minimum set of things that they need to support.

I am not convinced that we need to define any other entities and my thinkin=
g is that gateways are just a type of WebRTC Device and we don't need to ca=
ll them out explicitly.

Regards
Andy




> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Harald
> Alvestrand
> Sent: 18 August 2014 12:11
> To: rtcweb@ietf.org
> Subject: [rtcweb] WebRTC Device: Lowering the bar, or inventing a new
> term?
>=20
> When writing the gateway document, I found myself grappling with the
> "device" description.
>=20
> It is somewhat uncomfortable, in that as first defined, it requires
> full
> WebRTC functionality except for the API part - this means that it
> should
> support data channels, full ICE, bundling and so on.
>=20
> But for gateways, it makes sense to let go of a lot of these
> requirements. So a gateway is not a device. But what is the name for
> the
> thing that a gateway is, a device is, and a browser is, that names "the
> set of things that can successfully negotiate connectivity"? And what
> are the requirements on *that*?
>=20
> I see a couple of options:
>=20
> - Redefine "Device" to mean "that which can successfully communicate
> with a browser". It's free to negotiate away features if it can, but
> needs to interoperate with a browser that implements only the absolute
> minimum of what a browser has to implement.
>=20
> - Define a new class of "thing" which is "the thing that can talk to
> browsers" - of which a gateway has to be an exemplar.
>=20
> This is a kind of word game in some ways, but when we make definitions,
> we also define what's easy to talk about. And that can influence what
> we
> end up building.
>=20
> What do people think?
>=20
>     Harald
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From nobody Tue Aug 19 06:38:57 2014
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Date: Tue, 19 Aug 2014 09:38:52 -0400
From: Jim Barnett <1jhbarnett@gmail.com>
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Archived-At: http://mailarchive.ietf.org/arch/msg/rtcweb/f8_KfNwlM-XV4Ca1d7fHiPQ5LcI
Subject: Re: [rtcweb] WebRTC Device: Lowering the bar, or inventing a new term?
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I agree.  We need to define a) WebRTC browsers, which support the full 
protocol requirements and the API, b) WebRTC Endpoints/Devices, which 
support some minimum protocol set sufficient for establishing 
connectivity.  Is there also a need to define c) Full WebRTC Devices, 
which support more than the minimum protocol set?  What would be the use 
of definition c?

We know that we have to define the maximal set a) and the minimum usable 
set b).   If we can't think of a use for c) we can omit it and add extra 
profiles later if they turn out to be useful.

On 8/19/2014 6:19 AM, Hutton, Andrew wrote:
> My thinking is that there are only two entities we need to be concerned with.
>
> 1. WebRTC Browsers that implement both the IETF protocol requirements and the W3C API in full.
>
> 2. Things that can interoperate directly with WebRTC browsers but are not browsers. These have to implement at least a minimum set of the IETF protocol and it is irrelevant as to whether they have anything link a PeerConnection API internally. I was happy with calling these WebRTC Devices and think it is useful to define the minimum set of things that they need to support.
>
> I am not convinced that we need to define any other entities and my thinking is that gateways are just a type of WebRTC Device and we don't need to call them out explicitly.
>
> Regards
> Andy
>
>
>
>
>> -----Original Message-----
>> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Harald
>> Alvestrand
>> Sent: 18 August 2014 12:11
>> To: rtcweb@ietf.org
>> Subject: [rtcweb] WebRTC Device: Lowering the bar, or inventing a new
>> term?
>>
>> When writing the gateway document, I found myself grappling with the
>> "device" description.
>>
>> It is somewhat uncomfortable, in that as first defined, it requires
>> full
>> WebRTC functionality except for the API part - this means that it
>> should
>> support data channels, full ICE, bundling and so on.
>>
>> But for gateways, it makes sense to let go of a lot of these
>> requirements. So a gateway is not a device. But what is the name for
>> the
>> thing that a gateway is, a device is, and a browser is, that names "the
>> set of things that can successfully negotiate connectivity"? And what
>> are the requirements on *that*?
>>
>> I see a couple of options:
>>
>> - Redefine "Device" to mean "that which can successfully communicate
>> with a browser". It's free to negotiate away features if it can, but
>> needs to interoperate with a browser that implements only the absolute
>> minimum of what a browser has to implement.
>>
>> - Define a new class of "thing" which is "the thing that can talk to
>> browsers" - of which a gateway has to be an exemplar.
>>
>> This is a kind of word game in some ways, but when we make definitions,
>> we also define what's easy to talk about. And that can influence what
>> we
>> end up building.
>>
>> What do people think?
>>
>>      Harald
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From nobody Tue Aug 19 07:00:43 2014
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Subject: Re: [rtcweb] WebRTC Device: Lowering the bar, or inventing a new term?
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On 08/19/2014 03:38 PM, Jim Barnett wrote:
> I agree.  We need to define a) WebRTC browsers, which support the full 
> protocol requirements and the API, b) WebRTC Endpoints/Devices, which 
> support some minimum protocol set sufficient for establishing 
> connectivity.  Is there also a need to define c) Full WebRTC Devices, 
> which support more than the minimum protocol set?  What would be the 
> use of definition c?

An endpoint (I kind of start to like this concept/name) would be 
conformant as long as it was able to successfully negotiate away use of 
anything that makes life difficult.

It would be required to support DTLS-SRTP key handshakes and respond to 
ICE connectivity checks, plus whatever is MUST use in -rtp-usage 
(scanning, I'm not coming up with much - it's mostly MUST respond to, 
MUST implement (but not required to use) stuff). The RTP circuit breaker 
is also a MUST implement, and since it's a single-ended thing, it makes 
no sense if it isn't also MUST use.

Scanning, I'm not even sure a WebRTC endpoint is required to send RTCP - 
but I think it should be.
(Note: The -rtp-usage draft uses "WebRTC endpoint" to mean a 
fully-featured endpoint... obviously we need to harmonize at some point.)

> We know that we have to define the maximal set a) and the minimum 
> usable set b).   If we can't think of a use for c) we can omit it and 
> add extra profiles later if they turn out to be useful.

An example of c) would be a VC device based on a browser that can be 
plugged into any network (thus needing full ICE), but runs a captive 
application that cannot be made to run any other application.

It would be nice to mandate support in such a device for full ICE and 
the MTI codecs (something I think we will have trouble with mandating 
for gateways).
>
> On 8/19/2014 6:19 AM, Hutton, Andrew wrote:
>> My thinking is that there are only two entities we need to be 
>> concerned with.
>>
>> 1. WebRTC Browsers that implement both the IETF protocol requirements 
>> and the W3C API in full.
>>
>> 2. Things that can interoperate directly with WebRTC browsers but are 
>> not browsers. These have to implement at least a minimum set of the 
>> IETF protocol and it is irrelevant as to whether they have anything 
>> link a PeerConnection API internally. I was happy with calling these 
>> WebRTC Devices and think it is useful to define the minimum set of 
>> things that they need to support.
>>
>> I am not convinced that we need to define any other entities and my 
>> thinking is that gateways are just a type of WebRTC Device and we 
>> don't need to call them out explicitly.
>>
>> Regards
>> Andy
>>
>>
>>
>>
>>> -----Original Message-----
>>> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Harald
>>> Alvestrand
>>> Sent: 18 August 2014 12:11
>>> To: rtcweb@ietf.org
>>> Subject: [rtcweb] WebRTC Device: Lowering the bar, or inventing a new
>>> term?
>>>
>>> When writing the gateway document, I found myself grappling with the
>>> "device" description.
>>>
>>> It is somewhat uncomfortable, in that as first defined, it requires
>>> full
>>> WebRTC functionality except for the API part - this means that it
>>> should
>>> support data channels, full ICE, bundling and so on.
>>>
>>> But for gateways, it makes sense to let go of a lot of these
>>> requirements. So a gateway is not a device. But what is the name for
>>> the
>>> thing that a gateway is, a device is, and a browser is, that names "the
>>> set of things that can successfully negotiate connectivity"? And what
>>> are the requirements on *that*?
>>>
>>> I see a couple of options:
>>>
>>> - Redefine "Device" to mean "that which can successfully communicate
>>> with a browser". It's free to negotiate away features if it can, but
>>> needs to interoperate with a browser that implements only the absolute
>>> minimum of what a browser has to implement.
>>>
>>> - Define a new class of "thing" which is "the thing that can talk to
>>> browsers" - of which a gateway has to be an exemplar.
>>>
>>> This is a kind of word game in some ways, but when we make definitions,
>>> we also define what's easy to talk about. And that can influence what
>>> we
>>> end up building.
>>>
>>> What do people think?
>>>
>>>      Harald
>>>
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From nobody Tue Aug 19 07:06:36 2014
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To: Ted Hardie <ted.ietf@gmail.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, "Sean Turner" <turners@ieca.com>, Cullen Jennings <fluffy@cisco.com>
Thread-Topic: [rtcweb] WG Last Call for draft-ietf-rtcweb-stun-consent-freshness
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Date: Tue, 19 Aug 2014 10:26:02 -0400
From: Jim Barnett <1jhbarnett@gmail.com>
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Archived-At: http://mailarchive.ietf.org/arch/msg/rtcweb/lVnyBdJcn_qB3HaVozaTirsNSLg
Subject: Re: [rtcweb] WebRTC Device: Lowering the bar, or inventing a new term?
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Does the VC device support the full protocol requirements that are 
placed on the Browser?  If so, we can still get away with defining 
minimal protocol set, full protocol set, and then say Browser = full 
protocol set + API.

On 8/19/2014 10:00 AM, Harald Alvestrand wrote:
> On 08/19/2014 03:38 PM, Jim Barnett wrote:
>> I agree.  We need to define a) WebRTC browsers, which support the 
>> full protocol requirements and the API, b) WebRTC Endpoints/Devices, 
>> which support some minimum protocol set sufficient for establishing 
>> connectivity.  Is there also a need to define c) Full WebRTC Devices, 
>> which support more than the minimum protocol set?  What would be the 
>> use of definition c?
>
> An endpoint (I kind of start to like this concept/name) would be 
> conformant as long as it was able to successfully negotiate away use 
> of anything that makes life difficult.
>
> It would be required to support DTLS-SRTP key handshakes and respond 
> to ICE connectivity checks, plus whatever is MUST use in -rtp-usage 
> (scanning, I'm not coming up with much - it's mostly MUST respond to, 
> MUST implement (but not required to use) stuff). The RTP circuit 
> breaker is also a MUST implement, and since it's a single-ended thing, 
> it makes no sense if it isn't also MUST use.
>
> Scanning, I'm not even sure a WebRTC endpoint is required to send RTCP 
> - but I think it should be.
> (Note: The -rtp-usage draft uses "WebRTC endpoint" to mean a 
> fully-featured endpoint... obviously we need to harmonize at some point.)
>
>> We know that we have to define the maximal set a) and the minimum 
>> usable set b).   If we can't think of a use for c) we can omit it and 
>> add extra profiles later if they turn out to be useful.
>
> An example of c) would be a VC device based on a browser that can be 
> plugged into any network (thus needing full ICE), but runs a captive 
> application that cannot be made to run any other application.
>
> It would be nice to mandate support in such a device for full ICE and 
> the MTI codecs (something I think we will have trouble with mandating 
> for gateways).
>>
>> On 8/19/2014 6:19 AM, Hutton, Andrew wrote:
>>> My thinking is that there are only two entities we need to be 
>>> concerned with.
>>>
>>> 1. WebRTC Browsers that implement both the IETF protocol 
>>> requirements and the W3C API in full.
>>>
>>> 2. Things that can interoperate directly with WebRTC browsers but 
>>> are not browsers. These have to implement at least a minimum set of 
>>> the IETF protocol and it is irrelevant as to whether they have 
>>> anything link a PeerConnection API internally. I was happy with 
>>> calling these WebRTC Devices and think it is useful to define the 
>>> minimum set of things that they need to support.
>>>
>>> I am not convinced that we need to define any other entities and my 
>>> thinking is that gateways are just a type of WebRTC Device and we 
>>> don't need to call them out explicitly.
>>>
>>> Regards
>>> Andy
>>>
>>>
>>>
>>>
>>>> -----Original Message-----
>>>> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Harald
>>>> Alvestrand
>>>> Sent: 18 August 2014 12:11
>>>> To: rtcweb@ietf.org
>>>> Subject: [rtcweb] WebRTC Device: Lowering the bar, or inventing a new
>>>> term?
>>>>
>>>> When writing the gateway document, I found myself grappling with the
>>>> "device" description.
>>>>
>>>> It is somewhat uncomfortable, in that as first defined, it requires
>>>> full
>>>> WebRTC functionality except for the API part - this means that it
>>>> should
>>>> support data channels, full ICE, bundling and so on.
>>>>
>>>> But for gateways, it makes sense to let go of a lot of these
>>>> requirements. So a gateway is not a device. But what is the name for
>>>> the
>>>> thing that a gateway is, a device is, and a browser is, that names 
>>>> "the
>>>> set of things that can successfully negotiate connectivity"? And what
>>>> are the requirements on *that*?
>>>>
>>>> I see a couple of options:
>>>>
>>>> - Redefine "Device" to mean "that which can successfully communicate
>>>> with a browser". It's free to negotiate away features if it can, but
>>>> needs to interoperate with a browser that implements only the absolute
>>>> minimum of what a browser has to implement.
>>>>
>>>> - Define a new class of "thing" which is "the thing that can talk to
>>>> browsers" - of which a gateway has to be an exemplar.
>>>>
>>>> This is a kind of word game in some ways, but when we make 
>>>> definitions,
>>>> we also define what's easy to talk about. And that can influence what
>>>> we
>>>> end up building.
>>>>
>>>> What do people think?
>>>>
>>>>      Harald
>>>>
>>>> _______________________________________________
>>>> rtcweb mailing list
>>>> rtcweb@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From nobody Tue Aug 19 07:27:58 2014
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Subject: Re: [rtcweb] WebRTC Device: Lowering the bar, or inventing a new term?
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On 8/19/14 10:00 AM, Harald Alvestrand wrote:
> On 08/19/2014 03:38 PM, Jim Barnett wrote:
>> I agree.  We need to define a) WebRTC browsers, which support the full
>> protocol requirements and the API, b) WebRTC Endpoints/Devices, which
>> support some minimum protocol set sufficient for establishing
>> connectivity.  Is there also a need to define c) Full WebRTC Devices,
>> which support more than the minimum protocol set?  What would be the
>> use of definition c?
>
> An endpoint (I kind of start to like this concept/name) would be
> conformant as long as it was able to successfully negotiate away use of
> anything that makes life difficult.
>
> It would be required to support DTLS-SRTP key handshakes and respond to
> ICE connectivity checks, plus whatever is MUST use in -rtp-usage
> (scanning, I'm not coming up with much - it's mostly MUST respond to,
> MUST implement (but not required to use) stuff). The RTP circuit breaker
> is also a MUST implement, and since it's a single-ended thing, it makes
> no sense if it isn't also MUST use.

ISTM the requirement is that a foo (device, endpoint, whatever) MUST 
support at least one of:
- draft-ietf-rtcweb-data-channel
- draft-ietf-rtcweb-rtp-usage

There are quite plausible use cases for endpoints that support only data 
channels, or only rtp. But a device that doesn't support either can't 
reasonably be considered to support webrtc in any way.

	Thanks,
	Paul


> Scanning, I'm not even sure a WebRTC endpoint is required to send RTCP -
> but I think it should be.
> (Note: The -rtp-usage draft uses "WebRTC endpoint" to mean a
> fully-featured endpoint... obviously we need to harmonize at some point.)
>
>> We know that we have to define the maximal set a) and the minimum
>> usable set b).   If we can't think of a use for c) we can omit it and
>> add extra profiles later if they turn out to be useful.
>
> An example of c) would be a VC device based on a browser that can be
> plugged into any network (thus needing full ICE), but runs a captive
> application that cannot be made to run any other application.
>
> It would be nice to mandate support in such a device for full ICE and
> the MTI codecs (something I think we will have trouble with mandating
> for gateways).
>>
>> On 8/19/2014 6:19 AM, Hutton, Andrew wrote:
>>> My thinking is that there are only two entities we need to be
>>> concerned with.
>>>
>>> 1. WebRTC Browsers that implement both the IETF protocol requirements
>>> and the W3C API in full.
>>>
>>> 2. Things that can interoperate directly with WebRTC browsers but are
>>> not browsers. These have to implement at least a minimum set of the
>>> IETF protocol and it is irrelevant as to whether they have anything
>>> link a PeerConnection API internally. I was happy with calling these
>>> WebRTC Devices and think it is useful to define the minimum set of
>>> things that they need to support.
>>>
>>> I am not convinced that we need to define any other entities and my
>>> thinking is that gateways are just a type of WebRTC Device and we
>>> don't need to call them out explicitly.
>>>
>>> Regards
>>> Andy
>>>
>>>
>>>
>>>
>>>> -----Original Message-----
>>>> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Harald
>>>> Alvestrand
>>>> Sent: 18 August 2014 12:11
>>>> To: rtcweb@ietf.org
>>>> Subject: [rtcweb] WebRTC Device: Lowering the bar, or inventing a new
>>>> term?
>>>>
>>>> When writing the gateway document, I found myself grappling with the
>>>> "device" description.
>>>>
>>>> It is somewhat uncomfortable, in that as first defined, it requires
>>>> full
>>>> WebRTC functionality except for the API part - this means that it
>>>> should
>>>> support data channels, full ICE, bundling and so on.
>>>>
>>>> But for gateways, it makes sense to let go of a lot of these
>>>> requirements. So a gateway is not a device. But what is the name for
>>>> the
>>>> thing that a gateway is, a device is, and a browser is, that names "the
>>>> set of things that can successfully negotiate connectivity"? And what
>>>> are the requirements on *that*?
>>>>
>>>> I see a couple of options:
>>>>
>>>> - Redefine "Device" to mean "that which can successfully communicate
>>>> with a browser". It's free to negotiate away features if it can, but
>>>> needs to interoperate with a browser that implements only the absolute
>>>> minimum of what a browser has to implement.
>>>>
>>>> - Define a new class of "thing" which is "the thing that can talk to
>>>> browsers" - of which a gateway has to be an exemplar.
>>>>
>>>> This is a kind of word game in some ways, but when we make definitions,
>>>> we also define what's easy to talk about. And that can influence what
>>>> we
>>>> end up building.
>>>>
>>>> What do people think?
>>>>
>>>>      Harald
>>>>
>>>> _______________________________________________
>>>> rtcweb mailing list
>>>> rtcweb@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


From nobody Tue Aug 19 08:26:33 2014
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From: "Hutton, Andrew" <andrew.hutton@unify.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: rtcweb-transports, Suggested text for HTTP Connect usage.
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Subject: [rtcweb] rtcweb-transports, Suggested text for HTTP Connect usage.
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I promised in the Toronto meeting to provide suggested text for draft-ietf-=
rtcweb-transports once http://tools.ietf.org/html/draft-ietf-httpbis-tunnel=
-protocol-00 was adopted so here goes.

My suggestion is to replace the following text in http://tools.ietf.org/htm=
l/draft-ietf-rtcweb-transports-06#section-3.4

   Further discussion of the interaction of WebRTC with firewalls is
   contained in [I-D.hutton-rtcweb-nat-firewall-considerations].  This
   document makes no requirements on interacting with HTTP proxies or
   HTTP proxy configuration methods.

   The WebRTC implementation MAY support accessing the Internet through
   an HTTP proxy.  If it does so, it MUST support the "connect" header
   as specified in [I-D.hutton-httpbis-connect-protocol].


With the following

"In order to deal with the scenario in which the media must traverse a HTTP=
 Proxy then the HTTP Connect request (Section 4.3.6 of [RFC7231]<http://too=
ls.ietf.org/html/rfc7231>) MUST be supported.  The HTTP Proxy may require a=
uthentication and therefore proxy authentication as described in (Section 4=
.3.6 of [RFC7231]<http://tools.ietf.org/html/rfc7231>) and [RFC7235] MUST a=
lso be supported. In addition the HTTP Connect MUST include an indication o=
f the protocol being used with the HTTP Connect initiated tunnel as describ=
ed in [I.D.draft-ietf-httpbis-tunnel-protocol]".


Note:  I don't intend to maintain or progress [I-D.hutton-rtcweb-nat-firewa=
ll-considerations].


Regards
Andy





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<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
<meta name=3D"Generator" content=3D"Microsoft Exchange Server">
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</head>
<body>
<font face=3D"Calibri" size=3D"2"><span style=3D"font-size:11pt;">
<div>I promised in the Toronto meeting to provide suggested text for draft-=
ietf&#8211;rtcweb-transports once <a href=3D"http://tools.ietf.org/html/dra=
ft-ietf-httpbis-tunnel-protocol-00"><font color=3D"blue"><u>http://tools.ie=
tf.org/html/draft-ietf-httpbis-tunnel-protocol-00</u></font></a>
was adopted so here goes.</div>
<div>&nbsp;</div>
<div>My suggestion is to replace the following text in <a href=3D"http://to=
ols.ietf.org/html/draft-ietf-rtcweb-transports-06#section-3.4"><font color=
=3D"blue"><u>http://tools.ietf.org/html/draft-ietf-rtcweb-transports-06#sec=
tion-3.4</u></font></a> </div>
<div>&nbsp;</div>
<div>&nbsp;&nbsp; Further discussion of the interaction of WebRTC with fire=
walls is<br>

&nbsp;&nbsp; contained in [I-D.hutton-rtcweb-nat-firewall-considerations].&=
nbsp; This<br>

&nbsp;&nbsp; document makes no requirements on interacting with HTTP proxie=
s or<br>

&nbsp;&nbsp; HTTP proxy configuration methods.<br>

<br>

&nbsp;&nbsp; The WebRTC implementation MAY support accessing the Internet t=
hrough<br>

&nbsp;&nbsp; an HTTP proxy.&nbsp; If it does so, it MUST support the &quot;=
connect&quot; header<br>

&nbsp;&nbsp; as specified in [I-D.hutton-httpbis-connect-protocol].</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div>With the following</div>
<div>&nbsp;</div>
<div><i>&#8220;</i><i>In order to deal with the scenario in which the media=
 must traverse a HTTP Proxy then the HTTP Connect request (</i><a href=3D"h=
ttp://tools.ietf.org/html/rfc7231"><i>Section&nbsp;4.3.6 of [RFC7231]</i></=
a><i>) MUST be supported.&nbsp; The HTTP Proxy may
require authentication and therefore proxy authentication as described in (=
</i><a href=3D"http://tools.ietf.org/html/rfc7231"><i>Section&nbsp;4.3.6 of=
 [RFC7231]</i></a><i>) and [RFC7235] MUST also be supported. In addition th=
e HTTP Connect MUST include an indication
of the protocol being used with the HTTP Connect initiated tunnel as descri=
bed in [I.D.draft-ietf-httpbis-tunnel-protocol]</i><i>&#8221;</i><i>. </i><=
/div>
<div><font face=3D"Courier New" size=3D"2"><span style=3D"font-size:10pt;">=
&nbsp;</span></font></div>
<div><font face=3D"Courier New" size=3D"2"><span style=3D"font-size:10pt;">=
&nbsp;</span></font></div>
<div>Note:  I don&#8217;t intend to maintain or progress [I-D.hutton-rtcweb=
-nat-firewall-considerations].</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div>Regards</div>
<div>Andy<br>

</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
</span></font>
</body>
</html>

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From: Benjamin Schwartz <bemasc@webrtc.org>
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--089e0115f048b19b650500fece69
Content-Type: text/plain; charset=UTF-8

Hi rtcweb,

We've talked occasionally about various TURN proxy configuration mechanisms
and WebRTC TURN use cases, but it seems to me that we've never really
explained how they're all supposed to fit together.  I've written a draft
to try to fill in the details.

---------- Forwarded message ----------
From: <internet-drafts@ietf.org>
Date: Tue, Aug 19, 2014 at 10:39 AM
Subject: New Version Notification for draft-schwartz-rtcweb-return-00.txt
To: "Benjamin M. Schwartz" <bemasc@webrtc.org>



A new version of I-D, draft-schwartz-rtcweb-return-00.txt
has been successfully submitted by Benjamin M. Schwartz and posted to the
IETF repository.

Name:           draft-schwartz-rtcweb-return
Revision:       00
Title:          Recursively Encapsulated TURN (RETURN) for Connectivity and
Privacy in WebRTC
Document date:  2014-08-19
Group:          Individual Submission
Pages:          10
URL:
http://www.ietf.org/internet-drafts/draft-schwartz-rtcweb-return-00.txt
Status:
https://datatracker.ietf.org/doc/draft-schwartz-rtcweb-return/
Htmlized:       http://tools.ietf.org/html/draft-schwartz-rtcweb-return-00


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




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

--089e0115f048b19b650500fece69
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Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><span style=3D"font-family:arial,sans-serif;font-size:13px=
">Hi rtcweb,</span><div style=3D"font-family:arial,sans-serif;font-size:13p=
x"><br></div><div style=3D"font-family:arial,sans-serif;font-size:13px">We&=
#39;ve talked occasionally about various TURN proxy configuration mechanism=
s and WebRTC TURN use cases, but it seems to me that we&#39;ve never really=
 explained how they&#39;re all supposed to fit together. =C2=A0I&#39;ve wri=
tten a draft to try to fill in the details.</div>
<br><div class=3D"gmail_quote">---------- Forwarded message ----------<br>F=
rom: <b class=3D"gmail_sendername"></b> <span dir=3D"ltr">&lt;<a href=3D"ma=
ilto:internet-drafts@ietf.org">internet-drafts@ietf.org</a>&gt;</span><br>D=
ate: Tue, Aug 19, 2014 at 10:39 AM<br>
Subject: New Version Notification for draft-schwartz-rtcweb-return-00.txt<b=
r>To: &quot;Benjamin M. Schwartz&quot; &lt;<a href=3D"mailto:bemasc@webrtc.=
org">bemasc@webrtc.org</a>&gt;<br><br><br><br>
A new version of I-D, draft-schwartz-rtcweb-return-00.txt<br>
has been successfully submitted by Benjamin M. Schwartz and posted to the<b=
r>
IETF repository.<br>
<br>
Name:=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0draft-schwartz-rtcweb-return<=
br>
Revision:=C2=A0 =C2=A0 =C2=A0 =C2=A000<br>
Title:=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 Recursively Encapsulated TURN (RET=
URN) for Connectivity and Privacy in WebRTC<br>
Document date:=C2=A0 2014-08-19<br>
Group:=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 Individual Submission<br>
Pages:=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 10<br>
URL:=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 <a href=3D"http://www.ietf.or=
g/internet-drafts/draft-schwartz-rtcweb-return-00.txt" target=3D"_blank">ht=
tp://www.ietf.org/internet-drafts/draft-schwartz-rtcweb-return-00.txt</a><b=
r>
Status:=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0<a href=3D"https://datatracker.iet=
f.org/doc/draft-schwartz-rtcweb-return/" target=3D"_blank">https://datatrac=
ker.ietf.org/doc/draft-schwartz-rtcweb-return/</a><br>
Htmlized:=C2=A0 =C2=A0 =C2=A0 =C2=A0<a href=3D"http://tools.ietf.org/html/d=
raft-schwartz-rtcweb-return-00" target=3D"_blank">http://tools.ietf.org/htm=
l/draft-schwartz-rtcweb-return-00</a><br>
<br>
<br>
Abstract:<br>
=C2=A0 =C2=A0In the context of WebRTC, the concept of a local TURN proxy ha=
s been<br>
=C2=A0 =C2=A0suggested, but not reviewed in detail.=C2=A0 WebRTC applicatio=
ns are<br>
=C2=A0 =C2=A0already using TURN to enhance connectivity and privacy.=C2=A0 =
This<br>
=C2=A0 =C2=A0document explains how local TURN proxies and WebRTC applicatio=
ns can<br>
=C2=A0 =C2=A0work together.<br>
<br>
<br>
<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" target=3D"_blank">tools.ietf.org</a>.<br>
<br>
The IETF Secretariat<br>
<br>
</div><br></div>

--089e0115f048b19b650500fece69--


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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: "Hutton, Andrew" <andrew.hutton@unify.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: rtcweb-transports, Suggested text for HTTP Connect usage.
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"In order to deal with the scenario in which the media must traverse a HTTP=
 Proxy then the HTTP Connect request (Section 4.3.6 of [RFC7231]<http://too=
ls.ietf.org/html/rfc7231>) MUST be supported.  The HTTP Proxy may require a=
uthentication and therefore proxy authentication as described in (Section 4=
.3.6 of [RFC7231]<http://tools.ietf.org/html/rfc7231>) and [RFC7235] MUST a=
lso be supported. In addition the HTTP Connect MUST include an indication o=
f the protocol being used with the HTTP Connect initiated tunnel as describ=
ed in [I.D.draft-ietf-httpbis-tunnel-protocol]".

<Raju>
> ... In addition the HTTP Connect MUST include an indication of the protoc=
ol being...

Is the tunnel indication in CONNECT a "MUST"? or Can it be changed to "SHOU=
LD"?
The reason is, though typically unrecognized HTTP headers are ignored, what=
 if some proxies are sensitive to HTTP CONNECT's  unrecognized headers? In =
such a case excluding this header may just work fine.

</Raju>


BR
Raju

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<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">&nbsp;<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><i><span style=3D"font-size:11.0pt;font-family:&quot=
;Calibri&quot;,&quot;sans-serif&quot;">&#8220;In order to deal with the sce=
nario in which the media must traverse a HTTP Proxy then the HTTP Connect r=
equest (</span></i><span style=3D"font-size:11.0pt;font-family:&quot;Calibr=
i&quot;,&quot;sans-serif&quot;"><a href=3D"http://tools.ietf.org/html/rfc72=
31"><i>Section&nbsp;4.3.6
 of [RFC7231]</i></a><i>) MUST be supported.&nbsp; The HTTP Proxy may requi=
re authentication and therefore proxy authentication as described in (</i><=
a href=3D"http://tools.ietf.org/html/rfc7231"><i>Section&nbsp;4.3.6 of [RFC=
7231]</i></a><i>) and [RFC7235] MUST also be
 supported. In addition the HTTP Connect MUST include an indication of the =
protocol being used with the HTTP Connect initiated tunnel as described in =
[I.D.draft-ietf-httpbis-tunnel-protocol]&#8221;.
</i><o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;,&quot;serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span></=
p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">&lt;Raju&gt; <o:p></o:=
p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">&gt;</span><i><span st=
yle=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&qu=
ot;"> &#8230; In addition the HTTP Connect MUST include an indication of th=
e protocol being&#8230;<o:p></o:p></span></i></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Is the tunnel indicati=
on in CONNECT a &#8220;MUST&#8221;? or Can it be changed to &#8220;SHOULD&#=
8221;?<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">The reason is, though =
typically unrecognized HTTP headers are ignored, what if some proxies are s=
ensitive to HTTP CONNECT&#8217;s&nbsp; unrecognized headers? In such a case=
 excluding this header may just work fine.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">&lt;/Raju&gt;<o:p></o:=
p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">&nbsp;<span style=3D"color:#1F497D"><o:=
p></o:p></span></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">&nbsp;<o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">BR<o:p></o:p></span></=
p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Raju<o:p></o:p></span>=
</p>
</div>
</div>
</body>
</html>

--_000_E1FE4C082A89A246A11D7F32A95A17828E51BF51US70UWXCHMBA02z_--


From nobody Tue Aug 19 11:40:28 2014
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From: Martin Thomson <martin.thomson@gmail.com>
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Subject: Re: [rtcweb] rtcweb-transports, Suggested text for HTTP Connect usage.
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On 19 August 2014 08:26, Hutton, Andrew <andrew.hutton@unify.com> wrote:
> =E2=80=9CIn order to deal with the scenario in which the media must trave=
rse a HTTP
> Proxy then the HTTP Connect request (Section 4.3.6 of [RFC7231]) MUST be
> supported.  The HTTP Proxy may require authentication and therefore proxy
> authentication as described in (Section 4.3.6 of [RFC7231]) and [RFC7235]
> MUST also be supported. In addition the HTTP Connect MUST include an
> indication of the protocol being used with the HTTP Connect initiated tun=
nel
> as described in [I.D.draft-ietf-httpbis-tunnel-protocol]=E2=80=9D.


s/HTTP Connect/HTTP CONNECT/g

Use of lowercase "must" and "may require" are playing 2119 bingo,
maybe consider alternative words.

Making this mandatory on all endpoints is mainly just unnecessary.  As
we've seen, browsers are going to do this anyway because the economics
makes sense.  I don't see any reason to extend that mandate.  It's a
lot of work for a non-browser, for a relatively small gain.  I could
settle for SHOULD.

I think that putting a MUST on proxy authentication is a little of a
stretch for the same reasons that the traversal using CONNECT needs to
be MAY.


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Subject: Re: [rtcweb] WebRTC Device: Lowering the bar, or inventing a new term?
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On 19/08/2014 0:09, Silvia Pfeiffer wrote:
> On Tue, Aug 19, 2014 at 3:34 AM, Parthasarathi R
> <partha@parthasarathi.co.in> wrote:
>> Hi Harald & all,
>>
>> I'm seeing that we are discussing about four WebRTC entities in three term
>> which is confusing. I propose the following four entities to discuss:
>>
>> 1) WebRTC browser (full compliant Protocol + API) - as in
>> draft-ietf-rtcweb-overview-10
> To me, this is a WebRTC endpoint.
>
>
>> 2) WebRTC device (Full complaint protocol) - as in
>> draft-ietf-rtcweb-overview-10
> To me, this is a RTCWeb endpoint (it doesn't care about the browser
> API, but only the stuff that is defined in the rtcweb group of IEEE).
>
> Silvia.
>

Completely agree.

Also with this definition, an ORTC browser would still be an WebRTC 
endpoint.

Best regards
Sergio


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On 19/08/2014 16:27, Paul Kyzivat wrote:
> On 8/19/14 10:00 AM, Harald Alvestrand wrote:
>> On 08/19/2014 03:38 PM, Jim Barnett wrote:
>>> I agree.  We need to define a) WebRTC browsers, which support the full
>>> protocol requirements and the API, b) WebRTC Endpoints/Devices, which
>>> support some minimum protocol set sufficient for establishing
>>> connectivity.  Is there also a need to define c) Full WebRTC Devices,
>>> which support more than the minimum protocol set?  What would be the
>>> use of definition c?
>>
>> An endpoint (I kind of start to like this concept/name) would be
>> conformant as long as it was able to successfully negotiate away use of
>> anything that makes life difficult.
>>
>> It would be required to support DTLS-SRTP key handshakes and respond to
>> ICE connectivity checks, plus whatever is MUST use in -rtp-usage
>> (scanning, I'm not coming up with much - it's mostly MUST respond to,
>> MUST implement (but not required to use) stuff). The RTP circuit breaker
>> is also a MUST implement, and since it's a single-ended thing, it makes
>> no sense if it isn't also MUST use.
>
> ISTM the requirement is that a foo (device, endpoint, whatever) MUST 
> support at least one of:
> - draft-ietf-rtcweb-data-channel
> - draft-ietf-rtcweb-rtp-usage
>
> There are quite plausible use cases for endpoints that support only 
> data channels, or only rtp. But a device that doesn't support either 
> can't reasonably be considered to support webrtc in any way.
>
Right, but there are some other entities (gateways for example) that 
without fully implementing those requirements, are still able to talk to 
an WebRTC device/endpoint/whatever.
May I suggest that we follow the ICE naming and we call that entities, 
WebRTC lite endpoints? Just a thought.. ;)

Best regards
Sergio


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Archived-At: http://mailarchive.ietf.org/arch/msg/rtcweb/EUDNGCQYc5wFLeYA2A6eH2J2Nek
Subject: Re: [rtcweb] WebRTC Device: Lowering the bar, or inventing a new term?
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Hi Sergio/Silvia,

WebRTC browser and WebRTC device are also WebRTC endpoints as they conform
to all WebRTC end points requirements. 

Thanks
Partha

> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Sergio
> Garcia Murillo
> Sent: Wednesday, August 20, 2014 4:59 AM
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] WebRTC Device: Lowering the bar, or inventing a
> new term?
> 
> On 19/08/2014 0:09, Silvia Pfeiffer wrote:
> > On Tue, Aug 19, 2014 at 3:34 AM, Parthasarathi R
> > <partha@parthasarathi.co.in> wrote:
> >> Hi Harald & all,
> >>
> >> I'm seeing that we are discussing about four WebRTC entities in
> three term
> >> which is confusing. I propose the following four entities to
> discuss:
> >>
> >> 1) WebRTC browser (full compliant Protocol + API) - as in
> >> draft-ietf-rtcweb-overview-10
> > To me, this is a WebRTC endpoint.
> >
> >
> >> 2) WebRTC device (Full complaint protocol) - as in
> >> draft-ietf-rtcweb-overview-10
> > To me, this is a RTCWeb endpoint (it doesn't care about the browser
> > API, but only the stuff that is defined in the rtcweb group of IEEE).
> >
> > Silvia.
> >
> 
> Completely agree.
> 
> Also with this definition, an ORTC browser would still be an WebRTC
> endpoint.
> 
> Best regards
> Sergio
> 
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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Date: Wed, 20 Aug 2014 09:39:49 +0530
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From: Muthu Arul Mozhi Perumal <muthu.arul@gmail.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: Cullen Jennings <fluffy@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] WG Last Call for draft-ietf-rtcweb-stun-consent-freshness
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--001a11c240ea74b7f5050107c67b
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I believe draft-ietf-rtcweb-stun-consent-freshness deals with the WebRTC
browser. A WebRTC device (eg. a mobile phone) on the other hand, could run
a WebRTC browser and a JS and perform consent. Similarly, a WebRTC endpoint
(whatever that gets defined as) might also perform consent, while a WebRTC
gateway might not, and how consent applies to those should get specified in
the document(s) that define those requirements (note: the gateway referred
to in draft-ietf-rtcweb-stun-consent-freshness is any SIP gateway).

So, draft-ietf-rtcweb-stun-consent-freshness should just replace WebRTC
endpoint and WebRTC implementation with WebRTC browser, IMHO.

Muthu

On Tue, Aug 19, 2014 at 7:36 PM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

>  Hi,
>
>
>
> The draft says:
>
>
>
> =E2=80=9CWebRTC endpoints are required to support full ICE as specified i=
n
>
>                 section 3.4 of [I-D.
> =E2=80=8B=E2=80=8B
> ietf-rtcweb-transports].  However, when WebRTC
>
>                 endpoints interwork with other endpoints that support onl=
y
> ICE-lite
>
>                 (e.g., gateways) those endpoints will not generate consen=
t
> checks,
>
>                 but just respond to consent checks they receive.=E2=80=9D
>
>
>
> The draft also talks about =E2=80=9CWebRTC implementations=E2=80=9D and =
=E2=80=9CWebRTC browsers=E2=80=9D.
>
>
>
> We need to make sure that the terminology is aligned with
> draft-ietf-rtcweb-overview (for example,
> =E2=80=8B=E2=80=8B
> draft-ietf-rtcweb-overview does not talk about =E2=80=9CWebRTC endpoints=
=E2=80=9D), which
> also may be impacted based on the outcome of the current
> terminology/gateway discussion.
>
>
>
> Regards,
>
>
>
> Christer
>
>
>
>
>
> *From:* rtcweb [mailto:rtcweb-bounces@ietf.org] *On Behalf Of *Ted Hardie
> *Sent:* 14. elokuuta 2014 18:10
> *To:* rtcweb@ietf.org; Sean Turner; Cullen Jennings
> *Subject:* [rtcweb] WG Last Call for
> draft-ietf-rtcweb-stun-consent-freshness
>
>
>
> This starts a WG Last Call for draft-ietf-rtcweb-stun-consent-freshness
> (available at
> http://datatracker.ietf.org/doc/draft-ietf-rtcweb-stun-consent-freshness/=
).
> Please review the document and send comments to the list by September 10,
> 2014.
>
> thanks,
>
> Ted Hardie
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--001a11c240ea74b7f5050107c67b
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<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:arial,he=
lvetica,sans-serif;font-size:small">I believe=C2=A0draft-ietf-rtcweb-stun-c=
onsent-freshness deals with the WebRTC browser. A WebRTC device (eg. a mobi=
le phone) on the other hand, could run a WebRTC browser and a JS and perfor=
m consent. Similarly, a WebRTC endpoint (whatever that gets defined as) mig=
ht also perform consent, while a WebRTC gateway might not, and how consent =
applies to those should get specified in the document(s) that define those =
requirements (note: the gateway referred to in draft-ietf-rtcweb-stun-conse=
nt-freshness is any SIP gateway).</div>
<div class=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-seri=
f;font-size:small"><br></div><div class=3D"gmail_default" style=3D"font-fam=
ily:arial,helvetica,sans-serif;font-size:small">So, draft-ietf-rtcweb-stun-=
consent-freshness should just replace WebRTC endpoint and WebRTC implementa=
tion with WebRTC browser, IMHO.</div>
<div class=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-seri=
f;font-size:small"><br></div><div class=3D"gmail_default" style=3D"font-fam=
ily:arial,helvetica,sans-serif;font-size:small">Muthu</div><div class=3D"gm=
ail_extra">
<br><div class=3D"gmail_quote">On Tue, Aug 19, 2014 at 7:36 PM, 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:0px 0px 0px 0.8ex;bord=
er-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:soli=
d;padding-left:1ex">






<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif;color:rgb(31,73,125)">Hi,<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif;color:rgb(31,73,125)"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif;color:rgb(31,73,125)">The draft says:<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif;color:rgb(31,73,125)"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal" style=3D"text-indent:36pt"><span style=3D"font-size:=
11pt;font-family:Calibri,sans-serif;color:rgb(31,73,125)">=E2=80=9CWebRTC e=
ndpoints are required to support full ICE as specified in<u></u><u></u></sp=
an></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif;color:rgb(31,73,125)">=C2=A0=C2=A0 =C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 section 3.4 of [I-D.</span></p><=
div class=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-serif=
;font-size:small;display:inline">
=E2=80=8B=E2=80=8B</div>ietf-rtcweb-transports].=C2=A0 However, when WebRTC=
<u></u><u></u><p></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif;color:rgb(31,73,125)">=C2=A0=C2=A0 =C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 endpoints interwork with other e=
ndpoints that support only ICE-lite<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif;color:rgb(31,73,125)">=C2=A0=C2=A0 =C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 (e.g., gateways) those endpoints=
 will not generate consent checks,<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif;color:rgb(31,73,125)">=C2=A0=C2=A0 =C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 but just respond to consent chec=
ks they receive.=E2=80=9D<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif;color:rgb(31,73,125)"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif;color:rgb(31,73,125)">The draft also talks about =E2=80=9CWebRTC i=
mplementations=E2=80=9D and =E2=80=9CWebRTC browsers=E2=80=9D.<u></u><u></u=
></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif;color:rgb(31,73,125)"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif;color:rgb(31,73,125)">We need to make sure that the terminology is=
 aligned with draft-ietf-rtcweb-overview (for example, </span></p><div clas=
s=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-serif;font-si=
ze:small;display:inline">
=E2=80=8B=E2=80=8B</div>draft-ietf-rtcweb-overview does not talk about =E2=
=80=9CWebRTC endpoints=E2=80=9D),
 which also may be impacted based on the outcome of the current terminology=
/gateway discussion.<u></u><u></u><p></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif;color:rgb(31,73,125)"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif;color:rgb(31,73,125)">Regards,<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif;color:rgb(31,73,125)"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif;color:rgb(31,73,125)">Christer<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif;color:rgb(31,73,125)"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif;color:rgb(31,73,125)"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><b><span style=3D"font-size:10pt;font-family:Tahoma,=
sans-serif">From:</span></b><span style=3D"font-size:10pt;font-family:Tahom=
a,sans-serif"> rtcweb [mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org" ta=
rget=3D"_blank">rtcweb-bounces@ietf.org</a>]
<b>On Behalf Of </b>Ted Hardie<br>
<b>Sent:</b> 14. elokuuta 2014 18:10<br>
<b>To:</b> <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf=
.org</a>; Sean Turner; Cullen Jennings<br>
<b>Subject:</b> [rtcweb] WG Last Call for draft-ietf-rtcweb-stun-consent-fr=
eshness<u></u><u></u></span></p><div class=3D"">
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12pt"><span style=3D"font-fam=
ily:Garamond,serif">This starts a WG Last Call for draft-ietf-rtcweb-stun-c=
onsent-freshness (available at
<a href=3D"http://datatracker.ietf.org/doc/draft-ietf-rtcweb-stun-consent-f=
reshness/" target=3D"_blank">
http://datatracker.ietf.org/doc/draft-ietf-rtcweb-stun-consent-freshness/</=
a>).=C2=A0 Please review the document and send comments to the list by Sept=
ember 10, 2014.<u></u><u></u></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-family:Garamond,serif">thanks,<b=
r>
<br>
Ted Hardie<u></u><u></u></span></p>
</div>
</div>
</div></div>
</div>

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

--001a11c240ea74b7f5050107c67b--


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Subject: Re: [rtcweb] WG Last Call for draft-ietf-rtcweb-stun-consent-freshness
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On 08/20/2014 06:09 AM, Muthu Arul Mozhi Perumal wrote:
> I believe draft-ietf-rtcweb-stun-consent-freshness deals with the
> WebRTC browser. A WebRTC device (eg. a mobile phone) on the other
> hand, could run a WebRTC browser and a JS and perform consent.
> Similarly, a WebRTC endpoint (whatever that gets defined as) might
> also perform consent, while a WebRTC gateway might not, and how
> consent applies to those should get specified in the document(s) that
> define those requirements (note: the gateway referred to in
> draft-ietf-rtcweb-stun-consent-freshness is any SIP gateway).

Note: In terms of conformance requirements, "might" and "might not" are
completely equivalent (2119 style: MAY - there is no MAY NOT).

>
> So, draft-ietf-rtcweb-stun-consent-freshness should just replace
> WebRTC endpoint and WebRTC implementation with WebRTC browser, IMHO.
>
> Muthu
>
> On Tue, Aug 19, 2014 at 7:36 PM, Christer Holmberg
> <christer.holmberg@ericsson.com
> <mailto:christer.holmberg@ericsson.com>> wrote:
>
>     Hi,
>
>      
>
>     The draft says:
>
>      
>
>     â€œWebRTC endpoints are required to support full ICE as specified in
>
>                     section 3.4 of [I-D.
>
>     â€‹ â€‹
>     ietf-rtcweb-transports].  However, when WebRTC
>
>                     endpoints interwork with other endpoints that
>     support only ICE-lite
>
>                     (e.g., gateways) those endpoints will not generate
>     consent checks,
>
>                     but just respond to consent checks they receive.â€
>
>      
>
>     The draft also talks about â€œWebRTC implementationsâ€ and â€œWebRTC
>     browsersâ€.
>
>      
>
>     We need to make sure that the terminology is aligned with
>     draft-ietf-rtcweb-overview (for example,
>
>     â€‹ â€‹
>     draft-ietf-rtcweb-overview does not talk about â€œWebRTC
>     endpointsâ€), which also may be impacted based on the outcome of
>     the current terminology/gateway discussion.
>
>      
>
>     Regards,
>
>      
>
>     Christer
>
>      
>
>      
>
>     *From:*rtcweb [mailto:rtcweb-bounces@ietf.org
>     <mailto:rtcweb-bounces@ietf.org>] *On Behalf Of *Ted Hardie
>     *Sent:* 14. elokuuta 2014 18:10
>     *To:* rtcweb@ietf.org <mailto:rtcweb@ietf.org>; Sean Turner;
>     Cullen Jennings
>     *Subject:* [rtcweb] WG Last Call for
>     draft-ietf-rtcweb-stun-consent-freshness
>
>      
>
>     This starts a WG Last Call for
>     draft-ietf-rtcweb-stun-consent-freshness (available at
>     http://datatracker.ietf.org/doc/draft-ietf-rtcweb-stun-consent-freshness/). 
>     Please review the document and send comments to the list by
>     September 10, 2014.
>
>     thanks,
>
>     Ted Hardie
>
>
>     _______________________________________________
>     rtcweb mailing list
>     rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>     https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


-- 
Surveillance is pervasive. Go Dark.


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    <div class="moz-cite-prefix">On 08/20/2014 06:09 AM, Muthu Arul
      Mozhi Perumal wrote:<br>
    </div>
    <blockquote
cite="mid:CAKz0y8zycsyr9m4BA=-8xOaWkU+Sog5Mbz7K-oN3woqi++mVzg@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div class="gmail_default"
          style="font-family:arial,helvetica,sans-serif;font-size:small">I
          believeÂ draft-ietf-rtcweb-stun-consent-freshness deals with
          the WebRTC browser. A WebRTC device (eg. a mobile phone) on
          the other hand, could run a WebRTC browser and a JS and
          perform consent. Similarly, a WebRTC endpoint (whatever that
          gets defined as) might also perform consent, while a WebRTC
          gateway might not, and how consent applies to those should get
          specified in the document(s) that define those requirements
          (note: the gateway referred to in
          draft-ietf-rtcweb-stun-consent-freshness is any SIP gateway).</div>
      </div>
    </blockquote>
    <br>
    Note: In terms of conformance requirements, "might" and "might not"
    are completely equivalent (2119 style: MAY - there is no MAY NOT).<br>
    <br>
    <blockquote
cite="mid:CAKz0y8zycsyr9m4BA=-8xOaWkU+Sog5Mbz7K-oN3woqi++mVzg@mail.gmail.com"
      type="cite">
      <div dir="ltr">
        <div class="gmail_default"
          style="font-family:arial,helvetica,sans-serif;font-size:small"><br>
        </div>
        <div class="gmail_default"
          style="font-family:arial,helvetica,sans-serif;font-size:small">So,
          draft-ietf-rtcweb-stun-consent-freshness should just replace
          WebRTC endpoint and WebRTC implementation with WebRTC browser,
          IMHO.</div>
        <div class="gmail_default"
          style="font-family:arial,helvetica,sans-serif;font-size:small"><br>
        </div>
        <div class="gmail_default"
          style="font-family:arial,helvetica,sans-serif;font-size:small">Muthu</div>
        <div class="gmail_extra">
          <br>
          <div class="gmail_quote">On Tue, Aug 19, 2014 at 7:36 PM,
            Christer Holmberg <span dir="ltr">&lt;<a
                moz-do-not-send="true"
                href="mailto:christer.holmberg@ericsson.com"
                target="_blank">christer.holmberg@ericsson.com</a>&gt;</span>
            wrote:<br>
            <blockquote class="gmail_quote" style="margin:0px 0px 0px
0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
              <div link="blue" vlink="purple" lang="EN-US">
                <div>
                  <p class="MsoNormal"><span
style="font-size:11pt;font-family:Calibri,sans-serif;color:rgb(31,73,125)">Hi,</span></p>
                  <p class="MsoNormal"><span
style="font-size:11pt;font-family:Calibri,sans-serif;color:rgb(31,73,125)">Â </span></p>
                  <p class="MsoNormal"><span
style="font-size:11pt;font-family:Calibri,sans-serif;color:rgb(31,73,125)">The
                      draft says:</span></p>
                  <p class="MsoNormal"><span
style="font-size:11pt;font-family:Calibri,sans-serif;color:rgb(31,73,125)">Â </span></p>
                  <p class="MsoNormal" style="text-indent:36pt"><span
style="font-size:11pt;font-family:Calibri,sans-serif;color:rgb(31,73,125)">â€œWebRTC
                      endpoints are required to support full ICE as
                      specified in</span></p>
                  <p class="MsoNormal"><span
style="font-size:11pt;font-family:Calibri,sans-serif;color:rgb(31,73,125)">Â Â 
                      Â Â Â Â Â Â Â Â Â Â Â Â  section 3.4 of [I-D.</span></p>
                  <div class="gmail_default"
style="font-family:arial,helvetica,sans-serif;font-size:small;display:inline">â€‹
                    â€‹</div>
                  ietf-rtcweb-transports].Â  However, when WebRTC
                  <p class="MsoNormal"><span
style="font-size:11pt;font-family:Calibri,sans-serif;color:rgb(31,73,125)">Â Â 
                      Â Â Â Â Â Â Â Â Â Â Â Â  endpoints interwork with other
                      endpoints that support only ICE-lite</span></p>
                  <p class="MsoNormal"><span
style="font-size:11pt;font-family:Calibri,sans-serif;color:rgb(31,73,125)">Â Â 
                      Â Â Â Â Â Â Â Â Â Â Â Â  (e.g., gateways) those endpoints will
                      not generate consent checks,</span></p>
                  <p class="MsoNormal"><span
style="font-size:11pt;font-family:Calibri,sans-serif;color:rgb(31,73,125)">Â Â 
                      Â Â Â Â Â Â Â Â Â Â Â Â  but just respond to consent checks
                      they receive.â€</span></p>
                  <p class="MsoNormal"><span
style="font-size:11pt;font-family:Calibri,sans-serif;color:rgb(31,73,125)">Â </span></p>
                  <p class="MsoNormal"><span
style="font-size:11pt;font-family:Calibri,sans-serif;color:rgb(31,73,125)">The
                      draft also talks about â€œWebRTC implementationsâ€
                      and â€œWebRTC browsersâ€.</span></p>
                  <p class="MsoNormal"><span
style="font-size:11pt;font-family:Calibri,sans-serif;color:rgb(31,73,125)">Â </span></p>
                  <p class="MsoNormal"><span
style="font-size:11pt;font-family:Calibri,sans-serif;color:rgb(31,73,125)">We
                      need to make sure that the terminology is aligned
                      with draft-ietf-rtcweb-overview (for example, </span></p>
                  <div class="gmail_default"
style="font-family:arial,helvetica,sans-serif;font-size:small;display:inline">â€‹
                    â€‹</div>
                  draft-ietf-rtcweb-overview does not talk about â€œWebRTC
                  endpointsâ€), which also may be impacted based on the
                  outcome of the current terminology/gateway discussion.
                  <p class="MsoNormal"><span
style="font-size:11pt;font-family:Calibri,sans-serif;color:rgb(31,73,125)">Â </span></p>
                  <p class="MsoNormal"><span
style="font-size:11pt;font-family:Calibri,sans-serif;color:rgb(31,73,125)">Regards,</span></p>
                  <p class="MsoNormal"><span
style="font-size:11pt;font-family:Calibri,sans-serif;color:rgb(31,73,125)">Â </span></p>
                  <p class="MsoNormal"><span
style="font-size:11pt;font-family:Calibri,sans-serif;color:rgb(31,73,125)">Christer</span></p>
                  <p class="MsoNormal"><span
style="font-size:11pt;font-family:Calibri,sans-serif;color:rgb(31,73,125)">Â </span></p>
                  <p class="MsoNormal"><span
style="font-size:11pt;font-family:Calibri,sans-serif;color:rgb(31,73,125)">Â </span></p>
                  <p class="MsoNormal"><b><span
                        style="font-size:10pt;font-family:Tahoma,sans-serif">From:</span></b><span
style="font-size:10pt;font-family:Tahoma,sans-serif"> rtcweb [mailto:<a
                        moz-do-not-send="true"
                        href="mailto:rtcweb-bounces@ietf.org"
                        target="_blank">rtcweb-bounces@ietf.org</a>]
                      <b>On Behalf Of </b>Ted Hardie<br>
                      <b>Sent:</b> 14. elokuuta 2014 18:10<br>
                      <b>To:</b> <a moz-do-not-send="true"
                        href="mailto:rtcweb@ietf.org" target="_blank">rtcweb@ietf.org</a>;
                      Sean Turner; Cullen Jennings<br>
                      <b>Subject:</b> [rtcweb] WG Last Call for
                      draft-ietf-rtcweb-stun-consent-freshness</span></p>
                  <div class="">
                    <p class="MsoNormal">Â </p>
                    <div>
                      <div>
                        <p class="MsoNormal" style="margin-bottom:12pt"><span
                            style="font-family:Garamond,serif">This
                            starts a WG Last Call for
                            draft-ietf-rtcweb-stun-consent-freshness
                            (available at
                            <a moz-do-not-send="true"
href="http://datatracker.ietf.org/doc/draft-ietf-rtcweb-stun-consent-freshness/"
                              target="_blank">
http://datatracker.ietf.org/doc/draft-ietf-rtcweb-stun-consent-freshness/</a>).Â 
                            Please review the document and send comments
                            to the list by September 10, 2014.</span></p>
                      </div>
                      <div>
                        <p class="MsoNormal"><span
                            style="font-family:Garamond,serif">thanks,<br>
                            <br>
                            Ted Hardie</span></p>
                      </div>
                    </div>
                  </div>
                </div>
              </div>
              <br>
              _______________________________________________<br>
              rtcweb mailing list<br>
              <a moz-do-not-send="true" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
              <a moz-do-not-send="true"
                href="https://www.ietf.org/mailman/listinfo/rtcweb"
                target="_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
              <br>
            </blockquote>
          </div>
          <br>
        </div>
      </div>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_______________________________________________
rtcweb mailing list
<a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
    </blockquote>
    <br>
    <br>
    <pre class="moz-signature" cols="72">-- 
Surveillance is pervasive. Go Dark.
</pre>
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Subject: Re: [rtcweb] WebRTC Device: Lowering the bar, or inventing a new term?
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On 19 Aug 2014, at 15:00, Harald Alvestrand <harald@alvestrand.no> =
wrote:
> On 08/19/2014 03:38 PM, Jim Barnett wrote:
>> I agree.  We need to define a) WebRTC browsers, which support the =
full protocol requirements and the API, b) WebRTC Endpoints/Devices, =
which support some minimum protocol set sufficient for establishing =
connectivity.  Is there also a need to define c) Full WebRTC Devices, =
which support more than the minimum protocol set?  What would be the use =
of definition c?
>=20
> An endpoint (I kind of start to like this concept/name) would be =
conformant as long as it was able to successfully negotiate away use of =
anything that makes life difficult.
>=20
> It would be required to support DTLS-SRTP key handshakes and respond =
to ICE connectivity checks, plus whatever is MUST use in -rtp-usage =
(scanning, I'm not coming up with much - it's mostly MUST respond to, =
MUST implement (but not required to use) stuff). The RTP circuit breaker =
is also a MUST implement, and since it's a single-ended thing, it makes =
no sense if it isn't also MUST use.
>=20
> Scanning, I=92m not even sure a WebRTC endpoint is required to send =
RTCP - but I think it should be.

=93RTCP is a fundamental and integral part of RTP, and MUST be =
implemented in all WebRTC applications.=94 rtp-usage-16 section 4.1

> (Note: The -rtp-usage draft uses "WebRTC endpoint" to mean a =
fully-featured endpoint... obviously we need to harmonize at some =
point.)

We certainly need to make sure the terminology is consistent, but I =
would think that non-browser WebRTC endpoints would need to conform to =
rtp-usage, since that specifies what they might be sent by a WebRTC =
browser, what they can send to such a browser, and what needs to be =
negotiated.=20

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




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From nobody Wed Aug 20 03:26:30 2014
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To: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
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><Raju>=20
> ... In addition the HTTP Connect MUST include an indication of the protoc=
ol being...
>
>Is the tunnel indication in CONNECT a "MUST"? or Can it be changed to "SHO=
ULD"?
>The reason is, though typically unrecognized HTTP headers are ignored, wha=
t if some proxies are sensitive to >HTTP CONNECT's  unrecognized headers? I=
n such a case excluding this header may just work fine.
>
></Raju>

The consensus during the RTCWEB interim meeting in May was to my understand=
ing that this needed to be a MUST so that proxies are told that the tunnel =
will contain real-time media. My opinion is that we need browsers to be con=
sistent and this overrides the need to cater for broken proxies.







From nobody Wed Aug 20 03:38:29 2014
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On 08/20/2014 11:46 AM, Colin Perkins wrote:
> On 19 Aug 2014, at 15:00, Harald Alvestrand <harald@alvestrand.no> wrote:
>> On 08/19/2014 03:38 PM, Jim Barnett wrote:
>>> I agree.  We need to define a) WebRTC browsers, which support the full protocol requirements and the API, b) WebRTC Endpoints/Devices, which support some minimum protocol set sufficient for establishing connectivity.  Is there also a need to define c) Full WebRTC Devices, which support more than the minimum protocol set?  What would be the use of definition c?
>> An endpoint (I kind of start to like this concept/name) would be conformant as long as it was able to successfully negotiate away use of anything that makes life difficult.
>>
>> It would be required to support DTLS-SRTP key handshakes and respond to ICE connectivity checks, plus whatever is MUST use in -rtp-usage (scanning, I'm not coming up with much - it's mostly MUST respond to, MUST implement (but not required to use) stuff). The RTP circuit breaker is also a MUST implement, and since it's a single-ended thing, it makes no sense if it isn't also MUST use.
>>
>> Scanning, I’m not even sure a WebRTC endpoint is required to send RTCP - but I think it should be.
> “RTCP is a fundamental and integral part of RTP, and MUST be implemented in all WebRTC applications.” rtp-usage-16 section 4.1

Yep, I thought we wanted that too, but then I started to scan for 
mandatory-to-implement vs mandatory-to-use, and started being unsure; is 
it permitted to configure a WebRTC device to set RTCP-bandwidth to zero 
(the traditional means of turning off RTCP in a standards conformant 
manner, I think)?

Perhaps we should add a requirement on minimum RTCP bandwidth?
>
>> (Note: The -rtp-usage draft uses "WebRTC endpoint" to mean a fully-featured endpoint... obviously we need to harmonize at some point.)
> We certainly need to make sure the terminology is consistent, but I would think that non-browser WebRTC endpoints would need to conform to rtp-usage, since that specifies what they might be sent by a WebRTC browser, what they can send to such a browser, and what needs to be negotiated.
>
Yup, it was when I started to read it from a "what can we successfully 
negotiate away" viewpoint that I started to worry about what we had in 
fact required.




From nobody Wed Aug 20 03:48:10 2014
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On 20 Aug 2014, at 11:38, Harald Alvestrand <harald@alvestrand.no> =
wrote:
> On 08/20/2014 11:46 AM, Colin Perkins wrote:
>> On 19 Aug 2014, at 15:00, Harald Alvestrand <harald@alvestrand.no> =
wrote:
>>> On 08/19/2014 03:38 PM, Jim Barnett wrote:
>>>> I agree.  We need to define a) WebRTC browsers, which support the =
full protocol requirements and the API, b) WebRTC Endpoints/Devices, =
which support some minimum protocol set sufficient for establishing =
connectivity.  Is there also a need to define c) Full WebRTC Devices, =
which support more than the minimum protocol set?  What would be the use =
of definition c?
>>> An endpoint (I kind of start to like this concept/name) would be =
conformant as long as it was able to successfully negotiate away use of =
anything that makes life difficult.
>>>=20
>>> It would be required to support DTLS-SRTP key handshakes and respond =
to ICE connectivity checks, plus whatever is MUST use in -rtp-usage =
(scanning, I'm not coming up with much - it's mostly MUST respond to, =
MUST implement (but not required to use) stuff). The RTP circuit breaker =
is also a MUST implement, and since it's a single-ended thing, it makes =
no sense if it isn't also MUST use.
>>>=20
>>> Scanning, I=92m not even sure a WebRTC endpoint is required to send =
RTCP - but I think it should be.
>> =93RTCP is a fundamental and integral part of RTP, and MUST be =
implemented in all WebRTC applications.=94 rtp-usage-16 section 4.1
>=20
> Yep, I thought we wanted that too, but then I started to scan for =
mandatory-to-implement vs mandatory-to-use, and started being unsure; is =
it permitted to configure a WebRTC device to set RTCP-bandwidth to zero =
(the traditional means of turning off RTCP in a standards conformant =
manner, I think)?
>=20
> Perhaps we should add a requirement on minimum RTCP bandwidth?

I wouldn=92t know what minimum RTCP bandwidth to specify, but could =
certainly change the rtp-usage draft to say =93=85MUST be implemented =
and used=85=94 if that would help?=20

>>> (Note: The -rtp-usage draft uses "WebRTC endpoint" to mean a =
fully-featured endpoint... obviously we need to harmonize at some =
point.)
>> We certainly need to make sure the terminology is consistent, but I =
would think that non-browser WebRTC endpoints would need to conform to =
rtp-usage, since that specifies what they might be sent by a WebRTC =
browser, what they can send to such a browser, and what needs to be =
negotiated.
>>=20
> Yup, it was when I started to read it from a "what can we successfully =
negotiate away" viewpoint that I started to worry about what we had in =
fact required.

Please send concrete suggestions for changes, if there are things you =
think we should fix.


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




From nobody Wed Aug 20 03:57:04 2014
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Date: Wed, 20 Aug 2014 12:57:05 +0200
From: Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com>
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Subject: Re: [rtcweb] WebRTC Device: Lowering the bar, or inventing a new term?
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On 20/08/2014 12:47, Colin Perkins wrote:
> On 20 Aug 2014, at 11:38, Harald Alvestrand <harald@alvestrand.no> wrote:
>> On 08/20/2014 11:46 AM, Colin Perkins wrote:
>>> On 19 Aug 2014, at 15:00, Harald Alvestrand <harald@alvestrand.no> wrote:
>>>> On 08/19/2014 03:38 PM, Jim Barnett wrote:
>>>>> I agree.  We need to define a) WebRTC browsers, which support the full protocol requirements and the API, b) WebRTC Endpoints/Devices, which support some minimum protocol set sufficient for establishing connectivity.  Is there also a need to define c) Full WebRTC Devices, which support more than the minimum protocol set?  What would be the use of definition c?
>>>> An endpoint (I kind of start to like this concept/name) would be conformant as long as it was able to successfully negotiate away use of anything that makes life difficult.
>>>>
>>>> It would be required to support DTLS-SRTP key handshakes and respond to ICE connectivity checks, plus whatever is MUST use in -rtp-usage (scanning, I'm not coming up with much - it's mostly MUST respond to, MUST implement (but not required to use) stuff). The RTP circuit breaker is also a MUST implement, and since it's a single-ended thing, it makes no sense if it isn't also MUST use.
>>>>
>>>> Scanning, I’m not even sure a WebRTC endpoint is required to send RTCP - but I think it should be.
>>> “RTCP is a fundamental and integral part of RTP, and MUST be implemented in all WebRTC applications.” rtp-usage-16 section 4.1
>> Yep, I thought we wanted that too, but then I started to scan for mandatory-to-implement vs mandatory-to-use, and started being unsure; is it permitted to configure a WebRTC device to set RTCP-bandwidth to zero (the traditional means of turning off RTCP in a standards conformant manner, I think)?
>>
>> Perhaps we should add a requirement on minimum RTCP bandwidth?
> I wouldn’t know what minimum RTCP bandwidth to specify, but could certainly change the rtp-usage draft to say “…MUST be implemented and used…” if that would help?
>
Theoretically speaking, it could be possible that all RTCP packets sent 
by a peer could be lost (dropped by the network for example) and the 
communication would still possible (as long as the content 
freshness/keep alive is done via STUN/ICE). So in practice, it should be 
possible for a WebRTC endpoint communicate with  a peer not 
implementing/sending RTCP at all.

Best regards
Sergio


From nobody Wed Aug 20 05:10:21 2014
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From: Varun Singh <vsingh.ietf@gmail.com>
Date: Wed, 20 Aug 2014 15:09:56 +0300
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Subject: Re: [rtcweb] WebRTC Device: Lowering the bar, or inventing a new term?
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On Wed, Aug 20, 2014 at 1:57 PM, Sergio Garcia Murillo
<sergio.garcia.murillo@gmail.com> wrote:
> On 20/08/2014 12:47, Colin Perkins wrote:
>>
>> On 20 Aug 2014, at 11:38, Harald Alvestrand <harald@alvestrand.no> wrote=
:
>>>
>>> On 08/20/2014 11:46 AM, Colin Perkins wrote:
>>>>
>>>> On 19 Aug 2014, at 15:00, Harald Alvestrand <harald@alvestrand.no>
>>>> wrote:
>>>>>
>>>>> On 08/19/2014 03:38 PM, Jim Barnett wrote:
>>>>>>
>>>>>> I agree.  We need to define a) WebRTC browsers, which support the fu=
ll
>>>>>> protocol requirements and the API, b) WebRTC Endpoints/Devices, whic=
h
>>>>>> support some minimum protocol set sufficient for establishing connec=
tivity.
>>>>>> Is there also a need to define c) Full WebRTC Devices, which support=
 more
>>>>>> than the minimum protocol set?  What would be the use of definition =
c?
>>>>>
>>>>> An endpoint (I kind of start to like this concept/name) would be
>>>>> conformant as long as it was able to successfully negotiate away use =
of
>>>>> anything that makes life difficult.
>>>>>
>>>>> It would be required to support DTLS-SRTP key handshakes and respond =
to
>>>>> ICE connectivity checks, plus whatever is MUST use in -rtp-usage (sca=
nning,
>>>>> I'm not coming up with much - it's mostly MUST respond to, MUST imple=
ment
>>>>> (but not required to use) stuff). The RTP circuit breaker is also a M=
UST
>>>>> implement, and since it's a single-ended thing, it makes no sense if =
it
>>>>> isn't also MUST use.
>>>>>
>>>>> Scanning, I=E2=80=99m not even sure a WebRTC endpoint is required to =
send RTCP
>>>>> - but I think it should be.
>>>>
>>>> =E2=80=9CRTCP is a fundamental and integral part of RTP, and MUST be i=
mplemented
>>>> in all WebRTC applications.=E2=80=9D rtp-usage-16 section 4.1
>>>
>>> Yep, I thought we wanted that too, but then I started to scan for
>>> mandatory-to-implement vs mandatory-to-use, and started being unsure; i=
s it
>>> permitted to configure a WebRTC device to set RTCP-bandwidth to zero (t=
he
>>> traditional means of turning off RTCP in a standards conformant manner,=
 I
>>> think)?
>>>
>>> Perhaps we should add a requirement on minimum RTCP bandwidth?
>>
>> I wouldn=E2=80=99t know what minimum RTCP bandwidth to specify, but coul=
d
>> certainly change the rtp-usage draft to say =E2=80=9C=E2=80=A6MUST be im=
plemented and used=E2=80=A6=E2=80=9D
>> if that would help?
>>
> Theoretically speaking, it could be possible that all RTCP packets sent b=
y a
> peer could be lost (dropped by the network for example) and the
> communication would still possible (as long as the content freshness/keep
> alive is done via STUN/ICE). So in practice, it should be possible for a
> WebRTC endpoint communicate with  a peer not implementing/sending RTCP at
> all.
>

Since RTP Circuit Breaker is MTI as per rtp-usage.

Not receiving any RTCP could potentially trigger the circuit-breaker.
More so, if the stream is not sending any RTP packets in the
measurement period (between 3 RTCP reports).

Regards,
Varun


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



--=20
http://www.netlab.tkk.fi/~varun/


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Subject: Re: [rtcweb] WebRTC Device: Lowering the bar, or inventing a new term?
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On 8/20/14 6:47 AM, Colin Perkins wrote:
> On 20 Aug 2014, at 11:38, Harald Alvestrand <harald@alvestrand.no> wrote:
>> On 08/20/2014 11:46 AM, Colin Perkins wrote:
>>> On 19 Aug 2014, at 15:00, Harald Alvestrand <harald@alvestrand.no> wrote:
>>>> On 08/19/2014 03:38 PM, Jim Barnett wrote:
>>>>> I agree.  We need to define a) WebRTC browsers, which support the full protocol requirements and the API, b) WebRTC Endpoints/Devices, which support some minimum protocol set sufficient for establishing connectivity.  Is there also a need to define c) Full WebRTC Devices, which support more than the minimum protocol set?  What would be the use of definition c?
>>>> An endpoint (I kind of start to like this concept/name) would be conformant as long as it was able to successfully negotiate away use of anything that makes life difficult.
>>>>
>>>> It would be required to support DTLS-SRTP key handshakes and respond to ICE connectivity checks, plus whatever is MUST use in -rtp-usage (scanning, I'm not coming up with much - it's mostly MUST respond to, MUST implement (but not required to use) stuff). The RTP circuit breaker is also a MUST implement, and since it's a single-ended thing, it makes no sense if it isn't also MUST use.
>>>>
>>>> Scanning, I’m not even sure a WebRTC endpoint is required to send RTCP - but I think it should be.
>>> “RTCP is a fundamental and integral part of RTP, and MUST be implemented in all WebRTC applications.” rtp-usage-16 section 4.1
>>
>> Yep, I thought we wanted that too, but then I started to scan for mandatory-to-implement vs mandatory-to-use, and started being unsure; is it permitted to configure a WebRTC device to set RTCP-bandwidth to zero (the traditional means of turning off RTCP in a standards conformant manner, I think)?
>>
>> Perhaps we should add a requirement on minimum RTCP bandwidth?
>
> I wouldn’t know what minimum RTCP bandwidth to specify, but could certainly change the rtp-usage draft to say “…MUST be implemented and used…” if that would help?

That is fine *if RTP is being used*.

Use, or even implementation, of RTP should not be required in WebRTC 
devices that don't need it.

(I know that is what you meant.)

	Thanks,
	Paul


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Date: Wed, 20 Aug 2014 22:07:03 +0530
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Archived-At: http://mailarchive.ietf.org/arch/msg/rtcweb/UUhSVoP2PFMRs3_1ViXVfEPwMCE
Subject: Re: [rtcweb] WG Last Call for	draft-ietf-rtcweb-stun-consent-freshness
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Hi Muthu,

=20

I agree with Harald that WebRTC endpoint & WebRTC gateway may implement =
ice-lite and so, there is no need to send consent check. IIUC, the =
intention of the draft is to mention that Full ICE supporting WebRTC =
entities MUST support consent check.=20

=20

I think that the following changes will be sufficient for the below =
comment:

=20

1)      WebRTC endpoint & WebRTC implementation replaced by WebRTC =
devices. Please note that WebRTC browser is a webrtc device but not the =
other way around.=20

=20

2)      Add the statement that  Full ICE supporting WebRTC entities MUST =
support consent freshness. This statements will be useful for WebRTC =
Endpoint or WebRTC gateway which supports Full ICE.

=20

Thanks

Partha

=20

From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Harald =
Alvestrand
Sent: Wednesday, August 20, 2014 1:14 PM
To: rtcweb@ietf.org
Subject: Re: [rtcweb] WG Last Call for =
draft-ietf-rtcweb-stun-consent-freshness

=20

On 08/20/2014 06:09 AM, Muthu Arul Mozhi Perumal wrote:

I believe draft-ietf-rtcweb-stun-consent-freshness deals with the WebRTC =
browser. A WebRTC device (eg. a mobile phone) on the other hand, could =
run a WebRTC browser and a JS and perform consent. Similarly, a WebRTC =
endpoint (whatever that gets defined as) might also perform consent, =
while a WebRTC gateway might not, and how consent applies to those =
should get specified in the document(s) that define those requirements =
(note: the gateway referred to in =
draft-ietf-rtcweb-stun-consent-freshness is any SIP gateway).


Note: In terms of conformance requirements, "might" and "might not" are =
completely equivalent (2119 style: MAY - there is no MAY NOT).




=20

So, draft-ietf-rtcweb-stun-consent-freshness should just replace WebRTC =
endpoint and WebRTC implementation with WebRTC browser, IMHO.

=20

Muthu

=20

On Tue, Aug 19, 2014 at 7:36 PM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

Hi,

=20

The draft says:

=20

=E2=80=9CWebRTC endpoints are required to support full ICE as specified =
in

                section 3.4 of [I-D.

=E2=80=8B =E2=80=8B

ietf-rtcweb-transports].  However, when WebRTC=20

                endpoints interwork with other endpoints that support =
only ICE-lite

                (e.g., gateways) those endpoints will not generate =
consent checks,

                but just respond to consent checks they =
receive.=E2=80=9D

=20

The draft also talks about =E2=80=9CWebRTC implementations=E2=80=9D and =
=E2=80=9CWebRTC browsers=E2=80=9D.

=20

We need to make sure that the terminology is aligned with =
draft-ietf-rtcweb-overview (for example,=20

=E2=80=8B =E2=80=8B

draft-ietf-rtcweb-overview does not talk about =E2=80=9CWebRTC =
endpoints=E2=80=9D), which also may be impacted based on the outcome of =
the current terminology/gateway discussion.=20

=20

Regards,

=20

Christer

=20

=20

From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Ted Hardie
Sent: 14. elokuuta 2014 18:10
To: rtcweb@ietf.org; Sean Turner; Cullen Jennings
Subject: [rtcweb] WG Last Call for =
draft-ietf-rtcweb-stun-consent-freshness

=20

This starts a WG Last Call for draft-ietf-rtcweb-stun-consent-freshness =
(available at =
http://datatracker.ietf.org/doc/draft-ietf-rtcweb-stun-consent-freshness/=
).  Please review the document and send comments to the list by =
September 10, 2014.

thanks,

Ted Hardie


_______________________________________________
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
Surveillance is pervasive. Go Dark.

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<body bgcolor=3Dwhite lang=3DEN-US link=3Dblue vlink=3Dpurple>

<div class=3DSection1>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Hi Muthu,<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>I agree with Harald that WebRTC endpoint &amp; WebRTC =
gateway
may implement ice-lite and so, there is no need to send consent check. =
IIUC,
the intention of the draft is to mention that Full ICE supporting WebRTC
entities MUST support consent check. <o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>I think that the following changes will be sufficient for =
the
below comment:<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo1'><![if !supportLists]><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><span
style=3D'mso-list:Ignore'>1)<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>WebRTC endpoint &amp; WebRTC implementation replaced by =
WebRTC
devices. Please note that WebRTC browser is a webrtc device but not the =
other
way around. <o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo1'><![if !supportLists]><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><span
style=3D'mso-list:Ignore'>2)<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Add the statement that =C2=A0Full ICE supporting WebRTC =
entities MUST
support consent freshness. This statements will be useful for WebRTC =
Endpoint or
WebRTC gateway which supports Full ICE.<o:p></o:p></span></p>

<p class=3DMsoListParagraph><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Thanks<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Partha<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<div style=3D'border:none;border-left:solid blue 1.5pt;padding:0in 0in =
0in 4.0pt'>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt =
0in 0in 0in'>

<p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";
color:windowtext'>From:</span></b><span =
style=3D'font-size:10.0pt;font-family:
"Tahoma","sans-serif";color:windowtext'> rtcweb
[mailto:rtcweb-bounces@ietf.org] <b>On Behalf Of </b>Harald =
Alvestrand<br>
<b>Sent:</b> Wednesday, August 20, 2014 1:14 PM<br>
<b>To:</b> rtcweb@ietf.org<br>
<b>Subject:</b> Re: [rtcweb] WG Last Call for
draft-ietf-rtcweb-stun-consent-freshness<o:p></o:p></span></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<div>

<p class=3DMsoNormal>On 08/20/2014 06:09 AM, Muthu Arul Mozhi Perumal =
wrote:<o:p></o:p></p>

</div>

<blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'>

<div>

<div>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif"'>I
believe&nbsp;draft-ietf-rtcweb-stun-consent-freshness deals with the =
WebRTC
browser. A WebRTC device (eg. a mobile phone) on the other hand, could =
run a
WebRTC browser and a JS and perform consent. Similarly, a WebRTC =
endpoint
(whatever that gets defined as) might also perform consent, while a =
WebRTC
gateway might not, and how consent applies to those should get specified =
in the
document(s) that define those requirements (note: the gateway referred =
to in
draft-ietf-rtcweb-stun-consent-freshness is any SIP =
gateway).<o:p></o:p></span></p>

</div>

</div>

</blockquote>

<p class=3DMsoNormal><br>
Note: In terms of conformance requirements, &quot;might&quot; and =
&quot;might
not&quot; are completely equivalent (2119 style: MAY - there is no MAY =
NOT).<br>
<br>
<br>
<o:p></o:p></p>

<div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'>So,
draft-ietf-rtcweb-stun-consent-freshness should just replace WebRTC =
endpoint
and WebRTC implementation with WebRTC browser, =
IMHO.<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'>Muthu<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<div>

<p class=3DMsoNormal>On Tue, Aug 19, 2014 at 7:36 PM, Christer Holmberg =
&lt;<a
href=3D"mailto:christer.holmberg@ericsson.com" =
target=3D"_blank">christer.holmberg@ericsson.com</a>&gt;
wrote:<o:p></o:p></p>

<div>

<div>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Hi,</span><o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>&nbsp;</span><o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>The
draft says:</span><o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>&nbsp;</span><o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;
text-indent:.5in'><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>=E2=80=9CWebRTC endpoints are required to support full =
ICE as specified
in</span><o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>&nbsp;&nbsp;
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
section 3.4 of [I-D.</span><o:p></o:p></p>

<div>

<p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'>=E2=80=8B =
=E2=80=8B<o:p></o:p></span></p>

</div>

<p class=3DMsoNormal>ietf-rtcweb-transports].&nbsp; However, when WebRTC =
<o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>&nbsp;&nbsp;
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
endpoints interwork with other endpoints that support only =
ICE-lite</span><o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>&nbsp;&nbsp;
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
(e.g.,
gateways) those endpoints will not generate consent =
checks,</span><o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>&nbsp;&nbsp;
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
but
just respond to consent checks they =
receive.=E2=80=9D</span><o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>&nbsp;</span><o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>The
draft also talks about =E2=80=9CWebRTC implementations=E2=80=9D and =
=E2=80=9CWebRTC browsers=E2=80=9D.</span><o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>&nbsp;</span><o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>We
need to make sure that the terminology is aligned with
draft-ietf-rtcweb-overview (for example, </span><o:p></o:p></p>

<div>

<p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif"'>=E2=80=8B =
=E2=80=8B<o:p></o:p></span></p>

</div>

<p class=3DMsoNormal>draft-ietf-rtcweb-overview does not talk about =
=E2=80=9CWebRTC
endpoints=E2=80=9D), which also may be impacted based on the outcome of =
the current
terminology/gateway discussion. <o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>&nbsp;</span><o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Regards,</span><o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>&nbsp;</span><o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Christer</span><o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>&nbsp;</span><o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>&nbsp;</span><o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> rtcweb =
[mailto:<a
href=3D"mailto:rtcweb-bounces@ietf.org" =
target=3D"_blank">rtcweb-bounces@ietf.org</a>]
<b>On Behalf Of </b>Ted Hardie<br>
<b>Sent:</b> 14. elokuuta 2014 18:10<br>
<b>To:</b> <a href=3D"mailto:rtcweb@ietf.org" =
target=3D"_blank">rtcweb@ietf.org</a>;
Sean Turner; Cullen Jennings<br>
<b>Subject:</b> [rtcweb] WG Last Call for
draft-ietf-rtcweb-stun-consent-freshness</span><o:p></o:p></p>

<div>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>&nbsp;<o:p><=
/o:p></p>

<div>

<div>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;margin-bottom:12.0pt'><span
style=3D'font-family:"Garamond","serif"'>This starts a WG Last Call for
draft-ietf-rtcweb-stun-consent-freshness (available at <a
href=3D"http://datatracker.ietf.org/doc/draft-ietf-rtcweb-stun-consent-fr=
eshness/"
target=3D"_blank">http://datatracker.ietf.org/doc/draft-ietf-rtcweb-stun-=
consent-freshness/</a>).&nbsp;
Please review the document and send comments to the list by September =
10, 2014.</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style=3D'font-family:"Garamond","serif"'>thanks,<br>
<br>
Ted Hardie</span><o:p></o:p></p>

</div>

</div>

</div>

</div>

</div>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" =
target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><o:p></=
o:p></p>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

</div>

<p class=3DMsoNormal><br>
<br>
<br>
<o:p></o:p></p>

<pre>_______________________________________________<o:p></o:p></pre><pre=
>rtcweb mailing list<o:p></o:p></pre><pre><a
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><o:p></o:p></pre><pre>=
<a
href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.or=
g/mailman/listinfo/rtcweb</a><o:p></o:p></pre>

<p class=3DMsoNormal><br>
<br>
<br>
<o:p></o:p></p>

<pre>-- <o:p></o:p></pre><pre>Surveillance is pervasive. Go =
Dark.<o:p></o:p></pre></div>

</div>

</body>

</html>

------=_NextPart_000_001C_01CFBCC3.16858F10--


From nobody Wed Aug 20 21:16:27 2014
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Date: Thu, 21 Aug 2014 09:46:19 +0530
Message-ID: <CAKz0y8zNM3rc3XC6JqrK+d4hXiT5TomhNM+W2twg0+-83-pFow@mail.gmail.com>
From: Muthu Arul Mozhi Perumal <muthu.arul@gmail.com>
To: Parthasarathi R <partha@parthasarathi.co.in>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] WG Last Call for draft-ietf-rtcweb-stun-consent-freshness
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--f46d043c08ea86368105011bfb94
Content-Type: text/plain; charset=UTF-8
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The elementary entity that needs to perform consent is a WebRTC entity that
runs untrusted applications -- essentially a WebRTC browser. Other entities
may perform consent as a consequence of them running a browser or because
they want to use it to compute RTT or whatever and not simply because they
support full ICE. This elementary entity is the target for the consent
draft and I don't think the draft should convolute it any further. Of
course, adding some text that other WebRTC entities may perform consent for
the above said reasons could help, I think.

Muthu

On Wed, Aug 20, 2014 at 10:07 PM, Parthasarathi R <
partha@parthasarathi.co.in> wrote:

>  Hi Muthu,
>
>
>
> I agree with Harald that WebRTC endpoint & WebRTC gateway may implement
> ice-lite and so, there is no need to send consent check. IIUC, the
> intention of the draft is to mention that Full ICE supporting WebRTC
> entities MUST support consent check.
>
>
>
> I think that the following changes will be sufficient for the below
> comment:
>
>
>
> 1)      WebRTC endpoint & WebRTC implementation replaced by WebRTC
> devices. Please note that WebRTC browser is a webrtc device but not the
> other way around.
>
>
>
> 2)      Add the statement that  Full ICE supporting WebRTC entities MUST
> support consent freshness. This statements will be useful for WebRTC
> Endpoint or WebRTC gateway which supports Full ICE.
>
>
>
> Thanks
>
> Partha
>
>
>
> *From:* rtcweb [mailto:rtcweb-bounces@ietf.org] *On Behalf Of *Harald
> Alvestrand
> *Sent:* Wednesday, August 20, 2014 1:14 PM
> *To:* rtcweb@ietf.org
> *Subject:* Re: [rtcweb] WG Last Call for
> draft-ietf-rtcweb-stun-consent-freshness
>
>
>
> On 08/20/2014 06:09 AM, Muthu Arul Mozhi Perumal wrote:
>
>  I believe draft-ietf-rtcweb-stun-consent-freshness deals with the WebRTC
> browser. A WebRTC device (eg. a mobile phone) on the other hand, could ru=
n
> a WebRTC browser and a JS and perform consent. Similarly, a WebRTC endpoi=
nt
> (whatever that gets defined as) might also perform consent, while a WebRT=
C
> gateway might not, and how consent applies to those should get specified =
in
> the document(s) that define those requirements (note: the gateway referre=
d
> to in draft-ietf-rtcweb-stun-consent-freshness is any SIP gateway).
>
>
> Note: In terms of conformance requirements, "might" and "might not" are
> completely equivalent (2119 style: MAY - there is no MAY NOT).
>
>
>
>
> So, draft-ietf-rtcweb-stun-consent-freshness should just replace WebRTC
> endpoint and WebRTC implementation with WebRTC browser, IMHO.
>
>
>
> Muthu
>
>
>
> On Tue, Aug 19, 2014 at 7:36 PM, Christer Holmberg <
> christer.holmberg@ericsson.com> wrote:
>
> Hi,
>
>
>
> The draft says:
>
>
>
> =E2=80=9CWebRTC endpoints are required to support full ICE as specified i=
n
>
>                 section 3.4 of [I-D.
>
> =E2=80=8B =E2=80=8B
>
> ietf-rtcweb-transports].  However, when WebRTC
>
>                 endpoints interwork with other endpoints that support onl=
y
> ICE-lite
>
>                 (e.g., gateways) those endpoints will not generate consen=
t
> checks,
>
>                 but just respond to consent checks they receive.=E2=80=9D
>
>
>
> The draft also talks about =E2=80=9CWebRTC implementations=E2=80=9D and =
=E2=80=9CWebRTC browsers=E2=80=9D.
>
>
>
> We need to make sure that the terminology is aligned with
> draft-ietf-rtcweb-overview (for example,
>
> =E2=80=8B =E2=80=8B
>
> draft-ietf-rtcweb-overview does not talk about =E2=80=9CWebRTC endpoints=
=E2=80=9D), which
> also may be impacted based on the outcome of the current
> terminology/gateway discussion.
>
>
>
> Regards,
>
>
>
> Christer
>
>
>
>
>
> *From:* rtcweb [mailto:rtcweb-bounces@ietf.org] *On Behalf Of *Ted Hardie
> *Sent:* 14. elokuuta 2014 18:10
> *To:* rtcweb@ietf.org; Sean Turner; Cullen Jennings
> *Subject:* [rtcweb] WG Last Call for
> draft-ietf-rtcweb-stun-consent-freshness
>
>
>
> This starts a WG Last Call for draft-ietf-rtcweb-stun-consent-freshness
> (available at
> http://datatracker.ietf.org/doc/draft-ietf-rtcweb-stun-consent-freshness/=
).
> Please review the document and send comments to the list by September 10,
> 2014.
>
> thanks,
>
> Ted Hardie
>
>
> _______________________________________________
> 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
>
>
>
>
>  --
>
> Surveillance is pervasive. Go Dark.
>
>

--f46d043c08ea86368105011bfb94
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:arial,he=
lvetica,sans-serif;font-size:small">The elementary entity that needs to per=
form consent is a WebRTC entity that runs untrusted applications -- essenti=
ally a WebRTC browser. Other entities may perform consent as a consequence =
of them running a browser or because they want to use it to compute RTT or =
whatever and not simply because they support full ICE. This elementary enti=
ty is the target for the consent draft and I don&#39;t think the draft shou=
ld convolute it any further. Of course, adding some text that other WebRTC =
entities may perform consent for the above said reasons could help, I think=
.<br>
</div><div class=3D"gmail_default" style=3D"font-family:arial,helvetica,san=
s-serif;font-size:small"><br></div><div class=3D"gmail_default" style=3D"fo=
nt-family:arial,helvetica,sans-serif;font-size:small">Muthu</div><div class=
=3D"gmail_extra">
<br><div class=3D"gmail_quote">On Wed, Aug 20, 2014 at 10:07 PM, Parthasara=
thi R <span dir=3D"ltr">&lt;<a href=3D"mailto:partha@parthasarathi.co.in" t=
arget=3D"_blank">partha@parthasarathi.co.in</a>&gt;</span> wrote:<br><block=
quote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc=
 solid;padding-left:1ex">









<div bgcolor=3D"white" lang=3D"EN-US" link=3D"blue" vlink=3D"purple">

<div>

<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Hi Muthu,<u></u><u></u></=
span></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=C2=A0<u></u></spa=
n></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">I agree with Harald that =
WebRTC endpoint &amp; WebRTC gateway
may implement ice-lite and so, there is no need to send consent check. IIUC=
,
the intention of the draft is to mention that Full ICE supporting WebRTC
entities MUST support consent check. <u></u><u></u></span></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=C2=A0<u></u></spa=
n></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">I think that the followin=
g changes will be sufficient for the
below comment:<u></u><u></u></span></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=C2=A0<u></u></spa=
n></p>

<p><u></u><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&=
quot;sans-serif&quot;;color:#1f497d"><span>1)<span style=3D"font:7.0pt &quo=
t;Times New Roman&quot;">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0
</span></span></span><u></u><span style=3D"font-size:11.0pt;font-family:&qu=
ot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">WebRTC endpoint &amp=
; WebRTC implementation replaced by WebRTC
devices. Please note that WebRTC browser is a webrtc device but not the oth=
er
way around. <u></u><u></u></span></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=C2=A0<u></u></spa=
n></p>

<p><u></u><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&=
quot;sans-serif&quot;;color:#1f497d"><span>2)<span style=3D"font:7.0pt &quo=
t;Times New Roman&quot;">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0
</span></span></span><u></u><span style=3D"font-size:11.0pt;font-family:&qu=
ot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">Add the statement th=
at =C2=A0Full ICE supporting WebRTC entities MUST
support consent freshness. This statements will be useful for WebRTC Endpoi=
nt or
WebRTC gateway which supports Full ICE.<u></u><u></u></span></p>

<p><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sa=
ns-serif&quot;;color:#1f497d"><u></u>=C2=A0<u></u></span></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Thanks<u></u><u></u></spa=
n></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Partha<u></u><u></u></spa=
n></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=C2=A0<u></u></spa=
n></p>

<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">

<div>

<div style=3D"border:none;border-top:solid #b5c4df 1.0pt;padding:3.0pt 0in =
0in 0in">

<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext">From:</span></b><spa=
n style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif=
&quot;;color:windowtext"> rtcweb
[mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org" target=3D"_blank">rtcweb=
-bounces@ietf.org</a>] <b>On Behalf Of </b>Harald Alvestrand<br>
<b>Sent:</b> Wednesday, August 20, 2014 1:14 PM<br>
<b>To:</b> <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf=
.org</a><br>
<b>Subject:</b> Re: [rtcweb] WG Last Call for
draft-ietf-rtcweb-stun-consent-freshness<u></u><u></u></span></p>

</div>

</div><div><div class=3D"h5">

<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>

<div>

<p class=3D"MsoNormal">On 08/20/2014 06:09 AM, Muthu Arul Mozhi Perumal wro=
te:<u></u><u></u></p>

</div>

<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">

<div>

<div>

<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Arial&quot;,&quot;s=
ans-serif&quot;">I
believe=C2=A0draft-ietf-rtcweb-stun-consent-freshness deals with the WebRTC
browser. A WebRTC device (eg. a mobile phone) on the other hand, could run =
a
WebRTC browser and a JS and perform consent. Similarly, a WebRTC endpoint
(whatever that gets defined as) might also perform consent, while a WebRTC
gateway might not, and how consent applies to those should get specified in=
 the
document(s) that define those requirements (note: the gateway referred to i=
n
draft-ietf-rtcweb-stun-consent-freshness is any SIP gateway).<u></u><u></u>=
</span></p>

</div>

</div>

</blockquote>

<p class=3D"MsoNormal"><br>
Note: In terms of conformance requirements, &quot;might&quot; and &quot;mig=
ht
not&quot; are completely equivalent (2119 style: MAY - there is no MAY NOT)=
.<br>
<br>
<br>
<u></u><u></u></p>

<div>

<div>

<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Arial&quot;,&quot;s=
ans-serif&quot;"><u></u>=C2=A0<u></u></span></p>

</div>

<div>

<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Arial&quot;,&quot;s=
ans-serif&quot;">So,
draft-ietf-rtcweb-stun-consent-freshness should just replace WebRTC endpoin=
t
and WebRTC implementation with WebRTC browser, IMHO.<u></u><u></u></span></=
p>

</div>

<div>

<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Arial&quot;,&quot;s=
ans-serif&quot;"><u></u>=C2=A0<u></u></span></p>

</div>

<div>

<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Arial&quot;,&quot;s=
ans-serif&quot;">Muthu<u></u><u></u></span></p>

</div>

<div>

<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>

<div>

<p class=3D"MsoNormal">On Tue, Aug 19, 2014 at 7:36 PM, Christer Holmberg &=
lt;<a href=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">chri=
ster.holmberg@ericsson.com</a>&gt;
wrote:<u></u><u></u></p>

<div>

<div>

<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Hi,</span><u></u><u></u><=
/p>

<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">=C2=A0</span><u></u><u></=
u></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">The
draft says:</span><u></u><u></u></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">=C2=A0</span><u></u><u></=
u></p>

<p class=3D"MsoNormal" style=3D"text-indent:.5in"><span style=3D"font-size:=
11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d=
">=E2=80=9CWebRTC endpoints are required to support full ICE as specified
in</span><u></u><u></u></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">=C2=A0=C2=A0
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0
section 3.4 of [I-D.</span><u></u><u></u></p>

<div>

<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Arial&quot;,&quot;s=
ans-serif&quot;">=E2=80=8B =E2=80=8B<u></u><u></u></span></p>

</div>

<p class=3D"MsoNormal">ietf-rtcweb-transports].=C2=A0 However, when WebRTC =
<u></u><u></u></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">=C2=A0=C2=A0
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0
endpoints interwork with other endpoints that support only ICE-lite</span><=
u></u><u></u></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">=C2=A0=C2=A0
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 (e=
.g.,
gateways) those endpoints will not generate consent checks,</span><u></u><u=
></u></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">=C2=A0=C2=A0
=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0 bu=
t
just respond to consent checks they receive.=E2=80=9D</span><u></u><u></u><=
/p>

<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">=C2=A0</span><u></u><u></=
u></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">The
draft also talks about =E2=80=9CWebRTC implementations=E2=80=9D and =E2=80=
=9CWebRTC browsers=E2=80=9D.</span><u></u><u></u></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">=C2=A0</span><u></u><u></=
u></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">We
need to make sure that the terminology is aligned with
draft-ietf-rtcweb-overview (for example, </span><u></u><u></u></p>

<div>

<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Arial&quot;,&quot;s=
ans-serif&quot;">=E2=80=8B =E2=80=8B<u></u><u></u></span></p>

</div>

<p class=3D"MsoNormal">draft-ietf-rtcweb-overview does not talk about =E2=
=80=9CWebRTC
endpoints=E2=80=9D), which also may be impacted based on the outcome of the=
 current
terminology/gateway discussion. <u></u><u></u></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">=C2=A0</span><u></u><u></=
u></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Regards,</span><u></u><u>=
</u></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">=C2=A0</span><u></u><u></=
u></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Christer</span><u></u><u>=
</u></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">=C2=A0</span><u></u><u></=
u></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">=C2=A0</span><u></u><u></=
u></p>

<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> rtcweb [=
mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org" target=3D"_blank">rtcweb-=
bounces@ietf.org</a>]
<b>On Behalf Of </b>Ted Hardie<br>
<b>Sent:</b> 14. elokuuta 2014 18:10<br>
<b>To:</b> <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf=
.org</a>;
Sean Turner; Cullen Jennings<br>
<b>Subject:</b> [rtcweb] WG Last Call for
draft-ietf-rtcweb-stun-consent-freshness</span><u></u><u></u></p>

<div>

<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>

<div>

<div>

<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-f=
amily:&quot;Garamond&quot;,&quot;serif&quot;">This starts a WG Last Call fo=
r
draft-ietf-rtcweb-stun-consent-freshness (available at <a href=3D"http://da=
tatracker.ietf.org/doc/draft-ietf-rtcweb-stun-consent-freshness/" target=3D=
"_blank">http://datatracker.ietf.org/doc/draft-ietf-rtcweb-stun-consent-fre=
shness/</a>).=C2=A0
Please review the document and send comments to the list by September 10, 2=
014.</span><u></u><u></u></p>

</div>

<div>

<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Garamond&quot;,&quo=
t;serif&quot;">thanks,<br>
<br>
Ted Hardie</span><u></u><u></u></p>

</div>

</div>

</div>

</div>

</div>

<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></p>

</div>

<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>

</div>

</div>

<p class=3D"MsoNormal"><br>
<br>
<br>
<u></u><u></u></p>

<pre>_______________________________________________<u></u><u></u></pre><pr=
e>rtcweb mailing list<u></u><u></u></pre><pre><a href=3D"mailto:rtcweb@ietf=
.org" target=3D"_blank">rtcweb@ietf.org</a><u></u><u></u></pre><pre><a href=
=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">https:/=
/www.ietf.org/mailman/listinfo/rtcweb</a><u></u><u></u></pre>


<p class=3D"MsoNormal"><br>
<br>
<br>
<u></u><u></u></p>

<pre>-- <u></u><u></u></pre><pre>Surveillance is pervasive. Go Dark.<u></u>=
<u></u></pre></div></div></div>

</div>

</div>


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

--f46d043c08ea86368105011bfb94--


From nobody Thu Aug 21 04:42:43 2014
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From: Colin Perkins <csp@csperkins.org>
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Subject: Re: [rtcweb] WebRTC Device: Lowering the bar, or inventing a new term?
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On 20 Aug 2014, at 15:24, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
> On 8/20/14 6:47 AM, Colin Perkins wrote:
>> On 20 Aug 2014, at 11:38, Harald Alvestrand <harald@alvestrand.no> =
wrote:
>>> On 08/20/2014 11:46 AM, Colin Perkins wrote:
>>>> On 19 Aug 2014, at 15:00, Harald Alvestrand <harald@alvestrand.no> =
wrote:
>>>>> On 08/19/2014 03:38 PM, Jim Barnett wrote:
>>>>>> I agree.  We need to define a) WebRTC browsers, which support the =
full protocol requirements and the API, b) WebRTC Endpoints/Devices, =
which support some minimum protocol set sufficient for establishing =
connectivity.  Is there also a need to define c) Full WebRTC Devices, =
which support more than the minimum protocol set?  What would be the use =
of definition c?
>>>>> An endpoint (I kind of start to like this concept/name) would be =
conformant as long as it was able to successfully negotiate away use of =
anything that makes life difficult.
>>>>>=20
>>>>> It would be required to support DTLS-SRTP key handshakes and =
respond to ICE connectivity checks, plus whatever is MUST use in =
-rtp-usage (scanning, I'm not coming up with much - it's mostly MUST =
respond to, MUST implement (but not required to use) stuff). The RTP =
circuit breaker is also a MUST implement, and since it's a single-ended =
thing, it makes no sense if it isn't also MUST use.
>>>>>=20
>>>>> Scanning, I=92m not even sure a WebRTC endpoint is required to =
send RTCP - but I think it should be.
>>>> =93RTCP is a fundamental and integral part of RTP, and MUST be =
implemented in all WebRTC applications.=94 rtp-usage-16 section 4.1
>>>=20
>>> Yep, I thought we wanted that too, but then I started to scan for =
mandatory-to-implement vs mandatory-to-use, and started being unsure; is =
it permitted to configure a WebRTC device to set RTCP-bandwidth to zero =
(the traditional means of turning off RTCP in a standards conformant =
manner, I think)?
>>>=20
>>> Perhaps we should add a requirement on minimum RTCP bandwidth?
>>=20
>> I wouldn=92t know what minimum RTCP bandwidth to specify, but could =
certainly change the rtp-usage draft to say =93=85MUST be implemented =
and used=85=94 if that would help?
>=20
> That is fine *if RTP is being used*.

I am suggesting adding this to the RTP usage draft=85

> Use, or even implementation, of RTP should not be required in WebRTC =
devices that don=92t need it.

Perhaps, but if implementing RTP isn=92t a requirement, that needs to be =
captured somewhere. The rtcweb-overview-11 draft currently mandates RTP =
be implemented.=20


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




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On 20 August 2014 21:16, Muthu Arul Mozhi Perumal <muthu.arul@gmail.com> wrote:
> The elementary entity that needs to perform consent is a WebRTC entity that
> runs untrusted applications -- essentially a WebRTC browser.

That is incorrect.  Any entity that receives peer transport
information from elsewhere needs to perform consent.


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Subject: Re: [rtcweb] WG Last Call for draft-ietf-rtcweb-stun-consent-freshness
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--f46d043c80d6978201050127d894
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On Thu, Aug 21, 2014 at 2:21 PM, Martin Thomson <martin.thomson@gmail.com>
wrote:

> On 20 August 2014 21:16, Muthu Arul Mozhi Perumal <muthu.arul@gmail.com>
> wrote:
> > The elementary entity that needs to perform consent is a WebRTC entity
> that
> > runs untrusted applications -- essentially a WebRTC browser.
>
> That is incorrect.  Any entity that receives peer transport
> information from elsewhere needs to perform consent.
>
>
All entities receive peer transport information from elsewhere, including
gateways running ICE-Lite. Does it mean all of them need to perform consent?

Generally it would be trivial to use any public VoIP service to force it to
send RTP traffic anywhere. It is not like this is a new problem.
_____________
Roman Shpount

--f46d043c80d6978201050127d894
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div>On Thu, Aug 21, 2014 at 2:=
21 PM, Martin Thomson <span dir=3D"ltr">&lt;<a href=3D"mailto:martin.thomso=
n@gmail.com" target=3D"_blank">martin.thomson@gmail.com</a>&gt;</span> wrot=
e:<br></div>
<div class=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"margi=
n:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204=
);border-left-style:solid;padding-left:1ex">On 20 August 2014 21:16, Muthu =
Arul Mozhi Perumal &lt;<a href=3D"mailto:muthu.arul@gmail.com">muthu.arul@g=
mail.com</a>&gt; wrote:<br>

&gt; The elementary entity that needs to perform consent is a WebRTC entity=
 that<br>
&gt; runs untrusted applications -- essentially a WebRTC browser.<br>
<br>
That is incorrect.=C2=A0 Any entity that receives peer transport<br>
information from elsewhere needs to perform consent.<br><br></blockquote><d=
iv>=C2=A0</div><div>All entities receive peer transport information from el=
sewhere, including gateways running ICE-Lite. Does it mean all of them need=
 to perform consent?</div>
<div><br></div><div>Generally it would be trivial to use any public VoIP se=
rvice to force it to send RTP traffic anywhere. It is not like this is a ne=
w problem.</div><div>_____________<br>Roman Shpount</div><div>=C2=A0</div><=
/div>
</div></div>

--f46d043c80d6978201050127d894--


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On 21 August 2014 11:25, Roman Shpount <roman@telurix.com> wrote:
> All entities receive peer transport information from elsewhere, including
> gateways running ICE-Lite. Does it mean all of them need to perform consent?

That's the logical conclusion, yes.


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On 08 Aug 2014, at 22:27, Richard Barnes <rlb@ipv.sx> wrote:

> I have reviewed this document in preparation for IETF LC.  In general, =
it looks good, and I have requested LC.  There are some comments below =
that should be addressed along with IETF LC comments.
> --Richard
Hi Richard,

thank you very much for the review. See my comments in-line.

Best regards
Michael
>=20
>=20
> MAJOR:
>=20
> Section 4. "To avoid glare..."
> The term "glare" is undefined in this document, and will probably be =
unfamiliar to many readers.  Please add it to the Terminology section.
Right. It was introduced by a co-author and I hadn't read it before. =
Maybe the best is to
explicitly state what the problem is. So what about using

   To avoid collisions where both sides try to open a Data Channel with
   the same Stream Identifiers, each side MUST use Streams with either
   even or odd Stream Identifiers when sending a DATA_CHANNEL_OPEN
   message.  When using SCTP over DTLS
   [I-D.ietf-tsvwg-sctp-dtls-encaps], the method used to determine which
   side uses odd or even is based on the underlying DTLS connection
   role: the side acting as the DTLS client MUST use Streams with even
   Stream Identifiers, the side acting as the DTLS server MUST use
   Streams with odd Stream Identifiers.

   Note: There is no attempt to ensure uniqueness for the label; if both
   sides open a Data Channel labeled "x" at the same time, there will be
   two Data Channels labeled "x" - one on an even Stream pair, one on an
   odd pair.

   The protocol field is to ease cross-application interoperation
   ("federation") by identifying the user data being passed with an
   IANA-registered string ('WebSocket Subprotocol Name Registry' defined
   in [RFC6455]), and may be useful for homogeneous applications which
   may create more than one type of Data Channel.  Please note that
   there is also no attempt to ensure uniqueness for the protocol field.

>=20
> Section 4. "When using SCTP over DTLS..."
> This qualification is unnecessary, since data channels always use =
SCTP/DTLS/UDP:
> """
>    The DTLS encapsulation of SCTP packets as described in
>    [I-D.ietf-tsvwg-sctp-dtls-encaps] MUST be used.
> """
> So this should be "Since data channels use the DTLS encapsulation of =
SCTP..."
The text is written in a way that you could use DCEP over SCTP not over =
DTLS.
That is why we have the qualification cited above and also this text in =
the
security considerations:

   This protocol does not provide privacy, integrity or authentication.
   It needs to be used as part of a protocol suite that contains all
   these things.  Such a protocol suite is specified in
   [I-D.ietf-tsvwg-sctp-dtls-encaps].
So is it OK to keep the current text?
>=20
> Section 6. "... all parameters in the DATA_CHANNEL_OPEN message are =
valid"
> The receiver MUST fail if the parameters are invalid, but you never =
define what validation needs to be performed.  For example, is the DCEP =
layer supposed to verify that the Protocol and Label fields are valid =
UTF-8?
All parameters have to be checked (normal input validation of a =
receiver, I would say):
ChannelType: Is it one of the supported values?
Priority: Is it a value the receiver wants to work with? There are four =
values specified.
Reliability Parameter: Is it one of the supported values?
Label Length, Protocol Length: Does the packet contain exactly Label =
Length + Protocol Length + 12 bytes?
Label: Valid UTF-8?
Protocol: Valid UTF-8?

I would expect any receiver to do the above checks as normal input =
validation. If you prefer,
I can add explicit text.
>=20
> Section 10.2. [I-D.ietf-rtcweb-data-channel]
> This reference needs to be normative, since, after all, this is a =
protocol for negotiating data channels.
OK.
>=20
>=20
>=20
> MINOR / EDITORIAL:
>=20
> Section 5.1. [packet diagram]
> The diagram seems to imply that the protocol field starts on a 4-octet =
boundary, even though this is not true.  It might be good if you could =
find a way to reformat the diagram to make this clearer.
If you can point me to an example figure you like, please do. We used =
this kind
of figure in several RFC for SCTP and don't assume that the figure =
implies
that the packet starts on a 4 byte boundary. It just shows 32 bits in a =
row,
starting from the beginning of the packet.
>=20
> Section 5.1. "The suggested value of this field for IANA is 0x03"
> You're defining the initial contents of this registry.  Just say it.  =
Likewise for the same text in Section 5.2.
OK. What about using in Section 5.1

   Message Type: 1 byte (unsigned integer)
      This field holds the IANA defined message type for the
      DATA_CHANNEL_OPEN message.  The value of this field is 0x03 as
      specified in Section 8.2.1.

and in Section 5.2

   Message Type: 1 byte (unsigned integer)
      This field holds the IANA defined message type for the
      DATA_CHANNEL_ACK message.  The value of this field is 0x02 as
      specified in Section 8.2.1.
>=20
> Section 6. "an Stream identifier" -> "a stream identifier"
I would prefer to change it to a Stream Identifier, since we define =
'Stream Identifier' in the Terminology section.
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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Subject: Re: [rtcweb] Some final comments on draft-ietf-rtcweb-data-protocol-07
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On 11 Aug 2014, at 20:42, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:

> I just did another read-through of this draft, and identified a few =
things that I think need to be cleared up:
Hi Paul,

thanks for bringing these points up. See my comments in-line.

Best regards
Michael
>=20
> IIUC the data channel APIs expect that all channels, even those =
defined out of band, have a well defined set of properties. But those =
properties are defined in *this* draft that applies only to channels =
created using DCEP. OTOH, draft-ietf-rtcweb-data-channel-11 is much more =
vague about channel properties.
For out-of-band these properties don't have to be the same in both =
directions. That is why
this ID says:

   The Data Channel Establishment Protocol is a simple, low-overhead way
   to establish bidirectional Data Channels over an SCTP association
   with a consistent set of properties.

   The set of consistent properties includes:

For in-band the properties have to be the same in both directions.
>=20
> ISTM that most of the channel property specification should be moved =
to, or replicated in, draft-ietf-rtcweb-data-channel.
I can list them there, if you think it is clearer. But they don't have =
to be the same in both
directions.
>=20
> And this extends to the definitions for Channel Type, Priority, and =
Reliability since those also are intended to apply for the data channel =
API for channels defined out of band.
In draft-ietf-rtcweb-data-channel, Section 6.1, the reliability is =
described, in Section 6.4 the
priority is described. Maybe we should describe them all together in =
Section 6.4. Would that
address your issue?
>=20
> Alternatively, the data channel document could more extensively cross =
reference this document, or the two documents could be merged into one.
>=20
> One more thing - section 4 says:
>=20
>   Please note that
>   there is also no attempt to resolve protocol glare.
>=20
> I cannot figure out what this means. It is unambiguous which end is =
creating the channel, and that end is *declaring* the protocol to be =
used on it. There is no possibility of glare.
I changed the text to:

   To avoid collisions where both sides try to open a Data Channel with
   the same Stream Identifiers, each side MUST use Streams with either
   even or odd Stream Identifiers when sending a DATA_CHANNEL_OPEN
   message.  When using SCTP over DTLS
   [I-D.ietf-tsvwg-sctp-dtls-encaps], the method used to determine which
   side uses odd or even is based on the underlying DTLS connection
   role: the side acting as the DTLS client MUST use Streams with even
   Stream Identifiers, the side acting as the DTLS server MUST use
   Streams with odd Stream Identifiers.

   Note: There is no attempt to ensure uniqueness for the label; if both
   sides open a Data Channel labeled "x" at the same time, there will be
   two Data Channels labeled "x" - one on an even Stream pair, one on an
   odd pair.

   The protocol field is to ease cross-application interoperation
   ("federation") by identifying the user data being passed with an
   IANA-registered string ('WebSocket Subprotocol Name Registry' defined
   in [RFC6455]), and may be useful for homogeneous applications which
   may create more than one type of Data Channel.  Please note that
   there is also no attempt to ensure uniqueness for the protocol field.

Is this clearer?
>=20
> 	Thanks,
> 	Paul
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


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From: Jeremy Fuller <jeremy.fuller@genband.com>
To: "harald@alvestrand.no" <harald@alvestrand.no>
Thread-Topic: WebRTC gateway - draft-ietf-rtcweb-overview-11.txt
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Hi Harald,

With regard to your addition of the WebRTC gateway in the rtcweb overview. =
I appreciate signalling is out of scope for WebRTC, but as you say in your =
draft alvestrand-rtcweb-gateways-00.txt "any practical gateway needs to dea=
l with signalling". Without getting into too much detail or any implementat=
ion constraints, would it be possible to add a reference to signalling when=
 introducing the WebRTC gateway?

I have suggested some modified text below for your consideration:

Current Text  (section 1):
"A WebRTC gateway is something that mediates media traffic to non-WebRTC en=
tities.  It is like a device, but has certain restrictiions on where it can=
 operate, which means that some of the requirements can be relaxed."

Proposed Text (section 1):
"A WebRTC gateway is something that mediates media traffic, or signalling, =
or both, to non-WebRTC entities.  It is like a device, but has certain rest=
rictions on where it can operate, which means that some of the requirements=
 can be relaxed."

Regards,
Jeremy


-----Original Message-----
Date: Mon, 18 Aug 2014 04:19:16 -0700
From: internet-drafts@ietf.org
To: i-d-announce@ietf.org
Cc: rtcweb@ietf.org
Subject: [rtcweb] I-D Action: draft-ietf-rtcweb-overview-11.txt
Message-ID: <20140818111916.2857.63981.idtracker@ietfa.amsl.com>
Content-Type: text/plain; charset=3D"utf-8"


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

        Title           : Overview: Real Time Protocols for Browser-based A=
pplications
        Author          : Harald T. Alvestrand
=09Filename        : draft-ietf-rtcweb-overview-11.txt
=09Pages           : 22
=09Date            : 2014-08-18

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.

   This document is an Applicability Statement - it does not itself
   specify any protocol, but specifies which other specifications RTCWEB
   compliant implementations are supposed to follow.

   This document is a work item of the RTCWEB working group.


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

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

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-overview-11


Please note that it may take a couple of minutes from the time of submissio=
n 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/



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On 08/21/2014 10:24 PM, Jeremy Fuller wrote:
> Hi Harald,
>
> With regard to your addition of the WebRTC gateway in the rtcweb overview. I appreciate signalling is out of scope for WebRTC, but as you say in your draft alvestrand-rtcweb-gateways-00.txt "any practical gateway needs to deal with signalling". Without getting into too much detail or any implementation constraints, would it be possible to add a reference to signalling when introducing the WebRTC gateway?
>
> I have suggested some modified text below for your consideration:
>
> Current Text  (section 1):
> "A WebRTC gateway is something that mediates media traffic to non-WebRTC entities.  It is like a device, but has certain restrictiions on where it can operate, which means that some of the requirements can be relaxed."
>
> Proposed Text (section 1):
> "A WebRTC gateway is something that mediates media traffic, or signalling, or both, to non-WebRTC entities.  It is like a device, but has certain restrictions on where it can operate, which means that some of the requirements can be relaxed."

This does not work for me. A WebRTC gateway has to handle media, because 
it's the transport of media that WebRTC specifies. A pure signalling 
gateway would just transform the signalling between two WebRTC-capable 
endpoints (WebRTC endpoints?); this is what an application does, not 
what a WebRTC gateway does.

There might be a need to name the class of applications that transform 
signalling; I'm not sure.



>
> Regards,
> Jeremy
>
>
> -----Original Message-----
> Date: Mon, 18 Aug 2014 04:19:16 -0700
> From: internet-drafts@ietf.org
> To: i-d-announce@ietf.org
> Cc: rtcweb@ietf.org
> Subject: [rtcweb] I-D Action: draft-ietf-rtcweb-overview-11.txt
> Message-ID: <20140818111916.2857.63981.idtracker@ietfa.amsl.com>
> Content-Type: text/plain; charset="utf-8"
>
>
> A New Internet-Draft is available from the on-line Internet-Drafts directories.
>   This draft is a work item of the Real-Time Communication in WEB-browsers Working Group of the IETF.
>
>          Title           : Overview: Real Time Protocols for Browser-based Applications
>          Author          : Harald T. Alvestrand
> 	Filename        : draft-ietf-rtcweb-overview-11.txt
> 	Pages           : 22
> 	Date            : 2014-08-18
>
> 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.
>
>     This document is an Applicability Statement - it does not itself
>     specify any protocol, but specifies which other specifications RTCWEB
>     compliant implementations are supposed to follow.
>
>     This document is a work item of the RTCWEB working group.
>
>
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-overview/
>
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-rtcweb-overview-11
>
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-overview-11
>
>
> Please note that it may take a couple of minutes from the time of submission until the htmlized version and diff are available at tools.ietf.org.
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
>


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From: Muthu Arul Mozhi Perumal <muthu.arul@gmail.com>
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Subject: Re: [rtcweb] WG Last Call for draft-ietf-rtcweb-stun-consent-freshness
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--f46d043c07de27e51505013112ef
Content-Type: text/plain; charset=UTF-8

ICE-lite entities don't even perform connectivity checks. Requiring they
perform consent freshness doesn't seem to make sense to me (of course, we
can come up with a new spec ICE-lite-with-consent, but that's a different
problem).

I am not saying you can't spoof a public VoIP service to send RTP anywhere.
Legacy ICE entities perform connectivity checks today and not consent and
consent freshness. Do we want to make them more secure? Sure. It however is
a problem of different scope and need to be solved in MMUSIC or elsewhere,
IMHO.

draft-ietf-rtcweb-stun-consent-freshness is about making the WebRTC browser
more secure. It however allows an RTP endpoint (that also does ICE) to use
the mechanism to make it more secure or compute RTT or carry network
information or whatever. However, requiring every RTP endpoint perform it
seems asking too much.

My take:
WebRTC browser - MUST
WebRTC devide - SHOULD
Other RTP entities (including WebRTC gateway) - MAY

Thoughts?

Muthu

On Fri, Aug 22, 2014 at 1:06 AM, Martin Thomson <martin.thomson@gmail.com>
wrote:

> On 21 August 2014 11:25, Roman Shpount <roman@telurix.com> wrote:
> > All entities receive peer transport information from elsewhere, including
> > gateways running ICE-Lite. Does it mean all of them need to perform
> consent?
>
> That's the logical conclusion, yes.
>

--f46d043c07de27e51505013112ef
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:arial,he=
lvetica,sans-serif;font-size:small">ICE-lite entities don&#39;t even perfor=
m connectivity checks. Requiring they perform consent freshness doesn&#39;t=
 seem to make sense to me (of course, we can come up with a new spec ICE-li=
te-with-consent, but that&#39;s a different problem).</div>
<div class=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-seri=
f;font-size:small"><br></div><div class=3D"gmail_default" style=3D"font-fam=
ily:arial,helvetica,sans-serif;font-size:small">I am not saying you can&#39=
;t spoof a public VoIP service to send RTP anywhere. Legacy ICE entities pe=
rform connectivity checks today and not consent and consent freshness. Do w=
e want to make them more secure? Sure. It however is a problem of different=
 scope and need to be solved in MMUSIC or elsewhere, IMHO.</div>
<div class=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-seri=
f;font-size:small"><br></div><div class=3D"gmail_default" style><font face=
=3D"arial, helvetica, sans-serif">draft-ietf-rtcweb-stun-consent-freshness =
is about making the WebRTC browser more secure. It however allows an RTP en=
dpoint (that also does ICE) to use the mechanism to make it more secure or =
compute RTT or carry network information or whatever. However, requiring ev=
ery RTP endpoint perform it seems asking too much.</font><br>
</div><div class=3D"gmail_default" style><font face=3D"arial, helvetica, sa=
ns-serif"><br></font></div><div class=3D"gmail_default" style><font face=3D=
"arial, helvetica, sans-serif">My take:</font></div><div class=3D"gmail_def=
ault" style>
<font face=3D"arial, helvetica, sans-serif">WebRTC browser - MUST</font></d=
iv><div class=3D"gmail_default" style><font face=3D"arial, helvetica, sans-=
serif">WebRTC devide - SHOULD</font></div><div class=3D"gmail_default" styl=
e><font face=3D"arial, helvetica, sans-serif">Other RTP entities (including=
 WebRTC gateway) - MAY</font></div>
<div class=3D"gmail_default" style><font face=3D"arial, helvetica, sans-ser=
if"><br></font></div><div class=3D"gmail_default" style><font face=3D"arial=
, helvetica, sans-serif">Thoughts?</font></div><div class=3D"gmail_default"=
 style>
<font face=3D"arial, helvetica, sans-serif"><br></font></div><div class=3D"=
gmail_default" style><font face=3D"arial, helvetica, sans-serif">Muthu</fon=
t></div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Fri, A=
ug 22, 2014 at 1:06 AM, Martin Thomson <span dir=3D"ltr">&lt;<a href=3D"mai=
lto:martin.thomson@gmail.com" target=3D"_blank">martin.thomson@gmail.com</a=
>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><div class=3D"">On 21 August 2014 11:25, Roman Shpount &lt=
;<a href=3D"mailto:roman@telurix.com">roman@telurix.com</a>&gt; wrote:<br>

&gt; All entities receive peer transport information from elsewhere, includ=
ing<br>
&gt; gateways running ICE-Lite. Does it mean all of them need to perform co=
nsent?<br>
<br>
</div>That&#39;s the logical conclusion, yes.<br>
</blockquote></div><br></div></div>

--f46d043c07de27e51505013112ef--


From nobody Fri Aug 22 01:34:42 2014
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From: Jeremy Fuller <jeremy.fuller@genband.com>
To: Harald Alvestrand <harald@alvestrand.no>
Thread-Topic: WebRTC gateway - draft-ietf-rtcweb-overview-11.txt
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Subject: Re: [rtcweb] WebRTC gateway - draft-ietf-rtcweb-overview-11.txt
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Hi Harald,=20

It would be good to have something which explains to the reader that there =
exists functionality for transforming signalling. I am reasonably flexible =
on the term used. I proposed associating this functionality with the term W=
ebRTC Gateway to reduce the proliferation of new terms. =20

Regards,
Jeremy

-----Original Message-----
From: Harald Alvestrand [mailto:harald@alvestrand.no]=20
Sent: 21 August 2014 22:16
To: Jeremy Fuller
Cc: rtcweb@ietf.org
Subject: Re: WebRTC gateway - draft-ietf-rtcweb-overview-11.txt

On 08/21/2014 10:24 PM, Jeremy Fuller wrote:
> Hi Harald,
>
> With regard to your addition of the WebRTC gateway in the rtcweb overview=
. I appreciate signalling is out of scope for WebRTC, but as you say in you=
r draft alvestrand-rtcweb-gateways-00.txt "any practical gateway needs to d=
eal with signalling". Without getting into too much detail or any implement=
ation constraints, would it be possible to add a reference to signalling wh=
en introducing the WebRTC gateway?
>
> I have suggested some modified text below for your consideration:
>
> Current Text  (section 1):
> "A WebRTC gateway is something that mediates media traffic to non-WebRTC =
entities.  It is like a device, but has certain restrictiions on where it c=
an operate, which means that some of the requirements can be relaxed."
>
> Proposed Text (section 1):
> "A WebRTC gateway is something that mediates media traffic, or signalling=
, or both, to non-WebRTC entities.  It is like a device, but has certain re=
strictions on where it can operate, which means that some of the requiremen=
ts can be relaxed."

This does not work for me. A WebRTC gateway has to handle media, because it=
's the transport of media that WebRTC specifies. A pure signalling gateway =
would just transform the signalling between two WebRTC-capable endpoints (W=
ebRTC endpoints?); this is what an application does, not what a WebRTC gate=
way does.

There might be a need to name the class of applications that transform sign=
alling; I'm not sure.



>
> Regards,
> Jeremy
>
>
> -----Original Message-----
> Date: Mon, 18 Aug 2014 04:19:16 -0700
> From: internet-drafts@ietf.org
> To: i-d-announce@ietf.org
> Cc: rtcweb@ietf.org
> Subject: [rtcweb] I-D Action: draft-ietf-rtcweb-overview-11.txt
> Message-ID: <20140818111916.2857.63981.idtracker@ietfa.amsl.com>
> Content-Type: text/plain; charset=3D"utf-8"
>
>
> A New Internet-Draft is available from the on-line Internet-Drafts direct=
ories.
>   This draft is a work item of the Real-Time Communication in WEB-browser=
s Working Group of the IETF.
>
>          Title           : Overview: Real Time Protocols for Browser-base=
d Applications
>          Author          : Harald T. Alvestrand
> =09Filename        : draft-ietf-rtcweb-overview-11.txt
> =09Pages           : 22
> =09Date            : 2014-08-18
>
> 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.
>
>     This document is an Applicability Statement - it does not itself
>     specify any protocol, but specifies which other specifications RTCWEB
>     compliant implementations are supposed to follow.
>
>     This document is a work item of the RTCWEB working group.
>
>
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-overview/
>
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-rtcweb-overview-11
>
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-overview-11
>
>
> Please note that it may take a couple of minutes from the time of submiss=
ion 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/
>
>


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--f46d04428e3e35b13c050137efbc
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On Fri, Aug 22, 2014 at 1:25 AM, Muthu Arul Mozhi Perumal <
muthu.arul@gmail.com> wrote:

> draft-ietf-rtcweb-stun-consent-freshness is about making the WebRTC
> browser more secure. It however allows an RTP endpoint (that also does ICE)
> to use the mechanism to make it more secure or compute RTT or carry network
> information or whatever. However, requiring every RTP endpoint perform it
> seems asking too much.
>
> My take:
> WebRTC browser - MUST
> WebRTC devide - SHOULD
> Other RTP entities (including WebRTC gateway) - MAY
>
>
 I would say that all full ICE endpoint interworking with WebRTC MUST
implement consent freshness. ICE-LITE will not implement, but MUST respond
(unless someone defines it, but once you start sending STUN request you
might as well do full ICE, so I do not see much point in it). And to
conclude strongly encourage full ICE implementation for security and other
reasons.
_____________
Roman Shpount

--f46d04428e3e35b13c050137efbc
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div>On Fri, Aug 22, 2014 at 1:=
25 AM, Muthu Arul Mozhi Perumal <span dir=3D"ltr">&lt;<a href=3D"mailto:mut=
hu.arul@gmail.com" target=3D"_blank">muthu.arul@gmail.com</a>&gt;</span> wr=
ote:<br>
</div><div class=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D=
"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,2=
04,204);border-left-style:solid;padding-left:1ex"><div dir=3D"ltr"><div sty=
le=3D"font-family:arial,helvetica,sans-serif;font-size:small">
draft-ietf-rtcweb-stun-consent-freshness is about making the WebRTC browser=
 more secure. It however allows an RTP endpoint (that also does ICE) to use=
 the mechanism to make it more secure or compute RTT or carry network infor=
mation or whatever. However, requiring every RTP endpoint perform it seems =
asking too much.<br>
</div><div><font face=3D"arial, helvetica, sans-serif"><br></font></div><di=
v><font face=3D"arial, helvetica, sans-serif">My take:</font></div><div>
<font face=3D"arial, helvetica, sans-serif">WebRTC browser - MUST</font></d=
iv><div><font face=3D"arial, helvetica, sans-serif">WebRTC devide - SHOULD<=
/font></div><div><font face=3D"arial, helvetica, sans-serif">Other RTP enti=
ties (including WebRTC gateway) - MAY</font></div>

<div><br></div></div></blockquote><div>=C2=A0</div><div>=C2=A0I would say t=
hat all full ICE endpoint interworking with WebRTC MUST implement consent f=
reshness. ICE-LITE will not implement, but MUST respond (unless someone def=
ines it, but once you start sending STUN request you might as well do full =
ICE, so I do not see much point in it). And to conclude strongly encourage =
full ICE implementation for security and other reasons.</div>
<div>_____________<br>Roman Shpount</div><div>=C2=A0</div></div></div></div=
>

--f46d04428e3e35b13c050137efbc--


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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Fri, 22 Aug 2014 17:38:04 +0200
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Subject: [rtcweb] draft-ietf-rtcweb-transports
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Hi,

draft-ietf-rtcweb-transports-06 says:

   If TCP connections are used, RTP framing according to [RFC4571] MUST
   be used, both for the RTP packets and for the DTLS packets used to
   carry data channels.


Not just RTP and DTLS, but also for STUN packets must be framed with
RFC4571. And I do not understand why "DTLS packets used to carry data
channels". In fact all the DTLS packets (also the ones for the
handshake) must be framed as per RFC 4571.


So I suggest to improve the above text as follows:

   If TCP connections are used, RTP framing according to [RFC4571] MUST
   be used for the STUN packets, SRTP packets, SRTCP packets and DTLS
   packets.






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


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Roman Shpount <roman@telurix.com>, Muthu Arul Mozhi Perumal <muthu.arul@gmail.com>
Thread-Topic: [rtcweb] WG Last Call for draft-ietf-rtcweb-stun-consent-freshness
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Date: Fri, 22 Aug 2014 15:59:09 +0000
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Subject: Re: [rtcweb] WG Last Call for draft-ietf-rtcweb-stun-consent-freshness
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Hi,



An ICE-lite endpoint must already today *REPLY* to STUN requests - consent =
freshness does not change that.



But, I don't think an ICE-lite endpoint shall be mandated to *SEND* consent=
 freshness requests.



Regards,



Christer





________________________________
From: rtcweb [rtcweb-bounces@ietf.org] on behalf of Roman Shpount [roman@te=
lurix.com]
Sent: Friday, 22 August 2014 4:37 PM
To: Muthu Arul Mozhi Perumal
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] WG Last Call for draft-ietf-rtcweb-stun-consent-fresh=
ness

On Fri, Aug 22, 2014 at 1:25 AM, Muthu Arul Mozhi Perumal <muthu.arul@gmail=
.com<mailto:muthu.arul@gmail.com>> wrote:
draft-ietf-rtcweb-stun-consent-freshness is about making the WebRTC browser=
 more secure. It however allows an RTP endpoint (that also does ICE) to use=
 the mechanism to make it more secure or compute RTT or carry network infor=
mation or whatever. However, requiring every RTP endpoint perform it seems =
asking too much.

My take:
WebRTC browser - MUST
WebRTC devide - SHOULD
Other RTP entities (including WebRTC gateway) - MAY


 I would say that all full ICE endpoint interworking with WebRTC MUST imple=
ment consent freshness. ICE-LITE will not implement, but MUST respond (unle=
ss someone defines it, but once you start sending STUN request you might as=
 well do full ICE, so I do not see much point in it). And to conclude stron=
gly encourage full ICE implementation for security and other reasons.
_____________
Roman Shpount


--_000_7594FB04B1934943A5C02806D1A2204B1D427B68ESESSMB209erics_
Content-Type: text/html; charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<html dir=3D"ltr">
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Diso-8859-=
1">
<style id=3D"owaParaStyle">P {
	MARGIN-BOTTOM: 0px; MARGIN-TOP: 0px
}
</style>
</head>
<body fPStyle=3D"1" ocsi=3D"0">
<div style=3D"direction: ltr;font-family: Tahoma;color: #000000;font-size: =
10pt;">
<p>Hi,</p>
<p>&nbsp;</p>
<p>An ICE-lite endpoint must already today&nbsp;*REPLY* to STUN requests - =
consent freshness does not change that.</p>
<p>&nbsp;</p>
<p>But, I don't think an ICE-lite endpoint shall be mandated to *SEND* cons=
ent freshness requests.</p>
<p>&nbsp;</p>
<p>Regards,</p>
<p>&nbsp;</p>
<p>Christer</p>
<p>&nbsp;</p>
<p>&nbsp;</p>
<div style=3D"FONT-SIZE: 16px; FONT-FAMILY: Times New Roman; COLOR: #000000=
">
<hr tabindex=3D"-1">
<div id=3D"divRpF303059" style=3D"DIRECTION: ltr"><font color=3D"#000000" s=
ize=3D"2" face=3D"Tahoma"><b>From:</b> rtcweb [rtcweb-bounces@ietf.org] on =
behalf of Roman Shpount [roman@telurix.com]<br>
<b>Sent:</b> Friday, 22 August 2014 4:37 PM<br>
<b>To:</b> Muthu Arul Mozhi Perumal<br>
<b>Cc:</b> rtcweb@ietf.org<br>
<b>Subject:</b> Re: [rtcweb] WG Last Call for draft-ietf-rtcweb-stun-consen=
t-freshness<br>
</font><br>
</div>
<div></div>
<div>
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div>On Fri, Aug 22, 2014 at 1:25 AM, Muthu Arul Mozhi Perumal <span dir=3D=
"ltr">&lt;<a href=3D"mailto:muthu.arul@gmail.com" target=3D"_blank">muthu.a=
rul@gmail.com</a>&gt;</span> wrote:<br>
</div>
<div class=3D"gmail_quote">
<blockquote class=3D"gmail_quote" style=3D"PADDING-LEFT: 1ex; MARGIN: 0px 0=
px 0px 0.8ex; BORDER-LEFT: rgb(204,204,204) 1px solid">
<div dir=3D"ltr">
<div style=3D"FONT-SIZE: small; FONT-FAMILY: arial,helvetica,sans-serif">dr=
aft-ietf-rtcweb-stun-consent-freshness is about making the WebRTC browser m=
ore secure. It however allows an RTP endpoint (that also does ICE) to use t=
he mechanism to make it more secure
 or compute RTT or carry network information or whatever. However, requirin=
g every RTP endpoint perform it seems asking too much.<br>
</div>
<div><font face=3D"arial, helvetica, sans-serif"><br>
</font></div>
<div><font face=3D"arial, helvetica, sans-serif">My take:</font></div>
<div><font face=3D"arial, helvetica, sans-serif">WebRTC browser - MUST</fon=
t></div>
<div><font face=3D"arial, helvetica, sans-serif">WebRTC devide - SHOULD</fo=
nt></div>
<div><font face=3D"arial, helvetica, sans-serif">Other RTP entities (includ=
ing WebRTC gateway) - MAY</font></div>
<div><br>
</div>
</div>
</blockquote>
<div>&nbsp;</div>
<div>&nbsp;I would say that all full ICE endpoint interworking with WebRTC =
MUST implement consent freshness. ICE-LITE will not implement, but MUST res=
pond (unless someone defines it, but once you start sending STUN request yo=
u might as well do full ICE, so I do
 not see much point in it). And to conclude strongly encourage full ICE imp=
lementation for security and other reasons.</div>
<div>_____________<br>
Roman Shpount</div>
<div>&nbsp;</div>
</div>
</div>
</div>
</div>
</div>
</div>
</body>
</html>

--_000_7594FB04B1934943A5C02806D1A2204B1D427B68ESESSMB209erics_--


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From: "Ram Mohan R (rmohanr)" <rmohanr@cisco.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, Roman Shpount <roman@telurix.com>, Muthu Arul Mozhi Perumal <muthu.arul@gmail.com>
Thread-Topic: [rtcweb] WG Last Call for draft-ietf-rtcweb-stun-consent-freshness
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--_000_D01D6B42104F8rmohanrciscocom_
Content-Type: text/plain; charset="Windows-1252"
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Right. I think we had discussed this in the past and added the below text i=
n the draft

" WebRTC endpoints are required to support full ICE as specified in

   section 3.4 of [I-D.ietf-rtcweb-transports<http://tools.ietf.org/html/dr=
aft-ietf-rtcweb-stun-consent-freshness-05#ref-I-D.ietf-rtcweb-transports>].=
  However, when WebRTC
   endpoints interwork with other endpoints that support only ICE-lite
   (e.g. gateways) those endpoints will not generate consent checks, but
   just respond to consent checks they receive=94


Isn=92t this sufficient ?


Ram

From: Christer Holmberg <christer.holmberg@ericsson.com<mailto:christer.hol=
mberg@ericsson.com>>
Date: Friday, 22 August 2014 9:29 pm
To: Roman Shpount <roman@telurix.com<mailto:roman@telurix.com>>, Muthu Arul=
 Mozhi Perumal <muthu.arul@gmail.com<mailto:muthu.arul@gmail.com>>
Cc: "rtcweb@ietf.org<mailto:rtcweb@ietf.org>" <rtcweb@ietf.org<mailto:rtcwe=
b@ietf.org>>
Subject: Re: [rtcweb] WG Last Call for draft-ietf-rtcweb-stun-consent-fresh=
ness


Hi,



An ICE-lite endpoint must already today *REPLY* to STUN requests - consent =
freshness does not change that.



But, I don't think an ICE-lite endpoint shall be mandated to *SEND* consent=
 freshness requests.



Regards,



Christer





________________________________
From: rtcweb [rtcweb-bounces@ietf.org<mailto:rtcweb-bounces@ietf.org>] on b=
ehalf of Roman Shpount [roman@telurix.com<mailto:roman@telurix.com>]
Sent: Friday, 22 August 2014 4:37 PM
To: Muthu Arul Mozhi Perumal
Cc: rtcweb@ietf.org<mailto:rtcweb@ietf.org>
Subject: Re: [rtcweb] WG Last Call for draft-ietf-rtcweb-stun-consent-fresh=
ness

On Fri, Aug 22, 2014 at 1:25 AM, Muthu Arul Mozhi Perumal <muthu.arul@gmail=
.com<mailto:muthu.arul@gmail.com>> wrote:
draft-ietf-rtcweb-stun-consent-freshness is about making the WebRTC browser=
 more secure. It however allows an RTP endpoint (that also does ICE) to use=
 the mechanism to make it more secure or compute RTT or carry network infor=
mation or whatever. However, requiring every RTP endpoint perform it seems =
asking too much.

My take:
WebRTC browser - MUST
WebRTC devide - SHOULD
Other RTP entities (including WebRTC gateway) - MAY


 I would say that all full ICE endpoint interworking with WebRTC MUST imple=
ment consent freshness. ICE-LITE will not implement, but MUST respond (unle=
ss someone defines it, but once you start sending STUN request you might as=
 well do full ICE, so I do not see much point in it). And to conclude stron=
gly encourage full ICE implementation for security and other reasons.
_____________
Roman Shpount


--_000_D01D6B42104F8rmohanrciscocom_
Content-Type: text/html; charset="Windows-1252"
Content-ID: <84613A82A51CED40B7F71E68AE753488@emea.cisco.com>
Content-Transfer-Encoding: quoted-printable

<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3DWindows-1=
252">
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space; color: rgb(0, 0, 0); font-size: 14px; font-fami=
ly: Calibri, sans-serif;">
<div>Right. I think we had discussed this in the past and added the below t=
ext in the draft</div>
<div><br>
</div>
<div>&quot;<span style=3D"font-size: 1em;"> WebRTC endpoints are required t=
o support full ICE as specified in</span></div>
<pre class=3D"newpage" style=3D"font-size: 1em; margin-top: 0px; margin-bot=
tom: 0px; page-break-before: always;">   section 3.4 of [<a href=3D"http://=
tools.ietf.org/html/draft-ietf-rtcweb-stun-consent-freshness-05#ref-I-D.iet=
f-rtcweb-transports">I-D.ietf-rtcweb-transports</a>].  However, when WebRTC
   endpoints interwork with other endpoints that support only ICE-lite
   (e.g. gateways) those endpoints will not generate consent checks, but
   just respond to consent checks they receive=94</pre>
<pre class=3D"newpage" style=3D"font-size: 1em; margin-top: 0px; margin-bot=
tom: 0px; page-break-before: always;"><br></pre>
<pre class=3D"newpage" style=3D"font-size: 1em; margin-top: 0px; margin-bot=
tom: 0px; page-break-before: always;">Isn=92t this sufficient ?</pre>
<pre class=3D"newpage" style=3D"font-size: 1em; margin-top: 0px; margin-bot=
tom: 0px; page-break-before: always;"><br></pre>
<pre class=3D"newpage" style=3D"font-size: 1em; margin-top: 0px; margin-bot=
tom: 0px; page-break-before: always;">Ram</pre>
<div><br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<div style=3D"font-family:Calibri; font-size:11pt; text-align:left; color:b=
lack; BORDER-BOTTOM: medium none; BORDER-LEFT: medium none; PADDING-BOTTOM:=
 0in; PADDING-LEFT: 0in; PADDING-RIGHT: 0in; BORDER-TOP: #b5c4df 1pt solid;=
 BORDER-RIGHT: medium none; PADDING-TOP: 3pt">
<span style=3D"font-weight:bold">From: </span>Christer Holmberg &lt;<a href=
=3D"mailto:christer.holmberg@ericsson.com">christer.holmberg@ericsson.com</=
a>&gt;<br>
<span style=3D"font-weight:bold">Date: </span>Friday, 22 August 2014 9:29 p=
m<br>
<span style=3D"font-weight:bold">To: </span>Roman Shpount &lt;<a href=3D"ma=
ilto:roman@telurix.com">roman@telurix.com</a>&gt;, Muthu Arul Mozhi Perumal=
 &lt;<a href=3D"mailto:muthu.arul@gmail.com">muthu.arul@gmail.com</a>&gt;<b=
r>
<span style=3D"font-weight:bold">Cc: </span>&quot;<a href=3D"mailto:rtcweb@=
ietf.org">rtcweb@ietf.org</a>&quot; &lt;<a href=3D"mailto:rtcweb@ietf.org">=
rtcweb@ietf.org</a>&gt;<br>
<span style=3D"font-weight:bold">Subject: </span>Re: [rtcweb] WG Last Call =
for draft-ietf-rtcweb-stun-consent-freshness<br>
</div>
<div><br>
</div>
<div dir=3D"ltr"><style id=3D"owaParaStyle">P {
	MARGIN-BOTTOM: 0px; MARGIN-TOP: 0px
}
</style>
<div fpstyle=3D"1" ocsi=3D"0">
<div style=3D"direction: ltr;font-family: Tahoma;color: #000000;font-size: =
10pt;">
<p>Hi,</p>
<p>&nbsp;</p>
<p>An ICE-lite endpoint must already today&nbsp;*REPLY* to STUN requests - =
consent freshness does not change that.</p>
<p>&nbsp;</p>
<p>But, I don't think an ICE-lite endpoint shall be mandated to *SEND* cons=
ent freshness requests.</p>
<p>&nbsp;</p>
<p>Regards,</p>
<p>&nbsp;</p>
<p>Christer</p>
<p>&nbsp;</p>
<p>&nbsp;</p>
<div style=3D"FONT-SIZE: 16px; FONT-FAMILY: Times New Roman; COLOR: #000000=
">
<hr tabindex=3D"-1">
<div id=3D"divRpF303059" style=3D"DIRECTION: ltr"><font color=3D"#000000" s=
ize=3D"2" face=3D"Tahoma"><b>From:</b> rtcweb [<a href=3D"mailto:rtcweb-bou=
nces@ietf.org">rtcweb-bounces@ietf.org</a>] on behalf of Roman Shpount [<a =
href=3D"mailto:roman@telurix.com">roman@telurix.com</a>]<br>
<b>Sent:</b> Friday, 22 August 2014 4:37 PM<br>
<b>To:</b> Muthu Arul Mozhi Perumal<br>
<b>Cc:</b> <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<b>Subject:</b> Re: [rtcweb] WG Last Call for draft-ietf-rtcweb-stun-consen=
t-freshness<br>
</font><br>
</div>
<div></div>
<div>
<div dir=3D"ltr">
<div class=3D"gmail_extra">
<div>On Fri, Aug 22, 2014 at 1:25 AM, Muthu Arul Mozhi Perumal <span dir=3D=
"ltr">&lt;<a href=3D"mailto:muthu.arul@gmail.com" target=3D"_blank">muthu.a=
rul@gmail.com</a>&gt;</span> wrote:<br>
</div>
<div class=3D"gmail_quote">
<blockquote class=3D"gmail_quote" style=3D"PADDING-LEFT: 1ex; MARGIN: 0px 0=
px 0px 0.8ex; BORDER-LEFT: rgb(204,204,204) 1px solid">
<div dir=3D"ltr">
<div style=3D"FONT-SIZE: small; FONT-FAMILY: arial,helvetica,sans-serif">dr=
aft-ietf-rtcweb-stun-consent-freshness is about making the WebRTC browser m=
ore secure. It however allows an RTP endpoint (that also does ICE) to use t=
he mechanism to make it more secure
 or compute RTT or carry network information or whatever. However, requirin=
g every RTP endpoint perform it seems asking too much.<br>
</div>
<div><font face=3D"arial,helvetica,sans-serif"><br>
</font></div>
<div><font face=3D"arial,helvetica,sans-serif">My take:</font></div>
<div><font face=3D"arial,helvetica,sans-serif">WebRTC browser - MUST</font>=
</div>
<div><font face=3D"arial,helvetica,sans-serif">WebRTC devide - SHOULD</font=
></div>
<div><font face=3D"arial,helvetica,sans-serif">Other RTP entities (includin=
g WebRTC gateway) - MAY</font></div>
<div><br>
</div>
</div>
</blockquote>
<div>&nbsp;</div>
<div>&nbsp;I would say that all full ICE endpoint interworking with WebRTC =
MUST implement consent freshness. ICE-LITE will not implement, but MUST res=
pond (unless someone defines it, but once you start sending STUN request yo=
u might as well do full ICE, so I do
 not see much point in it). And to conclude strongly encourage full ICE imp=
lementation for security and other reasons.</div>
<div>_____________<br>
Roman Shpount</div>
<div>&nbsp;</div>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</span>
</body>
</html>

--_000_D01D6B42104F8rmohanrciscocom_--


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Date: Fri, 22 Aug 2014 12:39:25 -0400
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From: Roman Shpount <roman@telurix.com>
To: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
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Subject: Re: [rtcweb] WG Last Call for draft-ietf-rtcweb-stun-consent-freshness
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On Fri, Aug 22, 2014 at 12:21 PM, Makaraju, Maridi Raju (Raju) <
Raju.Makaraju@alcatel-lucent.com> wrote:

>  >WebRTC browser - MUST
>
> >WebRTC devide - SHOULD
>
> >Other RTP entities (including WebRTC gateway) - MAY
>
>
What I still do not understand, is why a full ICE end point would not send
consent messages? What is the reason for this? The security and operational
benefits of this are clear and the incremental work in implementing consent
is minimal.

Also, in what circumstances would ICE-lite end point ever send consent?
Where is this even defined? Also, if you are implementing sending STUN
messages, why would you not implement the full ICE end point? Once
implementation of sending and processing responses to STUN consent messages
is done, incremental work to implement full ICE is minimal.

And finally, what does any of this have to do with type of WebRTC end
point? Browsers and devices must implement full ICE and thus must implement
consent. Other RTP endpoints, may implement ICE-lite and skip consent, but
ideally should implement full ICE and consent. I also believe all of this
is already in the specification.
_____________
Roman Shpount

--e89a8f643366e4d87405013a7a59
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On F=
ri, Aug 22, 2014 at 12:21 PM, Makaraju, Maridi Raju (Raju) <span dir=3D"ltr=
">&lt;<a href=3D"mailto:Raju.Makaraju@alcatel-lucent.com" target=3D"_blank"=
>Raju.Makaraju@alcatel-lucent.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">





<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div><div class=3D"">
<p class=3D"MsoNormal"><span style=3D"font-family:Arial,sans-serif">&gt;Web=
RTC browser - MUST</span><u></u><u></u></p>
<p class=3D"MsoNormal"><span style=3D"font-family:Arial,sans-serif">&gt;Web=
RTC devide - SHOULD</span><u></u><u></u></p>
<p class=3D"MsoNormal"><span style=3D"font-family:Arial,sans-serif">&gt;Oth=
er RTP entities (including WebRTC gateway) - MAY</span><u></u><u></u></p>
<p class=3D"MsoNormal"></p></div></div></div></blockquote><div><br></div><d=
iv>What I still do not understand, is why a full ICE end point would not se=
nd consent messages? What is the reason for this? The security and operatio=
nal benefits of this are clear and the incremental work in implementing con=
sent is minimal.</div>
<div><br></div><div>Also, in what circumstances would ICE-lite end point ev=
er send consent? Where is this even defined? Also, if you are implementing =
sending STUN messages, why would you not implement the full ICE end point? =
Once implementation of sending and processing responses to STUN consent mes=
sages is done, incremental work to implement full ICE is minimal.</div>
<div><br></div><div>And finally, what does any of this have to do with type=
 of WebRTC end point? Browsers and devices must implement full ICE and thus=
 must implement consent. Other RTP endpoints, may implement ICE-lite and sk=
ip consent, but ideally should implement full ICE and consent. I also belie=
ve all of this is already in the specification.</div>
<div>_____________<br>Roman Shpount</div></div></div></div>

--e89a8f643366e4d87405013a7a59--


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From: Martin Thomson <martin.thomson@gmail.com>
To: Muthu Arul Mozhi Perumal <muthu.arul@gmail.com>
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Subject: Re: [rtcweb] WG Last Call for draft-ietf-rtcweb-stun-consent-freshness
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On 21 August 2014 22:25, Muthu Arul Mozhi Perumal <muthu.arul@gmail.com> wrote:
> ICE-lite entities don't even perform connectivity checks. Requiring they
> perform consent freshness doesn't seem to make sense to me (of course, we
> can come up with a new spec ICE-lite-with-consent, but that's a different
> problem).

I never suggested that.  What I suggested was that they SHOULD be
performing consent checks.  The fact that they don't is what I would
term a "legacy requirement".  I don't think that we should be
extending this exception to any other entities.

The only reason that ICE-lite gets away with not sending checks is
that it trusts both the signaling entities and its peers.  That's
dangerous.  We interoperate with ICE-lite only because doing otherwise
would exclude more entities than we are willing to exclude.

The current text is fine.  It opens a minimal hole that acknowledges
the existence of the bad actors, but doesn't attempt to bestow any
kind of validity or legitimacy on them.


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From nobody Fri Aug 22 13:34:01 2014
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From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [rtcweb] WebRTC Device: Lowering the bar, or inventing a new term?
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On 8/21/14 7:42 AM, Colin Perkins wrote:
> On 20 Aug 2014, at 15:24, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
>> On 8/20/14 6:47 AM, Colin Perkins wrote:
>>> On 20 Aug 2014, at 11:38, Harald Alvestrand <harald@alvestrand.no> wrote:
>>>> On 08/20/2014 11:46 AM, Colin Perkins wrote:
>>>>> On 19 Aug 2014, at 15:00, Harald Alvestrand <harald@alvestrand.no> wrote:
>>>>>> On 08/19/2014 03:38 PM, Jim Barnett wrote:
>>>>>>> I agree.  We need to define a) WebRTC browsers, which support the full protocol requirements and the API, b) WebRTC Endpoints/Devices, which support some minimum protocol set sufficient for establishing connectivity.  Is there also a need to define c) Full WebRTC Devices, which support more than the minimum protocol set?  What would be the use of definition c?
>>>>>> An endpoint (I kind of start to like this concept/name) would be conformant as long as it was able to successfully negotiate away use of anything that makes life difficult.
>>>>>>
>>>>>> It would be required to support DTLS-SRTP key handshakes and respond to ICE connectivity checks, plus whatever is MUST use in -rtp-usage (scanning, I'm not coming up with much - it's mostly MUST respond to, MUST implement (but not required to use) stuff). The RTP circuit breaker is also a MUST implement, and since it's a single-ended thing, it makes no sense if it isn't also MUST use.
>>>>>>
>>>>>> Scanning, I’m not even sure a WebRTC endpoint is required to send RTCP - but I think it should be.
>>>>> “RTCP is a fundamental and integral part of RTP, and MUST be implemented in all WebRTC applications.” rtp-usage-16 section 4.1
>>>>
>>>> Yep, I thought we wanted that too, but then I started to scan for mandatory-to-implement vs mandatory-to-use, and started being unsure; is it permitted to configure a WebRTC device to set RTCP-bandwidth to zero (the traditional means of turning off RTCP in a standards conformant manner, I think)?
>>>>
>>>> Perhaps we should add a requirement on minimum RTCP bandwidth?
>>>
>>> I wouldn’t know what minimum RTCP bandwidth to specify, but could certainly change the rtp-usage draft to say “…MUST be implemented and used…” if that would help?
>>
>> That is fine *if RTP is being used*.
>
> I am suggesting adding this to the RTP usage draft…
>
>> Use, or even implementation, of RTP should not be required in WebRTC devices that don’t need it.
>
> Perhaps, but if implementing RTP isn’t a requirement, that needs to be captured somewhere. The rtcweb-overview-11 draft currently mandates RTP be implemented.

It is appropriate for the overview to mandate RTP support *for browsers*.

But the current discussion is about how to frame requirements for 
compliant non-browser devices.

If there was an intent to define some non-browser *platform* for webrtc 
applications then it would probably be appropriate to mandate RTP 
support for them too. (AFAIK there is no plan to define such a platform.)

But other devices that only need data channels shouldn't be required to 
implement RTP.

	Thanks,
	Paul


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From: "Parthasarathi R" <partha@parthasarathi.co.in>
To: "'Ram Mohan R \(rmohanr\)'" <rmohanr@cisco.com>, "'Muthu Arul Mozhi Perumal'" <muthu.arul@gmail.com>
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Subject: Re: [rtcweb] WG Last Call for	draft-ietf-rtcweb-stun-consent-freshness
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Hi Ram/Muthu,

 

There is a terminology variation here. As per the current overview document,
WebRTC device supports full WebRTC protocol requirement which includes
consent-check. In case consent check is excluded from WebRTC device,  the
new definition has to be provided for "WebRTC device" in -overview draft. I
don't think that WebRTC device definition has to be changed now.

 

IMO, WebRTC device & WebRTC browser MUST support consent check. As discussed
in another mail thread, there is an need to define "WebRTC endpoint" which
relaxes the lot of WebRTC protocol requirement based on the specific
deployment like Enterprise, LTE EPC. "WebRTC Endpoint" & "WebRTC Gateway"
MAY support consent check.

 

I have highlighted Full ICE as a criteria for "WebRTC Endpoint" & "WebRTC
Gateway" in
http://www.ietf.org/mail-archive/web/rtcweb/current/msg12938.html as it
helps the implementers to decide whether consent check has to be implemented
in their "WebRTC Endpoint"/ "WebRTC Gateway" or not. As "WebRTC Endpoint" is
yet to defined and "WebRTC gateway" is just now proposed in -11 overview
draft,  I have proposed the text as

 

"Full ICE supporting WebRTC entities  MUST support consent freshness." 

 

Please let me your opinion on the same. 

 

Also, I agree with Muthu's statement to indicates the other requirement for
WebRTC entities to include consent check are:

"to use the mechanism to make it more secure or compute RTT or carry network
information" 

 

 

Thanks

Partha

 

 

 

From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Ram Mohan R
(rmohanr)
Sent: Friday, August 22, 2014 9:48 PM
To: Christer Holmberg; Roman Shpount; Muthu Arul Mozhi Perumal
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] WG Last Call for
draft-ietf-rtcweb-stun-consent-freshness

 

Right. I think we had discussed this in the past and added the below text in
the draft

 

" WebRTC endpoints are required to support full ICE as specified in

   section 3.4 of [I-D.ietf-rtcweb-transports
<http://tools.ietf.org/html/draft-ietf-rtcweb-stun-consent-freshness-05#ref-
I-D.ietf-rtcweb-transports> ].  However, when WebRTC
   endpoints interwork with other endpoints that support only ICE-lite
   (e.g. gateways) those endpoints will not generate consent checks, but
   just respond to consent checks they receive"


Isn't this sufficient ?


Ram

 

From: Christer Holmberg <christer.holmberg@ericsson.com>
Date: Friday, 22 August 2014 9:29 pm
To: Roman Shpount <roman@telurix.com>, Muthu Arul Mozhi Perumal
<muthu.arul@gmail.com>
Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] WG Last Call for
draft-ietf-rtcweb-stun-consent-freshness

 

Hi,

 

An ICE-lite endpoint must already today *REPLY* to STUN requests - consent
freshness does not change that.

 

But, I don't think an ICE-lite endpoint shall be mandated to *SEND* consent
freshness requests.

 

Regards,

 

Christer

 

 

  _____  

From: rtcweb [rtcweb-bounces@ietf.org] on behalf of Roman Shpount
[roman@telurix.com]
Sent: Friday, 22 August 2014 4:37 PM
To: Muthu Arul Mozhi Perumal
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] WG Last Call for
draft-ietf-rtcweb-stun-consent-freshness

On Fri, Aug 22, 2014 at 1:25 AM, Muthu Arul Mozhi Perumal
<muthu.arul@gmail.com> wrote:

draft-ietf-rtcweb-stun-consent-freshness is about making the WebRTC browser
more secure. It however allows an RTP endpoint (that also does ICE) to use
the mechanism to make it more secure or compute RTT or carry network
information or whatever. However, requiring every RTP endpoint perform it
seems asking too much.

 

My take:

WebRTC browser - MUST

WebRTC devide - SHOULD

Other RTP entities (including WebRTC gateway) - MAY

 

 

 I would say that all full ICE endpoint interworking with WebRTC MUST
implement consent freshness. ICE-LITE will not implement, but MUST respond
(unless someone defines it, but once you start sending STUN request you
might as well do full ICE, so I do not see much point in it). And to
conclude strongly encourage full ICE implementation for security and other
reasons.

_____________
Roman Shpount

 


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<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Hi Ram/Muthu,<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>There is a terminology variation here. As per the current
overview document, WebRTC device supports full WebRTC protocol =
requirement which
includes consent-check. In case consent check is excluded from WebRTC =
device, &nbsp;the
new definition has to be provided for &#8220;WebRTC device&#8221; in =
&#8211;overview
draft. I don&#8217;t think that WebRTC device definition has to be =
changed now.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>IMO, WebRTC device &amp; WebRTC browser MUST support =
consent
check. As discussed in another mail thread, there is an need to define =
&#8220;WebRTC
endpoint&#8221; which relaxes the lot of WebRTC protocol requirement =
based on
the specific deployment like Enterprise, LTE EPC. &#8220;WebRTC =
Endpoint&#8221;
&amp; &#8220;WebRTC Gateway&#8221; MAY support consent =
check.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>I have highlighted Full ICE as a criteria for =
&#8220;WebRTC
Endpoint&#8221; &amp; &#8220;WebRTC Gateway&#8221; in <a
href=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/msg12938.html=
">http://www.ietf.org/mail-archive/web/rtcweb/current/msg12938.html</a>
as it helps the implementers to decide whether consent check has to be
implemented in their &#8220;WebRTC Endpoint&#8221;/ &#8220;WebRTC =
Gateway&#8221;
or not. As &#8220;WebRTC Endpoint&#8221; is yet to defined and =
&#8220;WebRTC
gateway&#8221; is just now proposed in -11 overview draft, &nbsp;I have =
proposed
the text as<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>&#8220;Full ICE supporting WebRTC entities &nbsp;MUST =
support
consent freshness.&#8221; <o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Please let me your opinion on the same. =
<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Also, I agree with Muthu&#8217;s statement to indicates =
the
other requirement for WebRTC entities to include consent check =
are:<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>&#8220;</span><span =
style=3D'font-family:"Arial","sans-serif";
color:black'>to use the mechanism to make it more secure or compute RTT =
or
carry network information&#8221;</span><span style=3D'font-size:11.0pt;
font-family:"Calibri","sans-serif";color:#1F497D'> =
<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Thanks<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Partha<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<div style=3D'border:none;border-left:solid blue 1.5pt;padding:0in 0in =
0in 4.0pt'>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt =
0in 0in 0in'>

<p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> rtcweb
[mailto:rtcweb-bounces@ietf.org] <b>On Behalf Of </b>Ram Mohan R =
(rmohanr)<br>
<b>Sent:</b> Friday, August 22, 2014 9:48 PM<br>
<b>To:</b> Christer Holmberg; Roman Shpount; Muthu Arul Mozhi =
Perumal<br>
<b>Cc:</b> rtcweb@ietf.org<br>
<b>Subject:</b> Re: [rtcweb] WG Last Call for
draft-ietf-rtcweb-stun-consent-freshness<o:p></o:p></span></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.5pt;font-family:"Calibri","sans-serif";
color:black'>Right. I think we had discussed this in the past and added =
the
below text in the draft<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.5pt;font-family:"Calibri","sans-serif";
color:black'><o:p>&nbsp;</o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.5pt;font-family:"Calibri","sans-serif";
color:black'>&quot; WebRTC endpoints are required to support full ICE as
specified in<o:p></o:p></span></p>

</div>

<pre style=3D'page-break-before:always'><span =
style=3D'font-size:10.5pt;color:black'>&nbsp;&nbsp; section 3.4 of [<a
href=3D"http://tools.ietf.org/html/draft-ietf-rtcweb-stun-consent-freshne=
ss-05#ref-I-D.ietf-rtcweb-transports">I-D.ietf-rtcweb-transports</a>].&nb=
sp; However, when WebRTC<o:p></o:p></span></pre><pre
style=3D'page-break-before:always'><span =
style=3D'font-size:10.5pt;color:black'>&nbsp;&nbsp; endpoints interwork =
with other endpoints that support only =
ICE-lite<o:p></o:p></span></pre><pre
style=3D'page-break-before:always'><span =
style=3D'font-size:10.5pt;color:black'>&nbsp;&nbsp; (e.g. gateways) =
those endpoints will not generate consent checks, =
but<o:p></o:p></span></pre><pre
style=3D'page-break-before:always'><span =
style=3D'font-size:10.5pt;color:black'>&nbsp;&nbsp; just respond to =
consent checks they receive&#8221;<o:p></o:p></span></pre><span
style=3D'font-size:10.5pt;font-family:"Courier New";color:black'><br =
clear=3Dall
style=3D'page-break-before:always'>
</span><pre style=3D'page-break-before:always'><span =
style=3D'font-size:10.5pt;
color:black'>Isn&#8217;t this sufficient ?<o:p></o:p></span></pre><span
style=3D'font-size:10.5pt;font-family:"Courier New";color:black'><br =
clear=3Dall
style=3D'page-break-before:always'>
</span><pre style=3D'page-break-before:always'><span =
style=3D'font-size:10.5pt;
color:black'>Ram<o:p></o:p></span></pre>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.5pt;font-family:"Calibri","sans-serif";
color:black'><o:p>&nbsp;</o:p></span></p>

</div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt =
0in 0in 0in'>

<p class=3DMsoNormal><b><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:black'>From: </span></b><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:black'>Christer Holmberg &lt;<a
href=3D"mailto:christer.holmberg@ericsson.com">christer.holmberg@ericsson=
.com</a>&gt;<br>
<b>Date: </b>Friday, 22 August 2014 9:29 pm<br>
<b>To: </b>Roman Shpount &lt;<a =
href=3D"mailto:roman@telurix.com">roman@telurix.com</a>&gt;,
Muthu Arul Mozhi Perumal &lt;<a =
href=3D"mailto:muthu.arul@gmail.com">muthu.arul@gmail.com</a>&gt;<br>
<b>Cc: </b>&quot;<a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>&quot;
&lt;<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>&gt;<br>
<b>Subject: </b>Re: [rtcweb] WG Last Call for
draft-ietf-rtcweb-stun-consent-freshness<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.5pt;font-family:"Calibri","sans-serif";
color:black'><o:p>&nbsp;</o:p></span></p>

</div>

<div>

<div>

<div>

<p><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:black'>=
Hi,<o:p></o:p></span></p>

<p><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:black'>=
&nbsp;<o:p></o:p></span></p>

<p><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:black'>=
An
ICE-lite endpoint must already today&nbsp;*REPLY* to STUN requests - =
consent
freshness does not change that.<o:p></o:p></span></p>

<p><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:black'>=
&nbsp;<o:p></o:p></span></p>

<p><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:black'>=
But,
I don't think an ICE-lite endpoint shall be mandated to *SEND* consent
freshness requests.<o:p></o:p></span></p>

<p><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:black'>=
&nbsp;<o:p></o:p></span></p>

<p><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:black'>=
Regards,<o:p></o:p></span></p>

<p><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:black'>=
&nbsp;<o:p></o:p></span></p>

<p><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:black'>=
Christer<o:p></o:p></span></p>

<p><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:black'>=
&nbsp;<o:p></o:p></span></p>

<p><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:black'>=
&nbsp;<o:p></o:p></span></p>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><span
style=3D'color:black'>

<hr size=3D2 width=3D"100%" align=3Dcenter>

</span></div>

<div id=3DdivRpF303059>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><b><span =
style=3D'font-size:10.0pt;
font-family:"Tahoma","sans-serif";color:black'>From:</span></b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:black'>=
 rtcweb
[<a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a>] =
on
behalf of Roman Shpount [<a =
href=3D"mailto:roman@telurix.com">roman@telurix.com</a>]<br>
<b>Sent:</b> Friday, 22 August 2014 4:37 PM<br>
<b>To:</b> Muthu Arul Mozhi Perumal<br>
<b>Cc:</b> <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<b>Subject:</b> Re: [rtcweb] WG Last Call for =
draft-ietf-rtcweb-stun-consent-freshness</span><span
style=3D'color:black'><o:p></o:p></span></p>

</div>

<div>

<div>

<div>

<div>

<p class=3DMsoNormal><span style=3D'color:black'>On Fri, Aug 22, 2014 at =
1:25 AM,
Muthu Arul Mozhi Perumal &lt;<a href=3D"mailto:muthu.arul@gmail.com"
target=3D"_blank">muthu.arul@gmail.com</a>&gt; =
wrote:<o:p></o:p></span></p>

</div>

<div>

<blockquote style=3D'border:none;border-left:solid #CCCCCC =
1.0pt;padding:0in 0in 0in 6.0pt;
margin-left:4.8pt;margin-right:0in'>

<div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif";color:black'>draft-ietf-rtcweb-=
stun-consent-freshness
is about making the WebRTC browser more secure. It however allows an RTP
endpoint (that also does ICE) to use the mechanism to make it more =
secure or
compute RTT or carry network information or whatever. However, requiring =
every
RTP endpoint perform it seems asking too much.<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'color:black'><o:p>&nbsp;</o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif";color:black'>My
take:</span><span style=3D'color:black'><o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif";color:black'>WebRTC
browser - MUST</span><span style=3D'color:black'><o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif";color:black'>WebRTC
devide - SHOULD</span><span style=3D'color:black'><o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-family:"Arial","sans-serif";color:black'>Other
RTP entities (including WebRTC gateway) - MAY</span><span =
style=3D'color:black'><o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'color:black'><o:p>&nbsp;</o:p></span></p>

</div>

</div>

</blockquote>

<div>

<p class=3DMsoNormal><span =
style=3D'color:black'>&nbsp;<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span style=3D'color:black'>&nbsp;I would say that =
all full
ICE endpoint interworking with WebRTC MUST implement consent freshness.
ICE-LITE will not implement, but MUST respond (unless someone defines =
it, but
once you start sending STUN request you might as well do full ICE, so I =
do not
see much point in it). And to conclude strongly encourage full ICE
implementation for security and other reasons.<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span style=3D'color:black'>_____________<br>
Roman Shpount<o:p></o:p></span></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'color:black'>&nbsp;<o:p></o:p></span></p>

</div>

</div>

</div>

</div>

</div>

</div>

</div>

</div>

</div>

</div>

</div>

</body>

</html>

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Subject: Re: [rtcweb] WebRTC Device: Lowering the bar, or inventing a new term?
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Paul,

I agree with you that WebRTC Endpoint MAY implement data channel only and
not mandatory to implement RTP.

But I wish to differentiate between WebRTC Endpoint & WebRTC device wherein
WebRTC device MUST support RTP & data channel as it is full compliance to
WebRTC protocol requirements. Please let me know your opinion on the same.

Thanks
Partha

> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Paul Kyzivat
> Sent: Saturday, August 23, 2014 2:04 AM
> To: Colin Perkins
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] WebRTC Device: Lowering the bar, or inventing a
> new term?
> 
> On 8/21/14 7:42 AM, Colin Perkins wrote:
> > On 20 Aug 2014, at 15:24, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
> >> On 8/20/14 6:47 AM, Colin Perkins wrote:
> >>> On 20 Aug 2014, at 11:38, Harald Alvestrand <harald@alvestrand.no>
> wrote:
> >>>> On 08/20/2014 11:46 AM, Colin Perkins wrote:
> >>>>> On 19 Aug 2014, at 15:00, Harald Alvestrand
> <harald@alvestrand.no> wrote:
> >>>>>> On 08/19/2014 03:38 PM, Jim Barnett wrote:
> >>>>>>> I agree.  We need to define a) WebRTC browsers, which support
> the full protocol requirements and the API, b) WebRTC
> Endpoints/Devices, which support some minimum protocol set sufficient
> for establishing connectivity.  Is there also a need to define c) Full
> WebRTC Devices, which support more than the minimum protocol set?  What
> would be the use of definition c?
> >>>>>> An endpoint (I kind of start to like this concept/name) would be
> conformant as long as it was able to successfully negotiate away use of
> anything that makes life difficult.
> >>>>>>
> >>>>>> It would be required to support DTLS-SRTP key handshakes and
> respond to ICE connectivity checks, plus whatever is MUST use in -rtp-
> usage (scanning, I'm not coming up with much - it's mostly MUST respond
> to, MUST implement (but not required to use) stuff). The RTP circuit
> breaker is also a MUST implement, and since it's a single-ended thing,
> it makes no sense if it isn't also MUST use.
> >>>>>>
> >>>>>> Scanning, I'm not even sure a WebRTC endpoint is required to
> send RTCP - but I think it should be.
> >>>>> "RTCP is a fundamental and integral part of RTP, and MUST be
> implemented in all WebRTC applications." rtp-usage-16 section 4.1
> >>>>
> >>>> Yep, I thought we wanted that too, but then I started to scan for
> mandatory-to-implement vs mandatory-to-use, and started being unsure;
> is it permitted to configure a WebRTC device to set RTCP-bandwidth to
> zero (the traditional means of turning off RTCP in a standards
> conformant manner, I think)?
> >>>>
> >>>> Perhaps we should add a requirement on minimum RTCP bandwidth?
> >>>
> >>> I wouldn't know what minimum RTCP bandwidth to specify, but could
> certainly change the rtp-usage draft to say ".MUST be implemented and
> used." if that would help?
> >>
> >> That is fine *if RTP is being used*.
> >
> > I am suggesting adding this to the RTP usage draft.
> >
> >> Use, or even implementation, of RTP should not be required in WebRTC
> devices that don't need it.
> >
> > Perhaps, but if implementing RTP isn't a requirement, that needs to
> be captured somewhere. The rtcweb-overview-11 draft currently mandates
> RTP be implemented.
> 
> It is appropriate for the overview to mandate RTP support *for
> browsers*.
> 
> But the current discussion is about how to frame requirements for
> compliant non-browser devices.
> 
> If there was an intent to define some non-browser *platform* for webrtc
> applications then it would probably be appropriate to mandate RTP
> support for them too. (AFAIK there is no plan to define such a
> platform.)
> 
> But other devices that only need data channels shouldn't be required to
> implement RTP.
> 
> 	Thanks,
> 	Paul
> 
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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I've added some diagrams and made minor wording changes in response to
suggestions from Philip Hancke.

---------- Forwarded message ----------
From: <internet-drafts@ietf.org>
Date: Fri, Aug 22, 2014 at 6:54 PM
Subject: New Version Notification for draft-schwartz-rtcweb-return-01.txt
To: "Benjamin M. Schwartz" <bemasc@webrtc.org>



A new version of I-D, draft-schwartz-rtcweb-return-01.txt
has been successfully submitted by Benjamin M. Schwartz and posted to the
IETF repository.

Name:           draft-schwartz-rtcweb-return
Revision:       01
Title:          Recursively Encapsulated TURN (RETURN) for Connectivity and
Privacy in WebRTC
Document date:  2014-08-22
Group:          Individual Submission
Pages:          12
URL:
http://www.ietf.org/internet-drafts/draft-schwartz-rtcweb-return-01.txt
Status:
https://datatracker.ietf.org/doc/draft-schwartz-rtcweb-return/
Htmlized:       http://tools.ietf.org/html/draft-schwartz-rtcweb-return-01
Diff:
http://www.ietf.org/rfcdiff?url2=draft-schwartz-rtcweb-return-01

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




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

The IETF Secretariat

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<div dir=3D"ltr">I&#39;ve added some diagrams and made minor wording change=
s in response to suggestions from Philip Hancke.<br><br><div class=3D"gmail=
_quote">---------- Forwarded message ----------<br>From: <b class=3D"gmail_=
sendername"></b> <span dir=3D"ltr">&lt;<a href=3D"mailto:internet-drafts@ie=
tf.org">internet-drafts@ietf.org</a>&gt;</span><br>
Date: Fri, Aug 22, 2014 at 6:54 PM<br>Subject: New Version Notification for=
 draft-schwartz-rtcweb-return-01.txt<br>To: &quot;Benjamin M. Schwartz&quot=
; &lt;<a href=3D"mailto:bemasc@webrtc.org">bemasc@webrtc.org</a>&gt;<br><br=
>
<br><br>
A new version of I-D, draft-schwartz-rtcweb-return-01.txt<br>
has been successfully submitted by Benjamin M. Schwartz and posted to the<b=
r>
IETF repository.<br>
<br>
Name:=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0draft-schwartz-rtcweb-return<=
br>
Revision:=C2=A0 =C2=A0 =C2=A0 =C2=A001<br>
Title:=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 Recursively Encapsulated TURN (RET=
URN) for Connectivity and Privacy in WebRTC<br>
Document date:=C2=A0 2014-08-22<br>
Group:=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 Individual Submission<br>
Pages:=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 12<br>
URL:=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 <a href=3D"http://www.ietf.or=
g/internet-drafts/draft-schwartz-rtcweb-return-01.txt" target=3D"_blank">ht=
tp://www.ietf.org/internet-drafts/draft-schwartz-rtcweb-return-01.txt</a><b=
r>
Status:=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0<a href=3D"https://datatracker.iet=
f.org/doc/draft-schwartz-rtcweb-return/" target=3D"_blank">https://datatrac=
ker.ietf.org/doc/draft-schwartz-rtcweb-return/</a><br>
Htmlized:=C2=A0 =C2=A0 =C2=A0 =C2=A0<a href=3D"http://tools.ietf.org/html/d=
raft-schwartz-rtcweb-return-01" target=3D"_blank">http://tools.ietf.org/htm=
l/draft-schwartz-rtcweb-return-01</a><br>
Diff:=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0<a href=3D"http://www.ietf.or=
g/rfcdiff?url2=3Ddraft-schwartz-rtcweb-return-01" target=3D"_blank">http://=
www.ietf.org/rfcdiff?url2=3Ddraft-schwartz-rtcweb-return-01</a><br>
<br>
Abstract:<br>
=C2=A0 =C2=A0In the context of WebRTC, the concept of a local TURN proxy ha=
s been<br>
=C2=A0 =C2=A0suggested, but not reviewed in detail.=C2=A0 WebRTC applicatio=
ns are<br>
=C2=A0 =C2=A0already using TURN to enhance connectivity and privacy.=C2=A0 =
This<br>
=C2=A0 =C2=A0document explains how local TURN proxies and WebRTC applicatio=
ns can<br>
=C2=A0 =C2=A0work together.<br>
<br>
<br>
<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" target=3D"_blank">tools.ietf.org</a>.<br>
<br>
The IETF Secretariat<br>
<br>
</div><br></div>

--001a1135f63c3feaff05013fba96--


From nobody Fri Aug 22 16:53:28 2014
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References: <3DF911000D80F6459928AC50D3D742F601A509@gbplmail02.genband.com> <53F66180.6050308@alvestrand.no> <3DF911000D80F6459928AC50D3D742F601AA20@gbplmail02.genband.com>
From: Barry Dingle <btdingle@gmail.com>
Date: Sat, 23 Aug 2014 09:53:02 +1000
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--f46d043749cbd24c460501408af4
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Hi Harald,

I see Jeremy's point.

Gateways between a WebRTC environment and non-WebRTC environments need to
convert BOTH media and signalling. Media-only Gateways are usually referred
to as 'Media Gateways'. In draft-alvestrand-rtcweb-gateways-00 Section 1
paragraph 2, the description of 'the Gateway' refers only to the 'media'.

As the document later refers to the Signaling Model, I suggest that the
Introduction should also refer to 'Signalling' briefly also.

I suggest a new second paragraph in Section 1 that says:


*' When communicating between a WebRTC environment and non-WebRTC
environments, interworking of both the media and signaling is required. As
signaling protocols are outside the scope of WebRTC, this document only
considers signaling interworking at a high level'. *

To avoid confusion, when referring to 'Media only' functionality, the
document should use the term 'WebRTC *media* gateway' instead of 'WebRTC
gateway'.

Cheers,
/Barry Dingle



On Fri, Aug 22, 2014 at 6:34 PM, Jeremy Fuller <jeremy.fuller@genband.com>
wrote:

> Hi Harald,
>
> It would be good to have something which explains to the reader that there
> exists functionality for transforming signalling. I am reasonably flexible
> on the term used. I proposed associating this functionality with the term
> WebRTC Gateway to reduce the proliferation of new terms.
>
> Regards,
> Jeremy
>
> -----Original Message-----
> From: Harald Alvestrand [mailto:harald@alvestrand.no]
> Sent: 21 August 2014 22:16
> To: Jeremy Fuller
> Cc: rtcweb@ietf.org
> Subject: Re: WebRTC gateway - draft-ietf-rtcweb-overview-11.txt
>
> On 08/21/2014 10:24 PM, Jeremy Fuller wrote:
> > Hi Harald,
> >
> > With regard to your addition of the WebRTC gateway in the rtcweb
> overview. I appreciate signalling is out of scope for WebRTC, but as you
> say in your draft alvestrand-rtcweb-gateways-00.txt "any practical gateway
> needs to deal with signalling". Without getting into too much detail or any
> implementation constraints, would it be possible to add a reference to
> signalling when introducing the WebRTC gateway?
> >
> > I have suggested some modified text below for your consideration:
> >
> > Current Text  (section 1):
> > "A WebRTC gateway is something that mediates media traffic to non-WebRTC
> entities.  It is like a device, but has certain restrictiions on where it
> can operate, which means that some of the requirements can be relaxed."
> >
> > Proposed Text (section 1):
> > "A WebRTC gateway is something that mediates media traffic, or
> signalling, or both, to non-WebRTC entities.  It is like a device, but has
> certain restrictions on where it can operate, which means that some of the
> requirements can be relaxed."
>
> This does not work for me. A WebRTC gateway has to handle media, because
> it's the transport of media that WebRTC specifies. A pure signalling
> gateway would just transform the signalling between two WebRTC-capable
> endpoints (WebRTC endpoints?); this is what an application does, not what a
> WebRTC gateway does.
>
> There might be a need to name the class of applications that transform
> signalling; I'm not sure.
>
>
>
> >
> > Regards,
> > Jeremy
> >
> >
> > -----Original Message-----
> > Date: Mon, 18 Aug 2014 04:19:16 -0700
> > From: internet-drafts@ietf.org
> > To: i-d-announce@ietf.org
> > Cc: rtcweb@ietf.org
> > Subject: [rtcweb] I-D Action: draft-ietf-rtcweb-overview-11.txt
> > Message-ID: <20140818111916.2857.63981.idtracker@ietfa.amsl.com>
> > Content-Type: text/plain; charset="utf-8"
> >
> >
> > A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
> >   This draft is a work item of the Real-Time Communication in
> WEB-browsers Working Group of the IETF.
> >
> >          Title           : Overview: Real Time Protocols for
> Browser-based Applications
> >          Author          : Harald T. Alvestrand
> >       Filename        : draft-ietf-rtcweb-overview-11.txt
> >       Pages           : 22
> >       Date            : 2014-08-18
> >
> > 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.
> >
> >     This document is an Applicability Statement - it does not itself
> >     specify any protocol, but specifies which other specifications RTCWEB
> >     compliant implementations are supposed to follow.
> >
> >     This document is a work item of the RTCWEB working group.
> >
> >
> > The IETF datatracker status page for this draft is:
> > https://datatracker.ietf.org/doc/draft-ietf-rtcweb-overview/
> >
> > There's also a htmlized version available at:
> > http://tools.ietf.org/html/draft-ietf-rtcweb-overview-11
> >
> > A diff from the previous version is available at:
> > http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-overview-11
> >
> >
> > Please note that it may take a couple of minutes from the time of
> submission until the htmlized version and diff are available at
> tools.ietf.org.
> >
> > Internet-Drafts are also available by anonymous FTP at:
> > ftp://ftp.ietf.org/internet-drafts/
> >
> >
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--f46d043749cbd24c460501408af4
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<div dir=3D"ltr"><div>Hi Harald,<br><br>I see Jeremy&#39;s point.<br><br>Ga=
teways between a WebRTC environment and non-WebRTC environments need to con=
vert BOTH media and signalling. Media-only Gateways are usually referred to=
 as &#39;Media Gateways&#39;. In draft-alvestrand-rtcweb-gateways-00 Sectio=
n 1 paragraph 2, the description of &#39;the Gateway&#39; refers only to th=
e &#39;media&#39;. <br>

<br>As the document later refers to the Signaling Model, I suggest that the=
 Introduction should also refer to &#39;Signalling&#39; briefly also. <br>
<br>I suggest a new second paragraph in Section 1 that says:<br><br><i>&#39=
; When communicating between a WebRTC environment and non-WebRTC environmen=
ts,=20
interworking of both the media and signaling is required. As signaling prot=
ocols are outside the scope=20
of WebRTC, this document only considers signaling interworking at a high le=
vel&#39;. <br>
</i><div><br>
</div>To avoid confusion, when
 referring to &#39;Media only&#39; functionality, the document should use t=
he term &#39;WebRTC <b>media</b> gateway&#39; instead of &#39;WebRTC=20
gateway&#39;.<br><br></div><div><div class=3D"gmail_extra"><div><div dir=3D=
"ltr">Cheers,<br>/Barry Dingle<br><br></div></div>
<br><br><div class=3D"gmail_quote">On Fri, Aug 22, 2014 at 6:34 PM, Jeremy =
Fuller <span dir=3D"ltr">&lt;<a href=3D"mailto:jeremy.fuller@genband.com" t=
arget=3D"_blank">jeremy.fuller@genband.com</a>&gt;</span> wrote:<br><blockq=
uote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc =
solid;padding-left:1ex">

Hi Harald,<br>
<br>
It would be good to have something which explains to the reader that there =
exists functionality for transforming signalling. I am reasonably flexible =
on the term used. I proposed associating this functionality with the term W=
ebRTC Gateway to reduce the proliferation of new terms.<br>


<br>
Regards,<br>
Jeremy<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
-----Original Message-----<br>
From: Harald Alvestrand [mailto:<a href=3D"mailto:harald@alvestrand.no">har=
ald@alvestrand.no</a>]<br>
Sent: 21 August 2014 22:16<br>
To: Jeremy Fuller<br>
Cc: <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
Subject: Re: WebRTC gateway - draft-ietf-rtcweb-overview-11.txt<br>
<br>
On 08/21/2014 10:24 PM, Jeremy Fuller wrote:<br>
&gt; Hi Harald,<br>
&gt;<br>
&gt; With regard to your addition of the WebRTC gateway in the rtcweb overv=
iew. I appreciate signalling is out of scope for WebRTC, but as you say in =
your draft alvestrand-rtcweb-gateways-00.txt &quot;any practical gateway ne=
eds to deal with signalling&quot;. Without getting into too much detail or =
any implementation constraints, would it be possible to add a reference to =
signalling when introducing the WebRTC gateway?<br>


&gt;<br>
&gt; I have suggested some modified text below for your consideration:<br>
&gt;<br>
&gt; Current Text=C2=A0 (section 1):<br>
&gt; &quot;A WebRTC gateway is something that mediates media traffic to non=
-WebRTC entities.=C2=A0 It is like a device, but has certain restrictiions =
on where it can operate, which means that some of the requirements can be r=
elaxed.&quot;<br>


&gt;<br>
&gt; Proposed Text (section 1):<br>
&gt; &quot;A WebRTC gateway is something that mediates media traffic, or si=
gnalling, or both, to non-WebRTC entities.=C2=A0 It is like a device, but h=
as certain restrictions on where it can operate, which means that some of t=
he requirements can be relaxed.&quot;<br>


<br>
This does not work for me. A WebRTC gateway has to handle media, because it=
&#39;s the transport of media that WebRTC specifies. A pure signalling gate=
way would just transform the signalling between two WebRTC-capable endpoint=
s (WebRTC endpoints?); this is what an application does, not what a WebRTC =
gateway does.<br>


<br>
There might be a need to name the class of applications that transform sign=
alling; I&#39;m not sure.<br>
<br>
<br>
<br>
&gt;<br>
&gt; Regards,<br>
&gt; Jeremy<br>
&gt;<br>
&gt;<br>
&gt; -----Original Message-----<br>
&gt; Date: Mon, 18 Aug 2014 04:19:16 -0700<br>
&gt; From: <a href=3D"mailto:internet-drafts@ietf.org">internet-drafts@ietf=
.org</a><br>
&gt; To: <a href=3D"mailto:i-d-announce@ietf.org">i-d-announce@ietf.org</a>=
<br>
&gt; Cc: <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; Subject: [rtcweb] I-D Action: draft-ietf-rtcweb-overview-11.txt<br>
&gt; Message-ID: &lt;<a href=3D"mailto:20140818111916.2857.63981.idtracker@=
ietfa.amsl.com">20140818111916.2857.63981.idtracker@ietfa.amsl.com</a>&gt;<=
br>
&gt; Content-Type: text/plain; charset=3D&quot;utf-8&quot;<br>
&gt;<br>
&gt;<br>
&gt; A New Internet-Draft is available from the on-line Internet-Drafts dir=
ectories.<br>
&gt;=C2=A0 =C2=A0This draft is a work item of the Real-Time Communication i=
n WEB-browsers Working Group of the IETF.<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 Title=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0: Overview: Real Time Protocols for Browser-based Applications<br=
>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 Author=C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 : Harald T. Alvestrand<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0Filename=C2=A0 =C2=A0 =C2=A0 =C2=A0 : draft-=
ietf-rtcweb-overview-11.txt<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0Pages=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0: 22<br>
&gt;=C2=A0 =C2=A0 =C2=A0 =C2=A0Date=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 : 2014-08-18<br>
&gt;<br>
&gt; Abstract:<br>
&gt;=C2=A0 =C2=A0 =C2=A0This document gives an overview and context of a pr=
otocol suite<br>
&gt;=C2=A0 =C2=A0 =C2=A0intended for use with real-time applications that c=
an be deployed in<br>
&gt;=C2=A0 =C2=A0 =C2=A0browsers - &quot;real time communication on the Web=
&quot;.<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0It intends to serve as a starting and coordination =
point to make sure<br>
&gt;=C2=A0 =C2=A0 =C2=A0all the parts that are needed to achieve this goal =
are findable, and<br>
&gt;=C2=A0 =C2=A0 =C2=A0that the parts that belong in the Internet protocol=
 suite are fully<br>
&gt;=C2=A0 =C2=A0 =C2=A0specified and on the right publication track.<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0This document is an Applicability Statement - it do=
es not itself<br>
&gt;=C2=A0 =C2=A0 =C2=A0specify any protocol, but specifies which other spe=
cifications RTCWEB<br>
&gt;=C2=A0 =C2=A0 =C2=A0compliant implementations are supposed to follow.<b=
r>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0This document is a work item of the RTCWEB working =
group.<br>
&gt;<br>
&gt;<br>
&gt; The IETF datatracker status page for this draft is:<br>
&gt; <a href=3D"https://datatracker.ietf.org/doc/draft-ietf-rtcweb-overview=
/" target=3D"_blank">https://datatracker.ietf.org/doc/draft-ietf-rtcweb-ove=
rview/</a><br>
&gt;<br>
&gt; There&#39;s also a htmlized version available at:<br>
&gt; <a href=3D"http://tools.ietf.org/html/draft-ietf-rtcweb-overview-11" t=
arget=3D"_blank">http://tools.ietf.org/html/draft-ietf-rtcweb-overview-11</=
a><br>
&gt;<br>
&gt; A diff from the previous version is available at:<br>
&gt; <a href=3D"http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-overvi=
ew-11" target=3D"_blank">http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcw=
eb-overview-11</a><br>
&gt;<br>
&gt;<br>
&gt; Please note that it may take a couple of minutes from the time of subm=
ission until the htmlized version and diff are available at <a href=3D"http=
://tools.ietf.org" target=3D"_blank">tools.ietf.org</a>.<br>
&gt;<br>
&gt; Internet-Drafts are also available by anonymous FTP at:<br>
&gt; <a href=3D"ftp://ftp.ietf.org/internet-drafts/" target=3D"_blank">ftp:=
//ftp.ietf.org/internet-drafts/</a><br>
&gt;<br>
&gt;<br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div></div></div>

--f46d043749cbd24c460501408af4--


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--001a11c247e4ab0fa905016d93fe
Content-Type: text/plain; charset=UTF-8
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Yes, 4571 must be used for ICE connectivity checks, when using ICE-TCP.

However, TURN/TCP has its own framing for STUN messages.

I would simplify your sentence to:

   If ICE-TCP connections are used, framing according to [RFC4571] MUST
   be used for all packets, unless the content has its own framing
mechanism.


On Fri, Aug 22, 2014 at 8:38 AM, I=C3=B1aki Baz Castillo <ibc@aliax.net> wr=
ote:

> Hi,
>
> draft-ietf-rtcweb-transports-06 says:
>
>    If TCP connections are used, RTP framing according to [RFC4571] MUST
>    be used, both for the RTP packets and for the DTLS packets used to
>    carry data channels.
>
>
> Not just RTP and DTLS, but also for STUN packets must be framed with
> RFC4571. And I do not understand why "DTLS packets used to carry data
> channels". In fact all the DTLS packets (also the ones for the
> handshake) must be framed as per RFC 4571.
>
>
> So I suggest to improve the above text as follows:
>
>    If TCP connections are used, RTP framing according to [RFC4571] MUST
>    be used for the STUN packets, SRTP packets, SRTCP packets and DTLS
>    packets.
>
>
>
>
>
>
> --
> I=C3=B1aki Baz Castillo
> <ibc@aliax.net>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--001a11c247e4ab0fa905016d93fe
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Yes, 4571 must be used for ICE connectivity checks, when u=
sing ICE-TCP.<div><br></div><div>However, TURN/TCP has its own framing for =
STUN messages.</div><div><br></div><div>I would simplify your sentence to:<=
/div>

<div><br></div><div><span style=3D"font-family:arial,sans-serif;font-size:1=
3px">=C2=A0 =C2=A0If ICE-TCP connections are used, framing according to [RF=
C4571] MUST</span><br style=3D"font-family:arial,sans-serif;font-size:13px"=
><span style=3D"font-family:arial,sans-serif;font-size:13px">=C2=A0 =C2=A0b=
e used for all packets, unless the content has its own framing mechanism.</=
span><br>

</div></div><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">O=
n Fri, Aug 22, 2014 at 8:38 AM, I=C3=B1aki Baz Castillo <span dir=3D"ltr">&=
lt;<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.net</a>&gt;=
</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Hi,<br>
<br>
draft-ietf-rtcweb-transports-06 says:<br>
<br>
=C2=A0 =C2=A0If TCP connections are used, RTP framing according to [RFC4571=
] MUST<br>
=C2=A0 =C2=A0be used, both for the RTP packets and for the DTLS packets use=
d to<br>
=C2=A0 =C2=A0carry data channels.<br>
<br>
<br>
Not just RTP and DTLS, but also for STUN packets must be framed with<br>
RFC4571. And I do not understand why &quot;DTLS packets used to carry data<=
br>
channels&quot;. In fact all the DTLS packets (also the ones for the<br>
handshake) must be framed as per RFC 4571.<br>
<br>
<br>
So I suggest to improve the above text as follows:<br>
<br>
=C2=A0 =C2=A0If TCP connections are used, RTP framing according to [RFC4571=
] MUST<br>
=C2=A0 =C2=A0be used for the STUN packets, SRTP packets, SRTCP packets and =
DTLS<br>
=C2=A0 =C2=A0packets.<br>
<span class=3D"HOEnZb"><font color=3D"#888888"><br>
<br>
<br>
<br>
<br>
<br>
--<br>
I=C3=B1aki Baz Castillo<br>
&lt;<a href=3D"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt;<br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</font></span></blockquote></div><br></div>

--001a11c247e4ab0fa905016d93fe--


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Date: Mon, 25 Aug 2014 16:25:08 +0530
Message-ID: <CAKz0y8y3=0_-jwWiviR_Uj6tSq75NEq92Ocergc_NvFk2h_72Q@mail.gmail.com>
From: Muthu Arul Mozhi Perumal <muthu.arul@gmail.com>
To: Parthasarathi R <partha@parthasarathi.co.in>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] WG Last Call for draft-ietf-rtcweb-stun-consent-freshness
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I thought a "SHOULD perform" is more appropriate for a WebRTC device as it
doesn't implement the public API and might run a proprietary/native app
running in a constrained environment and the overhead of performing the
periodic consent might overweigh the benefits. However, I don't have an
issue of making it a MUST -- at least no one has objected so far. I hear
the following options from the discussions so far:

Option 1:
WebRTC browser - MUST
WebRTC device - MUST
Other (WebRTC/non-WebRTC) entities - MAY

Option 2:
Any WebRTC entity doing full ICE - MUST
Other (WebRTC/non-WebRTC) entities - MAY

Obviously, #2 is stronger than #1 and also covers a WebRTC gateway doing
full ICE. I would like to hear from others as well..

Muthu

On Sat, Aug 23, 2014 at 2:54 AM, Parthasarathi R <partha@parthasarathi.co.i=
n
> wrote:

>  Hi Ram/Muthu,
>
>
>
> There is a terminology variation here. As per the current overview
> document, WebRTC device supports full WebRTC protocol requirement which
> includes consent-check. In case consent check is excluded from WebRTC
> device,  the new definition has to be provided for =E2=80=9CWebRTC device=
=E2=80=9D in
> =E2=80=93overview draft. I don=E2=80=99t think that WebRTC device definit=
ion has to be
> changed now.
>
>
>
> IMO, WebRTC device & WebRTC browser MUST support consent check. As
> discussed in another mail thread, there is an need to define =E2=80=9CWeb=
RTC
> endpoint=E2=80=9D which relaxes the lot of WebRTC protocol requirement ba=
sed on the
> specific deployment like Enterprise, LTE EPC. =E2=80=9CWebRTC Endpoint=E2=
=80=9D & =E2=80=9CWebRTC
> Gateway=E2=80=9D MAY support consent check.
>
>
>
> I have highlighted Full ICE as a criteria for =E2=80=9CWebRTC Endpoint=E2=
=80=9D & =E2=80=9CWebRTC
> Gateway=E2=80=9D in
> http://www.ietf.org/mail-archive/web/rtcweb/current/msg12938.html as it
> helps the implementers to decide whether consent check has to be
> implemented in their =E2=80=9CWebRTC Endpoint=E2=80=9D/ =E2=80=9CWebRTC G=
ateway=E2=80=9D or not. As =E2=80=9CWebRTC
> Endpoint=E2=80=9D is yet to defined and =E2=80=9CWebRTC gateway=E2=80=9D =
is just now proposed in
> -11 overview draft,  I have proposed the text as
>
>
>
> =E2=80=9CFull ICE supporting WebRTC entities  MUST support consent freshn=
ess.=E2=80=9D
>
>
>
> Please let me your opinion on the same.
>
>
>
> Also, I agree with Muthu=E2=80=99s statement to indicates the other requi=
rement
> for WebRTC entities to include consent check are:
>
> =E2=80=9Cto use the mechanism to make it more secure or compute RTT or ca=
rry
> network information=E2=80=9D
>
>
>
>
>
> Thanks
>
> Partha
>
>
>
>
>
>
>
> *From:* rtcweb [mailto:rtcweb-bounces@ietf.org] *On Behalf Of *Ram Mohan
> R (rmohanr)
> *Sent:* Friday, August 22, 2014 9:48 PM
> *To:* Christer Holmberg; Roman Shpount; Muthu Arul Mozhi Perumal
>
> *Cc:* rtcweb@ietf.org
> *Subject:* Re: [rtcweb] WG Last Call for
> draft-ietf-rtcweb-stun-consent-freshness
>
>
>
> Right. I think we had discussed this in the past and added the below text
> in the draft
>
>
>
> " WebRTC endpoints are required to support full ICE as specified in
>
>    section 3.4 of [I-D.ietf-rtcweb-transports <http://tools.ietf.org/html=
/draft-ietf-rtcweb-stun-consent-freshness-05#ref-I-D.ietf-rtcweb-transports=
>].  However, when WebRTC
>
>    endpoints interwork with other endpoints that support only ICE-lite
>
>    (e.g. gateways) those endpoints will not generate consent checks, but
>
>    just respond to consent checks they receive=E2=80=9D
>
>
> Isn=E2=80=99t this sufficient ?
>
>
> Ram
>
>
>
> *From: *Christer Holmberg <christer.holmberg@ericsson.com>
> *Date: *Friday, 22 August 2014 9:29 pm
> *To: *Roman Shpount <roman@telurix.com>, Muthu Arul Mozhi Perumal <
> muthu.arul@gmail.com>
> *Cc: *"rtcweb@ietf.org" <rtcweb@ietf.org>
> *Subject: *Re: [rtcweb] WG Last Call for
> draft-ietf-rtcweb-stun-consent-freshness
>
>
>
> Hi,
>
>
>
> An ICE-lite endpoint must already today *REPLY* to STUN requests - consen=
t
> freshness does not change that.
>
>
>
> But, I don't think an ICE-lite endpoint shall be mandated to *SEND*
> consent freshness requests.
>
>
>
> Regards,
>
>
>
> Christer
>
>
>
>
>  ------------------------------
>
> *From:* rtcweb [rtcweb-bounces@ietf.org] on behalf of Roman Shpount [
> roman@telurix.com]
> *Sent:* Friday, 22 August 2014 4:37 PM
> *To:* Muthu Arul Mozhi Perumal
> *Cc:* rtcweb@ietf.org
> *Subject:* Re: [rtcweb] WG Last Call for
> draft-ietf-rtcweb-stun-consent-freshness
>
> On Fri, Aug 22, 2014 at 1:25 AM, Muthu Arul Mozhi Perumal <
> muthu.arul@gmail.com> wrote:
>
>  draft-ietf-rtcweb-stun-consent-freshness is about making the WebRTC
> browser more secure. It however allows an RTP endpoint (that also does IC=
E)
> to use the mechanism to make it more secure or compute RTT or carry netwo=
rk
> information or whatever. However, requiring every RTP endpoint perform it
> seems asking too much.
>
>
>
> My take:
>
> WebRTC browser - MUST
>
> WebRTC devide - SHOULD
>
> Other RTP entities (including WebRTC gateway) - MAY
>
>
>
>
>
>  I would say that all full ICE endpoint interworking with WebRTC MUST
> implement consent freshness. ICE-LITE will not implement, but MUST respon=
d
> (unless someone defines it, but once you start sending STUN request you
> might as well do full ICE, so I do not see much point in it). And to
> conclude strongly encourage full ICE implementation for security and othe=
r
> reasons.
>
> _____________
> Roman Shpount
>
>
>

--e89a8f3ba5b53111850501720513
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:arial,he=
lvetica,sans-serif;font-size:small">I thought a &quot;SHOULD perform&quot; =
is more appropriate for a WebRTC device as it doesn&#39;t implement the pub=
lic API and might run a proprietary/native app running in a constrained env=
ironment and the overhead of performing the periodic consent might overweig=
h the benefits. However, I don&#39;t have an issue of making it a MUST -- a=
t least no one has objected so far. I hear the following options from the d=
iscussions so far:</div>
<div class=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-seri=
f;font-size:small"><br></div><div class=3D"gmail_default" style=3D"font-fam=
ily:arial,helvetica,sans-serif;font-size:small">Option 1:</div><div class=
=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-serif;font-siz=
e:small">
WebRTC browser - MUST<br></div><div class=3D"gmail_default" style=3D"font-f=
amily:arial,helvetica,sans-serif;font-size:small">WebRTC device - MUST</div=
><div class=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-ser=
if;font-size:small">
Other (WebRTC/non-WebRTC) entities - MAY</div><div class=3D"gmail_default" =
style=3D"font-family:arial,helvetica,sans-serif;font-size:small"><br></div>=
<div class=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-seri=
f;font-size:small">
Option 2:</div><div class=3D"gmail_default" style=3D"font-family:arial,helv=
etica,sans-serif;font-size:small">Any WebRTC entity doing full ICE - MUST<b=
r></div><div class=3D"gmail_default" style=3D"font-family:arial,helvetica,s=
ans-serif;font-size:small">
Other (WebRTC/non-WebRTC) entities - MAY<br></div><div class=3D"gmail_defau=
lt" style=3D"font-family:arial,helvetica,sans-serif;font-size:small">=C2=A0=
</div><div class=3D"gmail_extra"><div class=3D"gmail_default" style=3D"font=
-family:arial,helvetica,sans-serif;font-size:small">
Obviously, #2 is stronger than #1 and also covers a WebRTC gateway doing fu=
ll ICE. I would like to hear from others as well..</div><div class=3D"gmail=
_default" style=3D"font-family:arial,helvetica,sans-serif;font-size:small">
<br></div><div class=3D"gmail_default" style=3D"font-family:arial,helvetica=
,sans-serif;font-size:small">Muthu</div><br><div class=3D"gmail_quote">On S=
at, Aug 23, 2014 at 2:54 AM, Parthasarathi R <span dir=3D"ltr">&lt;<a href=
=3D"mailto:partha@parthasarathi.co.in" target=3D"_blank">partha@parthasarat=
hi.co.in</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">









<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple" style=3D"word-wrap:break=
-word">

<div>

<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif;color:rgb(31,73,125)">Hi Ram/Muthu,<u></u><u></u></span></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif;color:rgb(31,73,125)"><u></u>=C2=A0<u></u></span></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif;color:rgb(31,73,125)">There is a terminology variation here. As pe=
r the current
overview document, WebRTC device supports full WebRTC protocol requirement =
which
includes consent-check. In case consent check is excluded from WebRTC devic=
e, =C2=A0the
new definition has to be provided for =E2=80=9CWebRTC device=E2=80=9D in =
=E2=80=93overview
draft. I don=E2=80=99t think that WebRTC device definition has to be change=
d now.<u></u><u></u></span></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif;color:rgb(31,73,125)"><u></u>=C2=A0<u></u></span></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif;color:rgb(31,73,125)">IMO, WebRTC device &amp; WebRTC browser MUST=
 support consent
check. As discussed in another mail thread, there is an need to define =E2=
=80=9CWebRTC
endpoint=E2=80=9D which relaxes the lot of WebRTC protocol requirement base=
d on
the specific deployment like Enterprise, LTE EPC. =E2=80=9CWebRTC Endpoint=
=E2=80=9D
&amp; =E2=80=9CWebRTC Gateway=E2=80=9D MAY support consent check.<u></u><u>=
</u></span></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif;color:rgb(31,73,125)"><u></u>=C2=A0<u></u></span></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif;color:rgb(31,73,125)">I have highlighted Full ICE as a criteria fo=
r =E2=80=9CWebRTC
Endpoint=E2=80=9D &amp; =E2=80=9CWebRTC Gateway=E2=80=9D in <a href=3D"http=
://www.ietf.org/mail-archive/web/rtcweb/current/msg12938.html" target=3D"_b=
lank">http://www.ietf.org/mail-archive/web/rtcweb/current/msg12938.html</a>
as it helps the implementers to decide whether consent check has to be
implemented in their =E2=80=9CWebRTC Endpoint=E2=80=9D/ =E2=80=9CWebRTC Gat=
eway=E2=80=9D
or not. As =E2=80=9CWebRTC Endpoint=E2=80=9D is yet to defined and =E2=80=
=9CWebRTC
gateway=E2=80=9D is just now proposed in -11 overview draft, =C2=A0I have p=
roposed
the text as<u></u><u></u></span></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif;color:rgb(31,73,125)"><u></u>=C2=A0<u></u></span></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif;color:rgb(31,73,125)">=E2=80=9CFull ICE supporting WebRTC entities=
 =C2=A0MUST support
consent freshness.=E2=80=9D <u></u><u></u></span></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif;color:rgb(31,73,125)"><u></u>=C2=A0<u></u></span></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif;color:rgb(31,73,125)">Please let me your opinion on the same. <u><=
/u><u></u></span></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif;color:rgb(31,73,125)"><u></u>=C2=A0<u></u></span></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif;color:rgb(31,73,125)">Also, I agree with Muthu=E2=80=99s statement=
 to indicates the
other requirement for WebRTC entities to include consent check are:<u></u><=
u></u></span></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif;color:rgb(31,73,125)">=E2=80=9C</span><span style=3D"font-family:A=
rial,sans-serif;color:black">to use the mechanism to make it more secure or=
 compute RTT or
carry network information=E2=80=9D</span><span style=3D"font-size:11pt;font=
-family:Calibri,sans-serif;color:rgb(31,73,125)"> <u></u><u></u></span></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif;color:rgb(31,73,125)"><u></u>=C2=A0<u></u></span></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif;color:rgb(31,73,125)"><u></u>=C2=A0<u></u></span></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif;color:rgb(31,73,125)">Thanks<u></u><u></u></span></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif;color:rgb(31,73,125)">Partha<u></u><u></u></span></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif;color:rgb(31,73,125)"><u></u>=C2=A0<u></u></span></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif;color:rgb(31,73,125)"><u></u>=C2=A0<u></u></span></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif;color:rgb(31,73,125)"><u></u>=C2=A0<u></u></span></p>

<div style=3D"border-style:none none none solid;border-left-color:blue;bord=
er-left-width:1.5pt;padding:0in 0in 0in 4pt">

<div>

<div style=3D"border-style:solid none none;border-top-color:rgb(181,196,223=
);border-top-width:1pt;padding:3pt 0in 0in">

<p class=3D"MsoNormal"><b><span style=3D"font-size:10pt;font-family:Tahoma,=
sans-serif">From:</span></b><span style=3D"font-size:10pt;font-family:Tahom=
a,sans-serif"> rtcweb
[mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org" target=3D"_blank">rtcweb=
-bounces@ietf.org</a>] <b>On Behalf Of </b>Ram Mohan R (rmohanr)<br>
<b>Sent:</b> Friday, August 22, 2014 9:48 PM<br>
<b>To:</b> Christer Holmberg; Roman Shpount; Muthu Arul Mozhi Perumal</span=
></p><div><div class=3D"h5"><br>
<b>Cc:</b> <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf=
.org</a><br>
<b>Subject:</b> Re: [rtcweb] WG Last Call for
draft-ietf-rtcweb-stun-consent-freshness<u></u><u></u></div></div><p></p>

</div>

</div><div><div class=3D"h5">

<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>

<div>

<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:Calibri,=
sans-serif;color:black">Right. I think we had discussed this in the past an=
d added the
below text in the draft<u></u><u></u></span></p>

</div>

<div>

<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:Calibri,=
sans-serif;color:black"><u></u>=C2=A0<u></u></span></p>

</div>

<div>

<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:Calibri,=
sans-serif;color:black">&quot; WebRTC endpoints are required to support ful=
l ICE as
specified in<u></u><u></u></span></p>

</div>

<pre><span style=3D"font-size:10.5pt;color:black">=C2=A0=C2=A0 section 3.4 =
of [<a href=3D"http://tools.ietf.org/html/draft-ietf-rtcweb-stun-consent-fr=
eshness-05#ref-I-D.ietf-rtcweb-transports" target=3D"_blank">I-D.ietf-rtcwe=
b-transports</a>].=C2=A0 However, when WebRTC<u></u><u></u></span></pre>
<pre><span style=3D"font-size:10.5pt;color:black">=C2=A0=C2=A0 endpoints in=
terwork with other endpoints that support only ICE-lite<u></u><u></u></span=
></pre><pre><span style=3D"font-size:10.5pt;color:black">=C2=A0=C2=A0 (e.g.=
 gateways) those endpoints will not generate consent checks, but<u></u><u><=
/u></span></pre>
<pre><span style=3D"font-size:10.5pt;color:black">=C2=A0=C2=A0 just respond=
 to consent checks they receive=E2=80=9D<u></u><u></u></span></pre><span st=
yle=3D"font-size:10.5pt;font-family:&#39;Courier New&#39;;color:black"><br =
clear=3D"all">
</span><pre><span style=3D"font-size:10.5pt;color:black">Isn=E2=80=99t this=
 sufficient ?<u></u><u></u></span></pre><span style=3D"font-size:10.5pt;fon=
t-family:&#39;Courier New&#39;;color:black"><br clear=3D"all">
</span><pre><span style=3D"font-size:10.5pt;color:black">Ram<u></u><u></u><=
/span></pre>

<div>

<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:Calibri,=
sans-serif;color:black"><u></u>=C2=A0<u></u></span></p>

</div>

<div style=3D"border-style:solid none none;border-top-color:rgb(181,196,223=
);border-top-width:1pt;padding:3pt 0in 0in">

<p class=3D"MsoNormal"><b><span style=3D"font-size:11pt;font-family:Calibri=
,sans-serif;color:black">From: </span></b><span style=3D"font-size:11pt;fon=
t-family:Calibri,sans-serif;color:black">Christer Holmberg &lt;<a href=3D"m=
ailto:christer.holmberg@ericsson.com" target=3D"_blank">christer.holmberg@e=
ricsson.com</a>&gt;<br>

<b>Date: </b>Friday, 22 August 2014 9:29 pm<br>
<b>To: </b>Roman Shpount &lt;<a href=3D"mailto:roman@telurix.com" target=3D=
"_blank">roman@telurix.com</a>&gt;,
Muthu Arul Mozhi Perumal &lt;<a href=3D"mailto:muthu.arul@gmail.com" target=
=3D"_blank">muthu.arul@gmail.com</a>&gt;<br>
<b>Cc: </b>&quot;<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcwe=
b@ietf.org</a>&quot;
&lt;<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a=
>&gt;<br>
<b>Subject: </b>Re: [rtcweb] WG Last Call for
draft-ietf-rtcweb-stun-consent-freshness<u></u><u></u></span></p>

</div>

<div>

<p class=3D"MsoNormal"><span style=3D"font-size:10.5pt;font-family:Calibri,=
sans-serif;color:black"><u></u>=C2=A0<u></u></span></p>

</div>

<div>

<div>

<div>

<p><span style=3D"font-size:10pt;font-family:Tahoma,sans-serif;color:black"=
>Hi,<u></u><u></u></span></p>

<p><span style=3D"font-size:10pt;font-family:Tahoma,sans-serif;color:black"=
>=C2=A0<u></u><u></u></span></p>

<p><span style=3D"font-size:10pt;font-family:Tahoma,sans-serif;color:black"=
>An
ICE-lite endpoint must already today=C2=A0*REPLY* to STUN requests - consen=
t
freshness does not change that.<u></u><u></u></span></p>

<p><span style=3D"font-size:10pt;font-family:Tahoma,sans-serif;color:black"=
>=C2=A0<u></u><u></u></span></p>

<p><span style=3D"font-size:10pt;font-family:Tahoma,sans-serif;color:black"=
>But,
I don&#39;t think an ICE-lite endpoint shall be mandated to *SEND* consent
freshness requests.<u></u><u></u></span></p>

<p><span style=3D"font-size:10pt;font-family:Tahoma,sans-serif;color:black"=
>=C2=A0<u></u><u></u></span></p>

<p><span style=3D"font-size:10pt;font-family:Tahoma,sans-serif;color:black"=
>Regards,<u></u><u></u></span></p>

<p><span style=3D"font-size:10pt;font-family:Tahoma,sans-serif;color:black"=
>=C2=A0<u></u><u></u></span></p>

<p><span style=3D"font-size:10pt;font-family:Tahoma,sans-serif;color:black"=
>Christer<u></u><u></u></span></p>

<p><span style=3D"font-size:10pt;font-family:Tahoma,sans-serif;color:black"=
>=C2=A0<u></u><u></u></span></p>

<p><span style=3D"font-size:10pt;font-family:Tahoma,sans-serif;color:black"=
>=C2=A0<u></u><u></u></span></p>

<div>

<div class=3D"MsoNormal" align=3D"center" style=3D"text-align:center"><span=
 style=3D"color:black">

<hr size=3D"2" width=3D"100%" align=3D"center">

</span></div>

<div>

<p class=3D"MsoNormal" style=3D"margin-bottom:12pt"><b><span style=3D"font-=
size:10pt;font-family:Tahoma,sans-serif;color:black">From:</span></b><span =
style=3D"font-size:10pt;font-family:Tahoma,sans-serif;color:black"> rtcweb
[<a href=3D"mailto:rtcweb-bounces@ietf.org" target=3D"_blank">rtcweb-bounce=
s@ietf.org</a>] on
behalf of Roman Shpount [<a href=3D"mailto:roman@telurix.com" target=3D"_bl=
ank">roman@telurix.com</a>]<br>
<b>Sent:</b> Friday, 22 August 2014 4:37 PM<br>
<b>To:</b> Muthu Arul Mozhi Perumal<br>
<b>Cc:</b> <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf=
.org</a><br>
<b>Subject:</b> Re: [rtcweb] WG Last Call for draft-ietf-rtcweb-stun-consen=
t-freshness</span><span style=3D"color:black"><u></u><u></u></span></p>

</div>

<div>

<div>

<div>

<div>

<p class=3D"MsoNormal"><span style=3D"color:black">On Fri, Aug 22, 2014 at =
1:25 AM,
Muthu Arul Mozhi Perumal &lt;<a href=3D"mailto:muthu.arul@gmail.com" target=
=3D"_blank">muthu.arul@gmail.com</a>&gt; wrote:<u></u><u></u></span></p>

</div>

<div>

<blockquote style=3D"border-style:none none none solid;border-left-color:rg=
b(204,204,204);border-left-width:1pt;padding:0in 0in 0in 6pt;margin-left:4.=
8pt;margin-right:0in">

<div>

<div>

<p class=3D"MsoNormal"><span style=3D"font-family:Arial,sans-serif;color:bl=
ack">draft-ietf-rtcweb-stun-consent-freshness
is about making the WebRTC browser more secure. It however allows an RTP
endpoint (that also does ICE) to use the mechanism to make it more secure o=
r
compute RTT or carry network information or whatever. However, requiring ev=
ery
RTP endpoint perform it seems asking too much.<u></u><u></u></span></p>

</div>

<div>

<p class=3D"MsoNormal"><span style=3D"color:black"><u></u>=C2=A0<u></u></sp=
an></p>

</div>

<div>

<p class=3D"MsoNormal"><span style=3D"font-family:Arial,sans-serif;color:bl=
ack">My
take:</span><span style=3D"color:black"><u></u><u></u></span></p>

</div>

<div>

<p class=3D"MsoNormal"><span style=3D"font-family:Arial,sans-serif;color:bl=
ack">WebRTC
browser - MUST</span><span style=3D"color:black"><u></u><u></u></span></p>

</div>

<div>

<p class=3D"MsoNormal"><span style=3D"font-family:Arial,sans-serif;color:bl=
ack">WebRTC
devide - SHOULD</span><span style=3D"color:black"><u></u><u></u></span></p>

</div>

<div>

<p class=3D"MsoNormal"><span style=3D"font-family:Arial,sans-serif;color:bl=
ack">Other
RTP entities (including WebRTC gateway) - MAY</span><span style=3D"color:bl=
ack"><u></u><u></u></span></p>

</div>

<div>

<p class=3D"MsoNormal"><span style=3D"color:black"><u></u>=C2=A0<u></u></sp=
an></p>

</div>

</div>

</blockquote>

<div>

<p class=3D"MsoNormal"><span style=3D"color:black">=C2=A0<u></u><u></u></sp=
an></p>

</div>

<div>

<p class=3D"MsoNormal"><span style=3D"color:black">=C2=A0I would say that a=
ll full
ICE endpoint interworking with WebRTC MUST implement consent freshness.
ICE-LITE will not implement, but MUST respond (unless someone defines it, b=
ut
once you start sending STUN request you might as well do full ICE, so I do =
not
see much point in it). And to conclude strongly encourage full ICE
implementation for security and other reasons.<u></u><u></u></span></p>

</div>

<div>

<p class=3D"MsoNormal"><span style=3D"color:black">_____________<br>
Roman Shpount<u></u><u></u></span></p>

</div>

<div>

<p class=3D"MsoNormal"><span style=3D"color:black">=C2=A0<u></u><u></u></sp=
an></p>

</div>

</div>

</div>

</div>

</div>

</div>

</div>

</div>

</div>

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

</div>

</div>


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

--e89a8f3ba5b53111850501720513--


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--e89a8f643366e57b7a0501738816
Content-Type: text/plain; charset=UTF-8

On Mon, Aug 25, 2014 at 6:55 AM, Muthu Arul Mozhi Perumal <
muthu.arul@gmail.com> wrote:

> I thought a "SHOULD perform" is more appropriate for a WebRTC device as it
> doesn't implement the public API and might run a proprietary/native app
> running in a constrained environment and the overhead of performing the
> periodic consent might overweigh the benefits. However, I don't have an
> issue of making it a MUST -- at least no one has objected so far. I hear
> the following options from the discussions so far:
>
> Option 1:
> WebRTC browser - MUST
> WebRTC device - MUST
> Other (WebRTC/non-WebRTC) entities - MAY
>
> Option 2:
> Any WebRTC entity doing full ICE - MUST
> Other (WebRTC/non-WebRTC) entities - MAY
>
>
>
To be clear, my opinion is

Any WebRTC entity implementing full ICE - MUST
Any WebRTC entity (such as WebRTC gateway) implementing ICE-lite -- MUST NOT

This way we are not inventing entity types and not defining behaviors which
would never be used (such as sending consent from an end point which would
send no STUN requests otherwise). This will also reduce the number of
interop cases from four (full ICE with consent, full iCE without consent,
ICE-lite, ICE-lite with consent) to only two (full ICE with consent,
ICE-lite).
_____________
Roman Shpount

--e89a8f643366e57b7a0501738816
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_extra"><div>On Mon, Aug 25, 2014 at 6:=
55 AM, Muthu Arul Mozhi Perumal <span dir=3D"ltr">&lt;<a href=3D"mailto:mut=
hu.arul@gmail.com" target=3D"_blank">muthu.arul@gmail.com</a>&gt;</span> wr=
ote:<br>
</div><div class=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D=
"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,2=
04,204);border-left-style:solid;padding-left:1ex"><div dir=3D"ltr"><div sty=
le=3D"font-family:arial,helvetica,sans-serif;font-size:small">
I thought a &quot;SHOULD perform&quot; is more appropriate for a WebRTC dev=
ice as it doesn&#39;t implement the public API and might run a proprietary/=
native app running in a constrained environment and the overhead of perform=
ing the periodic consent might overweigh the benefits. However, I don&#39;t=
 have an issue of making it a MUST -- at least no one has objected so far. =
I hear the following options from the discussions so far:</div>

<div style=3D"font-family:arial,helvetica,sans-serif;font-size:small"><br><=
/div><div style=3D"font-family:arial,helvetica,sans-serif;font-size:small">=
Option 1:</div><div style=3D"font-family:arial,helvetica,sans-serif;font-si=
ze:small">

WebRTC browser - MUST<br></div><div style=3D"font-family:arial,helvetica,sa=
ns-serif;font-size:small">WebRTC device - MUST</div><div style=3D"font-fami=
ly:arial,helvetica,sans-serif;font-size:small">
Other (WebRTC/non-WebRTC) entities - MAY</div><div style=3D"font-family:ari=
al,helvetica,sans-serif;font-size:small"><br></div><div style=3D"font-famil=
y:arial,helvetica,sans-serif;font-size:small">
Option 2:</div><div style=3D"font-family:arial,helvetica,sans-serif;font-si=
ze:small">Any WebRTC entity doing full ICE - MUST<br></div><div style=3D"fo=
nt-family:arial,helvetica,sans-serif;font-size:small">
Other (WebRTC/non-WebRTC) entities - MAY<br></div><div style=3D"font-family=
:arial,helvetica,sans-serif;font-size:small">=C2=A0</div><div class=3D"gmai=
l_extra"><div style=3D"font-family:arial,helvetica,sans-serif;font-size:sma=
ll"><br>
</div></div></div></blockquote><div><br></div><div>To be clear, my opinion =
is</div><div><br></div><div>Any WebRTC entity implementing full ICE - MUST<=
/div><div>Any WebRTC entity (such as WebRTC gateway) implementing ICE-lite =
-- MUST NOT</div>
<div><br></div><div>This way we are not inventing entity types and not defi=
ning behaviors which would never be used (such as sending consent from an e=
nd point which would send no STUN requests otherwise). This will also reduc=
e the number of interop cases from four (full ICE with consent, full iCE wi=
thout consent, ICE-lite, ICE-lite with consent) to only two (full ICE with =
consent, ICE-lite).</div>
<div>_____________<br>Roman Shpount</div><div>=C2=A0</div></div></div></div=
>

--e89a8f643366e57b7a0501738816--


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To: Roman Shpount <roman@telurix.com>, Muthu Arul Mozhi Perumal <muthu.arul@gmail.com>
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Subject: Re: [rtcweb] WG Last Call for draft-ietf-rtcweb-stun-consent-freshness
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From: Muthu Arul Mozhi Perumal <muthu.arul@gmail.com>
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Subject: Re: [rtcweb] WG Last Call for draft-ietf-rtcweb-stun-consent-freshness
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--047d7b6225ac782f1f0501751365
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An ICE-lite entity by definition doesn't generate STUN requests which means
it doesn't perform consent freshness as described in the draft. If it does,
it is not a ICE-lite entity. You have only two types of entities to interop
test with. I don't see any use of specifying a MUST NOT for ICE-lite.

Muthu

On Mon, Aug 25, 2014 at 6:13 PM, Roman Shpount <roman@telurix.com> wrote:

> On Mon, Aug 25, 2014 at 6:55 AM, Muthu Arul Mozhi Perumal <
> muthu.arul@gmail.com> wrote:
>
>> I thought a "SHOULD perform" is more appropriate for a WebRTC device as
>> it doesn't implement the public API and might run a proprietary/native app
>> running in a constrained environment and the overhead of performing the
>> periodic consent might overweigh the benefits. However, I don't have an
>> issue of making it a MUST -- at least no one has objected so far. I hear
>> the following options from the discussions so far:
>>
>> Option 1:
>> WebRTC browser - MUST
>> WebRTC device - MUST
>> Other (WebRTC/non-WebRTC) entities - MAY
>>
>> Option 2:
>> Any WebRTC entity doing full ICE - MUST
>> Other (WebRTC/non-WebRTC) entities - MAY
>>
>>
>>
> To be clear, my opinion is
>
> Any WebRTC entity implementing full ICE - MUST
> Any WebRTC entity (such as WebRTC gateway) implementing ICE-lite -- MUST
> NOT
>
> This way we are not inventing entity types and not defining behaviors
> which would never be used (such as sending consent from an end point which
> would send no STUN requests otherwise). This will also reduce the number of
> interop cases from four (full ICE with consent, full iCE without consent,
> ICE-lite, ICE-lite with consent) to only two (full ICE with consent,
> ICE-lite).
> _____________
> Roman Shpount
>
>

--047d7b6225ac782f1f0501751365
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:arial,he=
lvetica,sans-serif;font-size:small">An ICE-lite entity by definition doesn&=
#39;t generate STUN requests which means it doesn&#39;t perform consent fre=
shness as described in the draft. If it does, it is not a ICE-lite entity. =
You have only two types of entities to interop test with. I don&#39;t see a=
ny use of specifying a MUST NOT for ICE-lite.</div>
<div class=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-seri=
f;font-size:small"><br></div><div class=3D"gmail_default" style=3D"font-fam=
ily:arial,helvetica,sans-serif;font-size:small">Muthu</div><div class=3D"gm=
ail_extra">
<br><div class=3D"gmail_quote">On Mon, Aug 25, 2014 at 6:13 PM, Roman Shpou=
nt <span dir=3D"ltr">&lt;<a href=3D"mailto:roman@telurix.com" target=3D"_bl=
ank">roman@telurix.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_=
quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-=
color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D""><div>On Mon, Au=
g 25, 2014 at 6:55 AM, Muthu Arul Mozhi Perumal <span dir=3D"ltr">&lt;<a hr=
ef=3D"mailto:muthu.arul@gmail.com" target=3D"_blank">muthu.arul@gmail.com</=
a>&gt;</span> wrote:<br>

</div></div><div class=3D"gmail_quote"><div class=3D""><blockquote class=3D=
"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;borde=
r-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><di=
v dir=3D"ltr">
<div style=3D"font-family:arial,helvetica,sans-serif;font-size:small">
I thought a &quot;SHOULD perform&quot; is more appropriate for a WebRTC dev=
ice as it doesn&#39;t implement the public API and might run a proprietary/=
native app running in a constrained environment and the overhead of perform=
ing the periodic consent might overweigh the benefits. However, I don&#39;t=
 have an issue of making it a MUST -- at least no one has objected so far. =
I hear the following options from the discussions so far:</div>


<div style=3D"font-family:arial,helvetica,sans-serif;font-size:small"><br><=
/div><div style=3D"font-family:arial,helvetica,sans-serif;font-size:small">=
Option 1:</div><div style=3D"font-family:arial,helvetica,sans-serif;font-si=
ze:small">


WebRTC browser - MUST<br></div><div style=3D"font-family:arial,helvetica,sa=
ns-serif;font-size:small">WebRTC device - MUST</div><div style=3D"font-fami=
ly:arial,helvetica,sans-serif;font-size:small">
Other (WebRTC/non-WebRTC) entities - MAY</div><div style=3D"font-family:ari=
al,helvetica,sans-serif;font-size:small"><br></div><div style=3D"font-famil=
y:arial,helvetica,sans-serif;font-size:small">
Option 2:</div><div style=3D"font-family:arial,helvetica,sans-serif;font-si=
ze:small">Any WebRTC entity doing full ICE - MUST<br></div><div style=3D"fo=
nt-family:arial,helvetica,sans-serif;font-size:small">
Other (WebRTC/non-WebRTC) entities - MAY<br></div><div style=3D"font-family=
:arial,helvetica,sans-serif;font-size:small">=C2=A0</div><div class=3D"gmai=
l_extra"><div style=3D"font-family:arial,helvetica,sans-serif;font-size:sma=
ll"><br>

</div></div></div></blockquote><div><br></div></div><div>To be clear, my op=
inion is</div><div><br></div><div>Any WebRTC entity implementing full ICE -=
 MUST</div><div>Any WebRTC entity (such as WebRTC gateway) implementing ICE=
-lite -- MUST NOT</div>

<div><br></div><div>This way we are not inventing entity types and not defi=
ning behaviors which would never be used (such as sending consent from an e=
nd point which would send no STUN requests otherwise). This will also reduc=
e the number of interop cases from four (full ICE with consent, full iCE wi=
thout consent, ICE-lite, ICE-lite with consent) to only two (full ICE with =
consent, ICE-lite).</div>

<div>_____________<span class=3D""><font color=3D"#888888"><br>Roman Shpoun=
t</font></span></div><div>=C2=A0</div></div></div></div>
</blockquote></div><br></div></div>

--047d7b6225ac782f1f0501751365--


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Subject: Re: [rtcweb] WG Last Call for draft-ietf-rtcweb-stun-consent-freshness
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On Mon, Aug 25, 2014 at 10:33 AM, Muthu Arul Mozhi Perumal <
muthu.arul@gmail.com> wrote:

> An ICE-lite entity by definition doesn't generate STUN requests which
> means it doesn't perform consent freshness as described in the draft. If it
> does, it is not a ICE-lite entity. You have only two types of entities to
> interop test with. I don't see any use of specifying a MUST NOT for
> ICE-lite.
>
>
>
This is exactly what I am saying. All full ICE end points for WebRTC must
implement consent freshness. We should not specify consent freshness for
ICE-lite (as it is not specified right now), and we should not define a
special type of end points which will not send consent freshness requests
just because they are ICE-lite (since they already cannot send it).
_____________
Roman Shpount

--e89a8f643366131c190501754a23
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">=
On Mon, Aug 25, 2014 at 10:33 AM, Muthu Arul Mozhi Perumal <span dir=3D"ltr=
">&lt;<a href=3D"mailto:muthu.arul@gmail.com" target=3D"_blank">muthu.arul@=
gmail.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><div dir=3D"ltr"><div style=3D"font-family:arial,helvetica=
,sans-serif;font-size:small">
An ICE-lite entity by definition doesn&#39;t generate STUN requests which m=
eans it doesn&#39;t perform consent freshness as described in the draft. If=
 it does, it is not a ICE-lite entity. You have only two types of entities =
to interop test with. I don&#39;t see any use of specifying a MUST NOT for =
ICE-lite.</div>

<div style=3D"font-family:arial,helvetica,sans-serif;font-size:small"><br><=
/div><div style=3D"font-family:arial,helvetica,sans-serif;font-size:small">=
<br></div></div></blockquote><div><br></div><div>This is exactly what I am =
saying. All full ICE end points for WebRTC must implement consent freshness=
. We should not specify consent freshness for ICE-lite (as it is not specif=
ied right now), and we should not define a special type of end points which=
 will not send consent freshness requests just because they are ICE-lite (s=
ince they already cannot send it).</div>
<div>_____________<br>Roman Shpount</div><div>=C2=A0</div></div></div></div=
>

--e89a8f643366131c190501754a23--


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Date: Mon, 25 Aug 2014 20:22:09 +0530
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From: Muthu Arul Mozhi Perumal <muthu.arul@gmail.com>
To: Roman Shpount <roman@telurix.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] WG Last Call for draft-ietf-rtcweb-stun-consent-freshness
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--f46d043c08ead9e32705017554ef
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Agree.

On Mon, Aug 25, 2014 at 8:19 PM, Roman Shpount <roman@telurix.com> wrote:

>
> On Mon, Aug 25, 2014 at 10:33 AM, Muthu Arul Mozhi Perumal <
> muthu.arul@gmail.com> wrote:
>
>> An ICE-lite entity by definition doesn't generate STUN requests which
>> means it doesn't perform consent freshness as described in the draft. If it
>> does, it is not a ICE-lite entity. You have only two types of entities to
>> interop test with. I don't see any use of specifying a MUST NOT for
>> ICE-lite.
>>
>>
>>
> This is exactly what I am saying. All full ICE end points for WebRTC must
> implement consent freshness. We should not specify consent freshness for
> ICE-lite (as it is not specified right now), and we should not define a
> special type of end points which will not send consent freshness requests
> just because they are ICE-lite (since they already cannot send it).
> _____________
> Roman Shpount
>
>

--f46d043c08ead9e32705017554ef
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<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:arial,he=
lvetica,sans-serif;font-size:small">Agree.</div><div class=3D"gmail_extra">=
<br><div class=3D"gmail_quote">On Mon, Aug 25, 2014 at 8:19 PM, Roman Shpou=
nt <span dir=3D"ltr">&lt;<a href=3D"mailto:roman@telurix.com" target=3D"_bl=
ank">roman@telurix.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div class=3D"gmail_extra">=
<br><div class=3D"gmail_quote"><div class=3D"">On Mon, Aug 25, 2014 at 10:3=
3 AM, Muthu Arul Mozhi Perumal <span dir=3D"ltr">&lt;<a href=3D"mailto:muth=
u.arul@gmail.com" target=3D"_blank">muthu.arul@gmail.com</a>&gt;</span> wro=
te:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><div dir=3D"ltr"><div style=3D"font-family:arial,helvetica=
,sans-serif;font-size:small">

An ICE-lite entity by definition doesn&#39;t generate STUN requests which m=
eans it doesn&#39;t perform consent freshness as described in the draft. If=
 it does, it is not a ICE-lite entity. You have only two types of entities =
to interop test with. I don&#39;t see any use of specifying a MUST NOT for =
ICE-lite.</div>


<div style=3D"font-family:arial,helvetica,sans-serif;font-size:small"><br><=
/div><div style=3D"font-family:arial,helvetica,sans-serif;font-size:small">=
<br></div></div></blockquote><div><br></div></div><div>This is exactly what=
 I am saying. All full ICE end points for WebRTC must implement consent fre=
shness. We should not specify consent freshness for ICE-lite (as it is not =
specified right now), and we should not define a special type of end points=
 which will not send consent freshness requests just because they are ICE-l=
ite (since they already cannot send it).</div>

<div>_____________<span class=3D"HOEnZb"><font color=3D"#888888"><br>Roman =
Shpount</font></span></div><div>=C2=A0</div></div></div></div>
</blockquote></div><br></div></div>

--f46d043c08ead9e32705017554ef--


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From: Roman Shpount <roman@telurix.com>
To: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
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Subject: Re: [rtcweb] WG Last Call for draft-ietf-rtcweb-stun-consent-freshness
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On Fri, Aug 22, 2014 at 1:40 PM, Makaraju, Maridi Raju (Raju) <
Raju.Makaraju@alcatel-lucent.com> wrote:

>  *A WebRTC-device could be a native app or in an embedded device where
> there could be some limitations about memory, cpu, energy and/or bandwidt=
h.
> So, such a device could choose not to do consent freshness checks before
> sending data. Imagine an embedded device (Internet-of-things) sends some
> non real-time data (using webrtc data channel) every 30 secs; with consen=
t
> check it first has to send a STUN req for consent and wait for response
> then send the data. In this case the consent checks add delay and overhea=
d.
> IMHO, for WebRTC-device allowing the flexibility of no consent is a good
> idea; so a =E2=80=9CSHOULD=E2=80=9D is justified. But, I am ok to make co=
nsent for
> WebRTC-device a MUST too.*
>
>
>
First of all I do not believe this use case is something that was covered
by WebRTC normal use cases  and does not comply with the consent freshness
draft (where messages need to be sent regularly regardless of data being
sent). This would mean that consent freshness will not introduce additional
delay for sending data, but it will introduce additional battery, memory,
CPU, network and other resource usage.

Second,  there is a need to maintain the communication channel through the
NAT routers which will require regular messages in both directions,
Freshness requests serve this purpose well.

Third, it is useful to know that remote party is still present and ready to
receive data. Otherwise you need to rely on message timeouts which can be
misleading.

Finally, I am not sure if p2p webrtc data channel is the best way to
implement internet of things data communications. Most likely you will need
a more efficient protocol for such an app. Such protocol should not affect
normal use cases for WebRTC.

_____________
Roman Shpount

--f46d043c80d610129b05017570c0
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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><div>On Fri, Aug 22, 2014 a=
t 1:40 PM, Makaraju, Maridi Raju (Raju) <span dir=3D"ltr">&lt;<a href=3D"ma=
ilto:Raju.Makaraju@alcatel-lucent.com" target=3D"_blank">Raju.Makaraju@alca=
tel-lucent.com</a>&gt;</span> wrote:<br>
</div><div class=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D=
"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,2=
04,204);border-left-style:solid;padding-left:1ex">





<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div>
<div style=3D"border-style:none none none solid;border-left-color:blue;bord=
er-left-width:1.5pt;padding:0in 0in 0in 4pt"><div><div><div><div><p class=
=3D"MsoNormal"><b><i><span style=3D"font-size:11pt;font-family:Calibri,sans=
-serif;color:rgb(31,73,125)">A WebRTC-device could be a native app or in an=
 embedded device where there could be some limitations about memory, cpu, e=
nergy and/or bandwidth. So, such
 a device could choose not to do consent freshness checks before sending da=
ta. Imagine an embedded device (Internet-of-things) sends some non real-tim=
e data (using webrtc data channel) every 30 secs; with consent check it fir=
st has to send a STUN req for consent
 and wait for response then send the data. In this case the consent checks =
add delay and overhead. IMHO, for WebRTC-device allowing the flexibility of=
 no consent is a good idea; so a =E2=80=9CSHOULD=E2=80=9D is justified. But=
, I am ok to make consent for WebRTC-device a MUST
 too.</span></i></b><br></p>
<p class=3D"MsoNormal"><br></p></div></div></div></div></div></div></div></=
blockquote><div><br></div><div>First of all I do not believe this use case =
is something that was covered by WebRTC normal use cases =C2=A0and does not=
 comply with the consent freshness draft (where messages need to be sent re=
gularly regardless of data being sent). This would mean that consent freshn=
ess will not introduce additional delay for sending data, but it will intro=
duce additional battery, memory, CPU, network and other resource usage.</di=
v>
<div><br></div><div>Second,=C2=A0=C2=A0there is a need to maintain the comm=
unication channel through the NAT routers which will require regular messag=
es in both directions, Freshness requests serve this purpose well.<br></div=
><div>
<br></div><div>Third, it is useful to know that remote party is still prese=
nt and ready to receive data. Otherwise you need to rely on message timeout=
s which can be misleading.</div><div><div><br></div><div>Finally, I am not =
sure if p2p webrtc data channel is the best way to implement internet of th=
ings data communications. Most likely you will need a more efficient protoc=
ol for such an app. Such protocol should not affect normal use cases for We=
bRTC.</div>
</div><div><br></div><div>_____________<br>Roman Shpount</div><div>=C2=A0</=
div></div></div></div>

--f46d043c80d610129b05017570c0--


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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Muthu Arul Mozhi Perumal <muthu.arul@gmail.com>, Roman Shpount <roman@telurix.com>
Thread-Topic: [rtcweb] WG Last Call for draft-ietf-rtcweb-stun-consent-freshness
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Date: Mon, 25 Aug 2014 15:08:30 +0000
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Subject: Re: [rtcweb] WG Last Call for draft-ietf-rtcweb-stun-consent-freshness
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--_000_E1FE4C082A89A246A11D7F32A95A17828E52EB28US70UWXCHMBA02z_--


From nobody Mon Aug 25 09:13:36 2014
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Subject: [rtcweb] I-D Action: draft-ietf-rtcweb-rtp-usage-17.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 Working Group of the IETF.

        Title           : Web Real-Time Communication (WebRTC): Media Transport and Use of RTP
        Authors         : Colin Perkins
                          Magnus Westerlund
                          Joerg Ott
	Filename        : draft-ietf-rtcweb-rtp-usage-17.txt
	Pages           : 44
	Date            : 2014-08-25

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


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-17

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-rtp-usage-17


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

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


From nobody Mon Aug 25 09:30:14 2014
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As discussed last week, this version changes =93RTCP is a fundamental =
and integral part of RTP, and MUST be implemented in all WebRTC =
applications=94 to =93=85MUST be implemented and used=85=94 in Section =
4.1. There are no other technical changes.

Colin




On 25 Aug 2014, at 17:02, internet-drafts@ietf.org wrote:
> A New Internet-Draft is available from the on-line Internet-Drafts =
directories.
> This draft is a work item of the Real-Time Communication in =
WEB-browsers Working Group of the IETF.
>=20
>        Title           : Web Real-Time Communication (WebRTC): Media =
Transport and Use of RTP
>        Authors         : Colin Perkins
>                          Magnus Westerlund
>                          Joerg Ott
> 	Filename        : draft-ietf-rtcweb-rtp-usage-17.txt
> 	Pages           : 44
> 	Date            : 2014-08-25
>=20
> Abstract:
>   The Web Real-Time Communication (WebRTC) framework provides support
>   for direct interactive rich communication using audio, video, text,
>   collaboration, games, etc.  between two peers' web-browsers.  This
>   memo describes the media transport aspects of the WebRTC framework.
>   It specifies how the Real-time Transport Protocol (RTP) is used in
>   the WebRTC context, and gives requirements for which RTP features,
>   profiles, and extensions need to be supported.
>=20
>=20
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-rtp-usage/
>=20
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-17
>=20
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-rtp-usage-17
>=20
>=20
> 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.
>=20
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>=20
> _______________________________________________
> I-D-Announce mailing list
> I-D-Announce@ietf.org
> https://www.ietf.org/mailman/listinfo/i-d-announce
> Internet-Draft directories: http://www.ietf.org/shadow.html
> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt



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




From nobody Mon Aug 25 09:51:51 2014
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From: Roman Shpount <roman@telurix.com>
To: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
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Subject: Re: [rtcweb] WG Last Call for draft-ietf-rtcweb-stun-consent-freshness
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--089e01493c5460189e0501770071
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On Mon, Aug 25, 2014 at 11:08 AM, Makaraju, Maridi Raju (Raju) <
Raju.Makaraju@alcatel-lucent.com> wrote:

>  ICE RFC 5245 says ICE-lite clients don=E2=80=99t do candidate
> gathering+connectivity checks, it does NOT explicitly talks about NOT usi=
ng
> STUN for other purpose like consent freshness.
>
> Yes, consent freshness uses the same STUN messages. That should not be a
> reason to exclude ICE-lite clients from consent freshness checks.
>
>
>
> ICE-lite endpoints have same security concerns as ICE-full endpoints.
> IMHO, independent of STUN connectivity checks, the draft should allow
> ICE-lite client to do consent freshness, which means the other end
> (ICE-full or ICE-lite) should acknowledge the STUN requests sent by
> ICE-lite clients. I believe this is already supported by Chrome and Firef=
ox.
>
>
>

If someone goes through the trouble of implementing sending STUN bind
requests needed for consent freshness, why would not they implement full
ICE? Incremental work (ie state machine) is not that great.

Sending consent freshness or any other STUN requests is not prohibited by
ICE-lite clients, but it is not defined anywhere either. Let's not define
something which has no reason to exist.
_____________
Roman Shpount

--089e01493c5460189e0501770071
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On M=
on, Aug 25, 2014 at 11:08 AM, Makaraju, Maridi Raju (Raju) <span dir=3D"ltr=
">&lt;<a href=3D"mailto:Raju.Makaraju@alcatel-lucent.com" target=3D"_blank"=
>Raju.Makaraju@alcatel-lucent.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">





<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif;color:rgb(31,73,125)">ICE RFC 5245 says ICE-lite clients don=E2=80=
=99t do candidate gathering+connectivity checks, it does NOT explicitly tal=
ks about NOT using STUN for other purpose like
 consent freshness.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif;color:rgb(31,73,125)">Yes, consent freshness uses the same STUN me=
ssages. That should not be a reason to exclude ICE-lite clients from consen=
t freshness checks.<u></u><u></u></span></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif;color:rgb(31,73,125)"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif;color:rgb(31,73,125)">ICE-lite endpoints have same security concer=
ns as ICE-full endpoints. IMHO, independent of STUN connectivity checks, th=
e draft should allow ICE-lite client
 to do consent freshness, which means the other end (ICE-full or ICE-lite) =
should acknowledge the STUN requests sent by ICE-lite clients. I believe th=
is is already supported by Chrome and Firefox.<u></u><u></u></span></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11pt;font-family:Calibri,sa=
ns-serif;color:rgb(31,73,125)"><u></u>=C2=A0</span></p></div></div></blockq=
uote><div>=C2=A0</div><div>If someone goes through the trouble of implement=
ing sending STUN bind requests needed for consent freshness, why would not =
they implement full ICE? Incremental work (ie state machine) is not that gr=
eat.</div>
<div><br></div><div>Sending consent freshness or any other STUN requests is=
 not prohibited by ICE-lite clients, but it is not defined anywhere either.=
 Let&#39;s not define something which has no reason to exist.</div><div>
_____________<br>Roman Shpount</div><div>=C2=A0</div></div></div></div>

--089e01493c5460189e0501770071--


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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
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Thread-Topic: [rtcweb] WG Last Call for draft-ietf-rtcweb-stun-consent-freshness
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--_000_E1FE4C082A89A246A11D7F32A95A17828E52F290US70UWXCHMBA02z_--


From nobody Mon Aug 25 10:16:14 2014
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Date: Mon, 25 Aug 2014 10:15:50 -0700
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From: Martin Thomson <martin.thomson@gmail.com>
To: Roman Shpount <roman@telurix.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] WG Last Call for draft-ietf-rtcweb-stun-consent-freshness
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On 25 August 2014 05:43, Roman Shpount <roman@telurix.com> wrote:
> Any WebRTC entity implementing full ICE - MUST
> Any WebRTC entity (such as WebRTC gateway) implementing ICE-lite -- MUST NOT

Let's try this instead:

Any WebRTC entity MUST implement full ICE

Any entity performing full ICE MUST implement consent

Done.


From nobody Mon Aug 25 10:55:56 2014
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Date: Mon, 25 Aug 2014 13:54:17 -0400
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From: Roman Shpount <roman@telurix.com>
To: Martin Thomson <martin.thomson@gmail.com>
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Subject: Re: [rtcweb] WG Last Call for draft-ietf-rtcweb-stun-consent-freshness
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--089e01493c542a27d7050177e047
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On Mon, Aug 25, 2014 at 1:15 PM, Martin Thomson <martin.thomson@gmail.com>
wrote:

> Let's try this instead:
>
> Any WebRTC entity MUST implement full ICE
>
> Any entity performing full ICE MUST implement consent
>
> Done.
>

We are still going to get a fair number of WebRTC gateways that implement
ICE-lite who will disagree. Unless this group will outlaw ICE-lite the way
it did SDES-SRTP (for the very similar security reasons), the caveat for
ICE-lite should stay.
_____________
Roman Shpount

--089e01493c542a27d7050177e047
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On M=
on, Aug 25, 2014 at 1:15 PM, Martin Thomson <span dir=3D"ltr">&lt;<a href=
=3D"mailto:martin.thomson@gmail.com" target=3D"_blank">martin.thomson@gmail=
.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><div>Let&#39;s try this instead:<br></div>
<br>
Any WebRTC entity MUST implement full ICE<br>
<br>
Any entity performing full ICE MUST implement consent<br>
<br>
Done.<br></blockquote><div>=C2=A0</div></div><div>We are still going to get=
 a fair number of WebRTC gateways that implement ICE-lite who will disagree=
. Unless this group will outlaw ICE-lite the way it did SDES-SRTP (for the =
very similar security reasons), the caveat for ICE-lite should stay.</div>

<div>_____________<br>Roman Shpount</div><div><br></div></div></div>

--089e01493c542a27d7050177e047--


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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Roman Shpount <roman@telurix.com>, Martin Thomson <martin.thomson@gmail.com>
Thread-Topic: [rtcweb] WG Last Call for draft-ietf-rtcweb-stun-consent-freshness
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Date: Mon, 25 Aug 2014 14:20:05 -0400
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From: Roman Shpount <roman@telurix.com>
To: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
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Subject: Re: [rtcweb] WG Last Call for draft-ietf-rtcweb-stun-consent-freshness
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--089e01493c5470da100501783c13
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On Mon, Aug 25, 2014 at 1:08 PM, Makaraju, Maridi Raju (Raju) <
Raju.Makaraju@alcatel-lucent.com> wrote:

>  *IMO, ICE gathering, connectivity checks, prioritization etc. is much
> more work than doing just consent checks.*
>
> *ICE-lite entities (please note gateways in specific) don=E2=80=99t imple=
ment
> ICE-full because they can get away with ICE-lite due to presence of publi=
c
> ip. Why complicate an implementation unnecessarily?*
>
>
If the end-point is located on a public IP, ICE gathering and
prioritization are trivial. The benefits of aggressive nomination, which is
present in full ICE, on the other hand, are quite significant. ICE-full
should be recommended anyway.


> *IMHO, consent checks for ICE-lite have a security reason to exist; the
> same security reason why ICE-full clients needed it. Isn=E2=80=99t
> draft-ietf-rtcweb-stun-consent-freshness right place to define this?*
>
>
> I would prefer no mongrel ICE end points to be defined since I would
prefer never to test against them.
_____________
Roman Shpount

--089e01493c5470da100501783c13
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On M=
on, Aug 25, 2014 at 1:08 PM, Makaraju, Maridi Raju (Raju) <span dir=3D"ltr"=
>&lt;<a href=3D"mailto:Raju.Makaraju@alcatel-lucent.com" target=3D"_blank">=
Raju.Makaraju@alcatel-lucent.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">





<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div>
<div style=3D"border-style:none none none solid;border-left-color:blue;bord=
er-left-width:1.5pt;padding:0in 0in 0in 4pt"><div><div><div><div><p class=
=3D"MsoNormal"><b><i><span style=3D"font-size:11pt;font-family:Calibri,sans=
-serif;color:rgb(31,73,125)">IMO, ICE gathering, connectivity checks, prior=
itization etc. is much more work than doing just consent checks.<u></u><u><=
/u></span></i></b></p>


<p class=3D"MsoNormal"><b><i><span style=3D"font-size:11pt;font-family:Cali=
bri,sans-serif;color:rgb(31,73,125)">ICE-lite entities (please note gateway=
s in specific) don=E2=80=99t implement ICE-full because they can get away w=
ith ICE-lite due to presence of public ip. Why
 complicate an implementation unnecessarily?<u></u><u></u></span></i></b></=
p>
<p class=3D"MsoNormal"></p></div></div></div></div></div></div></div></bloc=
kquote><div><br></div><div>If the end-point is located on a public IP, ICE =
gathering and prioritization are trivial. The benefits of aggressive nomina=
tion, which is present in full ICE, on the other hand, are quite significan=
t. ICE-full should be recommended anyway.=C2=A0</div>

<div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px =
0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-l=
eft-style:solid;padding-left:1ex"><div lang=3D"EN-US" link=3D"blue" vlink=
=3D"purple">
<div>
<div style=3D"border-style:none none none solid;border-left-color:blue;bord=
er-left-width:1.5pt;padding:0in 0in 0in 4pt"><div><div><div><div><p class=
=3D"MsoNormal"><b><i><span style=3D"font-size:11pt;font-family:Calibri,sans=
-serif;color:rgb(31,73,125)">IMHO, consent checks for ICE-lite have a secur=
ity reason to exist; the same security reason why ICE-full clients needed i=
t. Isn=E2=80=99t draft-ietf-rtcweb-stun-consent-freshness
 right place to define this?</span></i></b><br></p></div><div>
<p class=3D"MsoNormal"><br></p></div></div></div></div></div></div></div></=
blockquote><div>I would prefer no mongrel ICE end points to be defined sinc=
e I would prefer never to test against them.</div><div>_____________<br>
Roman Shpount</div><div>=C2=A0</div></div></div></div>

--089e01493c5470da100501783c13--


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From: Martin Thomson <martin.thomson@gmail.com>
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On 25 August 2014 11:01, Makaraju, Maridi Raju (Raju)
<Raju.Makaraju@alcatel-lucent.com> wrote:
> I don=E2=80=99t think outlawing ICE-lite is acceptable.

I've not ever suggested that.  I'm only suggesting that ICE-lite =E2=88=89
WebRTC.  (That's U+2209, btw).

WebRTC devices might talk to non-WebRTC devices.  Some of those might
only do the minimal part of ICE that they can get away with.

Still, I don't see any reason to change any text here.


From nobody Mon Aug 25 14:30:22 2014
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From: "Rauschenbach, Uwe (NSN - DE/Munich)" <uwe.rauschenbach@nsn.com>
To: ext Martin Thomson <martin.thomson@gmail.com>, Roman Shpount <roman@telurix.com>
Thread-Topic: [rtcweb] WG Last Call for draft-ietf-rtcweb-stun-consent-freshness
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References: <CA+9kkMCZT1XW4LLaJ4Nq2DbrxD59cYnjLo5JXn9fjEb8pyamaQ@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D41CDC3@ESESSMB209.ericsson.se> <CAKz0y8zycsyr9m4BA=-8xOaWkU+Sog5Mbz7K-oN3woqi++mVzg@mail.gmail.com> <53F451CF.10705@alvestrand.no> <001b01cfbc94$fccd5310$f667f930$@co.in> <CAKz0y8zNM3rc3XC6JqrK+d4hXiT5TomhNM+W2twg0+-83-pFow@mail.gmail.com> <CABkgnnUnfB5bskH4zWRfBMdHbSoqftV5Fo_GEXoLt9XCH9Tt_w@mail.gmail.com> <CAD5OKxsT9Vdm0=tjk9WsLAH4ekbAizgyjm--168TrOf8UAYGZw@mail.gmail.com> <CABkgnnXUpibu8kWYmbJJJT2J3RNGXFV8LbceLijgG0U-pGY2xQ@mail.gmail.com> <CAKz0y8z_oBf2efavfOLgzqE1R8sZstefZ1tvwwJLkhRskXZERQ@mail.gmail.com> <CAD5OKxsSqA=cki_fSaqAPP0GXCv_kHr6571C+K9ze4ceHCGYdQ@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D427B68@ESESSMB209.ericsson.se> <D01D6B42.104F8%rmohanr@cisco.com> <009001cfbe4f$6b92f280$42b8d780$@co.in> <CAKz0y8y3=0_-jwWiviR_Uj6tSq75NEq92Ocergc_NvFk2h_72Q@mail.gmail.com> <CAD5OKxugaPq=cLppPUqT8CUADV-E2zDMcz6m3sUgKOnK-TdQfQ@mail.gmail.com>, <CABkgnnU6kv+=jQuxj4Ci-zYW9mHEfJ=6jqqrdqDTopPN06-kgw@mail.gmail.com>
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Subject: Re: [rtcweb] WG Last Call for draft-ietf-rtcweb-stun-consent-freshness
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I am not sure whether the -consent-freshness draft is the right place to re=
quire conformance to full ICE in order for something to be called a WebRTC =
endpoint. There are other drafts like -overview or -transport for this. The=
 -consent-freshness draft should only make the link between implemented ICE=
 flavor and implications for consent freshness support.

I therefore prefer Roman's proposal, but I am open to relaxing the MUST NOT=
.

Kind regards,
Uwe
________________________________
Von: ext Martin Thomson<mailto:martin.thomson@gmail.com>
Gesendet: =FD25.=FD08.=FD2014 19:16
An: Roman Shpount<mailto:roman@telurix.com>
Cc: rtcweb@ietf.org<mailto:rtcweb@ietf.org>
Betreff: Re: [rtcweb] WG Last Call for draft-ietf-rtcweb-stun-consent-fresh=
ness

On 25 August 2014 05:43, Roman Shpount <roman@telurix.com> wrote:
> Any WebRTC entity implementing full ICE - MUST
> Any WebRTC entity (such as WebRTC gateway) implementing ICE-lite -- MUST =
NOT

Let's try this instead:

Any WebRTC entity MUST implement full ICE

Any entity performing full ICE MUST implement consent

Done.

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

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<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dwindows-1=
256">
<meta name=3D"Generator" content=3D"Microsoft Exchange Server">
<!-- converted from text --><style><!-- .EmailQuote { margin-left: 1pt; pad=
ding-left: 4pt; border-left: #800000 2px solid; } --></style>
</head>
<body>
<div>
<div>
<div style=3D"font-size:11pt; font-family:Calibri,sans-serif">I am not sure=
 whether the -consent-freshness draft is the right place to require conform=
ance to full ICE in order for something to be called a WebRTC endpoint. The=
re are other drafts like -overview
 or -transport for this. The -consent-freshness draft should only make the =
link between implemented ICE flavor and implications for consent freshness =
support.<br>
<br>
I therefore prefer Roman's proposal, but I am open to relaxing the MUST NOT=
.<br>
<br>
Kind regards,<br>
Uwe </div>
</div>
<div dir=3D"ltr">
<hr>
<span style=3D"font-size:11pt; font-family:Calibri,sans-serif; font-weight:=
bold">Von:
</span><span style=3D"font-size:11pt; font-family:Calibri,sans-serif"><a hr=
ef=3D"mailto:martin.thomson@gmail.com">ext Martin Thomson</a></span><br>
<span style=3D"font-size:11pt; font-family:Calibri,sans-serif; font-weight:=
bold">Gesendet:
</span><span style=3D"font-size:11pt; font-family:Calibri,sans-serif">=FD25=
.=FD08.=FD2014 19:16</span><br>
<span style=3D"font-size:11pt; font-family:Calibri,sans-serif; font-weight:=
bold">An:
</span><span style=3D"font-size:11pt; font-family:Calibri,sans-serif"><a hr=
ef=3D"mailto:roman@telurix.com">Roman Shpount</a></span><br>
<span style=3D"font-size:11pt; font-family:Calibri,sans-serif; font-weight:=
bold">Cc:
</span><span style=3D"font-size:11pt; font-family:Calibri,sans-serif"><a hr=
ef=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a></span><br>
<span style=3D"font-size:11pt; font-family:Calibri,sans-serif; font-weight:=
bold">Betreff:
</span><span style=3D"font-size:11pt; font-family:Calibri,sans-serif">Re: [=
rtcweb] WG Last Call for draft-ietf-rtcweb-stun-consent-freshness</span><br=
>
<br>
</div>
</div>
<font size=3D"2"><span style=3D"font-size:10pt;">
<div class=3D"PlainText">On 25 August 2014 05:43, Roman Shpount &lt;roman@t=
elurix.com&gt; wrote:<br>
&gt; Any WebRTC entity implementing full ICE - MUST<br>
&gt; Any WebRTC entity (such as WebRTC gateway) implementing ICE-lite -- MU=
ST NOT<br>
<br>
Let's try this instead:<br>
<br>
Any WebRTC entity MUST implement full ICE<br>
<br>
Any entity performing full ICE MUST implement consent<br>
<br>
Done.<br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
rtcweb@ietf.org<br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.o=
rg/mailman/listinfo/rtcweb</a><br>
</div>
</span></font>
</body>
</html>

--_000_56C2F665D49E0341B9DF5938005ACDF819485063DEMUMBX005nsnin_--


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Date: Tue, 26 Aug 2014 09:44:50 +0530
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From: Muthu Arul Mozhi Perumal <muthu.arul@gmail.com>
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Subject: Re: [rtcweb] WG Last Call for draft-ietf-rtcweb-stun-consent-freshness
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On Mon, Aug 25, 2014 at 10:45 PM, Martin Thomson <martin.thomson@gmail.com>
wrote:

> On 25 August 2014 05:43, Roman Shpount <roman@telurix.com> wrote:
> > Any WebRTC entity implementing full ICE - MUST
> > Any WebRTC entity (such as WebRTC gateway) implementing ICE-lite -- MUS=
T
> NOT
>
> Let's try this instead:
>
> Any WebRTC entity MUST implement full ICE
>

=E2=80=8BThe transport draft already has the following:

=E2=80=8B   ICE [RFC5245] MUST be supported.  The implementation MUST be a =
full
   ICE implementation, not ICE-Lite.  A full ICE implementation allows
   interworking with both ICE and ICE-Lite implementations when they are
   deployed appropriately.

=E2=80=8BIt also says:

   This document focuses on the data transport protocols that are used
   by conforming implementations, including the protocols used for
   interaction with intermediate boxes such as firewalls, relays and NAT
   boxes.

"Conforming implementations" seem to mean any WebRTC entity. However, the
following limits it to only WebRTC devices (and leaves out WebRTC gateway,
in particular):

   This document describes requirements that apply to all WebRTC
   devices.=E2=80=8B

We should probably sort this out the "full ICE" requirement in the
transport draft.


> Any entity performing full ICE MUST implement consent
>
>
I guess you mean any WebRTC entity (there may be other SIP entities doing
full ICE and nothing to do with WebRTC). If so, it works for me..

Muthu=E2=80=8B



> Done.
>

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<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:arial,he=
lvetica,sans-serif;font-size:small"><span style=3D"font-family:arial">On Mo=
n, Aug 25, 2014 at 10:45 PM, Martin Thomson </span><span dir=3D"ltr" style=
=3D"font-family:arial">&lt;<a href=3D"mailto:martin.thomson@gmail.com" targ=
et=3D"_blank">martin.thomson@gmail.com</a>&gt;</span><span style=3D"font-fa=
mily:arial"> wrote:</span><br>
</div><div class=3D"gmail_extra"><div class=3D"gmail_quote"><blockquote cla=
ss=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;=
border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex=
"><div class=3D"">
On 25 August 2014 05:43, Roman Shpount &lt;<a href=3D"mailto:roman@telurix.=
com">roman@telurix.com</a>&gt; wrote:<br>
&gt; Any WebRTC entity implementing full ICE - MUST<br>
&gt; Any WebRTC entity (such as WebRTC gateway) implementing ICE-lite -- MU=
ST NOT<br>
<br>
</div>Let&#39;s try this instead:<br>
<br>
Any WebRTC entity MUST implement full ICE<br></blockquote><div>=C2=A0</div>=
<div><div class=3D"gmail_default" style=3D"font-family:arial,helvetica,sans=
-serif;font-size:small">=E2=80=8BThe transport draft already has the follow=
ing:</div><div class=3D"gmail_default" style=3D"font-family:arial,helvetica=
,sans-serif;font-size:small">
<br></div><div class=3D"gmail_default" style=3D"font-family:arial,helvetica=
,sans-serif;font-size:small">=E2=80=8B =C2=A0 ICE [RFC5245] MUST be support=
ed. =C2=A0The implementation MUST be a full<br></div><div class=3D"gmail_de=
fault" style=3D"font-family:arial,helvetica,sans-serif">
=C2=A0 =C2=A0ICE implementation, not ICE-Lite. =C2=A0A full ICE implementat=
ion allows</div><div class=3D"gmail_default" style=3D"font-family:arial,hel=
vetica,sans-serif">=C2=A0 =C2=A0interworking with both ICE and ICE-Lite imp=
lementations when they are</div>
<div class=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-seri=
f">=C2=A0 =C2=A0deployed appropriately.</div><br></div><div><div class=3D"g=
mail_default" style=3D"font-family:arial,helvetica,sans-serif;font-size:sma=
ll">=E2=80=8BIt also says:</div>
<div class=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-seri=
f;font-size:small"><br></div><div class=3D"gmail_default" style=3D"font-fam=
ily:arial,helvetica,sans-serif;font-size:small">=C2=A0 =C2=A0This document =
focuses on the data transport protocols that are used<br>
</div><div class=3D"gmail_default" style><font face=3D"arial, helvetica, sa=
ns-serif">=C2=A0 =C2=A0by conforming implementations, including the protoco=
ls used for</font></div><div class=3D"gmail_default" style><font face=3D"ar=
ial, helvetica, sans-serif">=C2=A0 =C2=A0interaction with intermediate boxe=
s such as firewalls, relays and NAT</font></div>
<div class=3D"gmail_default" style><font face=3D"arial, helvetica, sans-ser=
if">=C2=A0 =C2=A0boxes.</font></div><div class=3D"gmail_default" style><fon=
t face=3D"arial, helvetica, sans-serif"><br></font></div><div class=3D"gmai=
l_default" style>
<span style=3D"font-family:arial,helvetica,sans-serif">&quot;Conforming imp=
lementations&quot; seem to mean any WebRTC entity. However, the following l=
imits it to only WebRTC devices (and leaves out WebRTC gateway, in particul=
ar):</span></div>
<div class=3D"gmail_default" style><font face=3D"arial, helvetica, sans-ser=
if"><br></font></div><div class=3D"gmail_default" style><font face=3D"arial=
, helvetica, sans-serif">=C2=A0 =C2=A0This document describes requirements =
that apply to all WebRTC</font></div>
<div class=3D"gmail_default" style><font face=3D"arial, helvetica, sans-ser=
if">=C2=A0 =C2=A0devices.</font><span style=3D"font-family:arial,helvetica,=
sans-serif;font-size:small">=E2=80=8B</span></div></div><div class=3D"gmail=
_default" style><span style=3D"font-family:arial,helvetica,sans-serif;font-=
size:small"><br>
</span></div><div class=3D"gmail_default" style><span style=3D"font-family:=
arial,helvetica,sans-serif;font-size:small">We should probably sort this ou=
t the &quot;full ICE&quot; requirement in the transport draft.</span></div>
<div><br></div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0p=
x 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-lef=
t-style:solid;padding-left:1ex">
<br>
Any entity performing full ICE MUST implement consent<br>
<br></blockquote><div><br></div><div><div class=3D"gmail_default" style=3D"=
font-family:arial,helvetica,sans-serif;font-size:small">I guess you mean an=
y WebRTC entity (there may be other SIP entities doing full ICE and nothing=
 to do with WebRTC). If so, it works for me..=C2=A0</div>
<div class=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-seri=
f;font-size:small"><br></div><div class=3D"gmail_default" style=3D"font-fam=
ily:arial,helvetica,sans-serif;font-size:small">Muthu=E2=80=8B</div><br></d=
iv><div>=C2=A0</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex">
Done.<br>
</blockquote></div><br></div></div>

--001a11c3536a7a46380501808b02--


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Date: Tue, 26 Aug 2014 10:37:41 +0530
Message-ID: <CAKz0y8zO2JSNH8RV_n3qBFgQfA=g9nvHrwH0gJZPvhzaMATByw@mail.gmail.com>
From: Muthu Arul Mozhi Perumal <muthu.arul@gmail.com>
To: "Rauschenbach, Uwe (NSN - DE/Munich)" <uwe.rauschenbach@nsn.com>
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Subject: Re: [rtcweb] WG Last Call for draft-ietf-rtcweb-stun-consent-freshness
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On Tue, Aug 26, 2014 at 3:00 AM, Rauschenbach, Uwe (NSN - DE/Munich) <
uwe.rauschenbach@nsn.com> wrote:

>   I am not sure whether the -consent-freshness draft is the right place
> to require conformance to full ICE in order for something to be called a
> WebRTC endpoint.
>

=E2=80=8BAgree. That should probably go into the transport draft,


> There are other drafts like -overview or -transport for this. The
> -consent-freshness draft should only make the link between implemented IC=
E
> flavor and implications for consent freshness support.
>
> I therefore prefer Roman's proposal, but I am open to relaxing the MUST
> NOT.
>

=E2=80=8BRoman later clarified he just wants the consent draft not to speci=
fy any
additional requirement for ICE-lite. ICE-lite entities respond to consent
checks by definition. That's all.

Muthu=E2=80=8B


>
> Kind regards,
> Uwe
>  ------------------------------
> Von: ext Martin Thomson <martin.thomson@gmail.com>
> Gesendet: =E2=80=8E25.=E2=80=8E08.=E2=80=8E2014 19:16
> An: Roman Shpount <roman@telurix.com>
> Cc: rtcweb@ietf.org
> Betreff: Re: [rtcweb] WG Last Call for
> draft-ietf-rtcweb-stun-consent-freshness
>
>   On 25 August 2014 05:43, Roman Shpount <roman@telurix.com> wrote:
> > Any WebRTC entity implementing full ICE - MUST
> > Any WebRTC entity (such as WebRTC gateway) implementing ICE-lite -- MUS=
T
> NOT
>
> Let's try this instead:
>
> Any WebRTC entity MUST implement full ICE
>
> Any entity performing full ICE MUST implement consent
>
> Done.
>
> _______________________________________________
> 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
>
>

--001a11c240ea7980e6050181480f
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<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:arial,he=
lvetica,sans-serif;font-size:small"><span style=3D"font-family:arial">On Tu=
e, Aug 26, 2014 at 3:00 AM, Rauschenbach, Uwe (NSN - DE/Munich) </span><spa=
n dir=3D"ltr" style=3D"font-family:arial">&lt;<a href=3D"mailto:uwe.rausche=
nbach@nsn.com" target=3D"_blank">uwe.rauschenbach@nsn.com</a>&gt;</span><sp=
an style=3D"font-family:arial"> wrote:</span><br>

</div><div class=3D"gmail_extra"><div class=3D"gmail_quote"><blockquote cla=
ss=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;pa=
dding-left:1ex">





<div>
<div>
<div>
<div style=3D"font-size:11pt;font-family:Calibri,sans-serif">I am not sure =
whether the -consent-freshness draft is the right place to require conforma=
nce to full ICE in order for something to be called a WebRTC endpoint.</div=
>

</div></div></div></blockquote><div><br></div><div><div class=3D"gmail_defa=
ult" style=3D"font-family:arial,helvetica,sans-serif;font-size:small">=E2=
=80=8BAgree. That should probably go into the transport draft,</div></div><=
div>=C2=A0</div>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div><div><div><div style=3D"font-size:11pt;=
font-family:Calibri,sans-serif"> There are other drafts like -overview
 or -transport for this. The -consent-freshness draft should only make the =
link between implemented ICE flavor and implications for consent freshness =
support.<br>
<br>
I therefore prefer Roman&#39;s proposal, but I am open to relaxing the MUST=
 NOT.<br></div></div></div></div></blockquote><div><br></div><div><div clas=
s=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-serif;font-si=
ze:small">
=E2=80=8BRoman later clarified he just wants the consent draft not to speci=
fy any additional requirement for ICE-lite. ICE-lite entities respond to co=
nsent checks by definition. That&#39;s all.</div><div class=3D"gmail_defaul=
t" style=3D"font-family:arial,helvetica,sans-serif;font-size:small">
<br></div><div class=3D"gmail_default" style=3D"font-family:arial,helvetica=
,sans-serif;font-size:small">Muthu=E2=80=8B</div></div><div>=C2=A0</div><bl=
ockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #=
ccc solid;padding-left:1ex">

<div><div><div><div style=3D"font-size:11pt;font-family:Calibri,sans-serif"=
>
<br>
Kind regards,<br>
Uwe </div>
</div>
<div dir=3D"ltr">
<hr>
<span style=3D"font-size:11pt;font-family:Calibri,sans-serif;font-weight:bo=
ld">Von:
</span><span style=3D"font-size:11pt;font-family:Calibri,sans-serif"><a hre=
f=3D"mailto:martin.thomson@gmail.com" target=3D"_blank">ext Martin Thomson<=
/a></span><br>
<span style=3D"font-size:11pt;font-family:Calibri,sans-serif;font-weight:bo=
ld">Gesendet:
</span><span style=3D"font-size:11pt;font-family:Calibri,sans-serif">=E2=80=
=8E25.=E2=80=8E08.=E2=80=8E2014 19:16</span><br>
<span style=3D"font-size:11pt;font-family:Calibri,sans-serif;font-weight:bo=
ld">An:
</span><span style=3D"font-size:11pt;font-family:Calibri,sans-serif"><a hre=
f=3D"mailto:roman@telurix.com" target=3D"_blank">Roman Shpount</a></span><b=
r>
<span style=3D"font-size:11pt;font-family:Calibri,sans-serif;font-weight:bo=
ld">Cc:
</span><span style=3D"font-size:11pt;font-family:Calibri,sans-serif"><a hre=
f=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a></span><b=
r>
<span style=3D"font-size:11pt;font-family:Calibri,sans-serif;font-weight:bo=
ld">Betreff:
</span><span style=3D"font-size:11pt;font-family:Calibri,sans-serif">Re: [r=
tcweb] WG Last Call for draft-ietf-rtcweb-stun-consent-freshness</span><br>
<br>
</div>
</div>
<font><span style=3D"font-size:10pt">
<div><div><div>On 25 August 2014 05:43, Roman Shpount &lt;<a href=3D"mailto=
:roman@telurix.com" target=3D"_blank">roman@telurix.com</a>&gt; wrote:<br>
&gt; Any WebRTC entity implementing full ICE - MUST<br>
&gt; Any WebRTC entity (such as WebRTC gateway) implementing ICE-lite -- MU=
ST NOT<br>
<br>
Let&#39;s try this instead:<br>
<br>
Any WebRTC entity MUST implement full ICE<br>
<br>
Any entity performing full ICE MUST implement consent<br>
<br>
Done.<br>
<br></div></div><div>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div>
</span></font>
</div>

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

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From: Jeremy Fuller <jeremy.fuller@genband.com>
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Thread-Topic: [rtcweb] WebRTC gateway - draft-ietf-rtcweb-overview-11.txt
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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Martin Thomson <martin.thomson@gmail.com>
Thread-Topic: [rtcweb] WG Last Call for draft-ietf-rtcweb-stun-consent-freshness
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From: Muthu Arul Mozhi Perumal <muthu.arul@gmail.com>
To: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
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Subject: Re: [rtcweb] WG Last Call for draft-ietf-rtcweb-stun-consent-freshness
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--f46d043c07de3e85700501895c40
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On Tue, Aug 26, 2014 at 4:29 PM, Makaraju, Maridi Raju (Raju) <
Raju.Makaraju@alcatel-lucent.com> wrote:

> > On 25 August 2014 11:01, Makaraju, Maridi Raju (Raju)
> > <Raju.Makaraju@alcatel-lucent.com> wrote:
> > > I don=E2=80=99t think outlawing ICE-lite is acceptable.
> >
> > I've not ever suggested that.  I'm only suggesting that ICE-lite =E2=88=
=89
> > WebRTC.  (That's U+2209, btw).
> <Raju>
> WebRTC is defined by more than just ICE connectivity checks.
> IMHO, Categorizing a class of devices as NOT WebRTC just because of
> ICE-lite is not right, while there is so much other functionality (codecs=
,
> data channels) that is common. Also, the converse is not true either; jus=
t
> because a class of devices support ICE-full does not equate them to WebRT=
C
> class.
> So, IMHO there is no merit in (re)classifying WebRTC devices around
> ICE-lite.
> I also want to point out that ICE-lite is part of ICE RFC and also WebRTC
> clients already have a way to negotiate ICE-lite.
>
> In addition, draft-ietf-rtcweb-overview-11 added definition for 3 classes
> of webrtc devices with varying level of conformance while keeping the cor=
e
> webrtc conformance common. Won't rest of the specs like consent freshness=
,
> transports etc. refer to these classes of webrtc devices instead of just
> concentrating on just one type of webrtc device?
>

=E2=80=8BConsent freshness is applicable to any WebRTC entity supporting fu=
ll ICE.
A WebRTC browser/device as defined in the transport and overview drafts
support full ICE, so perform consent freshness. The requirements for other
WebRTC entities, including WebRTC gateway, is yet to be fully specified,
and the document that specifies them would also specify whether they
support full ICE or not (and hence whether they perform consent freshness
or not).

I don't see a problem with this approach.

If you have a better approach, do suggest the text.

Muthu =E2=80=8B


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

--f46d043c07de3e85700501895c40
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<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:arial,he=
lvetica,sans-serif;font-size:small"><span style=3D"font-family:arial">On Tu=
e, Aug 26, 2014 at 4:29 PM, Makaraju, Maridi Raju (Raju) </span><span dir=
=3D"ltr" style=3D"font-family:arial">&lt;<a href=3D"mailto:Raju.Makaraju@al=
catel-lucent.com" target=3D"_blank">Raju.Makaraju@alcatel-lucent.com</a>&gt=
;</span><span style=3D"font-family:arial"> wrote:</span><br>
</div><div class=3D"gmail_extra"><div class=3D"gmail_quote"><blockquote cla=
ss=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;pa=
dding-left:1ex"><div class=3D"">&gt; On 25 August 2014 11:01, Makaraju, Mar=
idi Raju (Raju)<br>

&gt; &lt;<a href=3D"mailto:Raju.Makaraju@alcatel-lucent.com">Raju.Makaraju@=
alcatel-lucent.com</a>&gt; wrote:<br>
&gt; &gt; I don=E2=80=99t think outlawing ICE-lite is acceptable.<br>
&gt;<br>
&gt; I&#39;ve not ever suggested that.=C2=A0 I&#39;m only suggesting that I=
CE-lite =E2=88=89<br>
&gt; WebRTC.=C2=A0 (That&#39;s U+2209, btw).<br>
</div>&lt;Raju&gt;<br>
WebRTC is defined by more than just ICE connectivity checks.<br>
IMHO, Categorizing a class of devices as NOT WebRTC just because of ICE-lit=
e is not right, while there is so much other functionality (codecs, data ch=
annels) that is common. Also, the converse is not true either; just because=
 a class of devices support ICE-full does not equate them to WebRTC class.<=
br>

So, IMHO there is no merit in (re)classifying WebRTC devices around ICE-lit=
e.<br>
I also want to point out that ICE-lite is part of ICE RFC and also WebRTC c=
lients already have a way to negotiate ICE-lite.<br>
<br>
In addition, draft-ietf-rtcweb-overview-11 added definition for 3 classes o=
f webrtc devices with varying level of conformance while keeping the core w=
ebrtc conformance common. Won&#39;t rest of the specs like consent freshnes=
s, transports etc. refer to these classes of webrtc devices instead of just=
 concentrating on just one type of webrtc device?<br>
</blockquote><div><br></div><div><div class=3D"gmail_default" style=3D"font=
-family:arial,helvetica,sans-serif;font-size:small">=E2=80=8BConsent freshn=
ess is applicable to any WebRTC entity supporting full ICE. A WebRTC browse=
r/device as defined in the transport and overview drafts support full ICE, =
so perform consent freshness. The requirements for other WebRTC entities, i=
ncluding WebRTC gateway, is yet to be fully specified, and the document tha=
t specifies them would also specify whether they support full ICE or not (a=
nd hence whether they perform consent freshness or not).</div>
<div class=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-seri=
f;font-size:small"><br></div><div class=3D"gmail_default" style=3D"font-fam=
ily:arial,helvetica,sans-serif;font-size:small">I don&#39;t see a problem w=
ith this approach.</div>
<div class=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-seri=
f;font-size:small"><br></div><div class=3D"gmail_default" style=3D"font-fam=
ily:arial,helvetica,sans-serif;font-size:small">If you have a better approa=
ch, do suggest the text.</div>
<div class=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-seri=
f;font-size:small"><br></div><div class=3D"gmail_default" style=3D"font-fam=
ily:arial,helvetica,sans-serif;font-size:small">Muthu =E2=80=8B</div></div>=
<div>=C2=A0</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<br>
&lt;/Raju&gt;<br>
<br>
BR<br>
<span class=3D"HOEnZb"><font color=3D"#888888">Raju<br>
</font></span><div class=3D"HOEnZb"><div class=3D"h5"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div></div>

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From: Martin Thomson <martin.thomson@gmail.com>
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Subject: Re: [rtcweb] WG Last Call for draft-ietf-rtcweb-stun-consent-freshness
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On 26 August 2014 07:45, Muthu Arul Mozhi Perumal <muthu.arul@gmail.com> wrote:
> Consent freshness is applicable to any WebRTC entity supporting full ICE. A
> WebRTC browser/device as defined in the transport and overview drafts
> support full ICE, so perform consent freshness. The requirements for other
> WebRTC entities, including WebRTC gateway, is yet to be fully specified, and
> the document that specifies them would also specify whether they support
> full ICE or not (and hence whether they perform consent freshness or not).


Actually, that's probably the best approach: note that this applies -
and can be useful for - any ICE implementation.  And leave it at that
for this document.  The transports or security docs might make some
statements about mandating this mechanism.


From nobody Tue Aug 26 12:22:39 2014
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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Muthu Arul Mozhi Perumal <muthu.arul@gmail.com>
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From nobody Tue Aug 26 12:39:41 2014
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On 08 Aug 2014, at 22:27, Richard Barnes <rlb@ipv.sx> wrote:

> I have reviewed this document in preparation for IETF Last Call.  I =
have gone ahead and requested last call, but there are some comments =
below that I would like to be addressed before the document goes to the =
IESG.
> --Richard
Richard,

thank you very much for the review. See my comments in-line.

Best regards
Michael
>=20
>=20
> MAJOR:
>=20
> Section 6.2.=20
> I'm very concerned about interoperability of data channels given that, =
e.g., (1) DCEP [draft-ietf-rtcweb-data-protocol] is not required, and =
(2) DCEP only works if it is used by both endpoints (it resets any =
streams that are used without DCEP negotiation).  In order to avoid =
interoperability=20
Streams are only reset, if they are already used.
> failure of this type, whatever mechanism is used to establish an SCTP =
association that is used for data channels must also negotiate the =
mechanism(s) that will be used to establish data channels over that =
association.  That could be directly (by negotiating some data channels =
at the same time as the association), or indirectly (by indicating that =
DCEP or some other protocol will be used), but either way, the endpoints =
of the association need to know how data channels will be negotiated =
before they can create any data channels.  This section needs to state =
this requirement so that applications using data channels can know that =
they need to meet it.
I don't think they need it at the time the association is created, they =
need this information
at the time a data channel is created. This is described in the first =
paragraph of section 6.5.
What about changing there:
   The details are out of scope of this document.
to
   The details are out of scope of this document.  Applications using
   data channels need to use the negotiation methods consistently on
   both end-points.
>=20
> Section 6.4.=20
> This section lists in passing several properties of a data channel, =
but doesn't give a full definition of all the things that need to be =
defined in order to define a data channel.  Fortunately, such a list is =
given in draft-ietf-rtcweb-data-protocol (Section 4); I would suggest =
moving that here.
I agree, this is much cleaner in draft-ietf-rtcweb-data-protocol. What =
about using

6.4.  WebRTC Data Channel Definition

   One strong wish is for the application-level API to be close to the
   API for WebSockets, which implies bidirectional streams of data and a
   textual field called 'label' used to identify the meaning of the data
   channel.

   The realization of a data channel is a pair of one incoming stream
   and one outgoing SCTP stream having the same SCTP stream identifier.
   How these SCTP stream identifiers are selected is protocol and
   implementation dependent.  This allows a bidirectional communication.

   Additionally, each data channel has the following properties in each
   direction:

   o  reliable or unreliable message transmission.  In case of
      unreliable transmissions, the same level of unreliability is used.
      Please note that in SCTP this is a property of an SCTP user
      message and not of an SCTP stream.

   o  in-order or out-of-order message delivery for message sent.
      Please note that in SCTP this is a property of an SCTP user
      message and not of an SCTP stream.

   o  A priority, which is a 2 byte unsigned integer.  These priorities
      MUST be interpreted as weighted-fair-queuing scheduling priorities
      per the definition of the corresponding stream scheduler
      supporting interleaving in [I-D.ietf-tsvwg-sctp-ndata].  For use
      in WebRTC, the values used SHOULD be one of 128 ("below normal"),
      256 ("normal"), 512 ("high") or 1024 ("extra high").

   o  an optional label.

   o  an optional protocol.

   Please note that for a data channel being negotiated with the
   protocol specified in [I-D.ietf-rtcweb-data-protocol] all of the
   above properties are the same in both directions.

Does this address your comment?
>=20
>=20
>=20
> MINOR / EDITORIAL:
>=20
> Figures 1 and 2 list ICE as a separate layer.  That seems incorrect, =
since ICE isn't actually involved once the connection is set up.
Doesn't ICE continue to run connectivity checks continuously and even =
allows changes
of the IP addresses being used for the communication?
>=20
> Section 6.4. "... in-sequence, out-of-sequence, reliable and =
unreliable as properties of channels"
> "... (as opposed to SCTP, which applies these concepts to messages)"
See above.
>=20
> Section 6.4. "... and waiting for onopen to fire ... meaning of the =
streams"
> This sentence doesn't really parse, and it's not clear why these =
API-level considerations are relevant here.
See above.
>=20
> Section 6.4. "stream SCTP identifier" -> "SCTP stream identifier" ?
Changed.
>=20
> Section 6.6. "The sender SHOULD disable the Nagle algorithm..."
> A reference to RFC 1122 would be helpful here.
OK. Added as an Informative Reference.
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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On 11 Aug 2014, at 20:42, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:

> I just did another read-through of this draft, and identified a few =
things that I think need to be cleared up:
Hi Paul,

thank you very much for your comments. See my answers in-line.

Best regards
Michael
>=20
> Association Usage:
>=20
> draft-ietf-mmusic-sctp-sdp specifies new SDP notation for describing =
SCTP associations. It is not specific to rtcweb. For SCTP m-lines, it is =
necessary to name an "association-usage". Support for rtcweb data =
channels is one possible usage.
>=20
> Currently draft-ietf-mmusic-sctp-sdp gives all of its examples using =
"webrtc-datachannel" as the association-usage. It does *not* currently =
define an IANA registry for association-usage, but it should. But it =
really is the wrong place to define this specific usage. Certainly =
draft-ietf-rtcweb-data-channel provides the *definition* of the usage. =
IMO it also ought to define and do the IANA registration of the name. =
(Obviously some coordination is required between this draft and the =
sctp-sdp draft, so that they are aligned on this.)
I really think that SDP related stuff should be in the MMUSIC ID =
including the IANA registrations.
>=20
> Also, can we PLEASE use a name that not specific to webrtc? This =
*will* be used in contexts where webrtc is not being used. (At least by =
CLUE.)
>=20
> I propose that the name be "data-channel".
Anyone against it? I could change the name consistently in both RTCWEB =
IDs.
>=20
> SCTP port:
>=20
> *Something* needs to talk about SCTP ports. Currently they aren't =
mentioned either here or in JSEP. Since this document makes the point =
that this will be a user layer stack, and talks about the layering over =
DTLS, I think they ought to be mentioned here. They also probably need =
to be mentioned in JSEP, as part of aligning to the most recent version =
of draft-ietf-mmusic-sctp-sdp.
>=20
> Only a little bit needs to be stated: that SCTP supports multiple SCTP =
ports, but that a particular SCTP association uses a single sctp port =
pair. (One port at each end of the SCTP association.) JSEP needs to =
specify how these ports are negotiated. (Via reference to =
draft-ietf-mmusic-sctp-sdp.)
What about adding
   SCTP uses the same port number concept as TCP and UDP do.  Therefore
   an SCTP association uses two port numbers, one at each SCTP end-
   point.
at the end of Section 5.
>=20
> Normative requirement for DCEP support:
>=20
> While this draft references ietf-rtcweb-data-protocol normatively, and =
talks about it, I can find no explicit statement that implementations of =
this draft MUST also implement DCEP. AFAIK negotiation of the sctp =
association-usage defined by this draft in intended to mean that DCEP is =
also supported.
If I remember it correctly, the decision was to take such a statement =
out of the document...
Such a statement was in the second paragraph of
http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-07#section-6.5
and was taken out. See
http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-08#section-6.5
>=20
> 	Thanks,
> 	Paul
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


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On 16 Aug 2014, at 08:44, Gunnar Hellstrom <gunnar.hellstrom@omnitor.se> =
wrote:

> We have discussed the lack of description of failure handling in =
draft-ietf-rtcweb-data-channel a couple of times.
>=20
> Explanations of mechanisms have been provided in mail answers that I =
have not seen documented elsewhere, but they have not been introduced in =
the draft.
>=20
> I think that for specification and implementation of the WebRTC API, =
there is a need for clear specification how the channels behave in =
failure situations.
>=20
> I suggest to insert a new chapter 6.7 in the draft with the following =
contents. =20
Hi Gunnar,

see my comments in-line.

Best regards
Michael
>=20
> =
-------------------------------------------------------------------Propose=
d new section =
6.7-----------------------------------------------------------------------=
--------------------
> 6.7 Transmission failure and error handling=20
>=20
> Failures can occur in an open channel by transmission error detection =
and by SCTP watchdog error indications.
> If transmission in a reliable channel fails, the association where the =
channel belongs will be aborted. If all retries for a watchdog expires, =
the corresponding association will be aborted.  All channels in an =
association that is aborted because of errors need to be closed by the =
party detecting the failure. Network error indications shall be provided =
to upper layers on all closed channels.
I think we could change Section
6.2.  Association Setup
to
6.2.  Association Handling
and state there that if an association is terminated, for example due to =
error cases, all data channels are closed.
>=20
> If retransmissions or transmission time is exceeded in a partially =
reliable channel, the transmission SHALL be allowed to continue with =
other data items. A transmission error indication SHALL be provided to =
upper layers on that channel.=20
I think the handling of the PR-SCTP stuff is clear. Whether this should =
be exposed in the API is a W3C issue, I think.
>=20
> If reception out-of order is detected from an SCTP channel for which =
ordered transmission is requested, the receiver SHALL close the data =
channel, and provide an transmission error indication to upper layers. =20=

I think we discussed this at the IETF and the conclusion was that this =
kind of check is not done.
>=20
> If a message with an unsupported PPID is received or some logical =
error is detected by the receiver, the receiver SHALL close the =
corresponding data channel.=20
This text is already in Section 6.6.
> =
-----------------------------------------------------------------------end=
 of =
proposal------------------------------------------------------------------=
---------------------------------------------

>=20
> I am not sure about what the two last kinds of errors should result in =
and hope that the experts can adjust the wording.
>=20
> Regards
>=20
> Gunnar
> Gunnar Hellstr=F6m
> Omnitor
> gunnar.hellstrom@omnitor.se
>=20
> On 2014-08-11 20:13, IESG Secretary wrote:
>> The IESG would like to retract the current Last Call on the following=20=

>> document:
>>=20
>> - 'WebRTC Data Channels'
>>   <draft-ietf-rtcweb-data-channel-11.txt> as Proposed Standard
>>=20
>> The document's state has been set back to "AD Evaluation."
>>=20
>>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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On 16 Aug 2014, at 14:01, Barry Dingle <btdingle@gmail.com> wrote:

> I agree that error handling is missing from =
draft-ietf-rtcweb-data-channel .=20
>=20
> Should the 'error' indications' in the proposed text be more specific? =
That is, should the error indications indicate the type of error. (The =
same error indication for all errors could be sent and be compliant with =
the currently proposed text.)=20
There is no way in transporting an error indication...

Best regards
Michael
>=20
> Note - I assume that you are proposing adding the new Section 6.7 to  =
draft-ietf-rtcweb-data-channel-11 . It already has a Section 6.7 titled =
"Closing a channel". I assume you mean a new Section 6.8.=20
>=20
> Cheers,
> /Barry Dingle
>=20
>=20
>=20
>=20
> On Sat, Aug 16, 2014 at 4:44 PM, Gunnar Hellstrom =
<gunnar.hellstrom@omnitor.se> wrote:
> We have discussed the lack of description of failure handling in =
draft-ietf-rtcweb-data-channel a couple of times.
>=20
> Explanations of mechanisms have been provided in mail answers that I =
have not seen documented elsewhere, but they have not been introduced in =
the draft.
>=20
> I think that for specification and implementation of the WebRTC API, =
there is a need for clear specification how the channels behave in =
failure situations.
>=20
> I suggest to insert a new chapter 6.7 in the draft with the following =
contents. =20
>=20
> =
-------------------------------------------------------------------Propose=
d new section =
6.7-----------------------------------------------------------------------=
--------------------
> 6.7 Transmission failure and error handling=20
>=20
> Failures can occur in an open channel by transmission error detection =
and by SCTP watchdog error indications.
> If transmission in a reliable channel fails, the association where the =
channel belongs will be aborted. If all retries for a watchdog expires, =
the corresponding association will be aborted.  All channels in an =
association that is aborted because of errors need to be closed by the =
party detecting the failure. Network error indications shall be provided =
to upper layers on all closed channels.
>=20
> If retransmissions or transmission time is exceeded in a partially =
reliable channel, the transmission SHALL be allowed to continue with =
other data items. A transmission error indication SHALL be provided to =
upper layers on that channel.=20
>=20
> If reception out-of order is detected from an SCTP channel for which =
ordered transmission is requested, the receiver SHALL close the data =
channel, and provide an transmission error indication to upper layers. =20=

>=20
> If a message with an unsupported PPID is received or some logical =
error is detected by the receiver, the receiver SHALL close the =
corresponding data channel.      =20
> =
-----------------------------------------------------------------------end=
 of =
proposal------------------------------------------------------------------=
---------------------------------------------
>=20
> I am not sure about what the two last kinds of errors should result in =
and hope that the experts can adjust the wording.
>=20
> Regards
>=20
> Gunnar
> Gunnar Hellstr=F6m
> Omnitor
> gunnar.hellstrom@omnitor.se
>=20
> On 2014-08-11 20:13, IESG Secretary wrote:
>> The IESG would like to retract the current Last Call on the following=20=

>> document:
>>=20
>> - 'WebRTC Data Channels'
>>   <draft-ietf-rtcweb-data-channel-11.txt> as Proposed Standard
>>=20
>> The document's state has been set back to "AD Evaluation."
>>=20
>>=20
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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On 18 Aug 2014, at 13:15, Harald Alvestrand <harald@alvestrand.no> =
wrote:

> Picking up a single point....
>=20
> On 08/16/2014 08:44 AM, Gunnar Hellstrom wrote:
>>=20
>> If a message with an unsupported PPID is received or some logical =
error is detected by the receiver, the receiver SHALL close the =
corresponding data channel.=20
>=20
> This struct me when thinking about the zero-length-data proposal: If =
we ever want to add another such feature *without* explicitly =
negotiating the use of the feature at setup time, we can't do this.
>=20
> If we want the possibility of un-negotiated extensions, we HAVE to say =
(somewhere)
>=20
>   If a message with an unsupported PPID is received, it is silently =
ignored.
>=20
> OTOH, if we're convinced we never need that kind of extensibility, any =
way we do this is fine.
Harald,

I think it is important to specify how unsupported PPIDs handled. Either =
close the data channel
or ignore the message.

What I don't understand how ignoring gives you extensibility. If one =
sides sends a new PPID, and
the other sides ignores it, the sender (at the SCTP level) thinks that =
the message was received
without any problem, since SCTP doesn't take the PPID into account. So =
user messages are lost.

Best regards
Michael
>=20
>              Harald
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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From: Martin Thomson <martin.thomson@gmail.com>
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Subject: Re: [rtcweb] Unsupported PPID (Re: draft-ietf-rtcweb-data-channel failure handling description)
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On 26 August 2014 13:36, Michael Tuexen
<Michael.Tuexen@lurchi.franken.de> wrote:
> What I don't understand how ignoring gives you extensibility. If one sides sends a new PPID, and
> the other sides ignores it, the sender (at the SCTP level) thinks that the message was received
> without any problem, since SCTP doesn't take the PPID into account. So user messages are lost.

I think that the use of a new PPID can be negotiated prior to use, so
there aren't big problems with any action recommended here.  Resetting
the stream might work, assuming that we can distinguish between an
error and a graceful close correctly, that is.


From nobody Tue Aug 26 14:23:30 2014
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Subject: Re: [rtcweb] draft-ietf-rtcweb-data-channel failure handling description
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On 2014-08-26 22:26, Michael Tuexen wrote:
> On 16 Aug 2014, at 08:44, Gunnar Hellstrom <gunnar.hellstrom@omnitor.se> wrote:
>
>> We have discussed the lack of description of failure handling in draft-ietf-rtcweb-data-channel a couple of times.
>>
>> Explanations of mechanisms have been provided in mail answers that I have not seen documented elsewhere, but they have not been introduced in the draft.
>>
>> I think that for specification and implementation of the WebRTC API, there is a need for clear specification how the channels behave in failure situations.
>>
>> I suggest to insert a new chapter 6.7 in the draft with the following contents.
> Hi Gunnar,
>
> see my comments in-line.
Hi Michael
and see my answers in-line
>
> Best regards
> Michael
>> -------------------------------------------------------------------Proposed new section 6.7-------------------------------------------------------------------------------------------
>> 6.7 Transmission failure and error handling
>>
>> Failures can occur in an open channel by transmission error detection and by SCTP watchdog error indications.
>> If transmission in a reliable channel fails, the association where the channel belongs will be aborted. If all retries for a watchdog expires, the corresponding association will be aborted.  All channels in an association that is aborted because of errors need to be closed by the party detecting the failure. Network error indications shall be provided to upper layers on all closed channels.
> I think we could change Section
> 6.2.  Association Setup
> to
> 6.2.  Association Handling
> and state there that if an association is terminated, for example due to error cases, all data channels are closed.
I meant 6.8.
I think a failure section is better placed after all sections on 
opening, closing and data transfer. Then it starts to be clear to the 
reader what can go wrong.
And I think failure handling is so important that it is worth having a 
section on its own with a heading that can be found in the contents.

When you say that all data channels are closed, can you then also tell 
how that is indicated to the application.

I also think that the basic reasons for Association termination by error 
should be explained briefly in the data-channel specification, so that 
the reader does not need to dig through the SCTP RFC to try to find 
them.  The watchdog failures and the exess retries in reliable channels, 
etc.   I cannot understand why you want to avoid explaining the mechanisms.
>> If retransmissions or transmission time is exceeded in a partially reliable channel, the transmission SHALL be allowed to continue with other data items. A transmission error indication SHALL be provided to upper layers on that channel.
> I think the handling of the PR-SCTP stuff is clear. Whether this should be exposed in the API is a W3C issue, I think.
I do not think it is clear. It merely mentions the names of the types of 
channels. The failure cases need also to be described.
And, it is a case for the API description and the data-channel spec to 
be aligned. What the W3C API spec specify will happen in case of failure 
needs to be supported by the data-channel spec. I have made proposals to 
WebRTC as well. Please align.
>> If reception out-of order is detected from an SCTP channel for which ordered transmission is requested, the receiver SHALL close the data channel, and provide an transmission error indication to upper layers.
> I think we discussed this at the IETF and the conclusion was that this kind of check is not done.
Well, why not describe what happens, so that further questions or 
misunderstandings are avoided?  You mean that what SCTP provides is also 
what the data-channel provides.

>> If a message with an unsupported PPID is received or some logical error is detected by the receiver, the receiver SHALL close the corresponding data channel.
> This text is already in Section 6.6.
Yes, there is a paragraph saying:
"

    If a message with an unsupported PPID is received or some error is
    detected by the receiver (for example, illegal ordering), the
    receiver SHOULD close the corresponding data channel.
"
Since my proposal was to create a separate failure handling section 6.8, my proposal would be to move all failure descriptions to
6.8, also the ones from 6.6.

A question: You usually say that the Association is aborted in case of error and all its channels closed.
But here it is said that just the channel is closed.  Please clarify.


Regards
/Gunnar

>> -----------------------------------------------------------------------end of proposal---------------------------------------------------------------------------------------------------------------
>> I am not sure about what the two last kinds of errors should result in and hope that the experts can adjust the wording.
>>
>> Regards
>>
>> Gunnar
>> Gunnar Hellström
>> Omnitor
>> gunnar.hellstrom@omnitor.se
>>
>> On 2014-08-11 20:13, IESG Secretary wrote:
>>> The IESG would like to retract the current Last Call on the following
>>> document:
>>>
>>> - 'WebRTC Data Channels'
>>>    <draft-ietf-rtcweb-data-channel-11.txt> as Proposed Standard
>>>
>>> The document's state has been set back to "AD Evaluation."
>>>
>>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb


From nobody Tue Aug 26 14:41:35 2014
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Subject: Re: [rtcweb] Unsupported PPID (Re: draft-ietf-rtcweb-data-channel failure handling description)
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On 26 Aug 2014, at 23:22, Martin Thomson <martin.thomson@gmail.com> =
wrote:

> On 26 August 2014 13:36, Michael Tuexen
> <Michael.Tuexen@lurchi.franken.de> wrote:
>> What I don't understand how ignoring gives you extensibility. If one =
sides sends a new PPID, and
>> the other sides ignores it, the sender (at the SCTP level) thinks =
that the message was received
>> without any problem, since SCTP doesn't take the PPID into account. =
So user messages are lost.
>=20
> I think that the use of a new PPID can be negotiated prior to use, so
> there aren't big problems with any action recommended here.  Resetting
> the stream might work, assuming that we can distinguish between an
> error and a graceful close correctly, that is.
You can't. It is just closed, there is error indication included.

Best regards
Michael
>=20


From nobody Tue Aug 26 14:44:39 2014
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Subject: Re: [rtcweb] Unsupported PPID (Re: draft-ietf-rtcweb-data-channel failure handling description)
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On 8/26/14 5:22 PM, Martin Thomson wrote:
> On 26 August 2014 13:36, Michael Tuexen
> <Michael.Tuexen@lurchi.franken.de> wrote:
>> What I don't understand how ignoring gives you extensibility. If one sides sends a new PPID, and
>> the other sides ignores it, the sender (at the SCTP level) thinks that the message was received
>> without any problem, since SCTP doesn't take the PPID into account. So user messages are lost.
>
> I think that the use of a new PPID can be negotiated prior to use, so
> there aren't big problems with any action recommended here.  Resetting
> the stream might work, assuming that we can distinguish between an
> error and a graceful close correctly, that is.

There isn't currently any way to distinguish graceful close from a reset 
due to error is there?

PPID 50 is currently being used just for DATA_CHANNEL_OPEN and 
Data_CHANNEL_ACK. But the message format has made provision for 512 
different message types. Some others could be used, now or in the 
future, for other housekeeping, such as giving a reason for a reset.

One approach would be to define DATA_CHANNEL_CLOSE, and specify that it 
be used just before reset when the close is graceful.

	Thanks,
	Paul


From nobody Tue Aug 26 15:01:07 2014
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Subject: Re: [rtcweb] Unsupported PPID (Re: draft-ietf-rtcweb-data-channel failure handling description)
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On 26 August 2014 14:44, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
> One approach would be to define DATA_CHANNEL_CLOSE, and specify that it be
> used just before reset when the close is graceful.

Or define a DATA_CHANNEL_ERROR, which would let you explain why.
Assuming that SCTP doesn't suffer from the same problems TCP does with
sending just prior to a close.


From nobody Wed Aug 27 04:08:12 2014
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Subject: Re: [rtcweb] Unsupported PPID (Re: draft-ietf-rtcweb-data-channel failure handling description)
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On 26 Aug 2014, at 23:44, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:

> On 8/26/14 5:22 PM, Martin Thomson wrote:
>> On 26 August 2014 13:36, Michael Tuexen
>> <Michael.Tuexen@lurchi.franken.de> wrote:
>>> What I don't understand how ignoring gives you extensibility. If one =
sides sends a new PPID, and
>>> the other sides ignores it, the sender (at the SCTP level) thinks =
that the message was received
>>> without any problem, since SCTP doesn't take the PPID into account. =
So user messages are lost.
>>=20
>> I think that the use of a new PPID can be negotiated prior to use, so
>> there aren't big problems with any action recommended here.  =
Resetting
>> the stream might work, assuming that we can distinguish between an
>> error and a graceful close correctly, that is.
>=20
> There isn't currently any way to distinguish graceful close from a =
reset due to error is there?
>=20
> PPID 50 is currently being used just for DATA_CHANNEL_OPEN and =
Data_CHANNEL_ACK. But the message format has made provision for 512 =
different message types. Some others could be used, now or in the =
future, for other housekeeping, such as giving a reason for a reset.
>=20
> One approach would be to define DATA_CHANNEL_CLOSE, and specify that =
it be used just before reset when the close is graceful.
What you are suggesting is to use the Data Channel Establishment =
Protocol also
for error signalling... This doesn't fit since for the error it doesn't =
matter
whether the channel was established with DCEP or by any other mechanism.

The way would be to define error messages with a different PPID and =
define the
message format. If I remember it correctly, we discussed this at an IETF =
meeting
and the conclusion was to not have such messages...
Is the WG now changing its mind? After WG LC?

Best regards
Michael
>=20
> 	Thanks,
> 	Paul
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


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Subject: Re: [rtcweb] Unsupported PPID (Re: draft-ietf-rtcweb-data-channel failure handling description)
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On 27 Aug 2014, at 00:01, Martin Thomson <martin.thomson@gmail.com> wrote:

> On 26 August 2014 14:44, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
>> One approach would be to define DATA_CHANNEL_CLOSE, and specify that it be
>> used just before reset when the close is graceful.
> 
> Or define a DATA_CHANNEL_ERROR, which would let you explain why.
This would require defining a simple packet format and register a PPID for
it. However, this was discussed in the past and the result was that no
error messages were wanted... Is this conclusion not valid anymore?
> Assuming that SCTP doesn't suffer from the same problems TCP does with
> sending just prior to a close.
What TCP problems are you referring to?

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


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Subject: Re: [rtcweb] Unsupported PPID (Re: draft-ietf-rtcweb-data-channel failure handling description)
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On 08/26/2014 11:22 PM, Martin Thomson wrote:
> On 26 August 2014 13:36, Michael Tuexen
> <Michael.Tuexen@lurchi.franken.de> wrote:
>> What I don't understand how ignoring gives you extensibility. If one sides sends a new PPID, and
>> the other sides ignores it, the sender (at the SCTP level) thinks that the message was received
>> without any problem, since SCTP doesn't take the PPID into account. So user messages are lost.
> I think that the use of a new PPID can be negotiated prior to use, so
> there aren't big problems with any action recommended here.  Resetting
> the stream might work, assuming that we can distinguish between an
> error and a graceful close correctly, that is.

If there is negotiation, there is no problem.

If there is no negotiation, starting to use the new feature will either 
lead to ignored messages or lead to a crashed connection.

Ignored messages are not a problem if they form part of an ignorable 
extension (for instance if someone decided to once in a while send 
checksums over all the data sent so far - doesn't matter to those that 
don't understand it, since it doesn't carry any extra data).

Crashed connections are usually a problem, so that will mean "no 
unnegotiated extensions, ever".

I'm happy with either way of doing it, as long as we're explicit about it.




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Subject: Re: [rtcweb] Unsupported PPID (Re: draft-ietf-rtcweb-data-channel failure handling description)
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> If there is negotiation, there is no problem.
>=20
> If there is no negotiation, starting to use the new feature will either
> lead to ignored messages or lead to a crashed connection.
>=20
> Ignored messages are not a problem if they form part of an ignorable
> extension (for instance if someone decided to once in a while send
> checksums over all the data sent so far - doesn't matter to those that
> don't understand it, since it doesn't carry any extra data).
>=20
> Crashed connections are usually a problem, so that will mean "no
> unnegotiated extensions, ever".
>=20
> I'm happy with either way of doing it, as long as we're explicit about it=
.
<Raju>
+1 for negotiation.
I think it is highly preferred to negotiate these additional PPIDs in all c=
ases as=20
anything other than a binary or text  PPID is additional capability. Use of=
 any additional
capabilities must at least be negotiated using JSEP.=20
One element of negotiation could be "what must be done for unrecognized PPI=
Ds?": ignore them or
abort the connection.

</Raju>


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Subject: Re: [rtcweb] draft-ietf-rtcweb-data-channel failure handling description
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On 26 Aug 2014, at 23:23, Gunnar Hellstrom <gunnar.hellstrom@omnitor.se> =
wrote:

> On 2014-08-26 22:26, Michael Tuexen wrote:
>> On 16 Aug 2014, at 08:44, Gunnar Hellstrom =
<gunnar.hellstrom@omnitor.se> wrote:
>>=20
>>> We have discussed the lack of description of failure handling in =
draft-ietf-rtcweb-data-channel a couple of times.
>>>=20
>>> Explanations of mechanisms have been provided in mail answers that I =
have not seen documented elsewhere, but they have not been introduced in =
the draft.
>>>=20
>>> I think that for specification and implementation of the WebRTC API, =
there is a need for clear specification how the channels behave in =
failure situations.
>>>=20
>>> I suggest to insert a new chapter 6.7 in the draft with the =
following contents.
>> Hi Gunnar,
>>=20
>> see my comments in-line.
> Hi Michael
> and see my answers in-line
>>=20
>> Best regards
>> Michael
>>> =
-------------------------------------------------------------------Propose=
d new section =
6.7-----------------------------------------------------------------------=
--------------------
>>> 6.7 Transmission failure and error handling
>>>=20
>>> Failures can occur in an open channel by transmission error =
detection and by SCTP watchdog error indications.
>>> If transmission in a reliable channel fails, the association where =
the channel belongs will be aborted. If all retries for a watchdog =
expires, the corresponding association will be aborted.  All channels in =
an association that is aborted because of errors need to be closed by =
the party detecting the failure. Network error indications shall be =
provided to upper layers on all closed channels.
>> I think we could change Section
>> 6.2.  Association Setup
>> to
>> 6.2.  Association Handling
>> and state there that if an association is terminated, for example due =
to error cases, all data channels are closed.
> I meant 6.8.
> I think a failure section is better placed after all sections on =
opening, closing and data transfer. Then it starts to be clear to the =
reader what can go wrong.
> And I think failure handling is so important that it is worth having a =
section on its own with a heading that can be found in the contents.
OK. What do others think? Should error handling be moved into a separate =
section?
>=20
> When you say that all data channels are closed, can you then also tell =
how that is indicated to the application.
I consider this an API issue. When looking at
http://dev.w3.org/2011/webrtc/editor/webrtc.html
wouldn't that mean to fire onclose() on all data channels of the peer =
connection?
>=20
> I also think that the basic reasons for Association termination by =
error should be explained briefly in the data-channel specification, so =
that the reader does not need to dig through the SCTP RFC to try to find =
them.  The watchdog failures and the exess retries in reliable channels, =
etc.   I cannot understand why you want to avoid explaining the =
mechanisms.
Could go into a failure section, if the WG agrees that we want to have =
some... Any opinions?
>>> If retransmissions or transmission time is exceeded in a partially =
reliable channel, the transmission SHALL be allowed to continue with =
other data items. A transmission error indication SHALL be provided to =
upper layers on that channel.
>> I think the handling of the PR-SCTP stuff is clear. Whether this =
should be exposed in the API is a W3C issue, I think.
> I do not think it is clear. It merely mentions the names of the types =
of channels. The failure cases need also to be described.
> And, it is a case for the API description and the data-channel spec to =
be aligned. What the W3C API spec specify will happen in case of failure =
needs to be supported by the data-channel spec. I have made proposals to =
WebRTC as well. Please align.
=46rom an SCTP point of view, you can get a notification at the sender =
side if a message is abandoned, but not at the receiver side.
I don't see that such an information is exposed to the JS application...
>>> If reception out-of order is detected from an SCTP channel for which =
ordered transmission is requested, the receiver SHALL close the data =
channel, and provide an transmission error indication to upper layers.
>> I think we discussed this at the IETF and the conclusion was that =
this kind of check is not done.
> Well, why not describe what happens, so that further questions or =
misunderstandings are avoided?  You mean that what SCTP provides is also =
what the data-channel provides.
So isn't the text cited by you below not enough?
>=20
>>> If a message with an unsupported PPID is received or some logical =
error is detected by the receiver, the receiver SHALL close the =
corresponding data channel.
>> This text is already in Section 6.6.
> Yes, there is a paragraph saying:
> "
>=20
>   If a message with an unsupported PPID is received or some error is
>   detected by the receiver (for example, illegal ordering), the
>   receiver SHOULD close the corresponding data channel.
> "
> Since my proposal was to create a separate failure handling section =
6.8, my proposal would be to move all failure descriptions to
> 6.8, also the ones from 6.6.
Understood. Can we get some agreement if we want this or not in the WG?
>=20
> A question: You usually say that the Association is aborted in case of =
error and all its channels closed.
> But here it is said that just the channel is closed.  Please clarify.
That is a good point... When I wrote this sentence, I was thinking about =
errors like
illegal ordering (which means the receiver expects ordered but got =
unordered, or expects
reliable transfer, but sees SSN holes...). I was not thinking about an =
indication that
the association is gone. So what about:

<t>If a message with an unsupported PPID is received or some error =
condition
related to the received message is detected by the receiver
(for example, illegal ordering),
the receiver SHOULD close the corresponding data channel.</t>

Best regards
Michael
>=20
>=20
> Regards
> /Gunnar
>=20
>>> =
-----------------------------------------------------------------------end=
 of =
proposal------------------------------------------------------------------=
---------------------------------------------
>>> I am not sure about what the two last kinds of errors should result =
in and hope that the experts can adjust the wording.
>>>=20
>>> Regards
>>>=20
>>> Gunnar
>>> Gunnar Hellstr=F6m
>>> Omnitor
>>> gunnar.hellstrom@omnitor.se
>>>=20
>>> On 2014-08-11 20:13, IESG Secretary wrote:
>>>> The IESG would like to retract the current Last Call on the =
following
>>>> document:
>>>>=20
>>>> - 'WebRTC Data Channels'
>>>>   <draft-ietf-rtcweb-data-channel-11.txt> as Proposed Standard
>>>>=20
>>>> The document's state has been set back to "AD Evaluation."
>>>>=20
>>>>=20
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
>=20


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Subject: Re: [rtcweb] Unsupported PPID (Re: draft-ietf-rtcweb-data-channel failure handling description)
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On 27 Aug 2014, at 13:16, Harald Alvestrand <harald@alvestrand.no> =
wrote:

> On 08/26/2014 11:22 PM, Martin Thomson wrote:
>> On 26 August 2014 13:36, Michael Tuexen
>> <Michael.Tuexen@lurchi.franken.de> wrote:
>>> What I don't understand how ignoring gives you extensibility. If one =
sides sends a new PPID, and
>>> the other sides ignores it, the sender (at the SCTP level) thinks =
that the message was received
>>> without any problem, since SCTP doesn't take the PPID into account. =
So user messages are lost.
>> I think that the use of a new PPID can be negotiated prior to use, so
>> there aren't big problems with any action recommended here.  =
Resetting
>> the stream might work, assuming that we can distinguish between an
>> error and a graceful close correctly, that is.
>=20
> If there is negotiation, there is no problem.
>=20
> If there is no negotiation, starting to use the new feature will =
either lead to ignored messages or lead to a crashed connection.
>=20
> Ignored messages are not a problem if they form part of an ignorable =
extension (for instance if someone decided to once in a while send =
checksums over all the data sent so far - doesn't matter to those that =
don't understand it, since it doesn't carry any extra data).
>=20
> Crashed connections are usually a problem, so that will mean "no =
unnegotiated extensions, ever".
>=20
> I'm happy with either way of doing it, as long as we're explicit about =
it.
I agree with you, don't have a preference. Up to now the consensus was =
to close the channel...
I can add a sentence to make clear that this means no unnegotiated  =
extensions.

Best regards
Michael
>=20
>=20
>=20
>=20


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Subject: Re: [rtcweb] Unsupported PPID (Re: draft-ietf-rtcweb-data-channel failure handling description)
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On 27 Aug 2014, at 13:38, Makaraju, Maridi Raju (Raju) =
<Raju.Makaraju@alcatel-lucent.com> wrote:

>> If there is negotiation, there is no problem.
>>=20
>> If there is no negotiation, starting to use the new feature will =
either
>> lead to ignored messages or lead to a crashed connection.
>>=20
>> Ignored messages are not a problem if they form part of an ignorable
>> extension (for instance if someone decided to once in a while send
>> checksums over all the data sent so far - doesn't matter to those =
that
>> don't understand it, since it doesn't carry any extra data).
>>=20
>> Crashed connections are usually a problem, so that will mean "no
>> unnegotiated extensions, ever".
>>=20
>> I'm happy with either way of doing it, as long as we're explicit =
about it.
> <Raju>
> +1 for negotiation.
> I think it is highly preferred to negotiate these additional PPIDs in =
all cases as=20
> anything other than a binary or text  PPID is additional capability. =
Use of any additional
> capabilities must at least be negotiated using JSEP.=20
> One element of negotiation could be "what must be done for =
unrecognized PPIDs?": ignore them or
> abort the connection.
Do we really need to negotiate that? The consensus up to now was to =
close the channel.
So if you negotiate the PPIDs and honor the result, you will never run =
into this issue...

Best regards
Michael
>=20
> </Raju>
>=20
>=20


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From: Martin Thomson <martin.thomson@gmail.com>
To: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
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Subject: Re: [rtcweb] Unsupported PPID (Re: draft-ietf-rtcweb-data-channel failure handling description)
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On 27 August 2014 04:09, Michael Tuexen
<Michael.Tuexen@lurchi.franken.de> wrote:
> This would require defining a simple packet format and register a PPID for
> it. However, this was discussed in the past and the result was that no
> error messages were wanted... Is this conclusion not valid anymore?

I have no recollection of discussing this, but will admit to not
reading every email on the topic.

>> Assuming that SCTP doesn't suffer from the same problems TCP does with
>> sending just prior to a close.
> What TCP problems are you referring to?

The sort where you send data followed immediately by a FIN (or RST)
and find that the data disappeared.  c.f.,
http://tools.ietf.org/html/rfc7230#page-57 (top)


From nobody Wed Aug 27 12:09:38 2014
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From: "Makaraju, Maridi Raju (Raju)" <Raju.Makaraju@alcatel-lucent.com>
To: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
Thread-Topic: [rtcweb] Unsupported PPID (Re: draft-ietf-rtcweb-data-channel failure handling description)
Thread-Index: AQHPwW1+7TUn3aZI7UycBQe10zH565vjp8uAgADpMwD//79ZMIAAW3uAgAAlAEA=
Date: Wed, 27 Aug 2014 19:08:47 +0000
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Subject: Re: [rtcweb] Unsupported PPID (Re: draft-ietf-rtcweb-data-channel failure handling description)
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> Do we really need to negotiate that? The consensus up to now was to close
> the channel.
> So if you negotiate the PPIDs and honor the result, you will never run in=
to
> this issue...
<Raju>
I like your response to Herald: "I can add a sentence to make clear that th=
is means no unnegotiated  extensions."
So, all un-negotiated extensions results in closing the data channel; which=
 leaves possibility for
negotiating extensions in future and handling them appropriately.
</Raju>


From nobody Wed Aug 27 12:40:56 2014
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Subject: Re: [rtcweb] Unsupported PPID (Re: draft-ietf-rtcweb-data-channel failure handling description)
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On 27 Aug 2014, at 20:46, Martin Thomson <martin.thomson@gmail.com> =
wrote:

> On 27 August 2014 04:09, Michael Tuexen
> <Michael.Tuexen@lurchi.franken.de> wrote:
>> This would require defining a simple packet format and register a =
PPID for
>> it. However, this was discussed in the past and the result was that =
no
>> error messages were wanted... Is this conclusion not valid anymore?
>=20
> I have no recollection of discussing this, but will admit to not
> reading every email on the topic.
I think this was discussed at an IETF meeting... Basically the questions =
was
if we want to define any protocol for data channels. DCEP is not covered =
by
this question, since it is a protocol to set channels up.
>=20
>>> Assuming that SCTP doesn't suffer from the same problems TCP does =
with
>>> sending just prior to a close.
>> What TCP problems are you referring to?
>=20
> The sort where you send data followed immediately by a FIN (or RST)
> and find that the data disappeared.  c.f.,
> http://tools.ietf.org/html/rfc7230#page-57 (top)
No, SCTP does not have this problem when resetting streams. Messages get =
delivered.
However, what can happen is that when using an unordered channel, that =
the message
sent last before resetting the channel, is not the one received as the =
last message
before the SCTP Stream Reset request is received...

Best regards
Michael
>=20


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On 08/27/2014 08:46 PM, Martin Thomson wrote:
> On 27 August 2014 04:09, Michael Tuexen
> <Michael.Tuexen@lurchi.franken.de> wrote:
>> This would require defining a simple packet format and register a PPID for
>> it. However, this was discussed in the past and the result was that no
>> error messages were wanted... Is this conclusion not valid anymore?
> I have no recollection of discussing this, but will admit to not
> reading every email on the topic.
>
>>> Assuming that SCTP doesn't suffer from the same problems TCP does with
>>> sending just prior to a close.
>> What TCP problems are you referring to?
> The sort where you send data followed immediately by a FIN (or RST)
> and find that the data disappeared.  c.f.,
> http://tools.ietf.org/html/rfc7230#page-57 (top)
Re TCP: I think the RFC 7230 problem is an application (or API) problem,
not a protocol problem.

This only happens if a data receiver uses the POSIX close(3) interface
to close a socket, instead of shutting it down gently (with shutdown(2)
using SHUT_WR, I think).

Anyone who sends RST is OK with data being lost. RST is a signal that
"I'm shutting this stuff down, I don't care whether your data arrived or
not".



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


-- 
Surveillance is pervasive. Go Dark.


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Subject: Re: [rtcweb] Unsupported PPID (Re: draft-ietf-rtcweb-data-channel failure handling description)
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There is an ambition to be as close as possible to Websocket.
Websocket is specified in RFC 6455.
It has an negotiation mechanism for extensions.  See chapter 9
And it closes channels in case of reception tht is not according to the 
protocol. ( e.g. Section 10.7 )

So, it could be wise to copy that behavior, to maintain the ambitions of 
keeping these data channel types similar.

/Gunnar

------------------------------------------------------------------------
On 2014-08-27 14:52, Michael Tuexen wrote:
> On 27 Aug 2014, at 13:38, Makaraju, Maridi Raju (Raju) <Raju.Makaraju@alcatel-lucent.com> wrote:
>
>>> If there is negotiation, there is no problem.
>>>
>>> If there is no negotiation, starting to use the new feature will either
>>> lead to ignored messages or lead to a crashed connection.
>>>
>>> Ignored messages are not a problem if they form part of an ignorable
>>> extension (for instance if someone decided to once in a while send
>>> checksums over all the data sent so far - doesn't matter to those that
>>> don't understand it, since it doesn't carry any extra data).
>>>
>>> Crashed connections are usually a problem, so that will mean "no
>>> unnegotiated extensions, ever".
>>>
>>> I'm happy with either way of doing it, as long as we're explicit about it.
>> <Raju>
>> +1 for negotiation.
>> I think it is highly preferred to negotiate these additional PPIDs in all cases as
>> anything other than a binary or text  PPID is additional capability. Use of any additional
>> capabilities must at least be negotiated using JSEP.
>> One element of negotiation could be "what must be done for unrecognized PPIDs?": ignore them or
>> abort the connection.
> Do we really need to negotiate that? The consensus up to now was to close the channel.
> So if you negotiate the PPIDs and honor the result, you will never run into this issue...
>
> Best regards
> Michael
>> </Raju>
>>
>>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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<html>
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    <meta content="text/html; charset=ISO-8859-1"
      http-equiv="Content-Type">
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  <body bgcolor="#FFFFFF" text="#000000">
    <div class="moz-cite-prefix">There is an ambition to be as close as
      possible to Websocket.<br>
      Websocket is specified in RFC 6455.<br>
      It has an negotiation mechanism for extensions.&nbsp; See chapter 9<br>
      And it closes channels in case of reception tht is not according
      to the protocol. ( e.g. Section 10.7 )<br>
      <br>
      So, it could be wise to copy that behavior, to maintain the
      ambitions of keeping these data channel types similar.<br>
      <br>
      /Gunnar<br>
      <br>
      <div class="moz-signature">
        <hr>On 2014-08-27 14:52, Michael Tuexen wrote:<br>
      </div>
    </div>
    <blockquote
      cite="mid:6104CD64-496D-406F-9067-A5226157B9FD@lurchi.franken.de"
      type="cite">
      <pre wrap="">On 27 Aug 2014, at 13:38, Makaraju, Maridi Raju (Raju) <a class="moz-txt-link-rfc2396E" href="mailto:Raju.Makaraju@alcatel-lucent.com">&lt;Raju.Makaraju@alcatel-lucent.com&gt;</a> wrote:

</pre>
      <blockquote type="cite">
        <blockquote type="cite">
          <pre wrap="">If there is negotiation, there is no problem.

If there is no negotiation, starting to use the new feature will either
lead to ignored messages or lead to a crashed connection.

Ignored messages are not a problem if they form part of an ignorable
extension (for instance if someone decided to once in a while send
checksums over all the data sent so far - doesn't matter to those that
don't understand it, since it doesn't carry any extra data).

Crashed connections are usually a problem, so that will mean "no
unnegotiated extensions, ever".

I'm happy with either way of doing it, as long as we're explicit about it.
</pre>
        </blockquote>
        <pre wrap="">&lt;Raju&gt;
+1 for negotiation.
I think it is highly preferred to negotiate these additional PPIDs in all cases as 
anything other than a binary or text  PPID is additional capability. Use of any additional
capabilities must at least be negotiated using JSEP. 
One element of negotiation could be "what must be done for unrecognized PPIDs?": ignore them or
abort the connection.
</pre>
      </blockquote>
      <pre wrap="">Do we really need to negotiate that? The consensus up to now was to close the channel.
So if you negotiate the PPIDs and honor the result, you will never run into this issue...

Best regards
Michael
</pre>
      <blockquote type="cite">
        <pre wrap="">
&lt;/Raju&gt;


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

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Martin Thomson <martin.thomson@gmail.com>, Muthu Arul Mozhi Perumal <muthu.arul@gmail.com>
Thread-Topic: [rtcweb] WG Last Call for draft-ietf-rtcweb-stun-consent-freshness
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Hi,

>> Consent freshness is applicable to any WebRTC entity supporting full=20
>> ICE. A WebRTC browser/device as defined in the transport and overview=20
>> drafts support full ICE, so perform consent freshness. The=20
>> requirements for other WebRTC entities, including WebRTC gateway, is=20
>> yet to be fully specified, and the document that specifies them would=20
>> also specify whether they support full ICE or not (and hence whether the=
y perform consent freshness or not).
>
> Actually, that's probably the best approach: note that this applies - and=
 can be useful for - any ICE implementation.  And leave=20
> it at that for this document.  The transports or security docs might make=
 some statements about mandating this mechanism.

As long as the WebRTC entity supporting full ICE does not assume that the r=
emote WebRTC entity (e.g. a WebRTC gateway) also supports full ICE (read: w=
on't perform consent freshness).

Regards,

Christer


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Subject: Re: [rtcweb] WG Last Call for draft-ietf-rtcweb-stun-consent-freshness
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On 29 August 2014 00:20, Christer Holmberg
<christer.holmberg@ericsson.com> wrote:
> As long as the WebRTC entity supporting full ICE does not assume that the remote WebRTC entity (e.g. a WebRTC gateway) also supports full ICE (read: won't perform consent freshness).

The only way that ICE uses the checks that a remote peer generates is
to populate peer reflexive candidates.  It doesn't apply to consent.
I think we're safe there.


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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Sun, 31 Aug 2014 13:47:47 +0200
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Subject: [rtcweb] Which RTCP packet types are valid in WebRTC?
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Hi, in order to correctly distinguish between RTP and RTCP, RFC 5761
(RTCP-mux) states:

> To allow this multiplexing, future RTCP packet type assignments SHOULD be=
 made after the current assignments in the range 209-223, then in the range=
 194-199, so that only the RTP payload types in the range 64-95 are blocked=
.


IANA has the following assignments for RTCP types:

http://www.iana.org/assignments/rtp-parameters/rtp-parameters.xhtml#rtp-par=
ameters-4

194 SMPTETCSMPTE time-code mapping [RFC5484]
195 IJExtended inter-arrival jitter report [RFC5450]
200 SRsender report [RFC3550]
201 RRreceiver report [RFC3550]
202 SDESsource description [RFC3550]
203 BYEgoodbye [RFC3550]
204 APPapplication-defined [RFC3550]
205 RTPFBGeneric RTP Feedback [RFC4585]
206 PSFBPayload-specific [RFC4585]
207 XRextended report [RFC3611]
208 AVBAVB RTCP packet ["Standard for Layer 3 Transport Protocol for
Time Sensitive Applications in Local Area Networks." Work in
progress.]
209 RSIReceiver Summary Information [RFC5760]
210 TOKENPort Mapping [RFC6284]
211I DMSIDMS Settings [RFC7272]


RFC 3550 defines RTCP packets types in 200-204, so I do not understand
the sentence above: "future RTCP packet type assignments SHOULD be
made after the current assignments in the range 209-223".

So what about 205-208? It is confusing IMHO.


Given the above information, is it safe to use the following check in
order to find RTCP packets when RTCP-mux is used?:


if (type >=3D 194 && type <=3D 223)
  // it is RTCP
else
  // it is not RTCP



Thanks a lot.


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


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Subject: Re: [rtcweb] draft-ietf-rtcweb-transports
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2014-08-25 7:36 GMT+02:00 Justin Uberti <juberti@google.com>:
> Yes, 4571 must be used for ICE connectivity checks, when using ICE-TCP.
>
> However, TURN/TCP has its own framing for STUN messages.
>
> I would simplify your sentence to:
>
>    If ICE-TCP connections are used, framing according to [RFC4571] MUST
>    be used for all packets, unless the content has its own framing
> mechanism.


Sure, I excluded TURN in my comment given that TURN has its own
framing mechanism.

Thanks a lot.

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


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Subject: Re: [rtcweb] Which RTCP packet types are valid in WebRTC?
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On 31 Aug 2014, at 12:47, I=F1aki Baz Castillo <ibc@aliax.net> wrote:
> Hi, in order to correctly distinguish between RTP and RTCP, RFC 5761
> (RTCP-mux) states:
>=20
>> To allow this multiplexing, future RTCP packet type assignments =
SHOULD be made after the current assignments in the range 209-223, then =
in the range 194-199, so that only the RTP payload types in the range =
64-95 are blocked.
>=20
>=20
> IANA has the following assignments for RTCP types:
>=20
> =
http://www.iana.org/assignments/rtp-parameters/rtp-parameters.xhtml#rtp-pa=
rameters-4
>=20
> 194 SMPTETCSMPTE time-code mapping [RFC5484]
> 195 IJExtended inter-arrival jitter report [RFC5450]
> 200 SRsender report [RFC3550]
> 201 RRreceiver report [RFC3550]
> 202 SDESsource description [RFC3550]
> 203 BYEgoodbye [RFC3550]
> 204 APPapplication-defined [RFC3550]
> 205 RTPFBGeneric RTP Feedback [RFC4585]
> 206 PSFBPayload-specific [RFC4585]
> 207 XRextended report [RFC3611]
> 208 AVBAVB RTCP packet ["Standard for Layer 3 Transport Protocol for
> Time Sensitive Applications in Local Area Networks." Work in
> progress.]
> 209 RSIReceiver Summary Information [RFC5760]
> 210 TOKENPort Mapping [RFC6284]
> 211I DMSIDMS Settings [RFC7272]
>=20
>=20
> RFC 3550 defines RTCP packets types in 200-204, so I do not understand
> the sentence above: "future RTCP packet type assignments SHOULD be
> made after the current assignments in the range 209-223".
>=20
> So what about 205-208? It is confusing IMHO.

RTCP packet types 205-208 were already assigned when the draft that =
formed RFC 5761 was submitted for publication. At that time, the current =
assignments were 194, 195, and 200-208. The phrase =93after the current =
assignments in the range 209-223=94 therefore makes sense, although =
maybe =93after the current assignments, in the range 209=97223=94 would =
have been clearer.

> Given the above information, is it safe to use the following check in
> order to find RTCP packets when RTCP-mux is used?:
>=20
>=20
> if (type >=3D 194 && type <=3D 223)
>  // it is RTCP
> else
>  // it is not RTCP

RTCP packet types 192 and 193 are assigned, so a better check would be =
to use the range 192-223 inclusive for RTCP packets.=20

Future RTCP packet type assignments outside that range are possible, but =
would seem unlikely.

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




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2014-08-31 15:24 GMT+02:00 Colin Perkins <csp@csperkins.org>:
> RTCP packet types 205-208 were already assigned when the draft that forme=
d RFC 5761 was submitted for publication. At that time, the current assignm=
ents were 194, 195, and 200-208. The phrase =E2=80=9Cafter the current assi=
gnments in the range 209-223=E2=80=9D therefore makes sense, although maybe=
 =E2=80=9Cafter the current assignments, in the range 209=E2=80=94223=E2=80=
=9D would have been clearer.
>
>> Given the above information, is it safe to use the following check in
>> order to find RTCP packets when RTCP-mux is used?:
>>
>>
>> if (type >=3D 194 && type <=3D 223)
>>  // it is RTCP
>> else
>>  // it is not RTCP
>
> RTCP packet types 192 and 193 are assigned, so a better check would be to=
 use the range 192-223 inclusive for RTCP packets.
>
> Future RTCP packet type assignments outside that range are possible, but =
would seem unlikely.

Thanks, so then I assume that detecting RTCP packets by checking their
packet type in the range 192-223 will not conflict with RTP packets
detection.

Thanks a lot.


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


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From: Colin Perkins <csp@csperkins.org>
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Subject: Re: [rtcweb] Which RTCP packet types are valid in WebRTC?
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On 31 Aug 2014, at 14:31, I=F1aki Baz Castillo <ibc@aliax.net> wrote:
> 2014-08-31 15:24 GMT+02:00 Colin Perkins <csp@csperkins.org>:
>> RTCP packet types 205-208 were already assigned when the draft that =
formed RFC 5761 was submitted for publication. At that time, the current =
assignments were 194, 195, and 200-208. The phrase =93after the current =
assignments in the range 209-223=94 therefore makes sense, although =
maybe =93after the current assignments, in the range 209=97223=94 would =
have been clearer.
>>=20
>>> Given the above information, is it safe to use the following check =
in
>>> order to find RTCP packets when RTCP-mux is used?:
>>>=20
>>>=20
>>> if (type >=3D 194 && type <=3D 223)
>>> // it is RTCP
>>> else
>>> // it is not RTCP
>>=20
>> RTCP packet types 192 and 193 are assigned, so a better check would =
be to use the range 192-223 inclusive for RTCP packets.
>>=20
>> Future RTCP packet type assignments outside that range are possible, =
but would seem unlikely.
>=20
> Thanks, so then I assume that detecting RTCP packets by checking their
> packet type in the range 192-223 will not conflict with RTP packets
> detection.

Don=92t assume =96 look at what=92s signalled.


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




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Subject: Re: [rtcweb] Which RTCP packet types are valid in WebRTC?
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2014-08-31 15:34 GMT+02:00 Colin Perkins <csp@csperkins.org>:
>> Thanks, so then I assume that detecting RTCP packets by checking their
>> packet type in the range 192-223 will not conflict with RTP packets
>> detection.
>
> Don=E2=80=99t assume =E2=80=93 look at what=E2=80=99s signalled.


Of course, I mean the case in which rtcp-mux is enabled.

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

