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From: "Richard Shockey" <richard@shockey.us>
To: "'Randall Gellens'" <randy@qualcomm.com>, "'John Leslie'" <john@jlc.net>, "'Randell Jesup'" <randell-ietf@jesup.org>
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
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So why, pray tell, did the IETF go through the grief of developing OPUS if
its most useful application will not mandate its implementation. 

SHOULD for 722 AMR-WB is very helpful in integration with Enterprise and
Mobile networks.

Its August .. clearly the silly season for technical discussions. 

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of
Randall Gellens
Sent: Friday, August 31, 2012 6:09 PM
To: John Leslie; Randell Jesup
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] RTCWEB needs an Internet Codec

At 11:12 AM -0400 8/31/12, John Leslie wrote:

>  Our issue here is Mandatory-to-Implement. It is very important to  
> have at least one MTI audio codec. I could live with that being G.711,  
> because I trust the market to _actually_ implement others.

Exactly.  The discussion has been going in my view off-track into debates
about which codec is best for which environments.  The real issue is
mandatory versus recommended.

We can pick G.711 as MTI and rely on implementers to support others.

--
Randall Gellens
Opinions are personal;    facts are suspect;    I speak for myself only
-------------- Randomly selected tag: --------------- One never sits in
hotel lobby chairs, my dear.  One never knows whom has been sitting in them
before one.
    --unknown
_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb


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At 1:52 PM -0400 9/3/12, Richard Shockey wrote:

>  So why, pray tell, did the IETF go through the grief of developing OPUS if
>  its most useful application will not mandate its implementation.

So OPUS won't be used unless it's mandated?

If OPUS has the benefits ascribed to it here, then developers will 
flock to it and it doesn't need to be mandated.  (If it doesn't have 
the benefits, then it shouldn't be mandated.)

>  SHOULD for 722 AMR-WB is very helpful in integration with Enterprise and
>  Mobile networks.
>
>  Its August .. clearly the silly season for technical discussions.
>
>  -----Original Message-----
>  From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of
>  Randall Gellens
>  Sent: Friday, August 31, 2012 6:09 PM
>  To: John Leslie; Randell Jesup
>  Cc: rtcweb@ietf.org
>  Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
>
>  At 11:12 AM -0400 8/31/12, John Leslie wrote:
>
>>   Our issue here is Mandatory-to-Implement. It is very important to 
>>  have at least one MTI audio codec. I could live with that being G.711, 
>>  because I trust the market to _actually_ implement others.
>
>  Exactly.  The discussion has been going in my view off-track into debates
>  about which codec is best for which environments.  The real issue is
>  mandatory versus recommended.
>
>  We can pick G.711 as MTI and rely on implementers to support others.
>
>  --
>  Randall Gellens
>  Opinions are personal;    facts are suspect;    I speak for myself only
>  -------------- Randomly selected tag: --------------- One never sits in
>  hotel lobby chairs, my dear.  One never knows whom has been sitting in them
>  before one.
>      --unknown
>  _______________________________________________
>  rtcweb mailing list
>  rtcweb@ietf.org
>  https://www.ietf.org/mailman/listinfo/rtcweb


-- 
Randall Gellens
Opinions are personal;    facts are suspect;    I speak for myself only
-------------- Randomly selected tag: ---------------
I Think We're All Bozos on This Bus.           --Firesign Theatre

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Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
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-----Original Message-----
From: Randall Gellens [mailto:randy@qualcomm.com] 
Sent: Monday, September 03, 2012 2:24 PM
To: Richard Shockey; 'John Leslie'; 'Randell Jesup'
Cc: rtcweb@ietf.org
Subject: RE: [rtcweb] RTCWEB needs an Internet Codec

At 1:52 PM -0400 9/3/12, Richard Shockey wrote:

>  So why, pray tell, did the IETF go through the grief of developing 
> OPUS if  its most useful application will not mandate its implementation.

So OPUS won't be used unless it's mandated?
 
[RS> ] Presumably we should try and make things better rather than just
accept the status quo which is the very definition of G.711


If OPUS has the benefits ascribed to it here, then developers will flock to
it and it doesn't need to be mandated.  (If it doesn't have the benefits,
then it shouldn't be mandated.)


 [RS> ] Humm should like a old argument for IPv6.



From basilgohar@librevideo.org  Mon Sep  3 20:13:08 2012
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Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
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History has shown time and again the companies with sufficient market 
power will opt to implement their own proprietary and/or patented 
formats, or formats which benefit them financially, over royalty free, 
widely-available formats, even when their own formats are technically 
inferiors.  Take, for example, Windows Media Audio, MP3, and AAC in the 
audio realm (in contrast to Vorbis) and Windows Media Video and also 
Quicktime formats (when, at the time, technically superior, more 
standardized formats exists, such as the MPEG family, though that has 
it's own problems).

Mandating the implementation a royalty free format is about the only way 
to get such corporations to implement it, even if it is technically 
superior, as the above examples demonstrate.

On 09/03/2012 02:23 PM, Randall Gellens wrote:
> At 1:52 PM -0400 9/3/12, Richard Shockey wrote:
>
>>  So why, pray tell, did the IETF go through the grief of developing 
>> OPUS if
>>  its most useful application will not mandate its implementation.
>
> So OPUS won't be used unless it's mandated?
>
> If OPUS has the benefits ascribed to it here, then developers will 
> flock to it and it doesn't need to be mandated.  (If it doesn't have 
> the benefits, then it shouldn't be mandated.)
>
>>  SHOULD for 722 AMR-WB is very helpful in integration with Enterprise 
>> and
>>  Mobile networks.
>>
>>  Its August .. clearly the silly season for technical discussions.
>>
>>  -----Original Message-----
>>  From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On 
>> Behalf Of
>>  Randall Gellens
>>  Sent: Friday, August 31, 2012 6:09 PM
>>  To: John Leslie; Randell Jesup
>>  Cc: rtcweb@ietf.org
>>  Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
>>
>>  At 11:12 AM -0400 8/31/12, John Leslie wrote:
>>
>>>   Our issue here is Mandatory-to-Implement. It is very important to 
>>>  have at least one MTI audio codec. I could live with that being 
>>> G.711,  because I trust the market to _actually_ implement others.
>>
>>  Exactly.  The discussion has been going in my view off-track into 
>> debates
>>  about which codec is best for which environments.  The real issue is
>>  mandatory versus recommended.
>>
>>  We can pick G.711 as MTI and rely on implementers to support others.
>>
>>  --
>>  Randall Gellens
>>  Opinions are personal;    facts are suspect;    I speak for myself only
>>  -------------- Randomly selected tag: --------------- One never sits in
>>  hotel lobby chairs, my dear.  One never knows whom has been sitting 
>> in them
>>  before one.
>>      --unknown
>>  _______________________________________________
>>  rtcweb mailing list
>>  rtcweb@ietf.org
>>  https://www.ietf.org/mailman/listinfo/rtcweb
>
>


-- 
Libre Video
http://librevideo.org


From sergio.garcia.murillo@gmail.com  Tue Sep  4 02:30:23 2012
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Date: Tue, 04 Sep 2012 11:30:19 +0200
From: Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com>
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Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
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Maybe an stupid question, but how it is planned to enforce the mandated 
codecs implementation? What prevents any of those "corporations" not 
implementing a mandated codec in their WebRTC products?

Best regards
Sergio

El 04/09/2012 5:13, Basil Mohamed Gohar escribió:
> History has shown time and again the companies with sufficient market 
> power will opt to implement their own proprietary and/or patented 
> formats, or formats which benefit them financially, over royalty free, 
> widely-available formats, even when their own formats are technically 
> inferiors.  Take, for example, Windows Media Audio, MP3, and AAC in 
> the audio realm (in contrast to Vorbis) and Windows Media Video and 
> also Quicktime formats (when, at the time, technically superior, more 
> standardized formats exists, such as the MPEG family, though that has 
> it's own problems).
>
> Mandating the implementation a royalty free format is about the only 
> way to get such corporations to implement it, even if it is 
> technically superior, as the above examples demonstrate.
>
> On 09/03/2012 02:23 PM, Randall Gellens wrote:
>> At 1:52 PM -0400 9/3/12, Richard Shockey wrote:
>>
>>>  So why, pray tell, did the IETF go through the grief of developing 
>>> OPUS if
>>>  its most useful application will not mandate its implementation.
>>
>> So OPUS won't be used unless it's mandated?
>>
>> If OPUS has the benefits ascribed to it here, then developers will 
>> flock to it and it doesn't need to be mandated.  (If it doesn't have 
>> the benefits, then it shouldn't be mandated.)
>>
>>>  SHOULD for 722 AMR-WB is very helpful in integration with 
>>> Enterprise and
>>>  Mobile networks.
>>>
>>>  Its August .. clearly the silly season for technical discussions.
>>>
>>>  -----Original Message-----
>>>  From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On 
>>> Behalf Of
>>>  Randall Gellens
>>>  Sent: Friday, August 31, 2012 6:09 PM
>>>  To: John Leslie; Randell Jesup
>>>  Cc: rtcweb@ietf.org
>>>  Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
>>>
>>>  At 11:12 AM -0400 8/31/12, John Leslie wrote:
>>>
>>>>   Our issue here is Mandatory-to-Implement. It is very important to 
>>>>  have at least one MTI audio codec. I could live with that being 
>>>> G.711,  because I trust the market to _actually_ implement others.
>>>
>>>  Exactly.  The discussion has been going in my view off-track into 
>>> debates
>>>  about which codec is best for which environments.  The real issue is
>>>  mandatory versus recommended.
>>>
>>>  We can pick G.711 as MTI and rely on implementers to support others.
>>>
>>>  --
>>>  Randall Gellens
>>>  Opinions are personal;    facts are suspect;    I speak for myself 
>>> only
>>>  -------------- Randomly selected tag: --------------- One never 
>>> sits in
>>>  hotel lobby chairs, my dear.  One never knows whom has been sitting 
>>> in them
>>>  before one.
>>>      --unknown
>>>  _______________________________________________
>>>  rtcweb mailing list
>>>  rtcweb@ietf.org
>>>  https://www.ietf.org/mailman/listinfo/rtcweb
>>
>>
>
>


From harald@alvestrand.no  Tue Sep  4 05:25:41 2012
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On 09/04/2012 11:30 AM, Sergio Garcia Murillo wrote:
> Maybe an stupid question, but how it is planned to enforce the 
> mandated codecs implementation? What prevents any of those 
> "corporations" not implementing a mandated codec in their WebRTC 
> products?

My first chance to quote the new, Web-page-format Tao of IETF:

"One more thing that is important for newcomers: the IETF in no way 
"runs the Internet", despite what some people mistakenly might say. The 
IETF makes voluntary standards that are often adopted by Internet users, 
but it does not control, or even patrol, the Internet. If your interest 
in the IETF is because you want to be part of the overseers, you may be 
badly disappointed by the IETF."

http://www.ietf.org/tao.html

There is no protocol police; anyone can implement a product that 
implements only part of an IETF standard. They just can't truthfully 
claim to have implemented that IETF standard.

>
> Best regards
> Sergio
>
> El 04/09/2012 5:13, Basil Mohamed Gohar escribió:
>> History has shown time and again the companies with sufficient market 
>> power will opt to implement their own proprietary and/or patented 
>> formats, or formats which benefit them financially, over royalty 
>> free, widely-available formats, even when their own formats are 
>> technically inferiors. Take, for example, Windows Media Audio, MP3, 
>> and AAC in the audio realm (in contrast to Vorbis) and Windows Media 
>> Video and also Quicktime formats (when, at the time, technically 
>> superior, more standardized formats exists, such as the MPEG family, 
>> though that has it's own problems).
>>
>> Mandating the implementation a royalty free format is about the only 
>> way to get such corporations to implement it, even if it is 
>> technically superior, as the above examples demonstrate.
>>
>> On 09/03/2012 02:23 PM, Randall Gellens wrote:
>>> At 1:52 PM -0400 9/3/12, Richard Shockey wrote:
>>>
>>>>  So why, pray tell, did the IETF go through the grief of developing 
>>>> OPUS if
>>>>  its most useful application will not mandate its implementation.
>>>
>>> So OPUS won't be used unless it's mandated?
>>>
>>> If OPUS has the benefits ascribed to it here, then developers will 
>>> flock to it and it doesn't need to be mandated.  (If it doesn't have 
>>> the benefits, then it shouldn't be mandated.)
>>>
>>>>  SHOULD for 722 AMR-WB is very helpful in integration with 
>>>> Enterprise and
>>>>  Mobile networks.
>>>>
>>>>  Its August .. clearly the silly season for technical discussions.
>>>>
>>>>  -----Original Message-----
>>>>  From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On 
>>>> Behalf Of
>>>>  Randall Gellens
>>>>  Sent: Friday, August 31, 2012 6:09 PM
>>>>  To: John Leslie; Randell Jesup
>>>>  Cc: rtcweb@ietf.org
>>>>  Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
>>>>
>>>>  At 11:12 AM -0400 8/31/12, John Leslie wrote:
>>>>
>>>>>   Our issue here is Mandatory-to-Implement. It is very important 
>>>>> to  have at least one MTI audio codec. I could live with that 
>>>>> being G.711,  because I trust the market to _actually_ implement 
>>>>> others.
>>>>
>>>>  Exactly.  The discussion has been going in my view off-track into 
>>>> debates
>>>>  about which codec is best for which environments.  The real issue is
>>>>  mandatory versus recommended.
>>>>
>>>>  We can pick G.711 as MTI and rely on implementers to support others.
>>>>
>>>>  --
>>>>  Randall Gellens
>>>>  Opinions are personal;    facts are suspect;    I speak for myself 
>>>> only
>>>>  -------------- Randomly selected tag: --------------- One never 
>>>> sits in
>>>>  hotel lobby chairs, my dear.  One never knows whom has been 
>>>> sitting in them
>>>>  before one.
>>>>      --unknown
>>>>  _______________________________________________
>>>>  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


--------------050308080108080204090904
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<html>
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    <meta content="text/html; charset=ISO-8859-1"
      http-equiv="Content-Type">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    <div class="moz-cite-prefix">On 09/04/2012 11:30 AM, Sergio Garcia
      Murillo wrote:<br>
    </div>
    <blockquote cite="mid:5045CA2B.2070406@gmail.com" type="cite">Maybe
      an stupid question, but how it is planned to enforce the mandated
      codecs implementation? What prevents any of those "corporations"
      not implementing a mandated codec in their WebRTC products?
      <br>
    </blockquote>
    <br>
    My first chance to quote the new, Web-page-format Tao of IETF:<br>
    <br>
    <meta http-equiv="content-type" content="text/html;
      charset=ISO-8859-1">
    <span style="color: rgb(0, 0, 0); font-family: verdana, helvetica,
      arial, sans-serif; font-size: 13px; font-style: normal;
      font-variant: normal; font-weight: normal; letter-spacing: normal;
      line-height: normal; orphans: 2; text-align: start; text-indent:
      0px; text-transform: none; white-space: normal; widows: 2;
      word-spacing: 0px; -webkit-text-size-adjust: auto;
      -webkit-text-stroke-width: 0px; display: inline !important; float:
      none; ">"One more thing that is important for newcomers: the IETF
      in no way "runs the Internet", despite what some people mistakenly
      might say. The IETF makes voluntary standards that are often
      adopted by Internet users, but it does not control, or even
      patrol, the Internet. If your interest in the IETF is because you
      want to be part of the overseers, you may be badly disappointed by
      the IETF.</span>"<br>
    <br>
    <meta http-equiv="content-type" content="text/html;
      charset=ISO-8859-1">
    <a href="http://www.ietf.org/tao.html">http://www.ietf.org/tao.html</a><br>
    <br>
    There is no protocol police; anyone can implement a product that
    implements only part of an IETF standard. They just can't truthfully
    claim to have implemented that IETF standard.<br>
    <br>
    <blockquote cite="mid:5045CA2B.2070406@gmail.com" type="cite">
      <br>
      Best regards
      <br>
      Sergio
      <br>
      <br>
      El 04/09/2012 5:13, Basil Mohamed Gohar escribi&oacute;:
      <br>
      <blockquote type="cite">History has shown time and again the
        companies with sufficient market power will opt to implement
        their own proprietary and/or patented formats, or formats which
        benefit them financially, over royalty free, widely-available
        formats, even when their own formats are technically inferiors.&nbsp;
        Take, for example, Windows Media Audio, MP3, and AAC in the
        audio realm (in contrast to Vorbis) and Windows Media Video and
        also Quicktime formats (when, at the time, technically superior,
        more standardized formats exists, such as the MPEG family,
        though that has it's own problems).
        <br>
        <br>
        Mandating the implementation a royalty free format is about the
        only way to get such corporations to implement it, even if it is
        technically superior, as the above examples demonstrate.
        <br>
        <br>
        On 09/03/2012 02:23 PM, Randall Gellens wrote:
        <br>
        <blockquote type="cite">At 1:52 PM -0400 9/3/12, Richard Shockey
          wrote:
          <br>
          <br>
          <blockquote type="cite">&nbsp;So why, pray tell, did the IETF go
            through the grief of developing OPUS if
            <br>
            &nbsp;its most useful application will not mandate its
            implementation.
            <br>
          </blockquote>
          <br>
          So OPUS won't be used unless it's mandated?
          <br>
          <br>
          If OPUS has the benefits ascribed to it here, then developers
          will flock to it and it doesn't need to be mandated.&nbsp; (If it
          doesn't have the benefits, then it shouldn't be mandated.)
          <br>
          <br>
          <blockquote type="cite">&nbsp;SHOULD for 722 AMR-WB is very helpful
            in integration with Enterprise and
            <br>
            &nbsp;Mobile networks.
            <br>
            <br>
            &nbsp;Its August .. clearly the silly season for technical
            discussions.
            <br>
            <br>
            &nbsp;-----Original Message-----
            <br>
            &nbsp;From: <a class="moz-txt-link-abbreviated" href="mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a>
            [<a class="moz-txt-link-freetext" href="mailto:rtcweb-bounces@ietf.org">mailto:rtcweb-bounces@ietf.org</a>] On Behalf Of
            <br>
            &nbsp;Randall Gellens
            <br>
            &nbsp;Sent: Friday, August 31, 2012 6:09 PM
            <br>
            &nbsp;To: John Leslie; Randell Jesup
            <br>
            &nbsp;Cc: <a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
            <br>
            &nbsp;Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
            <br>
            <br>
            &nbsp;At 11:12 AM -0400 8/31/12, John Leslie wrote:
            <br>
            <br>
            <blockquote type="cite">&nbsp; Our issue here is
              Mandatory-to-Implement. It is very important to &nbsp;have at
              least one MTI audio codec. I could live with that being
              G.711,&nbsp; because I trust the market to _actually_ implement
              others.
              <br>
            </blockquote>
            <br>
            &nbsp;Exactly.&nbsp; The discussion has been going in my view
            off-track into debates
            <br>
            &nbsp;about which codec is best for which environments.&nbsp; The real
            issue is
            <br>
            &nbsp;mandatory versus recommended.
            <br>
            <br>
            &nbsp;We can pick G.711 as MTI and rely on implementers to
            support others.
            <br>
            <br>
            &nbsp;--
            <br>
            &nbsp;Randall Gellens
            <br>
            &nbsp;Opinions are personal;&nbsp;&nbsp;&nbsp; facts are suspect;&nbsp;&nbsp;&nbsp; I speak for
            myself only
            <br>
            &nbsp;-------------- Randomly selected tag: --------------- One
            never sits in
            <br>
            &nbsp;hotel lobby chairs, my dear.&nbsp; One never knows whom has been
            sitting in them
            <br>
            &nbsp;before one.
            <br>
            &nbsp;&nbsp;&nbsp;&nbsp; --unknown
            <br>
            &nbsp;_______________________________________________
            <br>
            &nbsp;rtcweb mailing list
            <br>
            &nbsp;<a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
            <br>
            &nbsp;<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a>
            <br>
          </blockquote>
          <br>
          <br>
        </blockquote>
        <br>
        <br>
      </blockquote>
      <br>
      _______________________________________________
      <br>
      rtcweb mailing list
      <br>
      <a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
      <br>
      <a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a>
      <br>
    </blockquote>
    <br>
  </body>
</html>

--------------050308080108080204090904--

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On 09/04/2012 08:25 AM, Harald Alvestrand wrote:
> On 09/04/2012 11:30 AM, Sergio Garcia Murillo wrote:
>> Maybe an stupid question, but how it is planned to enforce the
>> mandated codecs implementation? What prevents any of those
>> "corporations" not implementing a mandated codec in their WebRTC
>> products?
>
> My first chance to quote the new, Web-page-format Tao of IETF:
>
> "One more thing that is important for newcomers: the IETF in no way
> "runs the Internet", despite what some people mistakenly might say.
> The IETF makes voluntary standards that are often adopted by Internet
> users, but it does not control, or even patrol, the Internet. If your
> interest in the IETF is because you want to be part of the overseers,
> you may be badly disappointed by the IETF."
>
> http://www.ietf.org/tao.html
>
> There is no protocol police; anyone can implement a product that
> implements only part of an IETF standard. They just can't truthfully
> claim to have implemented that IETF standard.

To the extent that the IETF has any power, it is an *indirect* power. 
If a standard is valued, then organisations (at their own discretion)
may start to write "Must implement RFCxxxx" into requests for proposals,
and choose (or not choose) to spend their money buying products that
conform or do not conform.

RFC's are littered with ignored mandatory features; similarly for RFC's
themselves.
                                    - Jim

>
>>
>> Best regards
>> Sergio
>>
>> El 04/09/2012 5:13, Basil Mohamed Gohar escribió:
>>> History has shown time and again the companies with sufficient
>>> market power will opt to implement their own proprietary and/or
>>> patented formats, or formats which benefit them financially, over
>>> royalty free, widely-available formats, even when their own formats
>>> are technically inferiors.  Take, for example, Windows Media Audio,
>>> MP3, and AAC in the audio realm (in contrast to Vorbis) and Windows
>>> Media Video and also Quicktime formats (when, at the time,
>>> technically superior, more standardized formats exists, such as the
>>> MPEG family, though that has it's own problems).
>>>
>>> Mandating the implementation a royalty free format is about the only
>>> way to get such corporations to implement it, even if it is
>>> technically superior, as the above examples demonstrate.
>>>
>>> On 09/03/2012 02:23 PM, Randall Gellens wrote:
>>>> At 1:52 PM -0400 9/3/12, Richard Shockey wrote:
>>>>
>>>>>  So why, pray tell, did the IETF go through the grief of
>>>>> developing OPUS if
>>>>>  its most useful application will not mandate its implementation.
>>>>
>>>> So OPUS won't be used unless it's mandated?
>>>>
>>>> If OPUS has the benefits ascribed to it here, then developers will
>>>> flock to it and it doesn't need to be mandated.  (If it doesn't
>>>> have the benefits, then it shouldn't be mandated.)
>>>>
>>>>>  SHOULD for 722 AMR-WB is very helpful in integration with
>>>>> Enterprise and
>>>>>  Mobile networks.
>>>>>
>>>>>  Its August .. clearly the silly season for technical discussions.
>>>>>
>>>>>  -----Original Message-----
>>>>>  From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
>>>>> Behalf Of
>>>>>  Randall Gellens
>>>>>  Sent: Friday, August 31, 2012 6:09 PM
>>>>>  To: John Leslie; Randell Jesup
>>>>>  Cc: rtcweb@ietf.org
>>>>>  Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
>>>>>
>>>>>  At 11:12 AM -0400 8/31/12, John Leslie wrote:
>>>>>
>>>>>>   Our issue here is Mandatory-to-Implement. It is very important
>>>>>> to  have at least one MTI audio codec. I could live with that
>>>>>> being G.711,  because I trust the market to _actually_ implement
>>>>>> others.
>>>>>
>>>>>  Exactly.  The discussion has been going in my view off-track into
>>>>> debates
>>>>>  about which codec is best for which environments.  The real issue is
>>>>>  mandatory versus recommended.
>>>>>
>>>>>  We can pick G.711 as MTI and rely on implementers to support others.
>>>>>
>>>>>  --
>>>>>  Randall Gellens
>>>>>  Opinions are personal;    facts are suspect;    I speak for
>>>>> myself only
>>>>>  -------------- Randomly selected tag: --------------- One never
>>>>> sits in
>>>>>  hotel lobby chairs, my dear.  One never knows whom has been
>>>>> sitting in them
>>>>>  before one.
>>>>>      --unknown
>>>>>  _______________________________________________
>>>>>  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 ron@debian.org  Tue Sep  4 14:01:02 2012
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Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
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On Tue, Sep 04, 2012 at 10:28:01AM -0400, Jim Gettys wrote:
> On 09/04/2012 08:25 AM, Harald Alvestrand wrote:
> > On 09/04/2012 11:30 AM, Sergio Garcia Murillo wrote:
> >> Maybe an stupid question, but how it is planned to enforce the
> >> mandated codecs implementation? What prevents any of those
> >> "corporations" not implementing a mandated codec in their WebRTC
> >> products?
> >
> > My first chance to quote the new, Web-page-format Tao of IETF:
> >
> > "One more thing that is important for newcomers: the IETF in no way
> > "runs the Internet", despite what some people mistakenly might say.
> > The IETF makes voluntary standards that are often adopted by Internet
> > users, but it does not control, or even patrol, the Internet. If your
> > interest in the IETF is because you want to be part of the overseers,
> > you may be badly disappointed by the IETF."
> >
> > http://www.ietf.org/tao.html
> >
> > There is no protocol police; anyone can implement a product that
> > implements only part of an IETF standard. They just can't truthfully
> > claim to have implemented that IETF standard.
> 
> To the extent that the IETF has any power, it is an *indirect* power. 
> If a standard is valued, then organisations (at their own discretion)
> may start to write "Must implement RFCxxxx" into requests for proposals,
> and choose (or not choose) to spend their money buying products that
> conform or do not conform.

Which in turn is of course precisely the compelling reason for Opus to be
explicitly specified here - so that _rtcweb_ *can* have a chance of being
a standard of value, which promises the people who follow it high-quality
results and interoperability with any others who implement the standard.

There are already plenty of Go It Alone, ad hoc solutions, which to date
have little to no market traction.  Anything which performs worse than
they do is unlikely to be attractive, and anything which doesn't promise
real interoperability doesn't solve the problem which has prevented them
from being widely used outside of small disjoint circles.

"The Market" has already decided.  It decided it wanted a Real standard
for this.  Our job here is to give them one that doesn't totally suck.

It's not about "forcing Opus on people who don't want it", it's about
making rtcweb actually work for all those people who are serious about
wanting something that actually works.  We're clearly not going to give
them that by saying: "Here's some valve-age technology that's been all
but obsolete for decades now - if you want this to be Any Good, you'll
have to figure that part out for yourselves.  Good Luck with that!"

However much valves might be the stuff of True Audiophile wet dreams,
that just won't fly.  And I'm pretty sure almost everyone here knows it.

If there's something _better_ than Opus for this, I'd love to hear all
about it.  But talk of things that are strictly and/or vastly worse is
kind of hard to take seriously.  I can see how some people might be
feeling a little frustrated at watching that tack push ever closer to
the wind.

  Luff and KISSes,
  Ron



From hoene@uni-tuebingen.de  Wed Sep  5 03:48:20 2012
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From: "Hoene" <hoene@uni-tuebingen.de>
To: "Jean-Marc Valin" <jmvalin@jmvalin.ca>, <rtcweb@ietf.org>
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Hi all,

please do not forget to forward any new characterization test results of
Opus to the Opus mailing so that I can collect them.
=09
These days, I hardly have time to follow all discussions on the rtcweb
mailing list.

Thanks

 Christian Hoene



-----Urspr=FCngliche Nachricht-----
Von: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] Im Auftrag =
von
Randell Jesup
Gesendet: Freitag, 31. August 2012 23:18
An: rtcweb@ietf.org
Betreff: Re: [rtcweb] Opus vs AMR-WB for packet loss

On 8/31/2012 4:50 PM, Jean-Marc Valin wrote:
> OK, here's something concrete for comparing Opus to AMR-WB in packet
> loss conditions. I've simulated 30% bursty packet loss in the
> following conditions:
> 1) AMR-WB mode 8 (23.85 kb/s)
> 2) Opus at 24 kb/s (no FEC)
> 3) Opus at 24 kb/s with FEC (24 kb/s is the total including FEC)
>
> I've uploaded the results at: http://jmvalin.ca/plc/
>
> I think everyone will agree that even without FEC, Opus sounds much
> better than AMR-WB. And when you turn FEC on, then you barely even
> notice the losses.

That's really, really good.  :-)  Especially with FEC.

--=20
Randell Jesup
randell-ietf@jesup.org
_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb


From randy@qualcomm.com  Wed Sep  5 16:31:35 2012
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To: Ron <ron@debian.org>, Cameron Byrne <cb.list6@gmail.com>
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At 6:30 AM +0930 9/5/12, Ron wrote:
>  It's not about "forcing Opus on people who don't want it", it's about
>  making rtcweb actually work for all those people who are serious about
>  wanting something that actually works.  We're clearly not going to give
>  them that by saying: "Here's some valve-age technology that's been all
>  but obsolete for decades now - if you want this to be Any Good, you'll
>  have to figure that part out for yourselves.  Good Luck with that!"
>
>  However much valves might be the stuff of True Audiophile wet dreams,
>  that just won't fly.  And I'm pretty sure almost everyone here knows it.


At 3:42 PM +0930 9/1/12, Ron wrote:
>  The question is what is the minimum
>  guarantee that any rtcweb user will have when talking to any other
>  rtcweb user, without first needing to sign consent forms and download
>  optional extras (possibly for a reasonably non-discriminatory fee,
>  and even possibly trojaned).


I'm puzzled by the repeated statements that imply that any 
non-mandatory codec can't be supported without separate downloads 
(now extended to include signed consent forms and fees).  That's 
simply not true.  Any implementation is free to support any codec 
out-of-the-box.

One of the goals of rtcweb is to encourage greater use of native 
codecs, and greater access to such codecs by applications such as 
those running in browsers.  These are worthy goals, since native 
codecs often have better performance within the environment. 
(Examples include AMR and EVRC on handsets.)  These codecs also can 
be supported out-of-the-box, no separate downloads, no signed forms.

The discussion is over which codecs MUST be supported.  Since 
mandatory codecs are required, with no choice, they should have the 
lowest possible burden to implement, taking into account all aspects 
of the implementation burden.  Then, implementers are free to 
consider which additional codecs they will support, taking into 
account the costs and benefits.

-- 
Randall Gellens
Opinions are personal;    facts are suspect;    I speak for myself only
-------------- Randomly selected tag: ---------------
The single biggest problem in communication is the illusion that it
has taken place.                                       --G. B. Shaw

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On Wed, Sep 5, 2012 at 4:29 PM, Randall Gellens <randy@qualcomm.com> wrote:
> The discussion is over which codecs MUST be supported.  Since mandatory
> codecs are required, with no choice, they should have the lowest possible
> burden to implement, taking into account all aspects of the implementation
> burden.

Speaking personally, I would put this slightly differently.  After
all, a codec with low implementation burdens may have very poor
bandwidth consumption characteristics, relatively inferior acoustic
range, or other problematic aspects which don't read on the
implementation burden.

As you note below, the overall problem is one of cost/benefit
analysis.  In the end, we need to find a way to avoid implementation
failure with a set of choices that represent the best overall
cost/benefit ratio in the contexts identified by our use cases.

>Then, implementers are free to consider which additional codecs
> they will support, taking into account the costs and benefits.

Absolutely.

regards,

Ted Hardie

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On 12-09-05 07:29 PM, Randall Gellens wrote:
> The discussion is over which codecs MUST be supported.  Since mandatory
> codecs are required, with no choice, they should have the lowest
> possible burden to implement, taking into account all aspects of the
> implementation burden.  Then, implementers are free to consider which
> additional codecs they will support, taking into account the costs and
> benefits.

Following the same reasoning, I think we should consider ROT13 as the
MTI encryption algorithm in rtcweb, since it's clearly the one with the
lowest possible burden to implement. Then, implementers would be free to
consider AES, 3DES, Blowfish or Enigma, taking into account the costs
and benefits.

	Jean-Marc

From ron@debian.org  Wed Sep  5 20:03:11 2012
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On Wed, Sep 05, 2012 at 04:29:28PM -0700, Randall Gellens wrote:
> At 6:30 AM +0930 9/5/12, Ron wrote:
> > It's not about "forcing Opus on people who don't want it", it's about
> > making rtcweb actually work for all those people who are serious about
> > wanting something that actually works.  We're clearly not going to give
> > them that by saying: "Here's some valve-age technology that's been all
> > but obsolete for decades now - if you want this to be Any Good, you'll
> > have to figure that part out for yourselves.  Good Luck with that!"
> >
> > However much valves might be the stuff of True Audiophile wet dreams,
> > that just won't fly.  And I'm pretty sure almost everyone here knows it.
> 
> 
> At 3:42 PM +0930 9/1/12, Ron wrote:
> > The question is what is the minimum
> > guarantee that any rtcweb user will have when talking to any other
> > rtcweb user, without first needing to sign consent forms and download
> > optional extras (possibly for a reasonably non-discriminatory fee,
> > and even possibly trojaned).
> 
> 
> I'm puzzled by the repeated statements that imply that any
> non-mandatory codec can't be supported without separate downloads
> (now extended to include signed consent forms and fees).  That's
> simply not true.

I'm not sure exactly what you find puzzling about the idea that if
non-mandatory codecs are _required_ for adequate performance, then
for people using different implementations, that provide different
subsets of all the available codecs by default, *someone* is almost
certain to have to add on something extra to enable interoperability
with adequate performance.

That should be simply self-evident.

Enough so that I'll leave deeper study of venn-diagrams as a homework
exercise for any still-bewildered readers.

>  Any implementation is free to support any codec out-of-the-box.

Especially since this point was repeated alongside the others with
approximately equal frequency.  I'm pretty sure we have a clear
consensus on that.  I'm a little puzzled myself why you trimmed
that from the messages you quoted, only to repeat it again.

But we have list archives for anyone who wants the full context.


> One of the goals of rtcweb is to encourage greater use of native
> codecs, and greater access to such codecs by applications such as
> those running in browsers.  These are worthy goals, since native
> codecs often have better performance within the environment.
> (Examples include AMR and EVRC on handsets.)  These codecs also can
> be supported out-of-the-box, no separate downloads, no signed forms.

The unspoken assumption there being "provided you both happen to be
using the same platform with the same native codecs", (and of course
provided your vendor already signed the forms and payed the fees to
obtain a licence for you for the specific examples you cite).

That's fine if you do happen to.  It's when you don't that the MTI
codecs are what is relevant for interoperability.

The tiny performance difference between AMR and Opus might be of
value to some people, and if both their devices support it, and
both of them have a licence to use it, then great.

But if rtcweb sucks if you don't have those things, then this group
will have failed to meet its charter.  Which does explicitly consider
interoperability as a deliverable (but doesn't seem to actually
mention anything about native codecs at all?).

It does mention that the IETF has a codec working group whose
work should be considered though.


> The discussion is over which codecs MUST be supported.  Since
> mandatory codecs are required, with no choice, they should have the
> lowest possible burden to implement, taking into account all aspects
> of the implementation burden.  Then, implementers are free to
> consider which additional codecs they will support, taking into
> account the costs and benefits.

The libopus reference implementation source can be freely download
by anyone, at any time, and redistributed at will by any party with
no other "burden" placed upon them to do anything else.

Much of the (small amount of) extra work needed to interface that to
many existing audio processing abstractions is already done too.


How does that compare to the burden of your preferred example AMR?

Let's see.
Wikipedia tells me:

 "The initial fee for professional content creation tools and
  "real-time channel" products is $6,500. The minimum annual
  royalty shall be $10,000, excluding the initial fee in year 1
  of the license agreement"

3GPP apparently provides a reference implementation, though it's
not clear to me at this stage if that implementation actually
provides the quality claimed by the people who say it is better
than Opus for the extreme low-bandwidth niche.

What does appear to be clear though is that I'm not able to
redistribute the source of that, so I couldn't actually provide
it with my rtcweb implementation if I also wanted to give my
users the source for that (whether they paid me for it or not).

If those people wanted to use it, they'd have to jump through
all the hoops themselves to obtain it and a licence to use it.


So it's not at all clear to me how Opus isn't convincingly the
winner of the "lowest possible burden" test you propose either.

"Lowest possible" is of course a silly extreme.  That would be
raw PCM (modulo endianness trouble).  So I'm assuming you meant
something more sensible than that like lowest practical.

One codec, which covers all ranges and is as near as possible
in this modern world to being completely unencumbered would
seem to be the obvious answer there.


Luckily for us, such a thing actually exists, and the IETF even
has change control over it.

  What more could we wish for,
  Ron



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Ron <ron@debian.org> wrote:
> 
> The libopus reference implementation source can be freely download
> by anyone, at any time, and redistributed at will by any party with
> no other "burden" placed upon them to do anything else.

   See:

www.opus-codec.org/downloads/

(the libopus.sourceforge.net is quite unrelated).

http://downloads.xiph.org/releases/opus/opus-1.0.0-rc.tar.gz

is the release-candidate. Note:
] 
] The codec is still under development, in the final stages preparing
] the first production release.

   I find this very encouraging, but not quite enough to justify MTI
today.

   Note also:

https://developer.mozilla.org/en-US/docs/Media_formats_supported_by_the_audio_and_video_elements#Ogg_Opus

which shows some level of support already in Mozilla (but not Chrome,
Explorer, Opera, or Safari)

   Here is a case where "Google's position is" could actually be
helpful -- were they to announce Opus support in Chrome...

--
John Leslie <john@jlc.net>

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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

On 09/06/2012 12:51 PM, John Leslie wrote:
> is the release-candidate. Note: ] ] The codec is still under 
> development, in the final stages preparing ] the first production 
> release.

Thanks for reporting this. It was old text that hadn't been updated in
a long time and which I have now fixed. In fact, the code has been in
a stable state for about one year now. The only reason we haven't
called it 1.0 yet is because we wanted to wait for the official RFC.
The good news is that RFC 6716 should be published any day now.

> which shows some level of support already in Mozilla (but not 
> Chrome, Explorer, Opera, or Safari)
> 
> Here is a case where "Google's position is" could actually be
> helpful -- were they to announce Opus support in Chrome...

Quoting from Harald Alvestrand's email on this topic:

> Since the number of people stating their opinion has been large,
> I'll just reiterate the opinion I had (and hummed for) in
> Vancouver: Opus and G.711 should be mandatory to implement for
> RTCWEB. Since the question of whether there's any value to making
> the decision now has been raised: The first interoperable products
> implementing RTCWEB are shipping within a very short timeframe.
> Those first implementations will shape the market for what's
> actually used in practice. In order to allow applications requiring
> high quality sound to be among the first ones developed, those
> first products need to include a common choice of a high quality
> codec.

I think that can be counted as Google going with Opus. As for
Microsoft, I haven't seen their official position, but I can only
assume that considering their investment in this codec, they'd
probably like to use it. Regarding Opera, some people have reported
Opus already working (even before Firefox) through GStreamer builds
that included Opus. I don't know what Opera's official position is on
Opus, but support is definitely trivial for them.

Cheers,

	Jean-Marc
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Jean-Marc Valin wrote:
> that included Opus. I don't know what Opera's official position is on
> Opus, but support is definitely trivial for them.

I spoke with Simon Peters at FOMS this week in Paris, and he told me 
that Opera did plan to enable Opus on platforms that do not provide 
their own version of gstreamer (where, of course, it is already enabled, 
providing they ship at least version 0.10.23).

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On Aug 8, 2012, at 9:28 AM, Harald Alvestrand <harald@alvestrand.no> wrote:

> Forking thread....
>=20
> On 08/08/2012 05:05 PM, Cameron Byrne wrote:
>>=20
>>=20
>> Is this thread  really about the ietf engineering a way to by-pass netwo=
rk policy set by network operators?
>>=20
> It's about getting stuff done in the presence of rules that hinder the "s=
imple way".
> Those rules may be set that way deliberately, by ignorance, because of im=
plementation limitations, or in error; there's no way to know.
>=20
>> I do not believe that is acceptable.
>>=20
> We've already done TURN, including TURN over TLS.

TURN was specifically designed to allow the network operator to be able to =
continue to control and enforce policies similar to what they currently do =
in their firewalls. You will note Kaufman strongly disagreed with this desi=
gn principle and would have done the permissions in TURN significantly diff=
erently - that would have actually made TURN faster and easier to use but w=
ould have removed this property. We choice to stay on the path of "not viol=
ating or circumventing the will of network administrators".=20

That said, I think the discussion about making media relay / tunnel protoco=
l that masquerades as HTTPS is leaping of the edge of the slippery slope. I=
 have mixed feelings about if that is wise or not but that's a long discuss=
ion.

>=20
> Old quote: "This is not about starting down the slippery slope; this is a=
bout how far we slide into the muck at the bottom".

Agree with that :-)=

From magnus.westerlund@ericsson.com  Fri Sep  7 08:11:11 2012
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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: [rtcweb] Consensus Statement for Re: Confirmation of consensus on audio codecs
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WG,

We chairs can confirm a strong consensus for both G.711 and Opus as
mandatory to implement audio codecs for RTCWeb based on the WG meeting
and the mailing list.

There have been individuals with other views. Several wanted G.711 only,
which is one of the alternative in the WG meeting's consensus decision.
These additional opinions have not altered the chairs view that there is
strong consensus for G.711 and Opus.

The Chairs take note of two other aspects which were raised in the
discussion on the mailing list;

First, that some argued for including also G.722 as mandatory to
implement codec and several more have proposed it to be listed as SHOULD
be implemented. Based on the consensus in the WG this is clearly not
mandatory to implement. When it comes recommending additional codecs to
be implemented there has been views expressed previously that any such
recommendations would not be particular useful and only increase the
amount of contention in the WG.

Based on the discussion to date, the chairs will run a consensus call on
whether the WG should recommend specific codecs which will not be
mandatory to implement (i.e. will the document contain SHOULDs as well
as MUSTs for codec implementation)".  If the response to that consensus
call indicates consensus for non-mandatory to implement recommendations,
we will run consensus calls on those codecs to be included at that level.

Secondly, there were views and suggestions that we should not make a
decision for mandatory to implement codecs at this point in time.  The
working group had previously indicated strong consensus to choose a
mandatory to implement codec in this space, to avoid interoperability
failure.  We do not believe that reconsideration of this decision is
required at this time, and we believe that the support for the current
consensus is strong, if not perfect.

The WG chairs
Ted, Cullen and Magnus



From ted.ietf@gmail.com  Fri Sep  7 08:27:55 2012
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From: Ted Hardie <ted.ietf@gmail.com>
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Subject: [rtcweb] Call for adoption of QoS draft
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Based on the feedback at the last working group meeting, the chairs
believe that there is general support for creating a document that
describes the QoS issues relating to RTCWEB.  We would like to call
for adoption of draft-jennings-rtcweb-qos-00 as the input draft.  Dan
Druta has agreed to serve as the WG document editor, should this
document move forward.

Please send comments to the list by  September 16, 2012

Ted and Magnus

From internet-drafts@ietf.org  Fri Sep  7 13:10:55 2012
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Subject: [rtcweb] I-D Action: draft-ietf-rtcweb-audio-00.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Real-Time Communication in WEB-browsers W=
orking Group of the IETF.

	Title           : WebRTC Audio Codec and Processing Requirements
	Author(s)       : Jean-Marc Valin
                          Cary Bran
	Filename        : draft-ietf-rtcweb-audio-00.txt
	Pages           : 6
	Date            : 2012-09-07

Abstract:
   This document outlines the audio codec and processing requirements
   for WebRTC client application and endpoint devices.


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

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


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


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Subject: [rtcweb] I-D Action: draft-ietf-rtcweb-data-channel-01.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Real-Time Communication in WEB-browsers W=
orking Group of the IETF.

	Title           : RTCWeb Datagram Connection
	Author(s)       : Randell Jesup
                          Salvatore Loreto
                          Michael Tuexen
	Filename        : draft-ietf-rtcweb-data-channel-01.txt
	Pages           : 12
	Date            : 2012-09-07

Abstract:
   The Web Real-Time Communication (WebRTC) working group is charged to
   provide protocol support for direct interactive rich communication
   using audio, video, and data between two peers' web-browsers.  This
   document describes 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 Browser to exchange generic
   data from peer to peer.


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-01

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-data-channel-01


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


From igor.faynberg@alcatel-lucent.com  Fri Sep  7 14:04:35 2012
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Subject: Re: [rtcweb] Call for adoption of QoS draft
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I am all for it, especially that with Dan's agreeing to edit it, the 
document will be in good hands!

Igor

On 9/7/2012 11:27 AM, Ted Hardie wrote:
> Based on the feedback at the last working group meeting, the chairs
> believe that there is general support for creating a document that
> describes the QoS issues relating to RTCWEB.  We would like to call
> for adoption of draft-jennings-rtcweb-qos-00 as the input draft.  Dan
> Druta has agreed to serve as the WG document editor, should this
> document move forward.
>
> Please send comments to the list by  September 16, 2012
>
> Ted and Magnus
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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Subject: Re: [rtcweb] Call for adoption of QoS draft
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Ted Hardie wrote:
> describes the QoS issues relating to RTCWEB.  We would like to call
> for adoption of draft-jennings-rtcweb-qos-00 as the input draft.  Dan

I support the publication of this draft as a working group item with Dan 
Druta as the editor.

From ekr@rtfm.com  Fri Sep  7 14:33:23 2012
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From: Eric Rescorla <ekr@rtfm.com>
Date: Fri, 7 Sep 2012 14:32:38 -0700
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Call for adoption of QoS draft
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+1

On Fri, Sep 7, 2012 at 2:29 PM, Timothy B. Terriberry
<tterriberry@mozilla.com> wrote:
> Ted Hardie wrote:
>>
>> describes the QoS issues relating to RTCWEB.  We would like to call
>> for adoption of draft-jennings-rtcweb-qos-00 as the input draft.  Dan
>
>
> I support the publication of this draft as a working group item with Dan
> Druta as the editor.
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From partha@parthasarathi.co.in  Fri Sep  7 18:14:04 2012
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From: "Parthasarathi R" <partha@parthasarathi.co.in>
To: "'Eric Rescorla'" <ekr@rtfm.com>, "'Timothy B. Terriberry'" <tterriberry@mozilla.com>
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Subject: Re: [rtcweb] Call for adoption of QoS draft
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+1

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of
Eric Rescorla
Sent: Saturday, September 08, 2012 3:03 AM
To: Timothy B. Terriberry
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Call for adoption of QoS draft

+1

On Fri, Sep 7, 2012 at 2:29 PM, Timothy B. Terriberry
<tterriberry@mozilla.com> wrote:
> Ted Hardie wrote:
>>
>> describes the QoS issues relating to RTCWEB.  We would like to call
>> for adoption of draft-jennings-rtcweb-qos-00 as the input draft.  Dan
>
>
> I support the publication of this draft as a working group item with Dan
> Druta as the editor.
>
> _______________________________________________
> 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 denglingli@chinamobile.com  Fri Sep  7 19:43:04 2012
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+1

Lingli =20



=3D=3D=3D=3D=3D=3D=3D=C4=FA=D4=DA=C0=B4=D0=C5=D6=D0=D0=B4=B5=BD=A3=BA=3D=3D=
=3D=3D=3D=3D=3D
+1

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of
Eric Rescorla
Sent: Saturday, September 08, 2012 3:03 AM
To: Timothy B. Terriberry
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Call for adoption of QoS draft

+1

On Fri, Sep 7, 2012 at 2:29 PM, Timothy B. Terriberry
<tterriberry@mozilla.com> wrote:
> Ted Hardie wrote:
>>
>> describes the QoS issues relating to RTCWEB.  We would like to call
>> for adoption of draft-jennings-rtcweb-qos-00 as the input draft.  Dan
>
>
> I support the publication of this draft as a working group item with Dan
> Druta as the editor.
>
> =5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=
=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=
=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=
=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb


=B5=CB=C1=E9=C0=F2
=D6=D0=B9=FA=D2=C6=B6=AF=CD=A8=D0=C5=D1=D0=BE=BF=D4=BA
denglingli@chinamobile.com=

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From: Toerless Eckert <eckert@cisco.com>
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+1

On Fri, Sep 07, 2012 at 08:27:33AM -0700, Ted Hardie wrote:
> Based on the feedback at the last working group meeting, the chairs
> believe that there is general support for creating a document that
> describes the QoS issues relating to RTCWEB.  We would like to call
> for adoption of draft-jennings-rtcweb-qos-00 as the input draft.  Dan
> Druta has agreed to serve as the WG document editor, should this
> document move forward.
> 
> Please send comments to the list by  September 16, 2012
> 
> Ted and Magnus
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From harald@alvestrand.no  Mon Sep 10 07:10:53 2012
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Subject: Re: [rtcweb] Call for adoption of QoS draft
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I support the adoption of this draft.

Comments separately.


On 09/07/2012 05:27 PM, Ted Hardie wrote:
> Based on the feedback at the last working group meeting, the chairs
> believe that there is general support for creating a document that
> describes the QoS issues relating to RTCWEB.  We would like to call
> for adoption of draft-jennings-rtcweb-qos-00 as the input draft.  Dan
> Druta has agreed to serve as the WG document editor, should this
> document move forward.
>
> Please send comments to the list by  September 16, 2012
>
> Ted and Magnus
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From harald@alvestrand.no  Mon Sep 10 07:15:12 2012
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Subject: [rtcweb] Comments on draft-jennings-rtcweb-qos (Re: Call for adoption of QoS draft)
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Since this draft is mercifully short, I did a quick review.

Notes:

- Pointers to documentation on how browsers are expected to be able to 
set QCI and WiFI markings would be a good addition. Compared to those, 
DSCP codepoint setting is well understood.

- I find it good that these 3 mechanisms are the only ones considered in 
the draft. I'm making the leap of faith that we intend to state that an 
implementation MUST implement DSCP codepoint marking, SHOULD implement 
QCI and WiFI markings when attached to appropriate interfaces, and that 
no other mechanism is going to get a MUST or SHOULD recommendation from 
the WG. If I'm right, can we make that explicit?

- The restriction to 3 levels of priority seems to me like a reasonable 
simplification. Again, I want to make sure that this is an explicit and 
supported choice of the group.

                Harald


On 09/07/2012 05:27 PM, Ted Hardie wrote:
> Based on the feedback at the last working group meeting, the chairs
> believe that there is general support for creating a document that
> describes the QoS issues relating to RTCWEB.  We would like to call
> for adoption of draft-jennings-rtcweb-qos-00 as the input draft.  Dan
> Druta has agreed to serve as the WG document editor, should this
> document move forward.
>
> Please send comments to the list by  September 16, 2012
>
> Ted and Magnus
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From martin.thomson@gmail.com  Mon Sep 10 09:38:18 2012
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Subject: Re: [rtcweb] Comments on draft-jennings-rtcweb-qos (Re: Call for adoption of QoS draft)
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On 10 September 2012 07:15, Harald Alvestrand <harald@alvestrand.no> wrote:
> - Pointers to documentation on how browsers are expected to be able to set
> QCI and WiFI markings would be a good addition. Compared to those, DSCP
> codepoint setting is well understood.

I agree that this would be useful, if the draft really did make
implementation (and use) of those markings a "SHOULD".  See below.

> - I find it good that these 3 mechanisms are the only ones considered in the
> draft. I'm making the leap of faith that we intend to state that an
> implementation MUST implement DSCP codepoint marking, SHOULD implement QCI
> and WiFI markings when attached to appropriate interfaces, and that no other
> mechanism is going to get a MUST or SHOULD recommendation from the WG. If
> I'm right, can we make that explicit?

What do people think about NOT including QCI/WiFi markings?  Is it not
possible for a wireless interface to examine DSCP markings in order to
determine the markings for the link?  We should endeavour to maintain
the proper abstractions.

> - The restriction to 3 levels of priority seems to me like a reasonable
> simplification. Again, I want to make sure that this is an explicit and
> supported choice of the group.

Like Cullen, I don't see much room for a fourth priority.  Nor do I
see any particular advantage in having the fourth.  2 just doesn't
seem like enough though.

From martin.thomson@gmail.com  Mon Sep 10 09:51:00 2012
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Subject: Re: [rtcweb] Call for adoption of QoS draft
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I think that the work is valuable and useful.

However, I have concerns about how priorities are communicated to the browser.

I know that this introduces an interesting case of where the
artificial division of labour between rtcweb and WebRTC breaks down,
but the recent ruling in the W3C implies that this would require SDP
work (unless there is something that I missed).  That would then imply
that this (new) work doesn't fit the rtweb charter and should go to
MMUSIC.

Did I miss a detail?

--Martin

On 10 September 2012 07:11, Harald Alvestrand <harald@alvestrand.no> wrote:
> I support the adoption of this draft.
>
> Comments separately.
>
>
>
> On 09/07/2012 05:27 PM, Ted Hardie wrote:
>>
>> Based on the feedback at the last working group meeting, the chairs
>> believe that there is general support for creating a document that
>> describes the QoS issues relating to RTCWEB.  We would like to call
>> for adoption of draft-jennings-rtcweb-qos-00 as the input draft.  Dan
>> Druta has agreed to serve as the WG document editor, should this
>> document move forward.
>>
>> Please send comments to the list by  September 16, 2012
>>
>> Ted and Magnus
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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On Mon, Sep 10, 2012 at 9:50 AM, Martin Thomson
<martin.thomson@gmail.com> wrote:
> I think that the work is valuable and useful.
>
> However, I have concerns about how priorities are communicated to the browser.
>
> I know that this introduces an interesting case of where the
> artificial division of labour between rtcweb and WebRTC breaks down,
> but the recent ruling in the W3C implies that this would require SDP
> work (unless there is something that I missed).  That would then imply
> that this (new) work doesn't fit the rtweb charter and should go to
> MMUSIC.

Why would this require SDP work?

I would assume that this would be a setting on the media stream/track.
or alternately, a hint.

-Ekr

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On 10 September 2012 09:54, Eric Rescorla <ekr@rtfm.com> wrote:
> Why would this require SDP work?
>
> I would assume that this would be a setting on the media stream/track.
> or alternately, a hint.

You could, but that means that you have a more complicated API to use.
 You have to set priority locally, signal the chosen priority, set the
signalled priority, etc...  As a session characteristic, I assumed
that you would want to put it in the SDP.  Or is this something that
is extra specific to WebRTC such that you don't want to provide this
to other SDP users?

--Martin

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On Mon, Sep 10, 2012 at 10:43 AM, Martin Thomson
<martin.thomson@gmail.com> wrote:
> On 10 September 2012 09:54, Eric Rescorla <ekr@rtfm.com> wrote:
>> Why would this require SDP work?
>>
>> I would assume that this would be a setting on the media stream/track.
>> or alternately, a hint.
>
> You could, but that means that you have a more complicated API to use.
>  You have to set priority locally, signal the chosen priority, set the
> signalled priority, etc...

It's not clear to me that these values need to be the same on both
sides.

But if it *is* true that we want that, then of course it should be in SDP so
that the functionality is available to endpoints that don't have anything
to do with WebRTC.

-Ekr

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From: James Polk <jmpolk@cisco.com>
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At 12:51 PM 9/10/2012, Eric Rescorla wrote:
>On Mon, Sep 10, 2012 at 10:43 AM, Martin Thomson
><martin.thomson@gmail.com> wrote:
> > On 10 September 2012 09:54, Eric Rescorla <ekr@rtfm.com> wrote:
> >> Why would this require SDP work?
> >>
> >> I would assume that this would be a setting on the media stream/track.
> >> or alternately, a hint.
> >
> > You could, but that means that you have a more complicated API to use.
> >  You have to set priority locally, signal the chosen priority, set the
> > signalled priority, etc...
>
>It's not clear to me that these values need to be the same on both
>sides.
>
>But if it *is* true that we want that, then of course it should be in SDP so
>that the functionality is available to endpoints that don't have anything
>to do with WebRTC.'

MMUSIC has a WG item that should provide this indication/hint, in a 
trafficclass label attribute. Browsers identifying a small set of 
labels from that effort should do the trick. See

http://www.ietf.org/internet-drafts/draft-ietf-mmusic-traffic-class-for-sdp-02.txt

James


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


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> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Eric Rescorla
> Sent: Monday, September 10, 2012 10:52 AM
> To: Martin Thomson
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Call for adoption of QoS draft
> 
> On Mon, Sep 10, 2012 at 10:43 AM, Martin Thomson
> <martin.thomson@gmail.com> wrote:
> > On 10 September 2012 09:54, Eric Rescorla <ekr@rtfm.com> wrote:
> >> Why would this require SDP work?
> >>
> >> I would assume that this would be a setting on the media
> stream/track.
> >> or alternately, a hint.
> >
> > You could, but that means that you have a more complicated API to
> use.
> >  You have to set priority locally, signal the chosen priority, set
> the
> > signalled priority, etc...
> 
> It's not clear to me that these values need to be the same on both
> sides.

They don't need to be the same.  But, once a packet comes into a
network that needs a certain DSCP value to get certain quality
of service, the packet needs to be treated accordingly.  

For example, let's say Alice and Bob are on different networks.
Someone decided Alice's network needs DSCP of "1" for its high 
priority traffic, and someone else decided Bob's network needs 
DSCP of "2" for high priority traffic.  When Alice and Bob
have a real time phone call with each other, their outgoing
packets will receive QoS treatment, but the incoming packets
won't.  

Do the network managers that configured Alice and Bob's network
need to use the same DSCP configuration, or do we want to
solve this some other way?

-d

> But if it *is* true that we want that, then of course it should be in
> SDP so
> that the functionality is available to endpoints that don't have
> anything
> to do with WebRTC.
> 
> -Ekr
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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On 10 September 2012 14:12, James Polk <jmpolk@cisco.com> wrote:
> MMUSIC has a WG item that should provide this indication/hint, in a
> trafficclass label attribute. Browsers identifying a small set of labels
> from that effort should do the trick. See
>
> http://www.ietf.org/internet-drafts/draft-ietf-mmusic-traffic-class-for-sdp-02.txt

Thanks James, that looks like what is needed.  It's not exactly the
same thing as the priority draft, so I'm not sure how I'd map all the
way from priority (1-3) to traffic class labels to the DSCP markings
described in Cullen's draft.

It looks like some more definition work is needed.

From jmpolk@cisco.com  Mon Sep 10 17:23:59 2012
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At 04:27 PM 9/10/2012, Martin Thomson wrote:
>On 10 September 2012 14:12, James Polk <jmpolk@cisco.com> wrote:
> > MMUSIC has a WG item that should provide this indication/hint, in a
> > trafficclass label attribute. Browsers identifying a small set of labels
> > from that effort should do the trick. See
> >
> > 
> http://www.ietf.org/internet-drafts/draft-ietf-mmusic-traffic-class-for-sdp-02.txt
>
>Thanks James, that looks like what is needed.  It's not exactly the
>same thing as the priority draft, so I'm not sure how I'd map all the
>way from priority (1-3) to traffic class labels to the DSCP markings
>described in Cullen's draft.

I'm not sure only 3 priorities are needed, though I believe 3 will 
get used. The point of the trafficclass labels (TCL) is to add a 
level of abstraction to what the actual DSCPs need to be by allowing 
each domain that is aware of the type of traffic choose the 
appropriate DSCP for that type of traffic. After all, some bright 
spark might just come along and suggest that even though RTCweb uses 
the same 5-tuple for all traffic, they propose that the audio packets 
use a different DSCP than the video packets.

Further, someone might consider the type of AV conference they are in 
to determine which DSCP they are to use. Say the difference between a 
small window av call and an immersive telepresence room experience? 
Both types of calls have the exact same types of packets (audio and 
video, and maybe preservation sharing or whiteboarding).

Another is Dan's point in which Alice uses her highest priority 
marking (DSCP 1) when she contacts Bob, who uses his highest priority 
marking (DSCP 2) for the same call. Now, which DSCP is the highest 
priority marking, 1 or 2? Does it depend on the network/domain the 
packet is in?

The TCL draft explores that, and we already have customer feedback 
that's caused us to allow greater flexibility than we originally planned.


>It looks like some more definition work is needed.

perhaps, and I'm not at all opposed to comments to the TCL draft 
within MMUSIC to address whatever folks think is important.

James



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From: Martin Thomson <martin.thomson@gmail.com>
To: Eric Rescorla <ekr@rtfm.com>, rtcweb@ietf.org
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Subject: [rtcweb] What is consent?
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What specific properties of a STUN Binding request does the browser
use to determine that the peer has consented to receive packets?

>From what I can infer, it's simply the existence of a response.  That
is insufficient.

-security says:
   [...]  ICE provides this verification as
   well, by using the STUN credentials as a form of per-session shared
   secret.  Those credentials are known to the Web application, but
   would need to also be known and used by the STUN-receiving element to
   be useful.

But that's not strictly correct.  The Binding response necessarily
does not include MESSAGE-INTEGRITY, which is the critical tie to
per-session credentials.

What then would prevent the browser from mistaking a response from a
STUN server as consent?

The STUN server doesn't need to check MESSAGE-INTEGRITY and USERNAME
is systematically ignored where it is not used, so we can't rely on
the server doing something helpful for itself.  I imagine that
stunserver.org could be made safe because it doesn't provide
FINGERPRINT in responses (it's 3489 compliant, not 5389).  But what a
5389 STUN server?

Hopefully, I'm just missing something.  But this has been bugging me.

--Martin

From martin.thomson@gmail.com  Mon Sep 10 17:32:42 2012
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On 10 September 2012 17:16, James Polk <jmpolk@cisco.com> wrote:
> At 04:27 PM 9/10/2012, Martin Thomson wrote:
>> It looks like some more definition work is needed.
>
> perhaps, and I'm not at all opposed to comments to the TCL draft within
> MMUSIC to address whatever folks think is important.

I don't think that this necessarily translates to TCL draft work.
More that it is becoming clear to me that the qos draft is both a)
new, and b) not in alignment with the TCL stuff.  I'd be happy to be
convinced otherwise, and I'm sure that it can be rectified.

From ekr@rtfm.com  Mon Sep 10 17:45:29 2012
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On Mon, Sep 10, 2012 at 5:30 PM, Martin Thomson
<martin.thomson@gmail.com> wrote:
> What specific properties of a STUN Binding request does the browser
> use to determine that the peer has consented to receive packets?
>
> From what I can infer, it's simply the existence of a response.  That
> is insufficient.
>
> -security says:
>    [...]  ICE provides this verification as
>    well, by using the STUN credentials as a form of per-session shared
>    secret.  Those credentials are known to the Web application, but
>    would need to also be known and used by the STUN-receiving element to
>    be useful.
>
> But that's not strictly correct.  The Binding response necessarily
> does not include MESSAGE-INTEGRITY, which is the critical tie to
> per-session credentials.

You lost me here.  RFC 5389 specifically requires that if the
client provided MESSAGE-INTEGRITY in the Binding Response
in S 10.1.2:

   If these checks pass, the agent continues to process the request or
   indication.  Any response generated by a server MUST include the
   MESSAGE-INTEGRITY attribute, computed using the password utilized to
   authenticate the request.  The response MUST NOT contain the USERNAME
   attribute.

And the client is required to check it in 10.1.3:

10.1.3.  Receiving a Response

   The client looks for the MESSAGE-INTEGRITY attribute in the response.
   If present, the client computes the message integrity over the
   response as defined in Section 15.4, using the same password it
   utilized for the request.  If the resulting value matches the
   contents of the MESSAGE-INTEGRITY attribute, the response is
   considered authenticated.  If the value does not match, or if
   MESSAGE-INTEGRITY was absent, the response MUST be discarded, as if
   it was never received.  This means that retransmits, if applicable,
   will continue.


-Ekr

From martin.thomson@gmail.com  Tue Sep 11 09:09:03 2012
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Clearly, I did miss something.  Obviously, this relates to the bit in
ICE where it requires that the STUN backward compatibility measures
are not used.

But it didn't really answer my question.

Clearly, an RFC 3489 STUN server will not pass this basic test.  That
means that we can safely exclude servers like stunserver.org from the
set of consenting peers.

I suspect that for practical reasons we have to tolerate missing
MESSAGE-INTEGRITY and FINGERPRINT.  That means we can at least use old
servers to collect server reflexive addresses.  For those old servers
it's trivial to use the absence of either parameter as an indication
that they do not provide consent.

But the question stands: for STUN servers that are 5389 compliant,
what distinguishes the STUN server from a consenting peer?

--Martin

p.s. I reached my conclusion for the following reasons:
1. Section 7.1.3.2 of RFC 5245, which I presumed to be
complete...which it is, sort of.
2. MESSAGE-INTEGRITY is only of questionable utility in the response
if you assume that the server is ignoring requests that don't have a
valid MESSAGE-INTEGRITY.

On 10 September 2012 17:44, Eric Rescorla <ekr@rtfm.com> wrote:
> You lost me here.  RFC 5389 specifically requires that if the
> client provided MESSAGE-INTEGRITY in the Binding Response
> in S 10.1.2:
>
>    If these checks pass, the agent continues to process the request or
>    indication.  Any response generated by a server MUST include the
>    MESSAGE-INTEGRITY attribute, computed using the password utilized to
>    authenticate the request.  The response MUST NOT contain the USERNAME
>    attribute.
>
> And the client is required to check it in 10.1.3:
>
> 10.1.3.  Receiving a Response
>
>    The client looks for the MESSAGE-INTEGRITY attribute in the response.
>    If present, the client computes the message integrity over the
>    response as defined in Section 15.4, using the same password it
>    utilized for the request.  If the resulting value matches the
>    contents of the MESSAGE-INTEGRITY attribute, the response is
>    considered authenticated.  If the value does not match, or if
>    MESSAGE-INTEGRITY was absent, the response MUST be discarded, as if
>    it was never received.  This means that retransmits, if applicable,
>    will continue.
>
>
> -Ekr

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From: Eric Rescorla <ekr@rtfm.com>
Date: Tue, 11 Sep 2012 09:22:36 -0700
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Subject: Re: [rtcweb] What is consent?
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On Tue, Sep 11, 2012 at 9:09 AM, Martin Thomson
<martin.thomson@gmail.com> wrote:
> Clearly, I did miss something.  Obviously, this relates to the bit in
> ICE where it requires that the STUN backward compatibility measures
> are not used.
>
> But it didn't really answer my question.
>
> Clearly, an RFC 3489 STUN server will not pass this basic test.  That
> means that we can safely exclude servers like stunserver.org from the
> set of consenting peers.
>
> I suspect that for practical reasons we have to tolerate missing
> MESSAGE-INTEGRITY and FINGERPRINT.  That means we can at least use old
> servers to collect server reflexive addresses.  For those old servers
> it's trivial to use the absence of either parameter as an indication
> that they do not provide consent.
>
> But the question stands: for STUN servers that are 5389 compliant,
> what distinguishes the STUN server from a consenting peer?
>
> --Martin
>
> p.s. I reached my conclusion for the following reasons:
> 1. Section 7.1.3.2 of RFC 5245, which I presumed to be
> complete...which it is, sort of.
> 2. MESSAGE-INTEGRITY is only of questionable utility in the response
> if you assume that the server is ignoring requests that don't have a
> valid MESSAGE-INTEGRITY.

I'm really not following this.

Responses from the server need to *contain* the MESSAGE-INTEGRITY
field and otherwise are not taken as evidence of consent. This field can
only be generated by a server that has the ICE credentials. So, obviously,
a legacy STUN server won't generate that.

-Ekr

From martin.thomson@gmail.com  Tue Sep 11 09:34:49 2012
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Subject: Re: [rtcweb] What is consent?
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On 11 September 2012 09:22, Eric Rescorla <ekr@rtfm.com> wrote:
> I'm really not following this.
>
> Responses from the server need to *contain* the MESSAGE-INTEGRITY
> field and otherwise are not taken as evidence of consent. This field can
> only be generated by a server that has the ICE credentials. So, obviously,
> a legacy STUN server won't generate that.

It's clear that I'm just being thick.

The STUN server (legacy or otherwise) won't have a password and wont
generate MESSAGE-INTEGRITY.

I apologise for wasting your (and everyone else's) time.

From harald@alvestrand.no  Tue Sep 11 09:49:46 2012
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Subject: Re: [rtcweb] What is consent?
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On 09/11/2012 06:34 PM, Martin Thomson wrote:
> On 11 September 2012 09:22, Eric Rescorla <ekr@rtfm.com> wrote:
>> I'm really not following this.
>>
>> Responses from the server need to *contain* the MESSAGE-INTEGRITY
>> field and otherwise are not taken as evidence of consent. This field can
>> only be generated by a server that has the ICE credentials. So, obviously,
>> a legacy STUN server won't generate that.
> It's clear that I'm just being thick.
>
> The STUN server (legacy or otherwise) won't have a password and wont
> generate MESSAGE-INTEGRITY.
>
> I apologise for wasting your (and everyone else's) time.
The terms might have gotten confused, since STUN is used in two modes:

- With a STUN server, to identify reflexive addresses
- With a peer, to verify ability to communicate, and to verify continued 
consent.

We don't want to mandate credentials for the first one.
We do want to mandate credentials for the second.

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


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Subject: Re: [rtcweb] What is consent?
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> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Harald Alvestrand
> Sent: Tuesday, September 11, 2012 9:50 AM
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] What is consent?
> 
> On 09/11/2012 06:34 PM, Martin Thomson wrote:
> > On 11 September 2012 09:22, Eric Rescorla <ekr@rtfm.com> wrote:
> >> I'm really not following this.
> >>
> >> Responses from the server need to *contain* the MESSAGE-INTEGRITY
> >> field and otherwise are not taken as evidence of consent. This field
> can
> >> only be generated by a server that has the ICE credentials. So,
> obviously,
> >> a legacy STUN server won't generate that.
> > It's clear that I'm just being thick.
> >
> > The STUN server (legacy or otherwise) won't have a password and wont
> > generate MESSAGE-INTEGRITY.
> >
> > I apologise for wasting your (and everyone else's) time.
> The terms might have gotten confused, since STUN is used in two modes:
> 
> - With a STUN server, to identify reflexive addresses
> - With a peer, to verify ability to communicate, and to verify
> continued
> consent.
> 
> We don't want to mandate credentials for the first one.
> We do want to mandate credentials for the second.

Right.  The confusion is that ICE uses the STUN protocol for
two purposes:  to collect reflexive addresses, and to
perform connectivity checks with peers.  In RTCWEB, we plan
to additionally use connectivity checks for "consent", so
the STUN protocol will be used for three different purposes.

(And four if we count TURN, which also uses the STUN packet
format.)

-d



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Subject: Re: [rtcweb] What is consent?
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Eric Rescorla said: 

"Responses from the server need to *contain* the MESSAGE-INTEGRITY field and
otherwise are not taken as evidence of consent. This field can only be
generated by a server that has the ICE credentials. So, obviously, a legacy
STUN server won't generate that."

[BA] While the presence of the MESSAGE-INTEGRITY field is a necessary
condition, is it sufficient to demonstrate consent? For example, does the
nominated flag need to be set to true?  RFC 5245 Section 7.1.3.2.4 says: 

7.1.3.2.4.  Updating the Nominated Flag

   If the agent was a controlling agent, and it had included a USE-
   CANDIDATE attribute in the Binding request, the valid pair generated
   from that check has its nominated flag set to true.  This flag
   indicates that this valid pair should be used for media if it is the
   highest-priority one amongst those whose nominated flag is set.  This
   may conclude ICE processing for this media stream or all media
   streams; see Section 8.

   If the agent is the controlled agent, the response may be the result
   of a triggered check that was sent in response to a request that
   itself had the USE-CANDIDATE attribute.  This case is described in
   Section 7.2.1.5, and may now result in setting the nominated flag for
   the pair learned from the original request.


 



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To: "'Bernard Aboba'" <bernard_aboba@hotmail.com>, "'Eric Rescorla'" <ekr@rtfm.com>
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Subject: Re: [rtcweb] What is consent?
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> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Bernard Aboba
> Sent: Tuesday, September 11, 2012 12:29 PM
> To: 'Eric Rescorla'
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] What is consent?
> 
> Eric Rescorla said:
> 
> "Responses from the server need to *contain* the MESSAGE-INTEGRITY
> field and
> otherwise are not taken as evidence of consent. This field can only be
> generated by a server that has the ICE credentials. So, obviously, a
> legacy
> STUN server won't generate that."
> 
> [BA] While the presence of the MESSAGE-INTEGRITY field is a necessary
> condition, is it sufficient to demonstrate consent? For example, does
> the
> nominated flag need to be set to true?  RFC 5245 Section 7.1.3.2.4
> says:
> 
> 7.1.3.2.4.  Updating the Nominated Flag
> 
>    If the agent was a controlling agent, and it had included a USE-
>    CANDIDATE attribute in the Binding request, the valid pair generated
>    from that check has its nominated flag set to true.  This flag
>    indicates that this valid pair should be used for media if it is the
>    highest-priority one amongst those whose nominated flag is set.
> This
>    may conclude ICE processing for this media stream or all media
>    streams; see Section 8.
> 
>    If the agent is the controlled agent, the response may be the result
>    of a triggered check that was sent in response to a request that
>    itself had the USE-CANDIDATE attribute.  This case is described in
>    Section 7.2.1.5, and may now result in setting the nominated flag
> for
>    the pair learned from the original request.

For ICE Mobility (draft-wing-mmusic-ice-mobility), we might want to 
keep other candidates available, but inactive.  Over those other
candidates we would not signal USE-CANDIDATE, but we would want to
be able to switch to the other candidate as quickly as possible
(ideally, switch over immediately).  Similar considerations might 
apply to multipath RTP (draft-singh-avtcore-mprtp), too.
 
-d



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Subject: Re: [rtcweb] What is consent?
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>> Dan Wing said:
>=20
> For ICE Mobility (draft-wing-mmusic-ice-mobility), we might want to=20
> keep other candidates available, but inactive.  Over those other
> candidates we would not signal USE-CANDIDATE, but we would want to
> be able to switch to the other candidate as quickly as possible
> (ideally, switch over immediately).  Similar considerations might=20
> apply to multipath RTP (draft-singh-avtcore-mprtp), too.
>=20
> -d

[BA] My question is whether the browser has a legitimate reason to send medi=
a to a destination that is not part of a valid pair for which the nominated f=
lag is set to true, as per RFC 5245 Section 7.1.3.2.4.  For mobility you can=
 still signal USE-CANDIDATE but not use the pair if another pair had higher p=
riority.=20=

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From bernard_aboba@hotmail.com  Wed Sep 12 05:10:56 2012
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Date: Wed, 12 Sep 2012 07:17:07 -0500
To: Lishitao <lishitao@huawei.com>
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Subject: Re: [rtcweb] What is consent?
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On Sep 12, 2012, at 1:51, "Lishitao" <lishitao@huawei.com> wrote:
> Dan Wing said:
>=20
> For ICE Mobility (draft-wing-mmusic-ice-mobility), we might want to=20
> keep other candidates available, but inactive.  Over those other
> candidates we would not signal USE-CANDIDATE, but we would want to
> be able to switch to the other candidate as quickly as possible
> (ideally, switch over immediately).  Similar considerations might=20
> apply to multipath RTP=20
> [BA] My question is whether the browser has a legitimate reason to send me=
dia to a destination that is not part of a valid pair for which the nominate=
d flag is set to true, as per RFC 5245 Section 7.1.3.2.4.  For mobility you c=
an still signal USE-CANDIDATE but not use the pair if another pair had highe=
r priority.=20
> =20
> =20
> I donot think rfc 5245 allows this, it only allow only one pair with the U=
SE-CANDIDATE attribute present.

[BA] But in that case you would want to use the pair, no?=

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From magnus.westerlund@ericsson.com  Wed Sep 12 06:32:16 2012
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Subject: Re: [rtcweb] Filling in details on "trickle ICE"
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WG,

Based on this discussion the chairs conclusion is that there is required
to have a definition of how trickle ICE is to work so that interoperable
implementations can be made.

This WG is clearly not chartered to develop such a specification. Thus a
possible way forward is to do the work in the MMUSIC WG and develop an
ICE mode for trickle. Such work would require someone to take the role
of being proponent and drive the work in MMUSIC.

Thus we chair wonder if there exist people willing to take on this effort?

We chairs will meantime discuss with our ADs and the MMUSIC WG chairs
how they see on starting work in MMUSIC for this.

Cheers

Magnus Westerlund
(As chair)

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
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Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
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From martin.thomson@gmail.com  Wed Sep 12 08:57:11 2012
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Subject: Re: [rtcweb] What is consent?
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There are a number of things that concern me regarding this.

Firstly, USE-CANDIDATE can be sent on multiple pairs over time.
Particularly if the first pair fails for some reason.  Blocking
nominations after the first would prevent failover, as well as things
like multipath or confusing cases where there are multiple components
using the same set of local candidates (c.f., BUNDLE backward
compatibility scenarios).

I don't think that USE-CANDIDATE is sufficient or necessary for
consent.  Nomination is asymmetric in nature; only the controlling
peer can send it.  The controlled peer has no way to know that the
nomination check wasn't spoofed.  After all, that is the reason both
peers send connectivity checks.  The controlled peer then can't attach
any real significance to the nomination.  That isn't inherently a
problem - the controlled peer is still making checks and it is those
checks that verify that the controlling peer is willing to receive
packets at those addresses.

However, the underlying concern Bernard alludes to is a real one.  Not
only does consent indicate that I can send an arbitrary volume of data
to the checked host, as a natural consequence of ICE checking, there
is a decent chance that I get a large number of options, all of which
accept that arbitrary volume of data.

--Martin

On 12 September 2012 05:17, Bernard Aboba <bernard_aboba@hotmail.com> wrote:
> On Sep 12, 2012, at 1:51, "Lishitao" <lishitao@huawei.com> wrote:
>
> Dan Wing said:
>
>
> For ICE Mobility (draft-wing-mmusic-ice-mobility), we might want to
> keep other candidates available, but inactive.  Over those other
> candidates we would not signal USE-CANDIDATE, but we would want to
> be able to switch to the other candidate as quickly as possible
> (ideally, switch over immediately).  Similar considerations might
> apply to multipath RTP
>
> [BA] My question is whether the browser has a legitimate reason to send
> media to a destination that is not part of a valid pair for which the
> nominated flag is set to true, as per RFC 5245 Section 7.1.3.2.4.  For
> mobility you can still signal USE-CANDIDATE but not use the pair if another
> pair had higher priority.
>
>
>
>
>
> I donot think rfc 5245 allows this, it only allow only one pair with the
> USE-CANDIDATE attribute present.
>
>
> [BA] But in that case you would want to use the pair, no?
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

From martin.thomson@gmail.com  Wed Sep 12 09:05:24 2012
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Subject: Re: [rtcweb] Filling in details on "trickle ICE"
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+1

I can't currently commit to writing specifications, but I'm certainly
willing to provide reviews.

On 12 September 2012 06:32, Magnus Westerlund
<magnus.westerlund@ericsson.com> wrote:
> WG,
>
> Based on this discussion the chairs conclusion is that there is required
> to have a definition of how trickle ICE is to work so that interoperable
> implementations can be made.
>
> This WG is clearly not chartered to develop such a specification. Thus a
> possible way forward is to do the work in the MMUSIC WG and develop an
> ICE mode for trickle. Such work would require someone to take the role
> of being proponent and drive the work in MMUSIC.
>
> Thus we chair wonder if there exist people willing to take on this effort=
?
>
> We chairs will meantime discuss with our ADs and the MMUSIC WG chairs
> how they see on starting work in MMUSIC for this.
>
> Cheers
>
> Magnus Westerlund
> (As chair)
>
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=C3=A4r=C3=B6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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Subject: Re: [rtcweb] What is consent?
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> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Martin Thomson
> Sent: Wednesday, September 12, 2012 8:57 AM
> To: Bernard Aboba
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] What is consent?
> 
> There are a number of things that concern me regarding this.
> 
> Firstly, USE-CANDIDATE can be sent on multiple pairs over time.
> Particularly if the first pair fails for some reason.  Blocking
> nominations after the first would prevent failover, as well as things
> like multipath or confusing cases where there are multiple components
> using the same set of local candidates (c.f., BUNDLE backward
> compatibility scenarios).
> 
> I don't think that USE-CANDIDATE is sufficient or necessary for
> consent.  Nomination is asymmetric in nature; only the controlling
> peer can send it.  The controlled peer has no way to know that the
> nomination check wasn't spoofed.  After all, that is the reason both
> peers send connectivity checks.  The controlled peer then can't attach
> any real significance to the nomination.  That isn't inherently a
> problem - the controlled peer is still making checks and it is those
> checks that verify that the controlling peer is willing to receive
> packets at those addresses.
> 
> However, the underlying concern Bernard alludes to is a real one.  Not
> only does consent indicate that I can send an arbitrary volume of data
> to the checked host, as a natural consequence of ICE checking, there
> is a decent chance that I get a large number of options, all of which
> accept that arbitrary volume of data.

The volume of data is separate from USE-CANDIDATE, though.  ICE does
not currently have a way to indicate bandwidth.  

There was a bandwidth extension for TURN, but it did not achieve
working group consensus and is not in the TURN RFC.  There is
a bandwidth extension to ICE that Microsoft documents and I believe
use in their products, http://msdn.microsoft.com/en-us/library/cc339480, 
MS-ICE2BWM and MS-TURNBWM.

-d


> --Martin
> 
> On 12 September 2012 05:17, Bernard Aboba <bernard_aboba@hotmail.com>
> wrote:
> > On Sep 12, 2012, at 1:51, "Lishitao" <lishitao@huawei.com> wrote:
> >
> > Dan Wing said:
> >
> >
> > For ICE Mobility (draft-wing-mmusic-ice-mobility), we might want to
> > keep other candidates available, but inactive.  Over those other
> > candidates we would not signal USE-CANDIDATE, but we would want to
> > be able to switch to the other candidate as quickly as possible
> > (ideally, switch over immediately).  Similar considerations might
> > apply to multipath RTP
> >
> > [BA] My question is whether the browser has a legitimate reason to
> send
> > media to a destination that is not part of a valid pair for which the
> > nominated flag is set to true, as per RFC 5245 Section 7.1.3.2.4.
> For
> > mobility you can still signal USE-CANDIDATE but not use the pair if
> another
> > pair had higher priority.
> >
> >
> >
> >
> >
> > I donot think rfc 5245 allows this, it only allow only one pair with
> the
> > USE-CANDIDATE attribute present.
> >
> >
> > [BA] But in that case you would want to use the pair, no?
> >
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
> >
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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On 12 September 2012 10:39, Dan Wing <dwing@cisco.com> wrote:
> The volume of data is separate from USE-CANDIDATE, though.  ICE does
> not currently have a way to indicate bandwidth.

Yes.

> There was a bandwidth extension for TURN, but it did not achieve
> working group consensus and is not in the TURN RFC.  There is
> a bandwidth extension to ICE that Microsoft documents and I believe
> use in their products, http://msdn.microsoft.com/en-us/library/cc339480,
> MS-ICE2BWM and MS-TURNBWM.

There are two bandwidth-related things in use.  One is related to
bandwidth estimation and not really pertinent to this.  The other
potentially relevant one is the bandwidth parameter (the one that
never reached consensus) is used to manage bandwidth for relay
allocations.

The latter is interesting to me, particularly if that parameter can be
applied to ICE.  There are some gotchas that need careful
consideration, particularly around this area of having consent on
multiple valid candidate pairs.

--Martin

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Subject: Re: [rtcweb] What is consent?
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On 09/12/2012 10:55 AM, Martin Thomson wrote:
> On 12 September 2012 10:39, Dan Wing <dwing@cisco.com> wrote:
>> The volume of data is separate from USE-CANDIDATE, though.  ICE does
>> not currently have a way to indicate bandwidth.
> Yes.
>
>> There was a bandwidth extension for TURN, but it did not achieve
>> working group consensus and is not in the TURN RFC.  There is
>> a bandwidth extension to ICE that Microsoft documents and I believe
>> use in their products, http://msdn.microsoft.com/en-us/library/cc339480,
>> MS-ICE2BWM and MS-TURNBWM.
> There are two bandwidth-related things in use.  One is related to
> bandwidth estimation and not really pertinent to this.  The other
> potentially relevant one is the bandwidth parameter (the one that
> never reached consensus) is used to manage bandwidth for relay
> allocations.
>
> The latter is interesting to me, particularly if that parameter can be
> applied to ICE.  There are some gotchas that need careful
> consideration, particularly around this area of having consent on
> multiple valid candidate pairs.
I may be hopelessly naive here, but isn't the b= parameter of the SDP 
negotiation supposed to give an upper limit on how much data the 
recipient expects to receive on a connection?

RFC 3264 section 5.1 (unicast offer):

    If the bandwidth attribute is present for a stream, it indicates the
    desired bandwidth that the offerer would like to receive.

Section 6.1 (unicast answer):

    The answerer MAY include a bandwidth attribute for any media stream;
    this indicates the bandwidth that the answerer would like the offerer
    to use when sending media.

(note that "media stream" here is != MediaStream)

The browser being used to stage an attack will of course not know this 
value, since its signalling is controlled by the attacker, but the 
browser being attacked will be able to tell the difference between a 
within-contract amount of data and an out-of-contract amount of data.

               Harald


From martin.thomson@gmail.com  Wed Sep 12 16:49:43 2012
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Subject: Re: [rtcweb] What is consent?
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On 12 September 2012 16:39, Harald Alvestrand <harald@alvestrand.no> wrote:
> I may be hopelessly naive here, but isn't the b= parameter of the SDP
> negotiation supposed to give an upper limit on how much data the recipient
> expects to receive on a connection?

SDP is completely unauthenticated.  We have to assume that it is
provided by a web attacker.

That doesn't mean that the SDP bandwidth isn't a limit, or that it
shouldn't be respected.  On the other hand, if it's set to 10
bazillion, it doesn't give an RTP sender permission to send that much
data toward a peer.

--Martin

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Subject: Re: [rtcweb] What is consent?
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On Sep 12, 2012, at 6:40 PM, "Harald Alvestrand" <harald@alvestrand.no> wrot=
e:
>=20
> The browser being used to stage an attack will of course not know this val=
ue, since its signalling is controlled by the attacker, but the browser bein=
g attacked will be able to tell the difference between a within-contract amo=
unt of data and an out-of-contract amount of data.
>=20
>              Harald

[BA] Correct, but how does the receiver turn off the spigot? Is this handled=
 via circuit breakers?=20=

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Subject: Re: [rtcweb] What is consent?
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By not returning consent checks.

On Wed, Sep 12, 2012 at 5:02 PM, Bernard Aboba
<bernard_aboba@hotmail.com> wrote:
> On Sep 12, 2012, at 6:40 PM, "Harald Alvestrand" <harald@alvestrand.no> w=
rote:
>>
>> The browser being used to stage an attack will of course not know this v=
alue, since its signalling is controlled by the attacker, but the browser b=
eing attacked will be able to tell the difference between a within-contract=
 amount of data and an out-of-contract amount of data.
>>
>>              Harald
>
> [BA] Correct, but how does the receiver turn off the spigot? Is this hand=
led via circuit breakers?
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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From: "Yanqiang (Michael)" <michael.yan@huawei.com>
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Thread-Topic: [rtcweb] Fw: New Version Notification for draft-yan-rtcweb-desktop-cloud-00.txt
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From harald@alvestrand.no  Fri Sep 14 11:10:24 2012
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Subject: [rtcweb] FYI: New Version Notification for draft-alvestrand-mmusic-msid-00.txt
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The MMUSIC chairs have asked me to prepare a version of the MSID draft 
that is named using the draft-<authorname>-mmusic-<subject> convention, 
in preparation for polling the MMUSIC WG about adopting the draft.

This is it. Comments welcome!

                Harald


-------- Original Message --------
Subject: 	New Version Notification for draft-alvestrand-mmusic-msid-00.txt
Date: 	Wed, 12 Sep 2012 16:27:09 -0700
From: 	internet-drafts@ietf.org
To: 	harald@alvestrand.no



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

Filename:	 draft-alvestrand-mmusic-msid
Revision:	 00
Title:		 Cross Session Stream Identification in the Session Description Protocol
Creation date:	 2012-09-12
WG ID:		 Individual Submission
Number of pages: 11
URL:             http://www.ietf.org/internet-drafts/draft-alvestrand-mmusic-msid-00.txt
Status:          http://datatracker.ietf.org/doc/draft-alvestrand-mmusic-msid
Htmlized:        http://tools.ietf.org/html/draft-alvestrand-mmusic-msid-00


Abstract:
    This document specifies a grouping mechanism for RTP media streams
    that can be used to specify relations betweeen media streams within
    different RTP sessions.

    This mechanism is used to signal the association between the RTP
    concept of SSRC and the WebRTC concept of "media stream" / "media
    stream track" using SDP signalling.

    This document is an input document for discussion.  It should be
    discussed in the RTCWEB WG list, rtcweb@ietf.org.


                                                                                   


The IETF Secretariat





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<html>
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    <meta http-equiv="content-type" content="text/html; charset=UTF-8">
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    The MMUSIC chairs have asked me to prepare a version of the MSID
    draft that is named using the
    draft-&lt;authorname&gt;-mmusic-&lt;subject&gt; convention, in
    preparation for polling the MMUSIC WG about adopting the draft.<br>
    <br>
    This is it. Comments welcome!<br>
    <br>
    Â Â Â Â Â Â Â Â Â Â Â Â Â Â  Harald<br>
    <div class="moz-forward-container"><br>
      <br>
      -------- Original 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-mmusic-msid-00.txt</td>
          </tr>
          <tr>
            <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Date: </th>
            <td>Wed, 12 Sep 2012 16:27:09 -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><a class="moz-txt-link-abbreviated" href="mailto:harald@alvestrand.no">harald@alvestrand.no</a></td>
          </tr>
        </tbody>
      </table>
      <br>
      <br>
      <pre>A new version of I-D, draft-alvestrand-mmusic-msid-00.txt
has been successfully submitted by Harald Alvestrand and posted to the
IETF repository.

Filename:	 draft-alvestrand-mmusic-msid
Revision:	 00
Title:		 Cross Session Stream Identification in the Session Description Protocol
Creation date:	 2012-09-12
WG ID:		 Individual Submission
Number of pages: 11
URL:             <a class="moz-txt-link-freetext" href="http://www.ietf.org/internet-drafts/draft-alvestrand-mmusic-msid-00.txt">http://www.ietf.org/internet-drafts/draft-alvestrand-mmusic-msid-00.txt</a>
Status:          <a class="moz-txt-link-freetext" href="http://datatracker.ietf.org/doc/draft-alvestrand-mmusic-msid">http://datatracker.ietf.org/doc/draft-alvestrand-mmusic-msid</a>
Htmlized:        <a class="moz-txt-link-freetext" href="http://tools.ietf.org/html/draft-alvestrand-mmusic-msid-00">http://tools.ietf.org/html/draft-alvestrand-mmusic-msid-00</a>


Abstract:
   This document specifies a grouping mechanism for RTP media streams
   that can be used to specify relations betweeen media streams within
   different RTP sessions.

   This mechanism is used to signal the association between the RTP
   concept of SSRC and the WebRTC concept of "media stream" / "media
   stream track" using SDP signalling.

   This document is an input document for discussion.  It should be
   discussed in the RTCWEB WG list, <a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>.


                                                                                  


The IETF Secretariat

</pre>
      <br>
      <br>
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    <br>
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--------------020000040500060802040703--

From fluffy@cisco.com  Mon Sep 17 15:16:16 2012
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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: "James Polk (jmpolk)" <jmpolk@cisco.com>
Thread-Topic: [rtcweb] Call for adoption of QoS draft
Thread-Index: AQHNlSIMVep5hW0c9kGqvpC6gcbbUg==
Date: Mon, 17 Sep 2012 22:16:14 +0000
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References: <CA+9kkMBo10T=EgRXmkeB1vfB6MdUMVeWUpZowoXdP=E_+rm+mQ@mail.gmail.com> <504DF4F4.9080401@alvestrand.no> <CABkgnnVGkBseamm-rE8KjnoK0u=NwZoHjwoffRQATE4YmURfBw@mail.gmail.com> <CABcZeBNZd+kWup5cVhLwVLNNOJg8tP=dO-JD0dqxtd5PtKyAmA@mail.gmail.com> <CABkgnnU2tQgXTwgFAdBP4_0L9AviEUZ2kGOg0NH5xVQM8qZybQ@mail.gmail.com> <CABcZeBNiQqGKey8H0wqLz3VA9s1=0wySCf=406OdO12sQykf4w@mail.gmail.com> <201209102113.q8ALD0qQ018125@mtv-core-2.cisco.com> <CABkgnnUj9Q1M6KqEFxSqOZfUNxR2XY4aS1vcxDKpXNsSd+g6=A@mail.gmail.com> <201209110016.q8B0GQ6x025835@mtv-core-2.cisco.com>
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Cc: "<rtcweb@ietf.org>" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Call for adoption of QoS draft
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On Sep 10, 2012, at 18:16 , James Polk wrote:

> At 04:27 PM 9/10/2012, Martin Thomson wrote:
>> On 10 September 2012 14:12, James Polk <jmpolk@cisco.com> wrote:
>>> MMUSIC has a WG item that should provide this indication/hint, in a
>>> trafficclass label attribute. Browsers identifying a small set of label=
s
>>> from that effort should do the trick. See
>>>=20
>>> http://www.ietf.org/internet-drafts/draft-ietf-mmusic-traffic-class-for=
-sdp-02.txt
>>=20
>> Thanks James, that looks like what is needed.  It's not exactly the
>> same thing as the priority draft, so I'm not sure how I'd map all the
>> way from priority (1-3) to traffic class labels to the DSCP markings
>> described in Cullen's draft.
>=20
> I'm not sure only 3 priorities are needed, though I believe 3 will get us=
ed. The point of the trafficclass labels (TCL) is to add a level of abstrac=
tion to what the actual DSCPs need to be by allowing each domain that is aw=
are of the type of traffic choose the appropriate DSCP for that type of tra=
ffic.

The problem with using TCL here is the browser has no idea how to find out =
the mapping from TCL to DSCP for the domain it is currently running in so i=
t can't use TCL to set the DSCP on the media packets. Subha and I talked ab=
out this a bunch and can not see any way to make TCL work.=20


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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Martin Thomson <martin.thomson@gmail.com>
Thread-Topic: [rtcweb] Comments on draft-jennings-rtcweb-qos (Re: Call for adoption of QoS draft)
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Subject: Re: [rtcweb] Comments on draft-jennings-rtcweb-qos (Re: Call for adoption of QoS draft)
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On Sep 10, 2012, at 10:38 , Martin Thomson wrote:

> On 10 September 2012 07:15, Harald Alvestrand <harald@alvestrand.no> wrot=
e:
>> - Pointers to documentation on how browsers are expected to be able to s=
et
>> QCI and WiFI markings would be a good addition. Compared to those, DSCP
>> codepoint setting is well understood.
>=20
> I agree that this would be useful, if the draft really did make
> implementation (and use) of those markings a "SHOULD".  See below.
>=20
>> - I find it good that these 3 mechanisms are the only ones considered in=
 the
>> draft. I'm making the leap of faith that we intend to state that an
>> implementation MUST implement DSCP codepoint marking, SHOULD implement Q=
CI
>> and WiFI markings when attached to appropriate interfaces, and that no o=
ther
>> mechanism is going to get a MUST or SHOULD recommendation from the WG. I=
f
>> I'm right, can we make that explicit?
>=20
> What do people think about NOT including QCI/WiFi markings?  Is it not
> possible for a wireless interface to examine DSCP markings in order to
> determine the markings for the link?  We should endeavour to maintain
> the proper abstractions.

On some OS is it is possible to see them but ignoring that =85 think for a =
second  about what these marketing are meant to be used for -  they are mor=
e meant to be seen by the network not the browser at the far end so not sur=
e this matters much if the some OS can't provide this information to the ap=
plication=20



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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Harald Alvestrand <harald@alvestrand.no>
Thread-Topic: [rtcweb] Comments on draft-jennings-rtcweb-qos (Re: Call for adoption of QoS draft)
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On Sep 10, 2012, at 8:15 , Harald Alvestrand wrote:

> Since this draft is mercifully short, I did a quick review.
>=20
> Notes:
>=20
> - Pointers to documentation on how browsers are expected to be able to se=
t QCI and WiFI markings would be a good addition. Compared to those, DSCP c=
odepoint setting is well understood.

I think the draft needs all the correct pointers to the specs for this the =
QCI / WIFI stuff as well as pointers to the normative mapping tables define=
d by the appropriate SDOs for them. If the device is sending a packet on a =
network that supports QCI or Wifi marking, it can use them.  I'm not really=
 sure what else to say. (Note that I don't think this could be a MUST use )=
.

>=20
> - I find it good that these 3 mechanisms are the only ones considered in =
the draft. I'm making the leap of faith that we intend to state that an imp=
lementation MUST implement DSCP code point

I have not tracked this all down but have been told that it might not be po=
ssible for a browser to do this in some versions of the windows operating s=
ystem so I suspect the DSCP would end up being SHOULD not a MUST  =20

> m unforotanly arking, SHOULD implement QCI and WiFI markings when attache=
d to appropriate interfaces, and that no other mechanism is going to get a =
MUST or SHOULD recommendation from the WG. If I'm right, can we make that e=
xplicit?
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Martin Thomson <martin.thomson@gmail.com>
Thread-Topic: [rtcweb] Call for adoption of QoS draft
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Date: Mon, 17 Sep 2012 22:24:31 +0000
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I'd suggest using hint for this.

And I think this is a topic for webrtc not this wg.=20


On Sep 10, 2012, at 11:43 , Martin Thomson wrote:

> On 10 September 2012 09:54, Eric Rescorla <ekr@rtfm.com> wrote:
>> Why would this require SDP work?
>>=20
>> I would assume that this would be a setting on the media stream/track.
>> or alternately, a hint.
>=20
> You could, but that means that you have a more complicated API to use.
> You have to set priority locally, signal the chosen priority, set the
> signalled priority, etc...  As a session characteristic, I assumed
> that you would want to put it in the SDP.  Or is this something that
> is extra specific to WebRTC such that you don't want to provide this
> to other SDP users?
>=20
> --Martin
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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Subject: Re: [rtcweb] Comments on draft-jennings-rtcweb-qos (Re: Call for adoption of QoS draft)
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On 17 September 2012 15:16, Cullen Jennings (fluffy) <fluffy@cisco.com> wro=
te:
>> What do people think about NOT including QCI/WiFi markings?  Is it not
>> possible for a wireless interface to examine DSCP markings in order to
>> determine the markings for the link?  We should endeavour to maintain
>> the proper abstractions.
>
> On some OS is it is possible to see them but ignoring that =E2=80=A6 thin=
k for a second  about what these marketing are meant to be used for -  they=
 are more meant to be seen by the network not the browser at the far end so=
 not sure this matters much if the some OS can't provide this information t=
o the application

I am not suggesting anything about reading the markings off packets.
Reading markings off is - as you say - close to useless.

I asked if it were possible to consider *not* specifying what link
layer markings are made for packets that are *sent*.  The suggestion
was that if DSCP markings were made, as suggested, the packets could
be marked on the way onto the link by the wireless driver using
whatever link-specific markings are appropriate.  That might lead to a
mapping from DSCP to QCI and WiFi markings, but not necessarily.  That
would be different to the straight-from-the-top application priority
to link-layer, as described in the draft.

From martin.thomson@gmail.com  Mon Sep 17 15:56:50 2012
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Subject: Re: [rtcweb] Comments on draft-jennings-rtcweb-qos (Re: Call for adoption of QoS draft)
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On 17 September 2012 15:19, Cullen Jennings (fluffy) <fluffy@cisco.com> wrote:
> I have not tracked this all down but have been told that it might not be possible for a browser to do this in some versions of the windows operating system so I suspect the DSCP would end up being SHOULD not a MUST

As I understand it, setting DSCP can be disabled by group policy.
This is an explicit request from certain network operators, usually
enterprises who have the ability to set such policies.

As a matter of interest, some of the same networks do still perform
traffic prioritization for audio and video traffic.  They do it based
on port ranges.  You could infer some requirements from that.

From harald@alvestrand.no  Mon Sep 17 16:45:54 2012
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Cc: "Cullen Jennings \(fluffy\)" <fluffy@cisco.com>, "<rtcweb@ietf.org>" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Comments on draft-jennings-rtcweb-qos (Re: Call for adoption of QoS draft)
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On 09/17/2012 03:56 PM, Martin Thomson wrote:
> On 17 September 2012 15:19, Cullen Jennings (fluffy) <fluffy@cisco.com> wrote:
>> I have not tracked this all down but have been told that it might not be possible for a browser to do this in some versions of the windows operating system so I suspect the DSCP would end up being SHOULD not a MUST
> As I understand it, setting DSCP can be disabled by group policy.
> This is an explicit request from certain network operators, usually
> enterprises who have the ability to set such policies.
>
> As a matter of interest, some of the same networks do still perform
> traffic prioritization for audio and video traffic.  They do it based
> on port ranges.  You could infer some requirements from that.
<choosing to not go with the witty repartee>

The normal case for client-server communication is that the server has 
fixed port numbers, and the client just uses the port number the OS 
gives it; the network's priority setting is then done on the server port 
number range.

There are "fun" issues with trying to set port numbers on clients, 
including how to ACL them (I don't know of an OS that can do that), and 
the problem of port number collision.
For client-client communication, I think DSCP has a better chance of 
succeeding (not a big chance. Just a better chance).


From Markus.Isomaki@nokia.com  Tue Sep 18 01:08:04 2012
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From: <Markus.Isomaki@nokia.com>
To: <martin.thomson@gmail.com>, <harald@alvestrand.no>
Thread-Topic: draft-jennings-rtcweb-qos: QCI "marking"?
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Subject: [rtcweb] draft-jennings-rtcweb-qos: QCI "marking"?
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Hi,

Martin Thomson wrote:
>
>On 10 September 2012 07:15, Harald Alvestrand <harald@alvestrand.no>
>wrote:
>> - Pointers to documentation on how browsers are expected to be able to
>> set QCI and WiFI markings would be a good addition. Compared to those,
>> DSCP codepoint setting is well understood.
>
<snip>
>
>What do people think about NOT including QCI/WiFi markings?  Is it not
>possible for a wireless interface to examine DSCP markings in order to
>determine the markings for the link?  We should endeavour to maintain the
>proper abstractions.
>

I'm not even sure if the browser or even the OS can influence much the QCI =
or bearer selection in LTE. I think the model is such that the "network" se=
ts up these special non-best-effort bearers, and gives the endpoint the cla=
ssifier rules related to them. The rules tell the endpoint how to map outgo=
ing packets to the bearers. Typically the rules are based on source/destina=
tion IP and port, and the transport protocol. Someone may be able to confir=
m, but I believe DSCP can be used too. But basically the rules come from th=
e network and the endpoint just follows them.=20

So, somehow the browser or the JS app would first have to convince the acce=
ss network provider to setup the special bearer so that the right flows get=
 classified to it. In IMS the SIP proxy reads the SDP and tells some networ=
k element what audio or video flows are being setup and what their 5-tuples=
 are, so the network is being able to setup the bearers correctly for them.=
 The endpoint merely waits for this setup and acts accordingly. This works =
as the SIP proxy is run by the access operator. In RTCWeb the same could wo=
rk if the calling site had a relationship with the access operator. I've he=
ard that there is some work by some operators to offer Web APIs for 3rd par=
ty apps to do the bearer setup requests more ad hoc. Most of this may be ou=
tside the scope of RTCWeb but I think the document should have some explana=
tion on how it works from the OS, browser, JS client, calling site, and mob=
ile operator perspective.=20

On the "transport" or "data path" level I don't think the browser needs to =
understand anything about QCIs. DSCP marking should be good enough.=20

Anyone with more LTE/mobile clue that could comment on this?=20

Markus=20

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On Sep 18, 2012 1:08 AM, <Markus.Isomaki@nokia.com> wrote:
>
> Hi,
>
> Martin Thomson wrote:
> >
> >On 10 September 2012 07:15, Harald Alvestrand <harald@alvestrand.no>
> >wrote:
> >> - Pointers to documentation on how browsers are expected to be able to
> >> set QCI and WiFI markings would be a good addition. Compared to those,
> >> DSCP codepoint setting is well understood.
> >
> <snip>
> >
> >What do people think about NOT including QCI/WiFi markings?  Is it not
> >possible for a wireless interface to examine DSCP markings in order to
> >determine the markings for the link?  We should endeavour to maintain the
> >proper abstractions.
> >
>
> I'm not even sure if the browser or even the OS can influence much the
QCI or bearer selection in LTE. I think the model is such that the
"network" sets up these special non-best-effort bearers, and gives the
endpoint the classifier rules related to them. The rules tell the endpoint
how to map outgoing packets to the bearers. Typically the rules are based
on source/destination IP and port, and the transport protocol. Someone may
be able to confirm, but I believe DSCP can be used too. But basically the
rules come from the network and the endpoint just follows them.
>
> So, somehow the browser or the JS app would first have to convince the
access network provider to setup the special bearer so that the right flows
get classified to it. In IMS the SIP proxy reads the SDP and tells some
network element what audio or video flows are being setup and what their
5-tuples are, so the network is being able to setup the bearers correctly
for them. The endpoint merely waits for this setup and acts accordingly.
This works as the SIP proxy is run by the access operator. In RTCWeb the
same could work if the calling site had a relationship with the access
operator. I've heard that there is some work by some operators to offer Web
APIs for 3rd party apps to do the bearer setup requests more ad hoc. Most
of this may be outside the scope of RTCWeb but I think the document should
have some explanation on how it works from the OS, browser, JS client,
calling site, and mobile operator perspective.
>
> On the "transport" or "data path" level I don't think the browser needs
to understand anything about QCIs. DSCP marking should be good enough.
>
> Anyone with more LTE/mobile clue that could comment on this?
>
> Markus
>

Your understanding is similar to mine.

Generally speaking, qos in 3gpp networks is network initiated and
controlled by a PCRF. The UEs are generally not trusted, as they may abuse
qos.

I am told that qos is an important differentiator for the operator run
services and consequently will not be generally available to 3rd party apps
such as webrtc.

I am generally sceptical about qos on the internets, and we may be getting
into a dangerous area if we start to assume networks will give anything
more than best effort to webrtc.

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

--f46d040714cf84688c04c9f9dabd
Content-Type: text/html; charset=ISO-8859-1
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<p><br>
On Sep 18, 2012 1:08 AM, &lt;<a href=3D"mailto:Markus.Isomaki@nokia.com">Ma=
rkus.Isomaki@nokia.com</a>&gt; wrote:<br>
&gt;<br>
&gt; Hi,<br>
&gt;<br>
&gt; Martin Thomson wrote:<br>
&gt; &gt;<br>
&gt; &gt;On 10 September 2012 07:15, Harald Alvestrand &lt;<a href=3D"mailt=
o:harald@alvestrand.no">harald@alvestrand.no</a>&gt;<br>
&gt; &gt;wrote:<br>
&gt; &gt;&gt; - Pointers to documentation on how browsers are expected to b=
e able to<br>
&gt; &gt;&gt; set QCI and WiFI markings would be a good addition. Compared =
to those,<br>
&gt; &gt;&gt; DSCP codepoint setting is well understood.<br>
&gt; &gt;<br>
&gt; &lt;snip&gt;<br>
&gt; &gt;<br>
&gt; &gt;What do people think about NOT including QCI/WiFi markings? =A0Is =
it not<br>
&gt; &gt;possible for a wireless interface to examine DSCP markings in orde=
r to<br>
&gt; &gt;determine the markings for the link? =A0We should endeavour to mai=
ntain the<br>
&gt; &gt;proper abstractions.<br>
&gt; &gt;<br>
&gt;<br>
&gt; I&#39;m not even sure if the browser or even the OS can influence much=
 the QCI or bearer selection in LTE. I think the model is such that the &qu=
ot;network&quot; sets up these special non-best-effort bearers, and gives t=
he endpoint the classifier rules related to them. The rules tell the endpoi=
nt how to map outgoing packets to the bearers. Typically the rules are base=
d on source/destination IP and port, and the transport protocol. Someone ma=
y be able to confirm, but I believe DSCP can be used too. But basically the=
 rules come from the network and the endpoint just follows them.<br>

&gt;<br>
&gt; So, somehow the browser or the JS app would first have to convince the=
 access network provider to setup the special bearer so that the right flow=
s get classified to it. In IMS the SIP proxy reads the SDP and tells some n=
etwork element what audio or video flows are being setup and what their 5-t=
uples are, so the network is being able to setup the bearers correctly for =
them. The endpoint merely waits for this setup and acts accordingly. This w=
orks as the SIP proxy is run by the access operator. In RTCWeb the same cou=
ld work if the calling site had a relationship with the access operator. I&=
#39;ve heard that there is some work by some operators to offer Web APIs fo=
r 3rd party apps to do the bearer setup requests more ad hoc. Most of this =
may be outside the scope of RTCWeb but I think the document should have som=
e explanation on how it works from the OS, browser, JS client, calling site=
, and mobile operator perspective.<br>

&gt;<br>
&gt; On the &quot;transport&quot; or &quot;data path&quot; level I don&#39;=
t think the browser needs to understand anything about QCIs. DSCP marking s=
hould be good enough.<br>
&gt;<br>
&gt; Anyone with more LTE/mobile clue that could comment on this?<br>
&gt;<br>
&gt; Markus<br>
&gt;</p>
<p>Your understanding is similar to mine.</p>
<p>Generally speaking, qos in 3gpp networks is network initiated and contro=
lled by a PCRF. The UEs are generally not trusted, as they may abuse qos.</=
p>
<p>I am told that qos is an important differentiator for the operator run s=
ervices and consequently will not be generally available to 3rd party apps =
such as webrtc.</p>
<p>I am generally sceptical about qos on the internets, and we may be getti=
ng into a dangerous area if we start to assume networks will give anything =
more than best effort to webrtc.</p>
<p>CB _______________________________________________<br>
&gt; rtcweb mailing list<br>
&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.i=
etf.org/mailman/listinfo/rtcweb</a><br>
</p>

--f46d040714cf84688c04c9f9dabd--

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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>, "public-webrtc@w3.org" <public-webrtc@w3.org>
Thread-Topic: Background info about what is going on WebRTC / RTCWeb
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I get lots of questions from folks trying to understand what is going on in=
 these working groups so I tried to put together a bit of a  presentation t=
hat is my best guess of what the system we are standardizing looks like and=
 helps provide the background info that people can better understand and pa=
rticipate in some of the debates going on. You can find it at

http://vimeo.com/cullenfluffyjennings/rtcwebexplained

It is probably wrong in some ways and I know there are parts some folks wou=
ld disagree with but it is just by best guess given the current information=
 on hand. If anyone wants to use the slides or parts of this for some other=
 presentation, just get drop and email.=20

Cullen


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On Sep 18, 2012, at 8:20 AM, Cullen Jennings <fluffy@cisco.com> wrote:

>=20
> I get lots of questions from folks trying to understand what is going on =
in these working groups so I tried to put together a bit of a  presentation=
 that is my best guess of what the system we are standardizing looks like a=
nd helps provide the background info that people can better understand and =
participate in some of the debates going on. You can find it at
>=20
> http://vimeo.com/cullenfluffyjennings/rtcwebexplained
>=20
> It is probably wrong in some ways and I know there are parts some folks w=
ould disagree with but it is just by best guess given the current informati=
on on hand. If anyone wants to use the slides or parts of this for some oth=
er presentation, just get drop and email.=20
>=20
> Cullen
>=20

Because a few people asked, there is a VP8 version at=20

http://dl.dropbox.com/u/17089001/RtcWeb/RtcWebTake1-360p.webm

PDF slides at http://dl.dropbox.com/u/17089001/RtcWeb/WebRTC_v8.pdf but be =
warned, they make no sense without a talk track to go with them.=20

Cullen

<rant> I've got all the video tools from adobe, apple, and avid but to crea=
te this I had to grab the right check in of ffmpeg from the SVN for ffmeg, =
then apply a patch from some other page, then deal with CLI options that ma=
kes cisco router configuration look downright friendly and simple. Ask me h=
ow I love living on the bleeding edge =85 </rant> =

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From: "Eggert, Lars" <lars@netapp.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
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Hi,

agree that the QCI stuff is probably not useful to discuss.

The DSCP content is almost fully redundant with what's in RFC4594 (which =
is in the references but not cited). And RFC4594 has a much better =
discussion on what DiffServ is and isn't.

For WLAN, 802.11e is what you want to point to, I guess.

So this doc - if it's even needed - could be MUCH shorter, by basically =
just saying "read RFC4594 and map RTCweb flows into the classes defined =
there like this" and do the equivalent for 802.11e.

Lars=

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From: Justin Uberti <juberti@google.com>
Date: Thu, 20 Sep 2012 12:00:01 -0700
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Subject: Re: [rtcweb] Filling in details on "trickle ICE"
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I have a few other things on my plate right now, but I'd be interested in
working on this.


On Wed, Sep 12, 2012 at 9:05 AM, Martin Thomson <martin.thomson@gmail.com>w=
rote:

> +1
>
> I can't currently commit to writing specifications, but I'm certainly
> willing to provide reviews.
>
> On 12 September 2012 06:32, Magnus Westerlund
> <magnus.westerlund@ericsson.com> wrote:
> > WG,
> >
> > Based on this discussion the chairs conclusion is that there is require=
d
> > to have a definition of how trickle ICE is to work so that interoperabl=
e
> > implementations can be made.
> >
> > This WG is clearly not chartered to develop such a specification. Thus =
a
> > possible way forward is to do the work in the MMUSIC WG and develop an
> > ICE mode for trickle. Such work would require someone to take the role
> > of being proponent and drive the work in MMUSIC.
> >
> > Thus we chair wonder if there exist people willing to take on this
> effort?
> >
> > We chairs will meantime discuss with our ADs and the MMUSIC WG chairs
> > how they see on starting work in MMUSIC for this.
> >
> > Cheers
> >
> > Magnus Westerlund
> > (As chair)
> >
> > ----------------------------------------------------------------------
> > Multimedia Technologies, Ericsson Research EAB/TVM
> > ----------------------------------------------------------------------
> > Ericsson AB                | Phone  +46 10 7148287
> > F=C3=A4r=C3=B6gatan 6                | Mobile +46 73 0949079
> > SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> > ----------------------------------------------------------------------
> >
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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I have a few other things on my plate right now, but I&#39;d be interested =
in working on this.<div class=3D"gmail_extra"><br><br><div class=3D"gmail_q=
uote">On Wed, Sep 12, 2012 at 9:05 AM, Martin Thomson <span dir=3D"ltr">&lt=
;<a href=3D"mailto:martin.thomson@gmail.com" target=3D"_blank">martin.thoms=
on@gmail.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">+1<br>
<br>
I can&#39;t currently commit to writing specifications, but I&#39;m certain=
ly<br>
willing to provide reviews.<br>
<br>
On 12 September 2012 06:32, Magnus Westerlund<br>
<div class=3D"HOEnZb"><div class=3D"h5">&lt;<a href=3D"mailto:magnus.wester=
lund@ericsson.com">magnus.westerlund@ericsson.com</a>&gt; wrote:<br>
&gt; WG,<br>
&gt;<br>
&gt; Based on this discussion the chairs conclusion is that there is requir=
ed<br>
&gt; to have a definition of how trickle ICE is to work so that interoperab=
le<br>
&gt; implementations can be made.<br>
&gt;<br>
&gt; This WG is clearly not chartered to develop such a specification. Thus=
 a<br>
&gt; possible way forward is to do the work in the MMUSIC WG and develop an=
<br>
&gt; ICE mode for trickle. Such work would require someone to take the role=
<br>
&gt; of being proponent and drive the work in MMUSIC.<br>
&gt;<br>
&gt; Thus we chair wonder if there exist people willing to take on this eff=
ort?<br>
&gt;<br>
&gt; We chairs will meantime discuss with our ADs and the MMUSIC WG chairs<=
br>
&gt; how they see on starting work in MMUSIC for this.<br>
&gt;<br>
&gt; Cheers<br>
&gt;<br>
&gt; Magnus Westerlund<br>
&gt; (As chair)<br>
&gt;<br>
&gt; ----------------------------------------------------------------------=
<br>
&gt; Multimedia Technologies, Ericsson Research EAB/TVM<br>
&gt; ----------------------------------------------------------------------=
<br>
&gt; Ericsson AB =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0| P=
hone =C2=A0<a href=3D"tel:%2B46%2010%207148287" value=3D"+46107148287">+46 =
10 7148287</a><br>
&gt; F=C3=A4r=C3=B6gatan 6 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0=
 =C2=A0| Mobile <a href=3D"tel:%2B46%2073%200949079" value=3D"+46730949079"=
>+46 73 0949079</a><br>
&gt; SE-164 80 Stockholm, Sweden| mailto: <a href=3D"mailto:magnus.westerlu=
nd@ericsson.com">magnus.westerlund@ericsson.com</a><br>
&gt; ----------------------------------------------------------------------=
<br>
&gt;<br>
&gt; _______________________________________________<br>
&gt; rtcweb mailing list<br>
&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
_______________________________________________<br>
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>

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Date: Thu, 20 Sep 2012 21:11:58 +0200
From: Emil Ivov <emcho@jitsi.org>
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Subject: Re: [rtcweb] Filling in details on "trickle ICE"
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I'd be willing to work on this (writing and/or reviewing).

Emil

On 12.09.12, 15:32, Magnus Westerlund wrote:
> WG,
>=20
> Based on this discussion the chairs conclusion is that there is require=
d
> to have a definition of how trickle ICE is to work so that interoperabl=
e
> implementations can be made.
>=20
> This WG is clearly not chartered to develop such a specification. Thus =
a
> possible way forward is to do the work in the MMUSIC WG and develop an
> ICE mode for trickle. Such work would require someone to take the role
> of being proponent and drive the work in MMUSIC.
>=20
> Thus we chair wonder if there exist people willing to take on this effo=
rt?
>=20
> We chairs will meantime discuss with our ADs and the MMUSIC WG chairs
> how they see on starting work in MMUSIC for this.
>=20
> Cheers
>=20
> Magnus Westerlund
> (As chair)
>=20
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=E4r=F6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20

--=20
Emil Ivov, Ph.D.                       67000 Strasbourg,
Project Lead                           France
Jitsi
emcho@jitsi.org                        PHONE: +33.1.77.62.43.30
https://jitsi.org                      FAX:   +33.1.77.62.47.31



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Date: Thu, 20 Sep 2012 17:06:56 -0700
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Subject: Re: [rtcweb] Filling in details on "trickle ICE"
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--20cf3074b9fcda8e3504ca2b054f
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Eric Rescorla and I discussed some of the tricky trickle issues and wrote
them down in this doc:
https://docs.google.com/document/d/1L5SP9YQxXRf-2KObnDI5rgRCRnOv9EbhXsHsVis=
Wp1Q/edit#.
I think that would be a good starting point for an I-D on this.


On Thu, Sep 20, 2012 at 12:11 PM, Emil Ivov <emcho@jitsi.org> wrote:

> I'd be willing to work on this (writing and/or reviewing).
>
> Emil
>
> On 12.09.12, 15:32, Magnus Westerlund wrote:
> > WG,
> >
> > Based on this discussion the chairs conclusion is that there is require=
d
> > to have a definition of how trickle ICE is to work so that interoperabl=
e
> > implementations can be made.
> >
> > This WG is clearly not chartered to develop such a specification. Thus =
a
> > possible way forward is to do the work in the MMUSIC WG and develop an
> > ICE mode for trickle. Such work would require someone to take the role
> > of being proponent and drive the work in MMUSIC.
> >
> > Thus we chair wonder if there exist people willing to take on this
> effort?
> >
> > We chairs will meantime discuss with our ADs and the MMUSIC WG chairs
> > how they see on starting work in MMUSIC for this.
> >
> > Cheers
> >
> > Magnus Westerlund
> > (As chair)
> >
> > ----------------------------------------------------------------------
> > Multimedia Technologies, Ericsson Research EAB/TVM
> > ----------------------------------------------------------------------
> > Ericsson AB                | Phone  +46 10 7148287
> > F=C3=A4r=C3=B6gatan 6                | Mobile +46 73 0949079
> > SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> > ----------------------------------------------------------------------
> >
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
> >
>
> --
> Emil Ivov, Ph.D.                       67000 Strasbourg,
> Project Lead                           France
> Jitsi
> emcho@jitsi.org                        PHONE: +33.1.77.62.43.30
> https://jitsi.org                      FAX:   +33.1.77.62.47.31
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--20cf3074b9fcda8e3504ca2b054f
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Eric Rescorla and I discussed some of the tricky trickle issues and wrote t=
hem down in this doc:=C2=A0<a href=3D"https://docs.google.com/document/d/1L=
5SP9YQxXRf-2KObnDI5rgRCRnOv9EbhXsHsVisWp1Q/edit#">https://docs.google.com/d=
ocument/d/1L5SP9YQxXRf-2KObnDI5rgRCRnOv9EbhXsHsVisWp1Q/edit#</a>. I think t=
hat would be a good starting point for an I-D on this.<div class=3D"gmail_e=
xtra">

<br><br><div class=3D"gmail_quote">On Thu, Sep 20, 2012 at 12:11 PM, Emil I=
vov <span dir=3D"ltr">&lt;<a href=3D"mailto:emcho@jitsi.org" target=3D"_bla=
nk">emcho@jitsi.org</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quo=
te" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"=
>

I&#39;d be willing to work on this (writing and/or reviewing).<br>
<br>
Emil<br>
<div><div class=3D"h5"><br>
On 12.09.12, 15:32, Magnus Westerlund wrote:<br>
&gt; WG,<br>
&gt;<br>
&gt; Based on this discussion the chairs conclusion is that there is requir=
ed<br>
&gt; to have a definition of how trickle ICE is to work so that interoperab=
le<br>
&gt; implementations can be made.<br>
&gt;<br>
&gt; This WG is clearly not chartered to develop such a specification. Thus=
 a<br>
&gt; possible way forward is to do the work in the MMUSIC WG and develop an=
<br>
&gt; ICE mode for trickle. Such work would require someone to take the role=
<br>
&gt; of being proponent and drive the work in MMUSIC.<br>
&gt;<br>
&gt; Thus we chair wonder if there exist people willing to take on this eff=
ort?<br>
&gt;<br>
&gt; We chairs will meantime discuss with our ADs and the MMUSIC WG chairs<=
br>
&gt; how they see on starting work in MMUSIC for this.<br>
&gt;<br>
&gt; Cheers<br>
&gt;<br>
&gt; Magnus Westerlund<br>
&gt; (As chair)<br>
&gt;<br>
&gt; ----------------------------------------------------------------------=
<br>
&gt; Multimedia Technologies, Ericsson Research EAB/TVM<br>
&gt; ----------------------------------------------------------------------=
<br>
&gt; Ericsson AB =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0| P=
hone =C2=A0<a href=3D"tel:%2B46%2010%207148287" value=3D"+46107148287">+46 =
10 7148287</a><br>
&gt; F=C3=A4r=C3=B6gatan 6 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0=
 =C2=A0| Mobile <a href=3D"tel:%2B46%2073%200949079" value=3D"+46730949079"=
>+46 73 0949079</a><br>
&gt; SE-164 80 Stockholm, Sweden| mailto: <a href=3D"mailto:magnus.westerlu=
nd@ericsson.com">magnus.westerlund@ericsson.com</a><br>
&gt; ----------------------------------------------------------------------=
<br>
&gt;<br>
&gt; _______________________________________________<br>
&gt; rtcweb mailing list<br>
&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
&gt;<br>
<br>
</div></div><div class=3D"im">--<br>
Emil Ivov, Ph.D. =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 =C2=A0 =C2=A0 67000 Strasbourg,<br>
Project Lead =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0=
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 France<br>
Jitsi<br>
<a href=3D"mailto:emcho@jitsi.org">emcho@jitsi.org</a> =C2=A0 =C2=A0 =C2=A0=
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0PHONE: <a hr=
ef=3D"tel:%2B33.1.77.62.43.30" value=3D"+33177624330">+33.1.77.62.43.30</a>=
<br>
</div><a href=3D"https://jitsi.org" target=3D"_blank">https://jitsi.org</a>=
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0FAX: =C2=A0 <a href=3D"tel:%2B33.1.77.62.47.31" value=3D"+33177624731">+=
33.1.77.62.47.31</a><br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div>

--20cf3074b9fcda8e3504ca2b054f--

From hannes.tschofenig@gmx.net  Fri Sep 21 05:09:21 2012
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Subject: Re: [rtcweb] draft-jennings-rtcweb-qos: QCI "marking"?
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Hi Cameron, Hi Markus,=20

your responses raise an interesting question: while everyone liked the =
idea presented in draft-jennings-rtcweb-qos it may have pretty limited =
practical value. There is the annoying layer issues (that we encounter =
in other areas in the IETF as well) that crosses our way. For cellular =
systems where the QoS mechanisms would provide the most benefits it does =
not work. If it would work then the QoS procedure would be available at =
the link layer during bearer establishment. For WLANs, where there are =
the least problems, you could also use the link layer specific QoS =
mechanisms (if at all needed). Same for Wimax.=20

On top of that there is the architectural issue (namely, how this is =
supposed to work e2e). Not talking about the architectural story does =
not make the problem go away.

Ciao
Hannes

On Sep 18, 2012, at 4:27 PM, Cameron Byrne wrote:

>=20
> On Sep 18, 2012 1:08 AM, <Markus.Isomaki@nokia.com> wrote:
> >
> > Hi,
> >
> > Martin Thomson wrote:
> > >
> > >On 10 September 2012 07:15, Harald Alvestrand =
<harald@alvestrand.no>
> > >wrote:
> > >> - Pointers to documentation on how browsers are expected to be =
able to
> > >> set QCI and WiFI markings would be a good addition. Compared to =
those,
> > >> DSCP codepoint setting is well understood.
> > >
> > <snip>
> > >
> > >What do people think about NOT including QCI/WiFi markings?  Is it =
not
> > >possible for a wireless interface to examine DSCP markings in order =
to
> > >determine the markings for the link?  We should endeavour to =
maintain the
> > >proper abstractions.
> > >
> >
> > I'm not even sure if the browser or even the OS can influence much =
the QCI or bearer selection in LTE. I think the model is such that the =
"network" sets up these special non-best-effort bearers, and gives the =
endpoint the classifier rules related to them. The rules tell the =
endpoint how to map outgoing packets to the bearers. Typically the rules =
are based on source/destination IP and port, and the transport protocol. =
Someone may be able to confirm, but I believe DSCP can be used too. But =
basically the rules come from the network and the endpoint just follows =
them.
> >
> > So, somehow the browser or the JS app would first have to convince =
the access network provider to setup the special bearer so that the =
right flows get classified to it. In IMS the SIP proxy reads the SDP and =
tells some network element what audio or video flows are being setup and =
what their 5-tuples are, so the network is being able to setup the =
bearers correctly for them. The endpoint merely waits for this setup and =
acts accordingly. This works as the SIP proxy is run by the access =
operator. In RTCWeb the same could work if the calling site had a =
relationship with the access operator. I've heard that there is some =
work by some operators to offer Web APIs for 3rd party apps to do the =
bearer setup requests more ad hoc. Most of this may be outside the scope =
of RTCWeb but I think the document should have some explanation on how =
it works from the OS, browser, JS client, calling site, and mobile =
operator perspective.
> >
> > On the "transport" or "data path" level I don't think the browser =
needs to understand anything about QCIs. DSCP marking should be good =
enough.
> >
> > Anyone with more LTE/mobile clue that could comment on this?
> >
> > Markus
> >
>=20
> Your understanding is similar to mine.
>=20
> Generally speaking, qos in 3gpp networks is network initiated and =
controlled by a PCRF. The UEs are generally not trusted, as they may =
abuse qos.
>=20
> I am told that qos is an important differentiator for the operator run =
services and consequently will not be generally available to 3rd party =
apps such as webrtc.
>=20
> I am generally sceptical about qos on the internets, and we may be =
getting into a dangerous area if we start to assume networks will give =
anything more than best effort to webrtc.
>=20
> CB _______________________________________________
> > 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 tterriberry@mozilla.com  Fri Sep 21 09:11:16 2012
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Date: Fri, 21 Sep 2012 09:11:14 -0700
From: "Timothy B. Terriberry" <tterriberry@mozilla.com>
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Following the success of Opus, we are hoping to organize a BoF to 
consider forming a working group to standardize a video codec. While 
this will obviously not be finished in time to be MTI for initial 
deployments of WebRTC (or something is drastically wrong), many here may 
have some interest in the idea.

Mailing list: https://www.ietf.org/mailman/listinfo/video-codec

Initial charter proposal: 
http://www.ietf.org/mail-archive/web/video-codec/current/msg00002.html

Please give comments and feedback!

From richard@shockey.us  Sat Sep 22 12:16:16 2012
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Subject: Re: [rtcweb] video-codec mailing list and charter
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+1  Outstanding idea.  Do it! 

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of
Timothy B. Terriberry
Sent: Friday, September 21, 2012 12:11 PM
To: rtcweb@ietf.org
Subject: [rtcweb] video-codec mailing list and charter

Following the success of Opus, we are hoping to organize a BoF to consider
forming a working group to standardize a video codec. While this will
obviously not be finished in time to be MTI for initial deployments of
WebRTC (or something is drastically wrong), many here may have some interest
in the idea.

Mailing list: https://www.ietf.org/mailman/listinfo/video-codec

Initial charter proposal: 
http://www.ietf.org/mail-archive/web/video-codec/current/msg00002.html

Please give comments and feedback!
_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb


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Subject: [rtcweb] BTW where are we at with MTI codecs?
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I do want to make it clear that SHOULD's should be included.

 

I really really believe 722 AMR WB needs to be included in order to properly
interoperate with  Public SIP Networks (the former PSTN) and soon to be
deployed Voice over LTE. 

 

Richard Shockey
Shockey Consulting
Chairman of the Board of Directors SIP Forum
PSTN Mobile: +1 703.593.2683
< <mailto:richard(at)shockey.us> mailto:richard(at)shockey.us>
skype-linkedin-facebook: rshockey101
http//www.sipforum.org

 


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</o:shapelayout></xml><![endif]--></head><body lang=3DEN-US link=3Dblue =
vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal>I do want =
to make it clear that SHOULD&#8217;s should be =
included.<o:p></o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal>I really really believe 722 AMR WB needs to be =
included in order to properly interoperate with &nbsp;Public SIP =
Networks (the former PSTN) and soon to be deployed Voice over LTE. =
<o:p></o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;margin-bottom:12.0pt'><span =
style=3D'font-size:10.0pt;font-family:"Times New Roman","serif"'>Richard =
Shockey<br>Shockey Consulting<br>Chairman of the Board of Directors SIP =
Forum<br>PSTN Mobile: +1 703.593.2683<br>&lt;<a =
href=3D"mailto:richard(at)shockey.us"><span =
style=3D'color:blue'>mailto:richard(at)shockey.us</span></a>&gt;<br>skype=
-linkedin-facebook: rshockey101<br>http//www.sipforum.org</span><span =
style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif"'><o:p></o:p></span></p><p =
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On Sun, Sep 23, 2012 at 5:16 AM, Richard Shockey <richard@shockey.us> wrote:
> +1  Outstanding idea.  Do it!

+1 of course!

From dromasca@avaya.com  Sun Sep 23 02:48:59 2012
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Thread-Topic: [rtcweb] BTW where are we at with MTI codecs?
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From: "Romascanu, Dan (Dan)" <dromasca@avaya.com>
To: "Richard Shockey" <richard@shockey.us>, <rtcweb@ietf.org>
Subject: Re: [rtcweb] BTW where are we at with MTI codecs?
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I believe that the answer to the question in the subject line was given =
in the message from the chairs at =
http://www.ietf.org/mail-archive/web/rtcweb/current/msg05267.html.

=20

So the question is rather - 'where we are with the non-MTI codecs?' (the =
SHOULD codecs)?=20

=20

The chairs wrote in the same message:=20

=20

=D8  Based on the discussion to date, the chairs will run a consensus =
call on

=D8  whether the WG should recommend specific codecs which will not be

=D8  mandatory to implement (i.e. will the document contain SHOULDs as =
well

=D8  as MUSTs for codec implementation)".  If the response to that =
consensus

=D8  call indicates consensus for non-mandatory to implement =
recommendations,

=D8  we will run consensus calls on those codecs to be included at that =
level.

=D8  =20

=20

I think that that consensus call was not run yet.=20

=20

Regards,

=20

Dan

=20

=20

=20

From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf =
Of Richard Shockey
Sent: Saturday, September 22, 2012 10:20 PM
To: rtcweb@ietf.org
Subject: [rtcweb] BTW where are we at with MTI codecs?

=20

I do want to make it clear that SHOULD's should be included.

=20

I really really believe 722 AMR WB needs to be included in order to =
properly interoperate with  Public SIP Networks (the former PSTN) and =
soon to be deployed Voice over LTE.=20

=20

Richard Shockey
Shockey Consulting
Chairman of the Board of Directors SIP Forum
PSTN Mobile: +1 703.593.2683
<mailto:richard(at)shockey.us>
skype-linkedin-facebook: rshockey101
http//www.sipforum.org

=20


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</o:shapelayout></xml><![endif]--></head><body lang=3DEN-US link=3Dblue =
vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal><span =
style=3D'color:#1F497D'>I believe that the answer to the question in the =
subject line was given in the message from the chairs at <a =
href=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/msg05267.html=
">http://www.ietf.org/mail-archive/web/rtcweb/current/msg05267.html</a>.<=
o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span style=3D'color:#1F497D'>So the question is =
rather &#8211; &#8216;where we are with the non-MTI codecs?&#8217; (the =
SHOULD codecs)? <o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span style=3D'color:#1F497D'>The chairs wrote in the =
same message: <o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'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:10.0pt;font-family:Wingdings'><span =
style=3D'mso-list:Ignore'>=D8<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp; </span></span></span><![endif]><span =
dir=3DLTR></span><span style=3D'font-size:10.0pt;font-family:"Courier =
New"'>Based on the discussion to date, the chairs will run a consensus =
call on<o:p></o:p></span></p><p class=3DMsoListParagraph =
style=3D'text-indent:-.25in;mso-list:l0 level1 lfo1'><![if =
!supportLists]><span =
style=3D'font-size:10.0pt;font-family:Wingdings'><span =
style=3D'mso-list:Ignore'>=D8<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp; </span></span></span><![endif]><span =
dir=3DLTR></span><span style=3D'font-size:10.0pt;font-family:"Courier =
New"'>whether the WG should recommend specific codecs which will not =
be<o:p></o:p></span></p><p class=3DMsoListParagraph =
style=3D'text-indent:-.25in;mso-list:l0 level1 lfo1'><![if =
!supportLists]><span =
style=3D'font-size:10.0pt;font-family:Wingdings'><span =
style=3D'mso-list:Ignore'>=D8<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp; </span></span></span><![endif]><span =
dir=3DLTR></span><span style=3D'font-size:10.0pt;font-family:"Courier =
New"'>mandatory to implement (i.e. will the document contain SHOULDs as =
well<o:p></o:p></span></p><p class=3DMsoListParagraph =
style=3D'text-indent:-.25in;mso-list:l0 level1 lfo1'><![if =
!supportLists]><span =
style=3D'font-size:10.0pt;font-family:Wingdings'><span =
style=3D'mso-list:Ignore'>=D8<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp; </span></span></span><![endif]><span =
dir=3DLTR></span><span style=3D'font-size:10.0pt;font-family:"Courier =
New"'>as MUSTs for codec implementation)&quot;.=A0 If the response to =
that consensus<o:p></o:p></span></p><p class=3DMsoListParagraph =
style=3D'text-indent:-.25in;mso-list:l0 level1 lfo1'><![if =
!supportLists]><span =
style=3D'font-size:10.0pt;font-family:Wingdings'><span =
style=3D'mso-list:Ignore'>=D8<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp; </span></span></span><![endif]><span =
dir=3DLTR></span><span style=3D'font-size:10.0pt;font-family:"Courier =
New"'>call indicates consensus for non-mandatory to implement =
recommendations,<o:p></o:p></span></p><p class=3DMsoListParagraph =
style=3D'text-indent:-.25in;mso-list:l0 level1 lfo1'><![if =
!supportLists]><span =
style=3D'font-size:10.0pt;font-family:Wingdings'><span =
style=3D'mso-list:Ignore'>=D8<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp; </span></span></span><![endif]><span =
dir=3DLTR></span><span style=3D'font-size:10.0pt;font-family:"Courier =
New"'>we will run consensus calls on those codecs to be included at that =
level.<o:p></o:p></span></p><p class=3DMsoListParagraph =
style=3D'text-indent:-.25in;mso-list:l0 level1 lfo1'><![if =
!supportLists]><span =
style=3D'font-size:10.0pt;font-family:Wingdings'><span =
style=3D'mso-list:Ignore'>=D8<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp; </span></span></span><![endif]><span =
dir=3DLTR></span><span style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span style=3D'color:#1F497D'>I think that that =
consensus call was not run yet. <o:p></o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'>Regards,<o:p></o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'>Dan<o:p></o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'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-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] <b>On Behalf Of =
</b>Richard Shockey<br><b>Sent:</b> Saturday, September 22, 2012 10:20 =
PM<br><b>To:</b> rtcweb@ietf.org<br><b>Subject:</b> [rtcweb] BTW where =
are we at with MTI codecs?<o:p></o:p></span></p></div></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>I do want to =
make it clear that SHOULD&#8217;s should be included.<o:p></o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>I really =
really believe 722 AMR WB needs to be included in order to properly =
interoperate with &nbsp;Public SIP Networks (the former PSTN) and soon =
to be deployed Voice over LTE. <o:p></o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;margin-bottom:12.0pt'><span =
style=3D'font-size:10.0pt;font-family:"Times New Roman","serif"'>Richard =
Shockey<br>Shockey Consulting<br>Chairman of the Board of Directors SIP =
Forum<br>PSTN Mobile: +1 703.593.2683<br>&lt;<a =
href=3D"mailto:richard(at)shockey.us">mailto:richard(at)shockey.us</a>&gt=
;<br>skype-linkedin-facebook: =
rshockey101<br>http//www.sipforum.org</span><span =
style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif"'><o:p></o:p></span></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div></div></body></html>
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To: <rtcweb@ietf.org>
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Subject: Re: [rtcweb] BTW where are we at with MTI codecs?
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I understand that ..=20

=20

It would be helpful to run the consensus call  ASAP so we could move on =
to
something more  =94interesting=94 like resolving the offer/answer  =
model. =20

=20

I was archiving some of my email today to free up disk space.=20

=20

From: Romascanu, Dan (Dan) [mailto:dromasca@avaya.com]=20
Sent: Sunday, September 23, 2012 5:49 AM
To: Richard Shockey; rtcweb@ietf.org
Subject: RE: [rtcweb] BTW where are we at with MTI codecs?

=20

I believe that the answer to the question in the subject line was given =
in
the message from the chairs at
http://www.ietf.org/mail-archive/web/rtcweb/current/msg05267.html.

=20

So the question is rather =96 =91where we are with the non-MTI =
codecs?=92 (the
SHOULD codecs)?=20

=20

The chairs wrote in the same message:=20

=20

=D8  Based on the discussion to date, the chairs will run a consensus =
call on

=D8  whether the WG should recommend specific codecs which will not be

=D8  mandatory to implement (i.e. will the document contain SHOULDs as =
well

=D8  as MUSTs for codec implementation)".  If the response to that =
consensus

=D8  call indicates consensus for non-mandatory to implement =
recommendations,

=D8  we will run consensus calls on those codecs to be included at that =
level.

=D8  =20

=20

I think that that consensus call was not run yet.=20

=20

Regards,

=20

Dan

=20

=20

=20

From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf =
Of
Richard Shockey
Sent: Saturday, September 22, 2012 10:20 PM
To: rtcweb@ietf.org
Subject: [rtcweb] BTW where are we at with MTI codecs?

=20

I do want to make it clear that SHOULD=92s should be included.

=20

I really really believe 722 AMR WB needs to be included in order to =
properly
interoperate with  Public SIP Networks (the former PSTN) and soon to be
deployed Voice over LTE.=20

=20

Richard Shockey
Shockey Consulting
Chairman of the Board of Directors SIP Forum
PSTN Mobile: +1 703.593.2683
<mailto:richard(at)shockey.us>
skype-linkedin-facebook: rshockey101
http//www.sipforum.org

=20


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vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal><span =
style=3D'color:#1F497D'>I understand that .. <o:p></o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span style=3D'color:#1F497D'>It would be helpful to =
run the consensus call =A0ASAP so we could move on to something more =
=A0&#8221;interesting&#8221; like resolving the offer/answer =A0model. =
=A0<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span style=3D'color:#1F497D'>I was archiving some of =
my email today to free up disk space. <o:p></o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'><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:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> =
Romascanu, Dan (Dan) [mailto:dromasca@avaya.com] <br><b>Sent:</b> =
Sunday, September 23, 2012 5:49 AM<br><b>To:</b> Richard Shockey; =
rtcweb@ietf.org<br><b>Subject:</b> RE: [rtcweb] BTW where are we at with =
MTI codecs?<o:p></o:p></span></p></div></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal><span =
style=3D'color:#1F497D'>I believe that the answer to the question in the =
subject line was given in the message from the chairs at <a =
href=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/msg05267.html=
">http://www.ietf.org/mail-archive/web/rtcweb/current/msg05267.html</a>.<=
o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span style=3D'color:#1F497D'>So the question is =
rather &#8211; &#8216;where we are with the non-MTI codecs?&#8217; (the =
SHOULD codecs)? <o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span style=3D'color:#1F497D'>The chairs wrote in the =
same message: <o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'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:10.0pt;font-family:Wingdings'><span =
style=3D'mso-list:Ignore'>=D8<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp; </span></span></span><![endif]><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>Based on the =
discussion to date, the chairs will run a consensus call =
on<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:10.0pt;font-family:Wingdings'><span =
style=3D'mso-list:Ignore'>=D8<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp; </span></span></span><![endif]><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>whether the WG =
should recommend specific codecs which will not =
be<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:10.0pt;font-family:Wingdings'><span =
style=3D'mso-list:Ignore'>=D8<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp; </span></span></span><![endif]><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>mandatory to =
implement (i.e. will the document contain SHOULDs as =
well<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:10.0pt;font-family:Wingdings'><span =
style=3D'mso-list:Ignore'>=D8<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp; </span></span></span><![endif]><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>as MUSTs for codec =
implementation)&quot;.&nbsp; If the response to that =
consensus<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:10.0pt;font-family:Wingdings'><span =
style=3D'mso-list:Ignore'>=D8<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp; </span></span></span><![endif]><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>call indicates =
consensus for non-mandatory to implement =
recommendations,<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:10.0pt;font-family:Wingdings'><span =
style=3D'mso-list:Ignore'>=D8<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp; </span></span></span><![endif]><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>we will run =
consensus calls on those codecs to be included at that =
level.<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:10.0pt;font-family:Wingdings'><span =
style=3D'mso-list:Ignore'>=D8<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp; </span></span></span><![endif]><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span style=3D'color:#1F497D'>I think that that =
consensus call was not run yet. <o:p></o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'>Regards,<o:p></o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'>Dan<o:p></o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'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"'> =
<a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a> =
[<a =
href=3D"mailto:rtcweb-bounces@ietf.org">mailto:rtcweb-bounces@ietf.org</a=
>] <b>On Behalf Of </b>Richard Shockey<br><b>Sent:</b> Saturday, =
September 22, 2012 10:20 PM<br><b>To:</b> <a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br><b>Subject:</b> =
[rtcweb] BTW where are we at with MTI =
codecs?<o:p></o:p></span></p></div></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>I do want to =
make it clear that SHOULD&#8217;s should be included.<o:p></o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>I really =
really believe 722 AMR WB needs to be included in order to properly =
interoperate with &nbsp;Public SIP Networks (the former PSTN) and soon =
to be deployed Voice over LTE. <o:p></o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;margin-bottom:12.0pt'><span =
style=3D'font-size:10.0pt;font-family:"Times New Roman","serif"'>Richard =
Shockey<br>Shockey Consulting<br>Chairman of the Board of Directors SIP =
Forum<br>PSTN Mobile: +1 703.593.2683<br>&lt;<a =
href=3D"mailto:richard(at)shockey.us">mailto:richard(at)shockey.us</a>&gt=
;<br>skype-linkedin-facebook: =
rshockey101<br>http//www.sipforum.org</span><span =
style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif"'><o:p></o:p></span></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div></div></body></html>
------=_NextPart_000_0061_01CD997B.E335E680--


From basilgohar@librevideo.org  Sun Sep 23 12:09:37 2012
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On 09/21/2012 12:11 PM, Timothy B. Terriberry wrote:
> Following the success of Opus, we are hoping to organize a BoF to 
> consider forming a working group to standardize a video codec. While 
> this will obviously not be finished in time to be MTI for initial 
> deployments of WebRTC (or something is drastically wrong), many here 
> may have some interest in the idea.
>
> Mailing list: https://www.ietf.org/mailman/listinfo/video-codec
>
> Initial charter proposal: 
> http://www.ietf.org/mail-archive/web/video-codec/current/msg00002.html
>
> Please give comments and feedback!
While I did not have time to go through the entire charter proposal, in 
principle from what I read and what I see described here, this is, to 
me, the next priority that needs to be address, so a strong +1 from me.

-- 
Libre Video
http://librevideo.org


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+1 - Thx Timothy.

*Erik Lagerway <http://ca.linkedin.com/in/lagerway> |
*Hookflash<http://hookflash.com>
* | m 1.604.562.8647*
****


On Fri, Sep 21, 2012 at 9:11 AM, Timothy B. Terriberry <
tterriberry@mozilla.com> wrote:

> Following the success of Opus, we are hoping to organize a BoF to consider
> forming a working group to standardize a video codec. While this will
> obviously not be finished in time to be MTI for initial deployments of
> WebRTC (or something is drastically wrong), many here may have some
> interest in the idea.
>
> Mailing list: https://www.ietf.org/mailman/**listinfo/video-codec<https://www.ietf.org/mailman/listinfo/video-codec>
>
> Initial charter proposal: http://www.ietf.org/mail-**
> archive/web/video-codec/**current/msg00002.html<http://www.ietf.org/mail-archive/web/video-codec/current/msg00002.html>
>
> Please give comments and feedback!
> ______________________________**_________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/**listinfo/rtcweb<https://www.ietf.org/mailman/listinfo/rtcweb>
>

--485b390f7a24df2e7604ca63676b
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+1 - Thx Timothy.<br><br clear=3D"all"><font color=3D"#943634" face=3D"aria=
l, sans-serif"><span style=3D"border-collapse:collapse;line-height:14px"><s=
pan style=3D"border-collapse:separate;color:rgb(0,0,0);font-family:arial;li=
ne-height:normal"><span style=3D"font-family:arial,sans-serif;border-collap=
se:collapse;color:rgb(148,54,52);line-height:14px"><b><span style=3D"color:=
rgb(0,0,0);font-weight:normal;font-size:13px"><font color=3D"#943634"><span=
 style=3D"font-size:small"><b><span style=3D"color:rgb(0,0,0);font-weight:n=
ormal;font-size:13px"><span style=3D"font-size:8pt;line-height:12px;color:g=
ray"><a href=3D"http://ca.linkedin.com/in/lagerway" target=3D"_blank"><span=
 style=3D"color:rgb(204,0,0)">Erik Lagerway</span></a> | </span></span></b>=
</span></font></span></b><a href=3D"http://hookflash.com" target=3D"_blank"=
><span style=3D"color:rgb(0,0,0);font-weight:normal;font-size:13px"><font c=
olor=3D"#943634"><span style=3D"font-size:small"><span style=3D"color:rgb(0=
,0,0);font-weight:normal;font-size:13px"><span style=3D"font-size:8pt;line-=
height:12px;color:gray"></span></span></span></font></span><span style=3D"c=
olor:rgb(0,0,0);font-weight:normal;font-size:13px"><font color=3D"#943634">=
<span style=3D"font-size:small"><span style=3D"color:rgb(0,0,0);font-weight=
:normal;font-size:13px"><span style=3D"font-size:8pt;line-height:12px;color=
:gray"><span style=3D"color:rgb(51,51,51)">Hookflash</span></span></span></=
span></font></span></a><span style=3D"color:rgb(0,0,0);font-weight:normal;f=
ont-size:13px"><font color=3D"#943634"><span style=3D"font-size:small"><spa=
n style=3D"color:rgb(0,0,0);font-weight:normal;font-size:13px"><span style=
=3D"font-size:8pt;line-height:12px;color:gray"></span></span></span></font>=
</span><b><span style=3D"color:rgb(0,0,0);font-weight:normal;font-size:13px=
"><font color=3D"#943634"><span style=3D"font-size:small"><b><span style=3D=
"color:rgb(0,0,0);font-weight:normal;font-size:13px"><span style=3D"font-si=
ze:8pt;line-height:12px;color:gray"> | m 1.604.562.8647</span></span></b></=
span></font></span></b></span></span></span></font><br>
<font color=3D"#943634" face=3D"arial, sans-serif"><span style=3D"border-co=
llapse:collapse;line-height:14px"><span style=3D"border-collapse:separate;c=
olor:rgb(0,0,0);font-family:arial;line-height:normal"><span style=3D"font-f=
amily:arial,sans-serif;border-collapse:collapse;color:rgb(148,54,52);line-h=
eight:14px"><b><span style=3D"color:rgb(0,0,0);font-weight:normal;font-size=
:13px"><font color=3D"#943634"><span style=3D"font-size:small"><b><span sty=
le=3D"color:rgb(0,0,0);font-weight:normal;font-size:13px"><span style=3D"fo=
nt-size:10pt;line-height:14px;color:rgb(148,54,52)"></span><span style=3D"f=
ont-size:8pt;line-height:12px"></span></span></b></span></font></span></b><=
/span></span></span></font><font color=3D"#943634" face=3D"arial, sans-seri=
f"><span style=3D"border-collapse:collapse;line-height:14px"><span style=3D=
"border-collapse:separate;color:rgb(0,0,0);font-family:arial;line-height:no=
rmal"></span></span></font><font color=3D"#943634" face=3D"arial, sans-seri=
f"><span style=3D"border-collapse:collapse;line-height:14px"><span style=3D=
"border-collapse:separate;color:rgb(0,0,0);font-family:arial;line-height:no=
rmal"><span style=3D"font-family:arial,sans-serif;border-collapse:collapse;=
color:rgb(148,54,52);line-height:14px"><b><span style=3D"color:rgb(0,0,0);f=
ont-weight:normal;font-size:13px"><font color=3D"#943634"><span style=3D"fo=
nt-size:small"><b><span style=3D"color:rgb(0,0,0);font-weight:normal;font-s=
ize:13px"><span style=3D"font-size:10pt;line-height:14px;color:rgb(148,54,5=
2)"></span><span style=3D"font-size:8pt;line-height:12px"></span></span></b=
></span></font></span></b></span></span></span></font><font color=3D"#94363=
4" face=3D"arial, sans-serif"><span style=3D"border-collapse:collapse;line-=
height:14px"><span style=3D"border-collapse:separate;color:rgb(0,0,0);font-=
family:arial;line-height:normal"></span></span></font><br>

<br><br><div class=3D"gmail_quote">On Fri, Sep 21, 2012 at 9:11 AM, Timothy=
 B. Terriberry <span dir=3D"ltr">&lt;<a href=3D"mailto:tterriberry@mozilla.=
com" target=3D"_blank">tterriberry@mozilla.com</a>&gt;</span> wrote:<br><bl=
ockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #=
ccc solid;padding-left:1ex">
Following the success of Opus, we are hoping to organize a BoF to consider =
forming a working group to standardize a video codec. While this will obvio=
usly not be finished in time to be MTI for initial deployments of WebRTC (o=
r something is drastically wrong), many here may have some interest in the =
idea.<br>

<br>
Mailing list: <a href=3D"https://www.ietf.org/mailman/listinfo/video-codec"=
 target=3D"_blank">https://www.ietf.org/mailman/<u></u>listinfo/video-codec=
</a><br>
<br>
Initial charter proposal: <a href=3D"http://www.ietf.org/mail-archive/web/v=
ideo-codec/current/msg00002.html" target=3D"_blank">http://www.ietf.org/mai=
l-<u></u>archive/web/video-codec/<u></u>current/msg00002.html</a><br>
<br>
Please give comments and feedback!<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>

--485b390f7a24df2e7604ca63676b--

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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Hannes Tschofenig <hannes.tschofenig@gmx.net>
Thread-Topic: [rtcweb] draft-jennings-rtcweb-qos: QCI "marking"?
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Subject: Re: [rtcweb] draft-jennings-rtcweb-qos: QCI "marking"?
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I don't think the DSCP stuff in the draft is trying to solve any end to end=
 problems or even make any guarantees about QoS anywhere. But the draft doe=
s clearly state where this does sometimes help (like the local DSL uplink f=
rom my house) and points out it won't help for something like end to end.=20

On the topic of the QCI on smart phones. I agree that untrusted 3rd party a=
pplications are probably not going to be able to request QoS on most carrie=
r networks. However, some carriers are clearly looking at the idea that the=
re could be carrier provided RTCWeb applications that have a trust relation=
ship that allows them to get different treatment. I don't claim to be a LTE=
 expert but luckily Dan is an architect at a major carrier with strong inte=
rest in LTE so I'm pretty sure he can get this straightened out.=20




On Sep 21, 2012, at 6:09 AM, Hannes Tschofenig <hannes.tschofenig@gmx.net> =
wrote:

> Hi Cameron, Hi Markus,=20
>=20
> your responses raise an interesting question: while everyone liked the id=
ea presented in draft-jennings-rtcweb-qos it may have pretty limited practi=
cal value. There is the annoying layer issues (that we encounter in other a=
reas in the IETF as well) that crosses our way. For cellular systems where =
the QoS mechanisms would provide the most benefits it does not work. If it =
would work then the QoS procedure would be available at the link layer duri=
ng bearer establishment. For WLANs, where there are the least problems, you=
 could also use the link layer specific QoS mechanisms (if at all needed). =
Same for Wimax.=20
>=20
> On top of that there is the architectural issue (namely, how this is supp=
osed to work e2e). Not talking about the architectural story does not make =
the problem go away.
>=20
> Ciao
> Hannes
>=20
> On Sep 18, 2012, at 4:27 PM, Cameron Byrne wrote:
>=20
>>=20
>> On Sep 18, 2012 1:08 AM, <Markus.Isomaki@nokia.com> wrote:
>>>=20
>>> Hi,
>>>=20
>>> Martin Thomson wrote:
>>>>=20
>>>> On 10 September 2012 07:15, Harald Alvestrand <harald@alvestrand.no>
>>>> wrote:
>>>>> - Pointers to documentation on how browsers are expected to be able t=
o
>>>>> set QCI and WiFI markings would be a good addition. Compared to those=
,
>>>>> DSCP codepoint setting is well understood.
>>>>=20
>>> <snip>
>>>>=20
>>>> What do people think about NOT including QCI/WiFi markings?  Is it not
>>>> possible for a wireless interface to examine DSCP markings in order to
>>>> determine the markings for the link?  We should endeavour to maintain =
the
>>>> proper abstractions.
>>>>=20
>>>=20
>>> I'm not even sure if the browser or even the OS can influence much the =
QCI or bearer selection in LTE. I think the model is such that the "network=
" sets up these special non-best-effort bearers, and gives the endpoint the=
 classifier rules related to them. The rules tell the endpoint how to map o=
utgoing packets to the bearers. Typically the rules are based on source/des=
tination IP and port, and the transport protocol. Someone may be able to co=
nfirm, but I believe DSCP can be used too. But basically the rules come fro=
m the network and the endpoint just follows them.
>>>=20
>>> So, somehow the browser or the JS app would first have to convince the =
access network provider to setup the special bearer so that the right flows=
 get classified to it. In IMS the SIP proxy reads the SDP and tells some ne=
twork element what audio or video flows are being setup and what their 5-tu=
ples are, so the network is being able to setup the bearers correctly for t=
hem. The endpoint merely waits for this setup and acts accordingly. This wo=
rks as the SIP proxy is run by the access operator. In RTCWeb the same coul=
d work if the calling site had a relationship with the access operator. I'v=
e heard that there is some work by some operators to offer Web APIs for 3rd=
 party apps to do the bearer setup requests more ad hoc. Most of this may b=
e outside the scope of RTCWeb but I think the document should have some exp=
lanation on how it works from the OS, browser, JS client, calling site, and=
 mobile operator perspective.
>>>=20
>>> On the "transport" or "data path" level I don't think the browser needs=
 to understand anything about QCIs. DSCP marking should be good enough.
>>>=20
>>> Anyone with more LTE/mobile clue that could comment on this?
>>>=20
>>> Markus
>>>=20
>>=20
>> Your understanding is similar to mine.
>>=20
>> Generally speaking, qos in 3gpp networks is network initiated and contro=
lled by a PCRF. The UEs are generally not trusted, as they may abuse qos.
>>=20
>> I am told that qos is an important differentiator for the operator run s=
ervices and consequently will not be generally available to 3rd party apps =
such as webrtc.
>>=20
>> I am generally sceptical about qos on the internets, and we may be getti=
ng into a dangerous area if we start to assume networks will give anything =
more than best effort to webrtc.
>>=20
>> CB _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Justin Uberti <juberti@google.com>
Thread-Topic: [rtcweb] Filling in details on "trickle ICE"
Thread-Index: AQHNmeAZG9HIi6Jp2Eqo3+7YsAvZGw==
Date: Sun, 23 Sep 2012 23:06:44 +0000
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Filling in details on "trickle ICE"
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Just read that and I like it.=20

On Sep 20, 2012, at 6:06 PM, Justin Uberti <juberti@google.com> wrote:

> Eric Rescorla and I discussed some of the tricky trickle issues and wrote=
 them down in this doc: https://docs.google.com/document/d/1L5SP9YQxXRf-2KO=
bnDI5rgRCRnOv9EbhXsHsVisWp1Q/edit#. I think that would be a good starting p=
oint for an I-D on this.
>=20
>=20
> On Thu, Sep 20, 2012 at 12:11 PM, Emil Ivov <emcho@jitsi.org> wrote:
> I'd be willing to work on this (writing and/or reviewing).
>=20
> Emil
>=20
> On 12.09.12, 15:32, Magnus Westerlund wrote:
> > WG,
> >
> > Based on this discussion the chairs conclusion is that there is require=
d
> > to have a definition of how trickle ICE is to work so that interoperabl=
e
> > implementations can be made.
> >
> > This WG is clearly not chartered to develop such a specification. Thus =
a
> > possible way forward is to do the work in the MMUSIC WG and develop an
> > ICE mode for trickle. Such work would require someone to take the role
> > of being proponent and drive the work in MMUSIC.
> >
> > Thus we chair wonder if there exist people willing to take on this effo=
rt?
> >
> > We chairs will meantime discuss with our ADs and the MMUSIC WG chairs
> > how they see on starting work in MMUSIC for this.
> >
> > Cheers
> >
> > Magnus Westerlund
> > (As chair)
> >
> > ----------------------------------------------------------------------
> > Multimedia Technologies, Ericsson Research EAB/TVM
> > ----------------------------------------------------------------------
> > Ericsson AB                | Phone  +46 10 7148287
> > F=E4r=F6gatan 6                | Mobile +46 73 0949079
> > SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> > ----------------------------------------------------------------------
> >
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
> >
>=20
> --
> Emil Ivov, Ph.D.                       67000 Strasbourg,
> Project Lead                           France
> Jitsi
> emcho@jitsi.org                        PHONE: +33.1.77.62.43.30
> https://jitsi.org                      FAX:   +33.1.77.62.47.31
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: "Eggert, Lars" <lars@netapp.com>
Thread-Topic: [rtcweb] Comments on draft-jennings-rtcweb-qos (Re: Call for adoption of QoS draft)
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Date: Sun, 23 Sep 2012 23:23:59 +0000
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Subject: Re: [rtcweb] Comments on draft-jennings-rtcweb-qos (Re: Call for adoption of QoS draft)
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On Sep 20, 2012, at 2:17 AM, "Eggert, Lars" <lars@netapp.com> wrote:

> Hi,
>=20
> agree that the QCI stuff is probably not useful to discuss.
>=20
> The DSCP content is almost fully redundant with what's in RFC4594 (which =
is in the references but not cited). And RFC4594 has a much better discussi=
on on what DiffServ is and isn't.

So this is trying to work within the bounds of 4594 (and clearly needs to r=
eference that) but I think this narrows things down a bit more that 4594 mi=
ght or at least tries to clarify things a bit .=20

For example 4594 says use "Multimedia Conferencing service class is best su=
ited =85 video conferencing service"

It also says "Real-Time Interactive is intended for =85 and video conferenc=
ing applications"

It also suggest there are times the right thing for video might be "Multime=
dia Streaming service class"

As a result we see some major vendors shipping video phone like devices usi=
ng CSx class while others are using AFx set.=20

So this may be 100% redundant with 4594 depending on your point of view - o=
r it might not be redundant -  but we need one clear unambiguous thing abou=
t what needs to happen. If it can be reduced to two sentences, I'm all for =
that.=20


>=20
> For WLAN, 802.11e is what you want to point to, I guess.
>=20
> So this doc - if it's even needed - could be MUCH shorter, by basically j=
ust saying "read RFC4594 and map RTCweb flows into the classes defined ther=
e like this" and do the equivalent for 802.11e.

It also forms the requirement for the three priority levels at a JS layer a=
nd pointed out we don't want the JS setting if a stream is says audio or vi=
deo.


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


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From: "Tirumaleswar Reddy (tireddy)" <tireddy@cisco.com>
To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>, Hannes Tschofenig <hannes.tschofenig@gmx.net>
Thread-Topic: [rtcweb] draft-jennings-rtcweb-qos: QCI "marking"?
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Subject: Re: [rtcweb] draft-jennings-rtcweb-qos: QCI "marking"?
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Hi Cullen -

You may want to look into (3GPP TS 23.401 V11.3.0) which permits UE to requ=
est bearer resource modification.

Section 5.4.5

"The UE requested bearer resource modification procedure for an E-UTRAN is =
depicted in figure 5.4.5-1. The procedure allows the UE to request for a mo=
dification of bearer resources (e.g. allocation or release of resources) fo=
r one traffic flow aggregate with a specific QoS demand."

--Tiru.

> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf =
Of
> Cullen Jennings (fluffy)
> Sent: Monday, September 24, 2012 4:35 AM
> To: Hannes Tschofenig
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] draft-jennings-rtcweb-qos: QCI "marking"?
>=20
>=20
> I don't think the DSCP stuff in the draft is trying to solve any end to e=
nd
> problems or even make any guarantees about QoS anywhere. But the draft do=
es
> clearly state where this does sometimes help (like the local DSL uplink f=
rom
> my house) and points out it won't help for something like end to end.
>=20
> On the topic of the QCI on smart phones. I agree that untrusted 3rd party
> applications are probably not going to be able to request QoS on most car=
rier
> networks. However, some carriers are clearly looking at the idea that the=
re
> could be carrier provided RTCWeb applications that have a trust relations=
hip
> that allows them to get different treatment. I don't claim to be a LTE ex=
pert
> but luckily Dan is an architect at a major carrier with strong interest i=
n LTE
> so I'm pretty sure he can get this straightened out.
>=20
>=20
>=20
>=20
> On Sep 21, 2012, at 6:09 AM, Hannes Tschofenig <hannes.tschofenig@gmx.net=
>
> wrote:
>=20
> > Hi Cameron, Hi Markus,
> >
> > your responses raise an interesting question: while everyone liked the =
idea
> presented in draft-jennings-rtcweb-qos it may have pretty limited practic=
al
> value. There is the annoying layer issues (that we encounter in other are=
as in
> the IETF as well) that crosses our way. For cellular systems where the Qo=
S
> mechanisms would provide the most benefits it does not work. If it would =
work
> then the QoS procedure would be available at the link layer during bearer
> establishment. For WLANs, where there are the least problems, you could a=
lso
> use the link layer specific QoS mechanisms (if at all needed). Same for W=
imax.
> >
> > On top of that there is the architectural issue (namely, how this is
> supposed to work e2e). Not talking about the architectural story does not=
 make
> the problem go away.
> >
> > Ciao
> > Hannes
> >
> > On Sep 18, 2012, at 4:27 PM, Cameron Byrne wrote:
> >
> >>
> >> On Sep 18, 2012 1:08 AM, <Markus.Isomaki@nokia.com> wrote:
> >>>
> >>> Hi,
> >>>
> >>> Martin Thomson wrote:
> >>>>
> >>>> On 10 September 2012 07:15, Harald Alvestrand <harald@alvestrand.no>
> >>>> wrote:
> >>>>> - Pointers to documentation on how browsers are expected to be able=
 to
> >>>>> set QCI and WiFI markings would be a good addition. Compared to tho=
se,
> >>>>> DSCP codepoint setting is well understood.
> >>>>
> >>> <snip>
> >>>>
> >>>> What do people think about NOT including QCI/WiFi markings?  Is it n=
ot
> >>>> possible for a wireless interface to examine DSCP markings in order =
to
> >>>> determine the markings for the link?  We should endeavour to maintai=
n the
> >>>> proper abstractions.
> >>>>
> >>>
> >>> I'm not even sure if the browser or even the OS can influence much th=
e QCI
> or bearer selection in LTE. I think the model is such that the "network" =
sets
> up these special non-best-effort bearers, and gives the endpoint the
> classifier rules related to them. The rules tell the endpoint how to map
> outgoing packets to the bearers. Typically the rules are based on
> source/destination IP and port, and the transport protocol. Someone may b=
e
> able to confirm, but I believe DSCP can be used too. But basically the ru=
les
> come from the network and the endpoint just follows them.
> >>>
> >>> So, somehow the browser or the JS app would first have to convince th=
e
> access network provider to setup the special bearer so that the right flo=
ws
> get classified to it. In IMS the SIP proxy reads the SDP and tells some
> network element what audio or video flows are being setup and what their =
5-
> tuples are, so the network is being able to setup the bearers correctly f=
or
> them. The endpoint merely waits for this setup and acts accordingly. This
> works as the SIP proxy is run by the access operator. In RTCWeb the same =
could
> work if the calling site had a relationship with the access operator. I'v=
e
> heard that there is some work by some operators to offer Web APIs for 3rd
> party apps to do the bearer setup requests more ad hoc. Most of this may =
be
> outside the scope of RTCWeb but I think the document should have some
> explanation on how it works from the OS, browser, JS client, calling site=
, and
> mobile operator perspective.
> >>>
> >>> On the "transport" or "data path" level I don't think the browser nee=
ds to
> understand anything about QCIs. DSCP marking should be good enough.
> >>>
> >>> Anyone with more LTE/mobile clue that could comment on this?
> >>>
> >>> Markus
> >>>
> >>
> >> Your understanding is similar to mine.
> >>
> >> Generally speaking, qos in 3gpp networks is network initiated and
> controlled by a PCRF. The UEs are generally not trusted, as they may abus=
e
> qos.
> >>
> >> I am told that qos is an important differentiator for the operator run
> services and consequently will not be generally available to 3rd party ap=
ps
> such as webrtc.
> >>
> >> I am generally sceptical about qos on the internets, and we may be get=
ting
> into a dangerous area if we start to assume networks will give anything m=
ore
> than best effort to webrtc.
> >>
> >> CB _______________________________________________
> >>> 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
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From magnus.westerlund@ericsson.com  Mon Sep 24 01:19:24 2012
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To: Richard Shockey <richard@shockey.us>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] BTW where are we at with MTI codecs?
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WG,

We are sorry that we haven't started the consensus call. It was intended
to be started already, but some issues have made us hold the call. We
hope to be able to start it soon.

Cheers

Magnus Westerlund
WG chair.



On 2012-09-23 17:09, Richard Shockey wrote:
> I understand that ..
> 
>  
> 
> It would be helpful to run the consensus call  ASAP so we could move on
> to something more  ”interesting” like resolving the offer/answer  model.  
> 
>  
> 
> I was archiving some of my email today to free up disk space.
> 
>  
> 
> *From:*Romascanu, Dan (Dan) [mailto:dromasca@avaya.com]
> *Sent:* Sunday, September 23, 2012 5:49 AM
> *To:* Richard Shockey; rtcweb@ietf.org
> *Subject:* RE: [rtcweb] BTW where are we at with MTI codecs?
> 
>  
> 
> I believe that the answer to the question in the subject line was given
> in the message from the chairs at
> http://www.ietf.org/mail-archive/web/rtcweb/current/msg05267.html.
> 
>  
> 
> So the question is rather – ‘where we are with the non-MTI codecs?’ (the
> SHOULD codecs)?
> 
>  
> 
> The chairs wrote in the same message:
> 
>  
> 
> Ø  Based on the discussion to date, the chairs will run a consensus call on
> 
> Ø  whether the WG should recommend specific codecs which will not be
> 
> Ø  mandatory to implement (i.e. will the document contain SHOULDs as well
> 
> Ø  as MUSTs for codec implementation)".  If the response to that consensus
> 
> Ø  call indicates consensus for non-mandatory to implement recommendations,
> 
> Ø  we will run consensus calls on those codecs to be included at that level.
> 
> Ø   
> 
>  
> 
> I think that that consensus call was not run yet.
> 
>  
> 
> Regards,
> 
>  
> 
> Dan
> 
>  
> 
>  
> 
>  
> 
> *From:*rtcweb-bounces@ietf.org <mailto:rtcweb-bounces@ietf.org>
> [mailto:rtcweb-bounces@ietf.org] *On Behalf Of *Richard Shockey
> *Sent:* Saturday, September 22, 2012 10:20 PM
> *To:* rtcweb@ietf.org <mailto:rtcweb@ietf.org>
> *Subject:* [rtcweb] BTW where are we at with MTI codecs?
> 
>  
> 
> I do want to make it clear that SHOULD’s should be included.
> 
>  
> 
> I really really believe 722 AMR WB needs to be included in order to
> properly interoperate with  Public SIP Networks (the former PSTN) and
> soon to be deployed Voice over LTE.
> 
>  
> 
> Richard Shockey
> Shockey Consulting
> Chairman of the Board of Directors SIP Forum
> PSTN Mobile: +1 703.593.2683
> <mailto:richard(at)shockey.us>
> skype-linkedin-facebook: rshockey101
> http//www.sipforum.org
> 
>  
> 


-- 

Magnus Westerlund

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


From christer.holmberg@ericsson.com  Wed Sep 26 01:24:39 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Wed, 26 Sep 2012 10:24:34 +0200
Thread-Topic: Draft new version/working group: draft-westerlund-mmusic-max-ssrc-00
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Subject: [rtcweb] FW: Draft new version/working group: draft-westerlund-mmusic-max-ssrc-00
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FYI,

I've submitted the max-ssrc draft to MMUSIC. The mechanism may be of intere=
st for RTCWEB.

Regards,

Christer

(Associated IPR: https://datatracker.ietf.org/ipr/1639/)


From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf Of=
 Christer Holmberg
Sent: 26. syyskuuta 2012 10:23
To: mmusic@ietf.org
Subject: [MMUSIC] Draft new version/working group: draft-westerlund-mmusic-=
max-ssrc-00

Hi,

We have submitted a draft, new to MMUSIC: draft-westerlund-mmusic-max-ssrc.

The draft SDP entities to, within an m- line, indicate sending and receivin=
g capabilities for multiple streams, based on used payload types.

Earlier versions of the draft were submitted to AVTCORE. However, we think =
MMUSIC is more appropriate, due to the new SDP attributes etc.

Regards,

Christer

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<html xmlns:v=3D"urn:schemas-microsoft-com:vml" xmlns:o=3D"urn:schemas-micr=
osoft-com:office:office" xmlns:w=3D"urn:schemas-microsoft-com:office:word" =
xmlns:m=3D"http://schemas.microsoft.com/office/2004/12/omml" xmlns=3D"http:=
//www.w3.org/TR/REC-html40"><head><META HTTP-EQUIV=3D"Content-Type" CONTENT=
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<o:shapedefaults v:ext=3D"edit" spidmax=3D"1026" />
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<o:idmap v:ext=3D"edit" data=3D"1" />
</o:shapelayout></xml><![endif]--></head><body lang=3DEN-US link=3Dblue vli=
nk=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal><span style=3D'c=
olor:#1F497D'>FYI,<o:p></o:p></span></p><p class=3DMsoNormal><span style=3D=
'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span styl=
e=3D'color:#1F497D'>I&#8217;ve submitted the max-ssrc draft to MMUSIC. The =
mechanism may be of interest for RTCWEB.<o:p></o:p></span></p><p class=3DMs=
oNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p class=
=3DMsoNormal><span style=3D'color:#1F497D'>Regards,<o:p></o:p></span></p><p=
 class=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></=
p><p class=3DMsoNormal><span style=3D'color:#1F497D'>Christer<o:p></o:p></s=
pan></p><p class=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p=
></span></p><p class=3DMsoNormal><span style=3D'color:#1F497D'>(Associated =
IPR: https://datatracker.ietf.org/ipr/1639/)<o:p></o:p></span></p><p class=
=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p c=
lass=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p>=
<div><div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt=
 0cm 0cm 0cm'><p class=3DMsoNormal><b><span style=3D'font-size:10.0pt;font-=
family:"Tahoma","sans-serif"'>From:</span></b><span style=3D'font-size:10.0=
pt;font-family:"Tahoma","sans-serif"'> mmusic-bounces@ietf.org [mailto:mmus=
ic-bounces@ietf.org] <b>On Behalf Of </b>Christer Holmberg<br><b>Sent:</b> =
26. syyskuuta 2012 10:23<br><b>To:</b> mmusic@ietf.org<br><b>Subject:</b> [=
MMUSIC] Draft new version/working group: draft-westerlund-mmusic-max-ssrc-0=
0<o:p></o:p></span></p></div></div><p class=3DMsoNormal><o:p>&nbsp;</o:p></=
p><p class=3DMsoNormal><span lang=3DFI>Hi,<o:p></o:p></span></p><p class=3D=
MsoNormal><span lang=3DFI><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal>=
We have submitted a draft, new to MMUSIC: draft-westerlund-mmusic-max-ssrc.=
<o:p></o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNorm=
al>The draft SDP entities to, within an m- line, indicate sending and recei=
ving capabilities for multiple streams, based on used payload types.<o:p></=
o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>Earl=
ier versions of the draft were submitted to AVTCORE. However, we think MMUS=
IC is more appropriate, due to the new SDP attributes etc.<o:p></o:p></p><p=
 class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>Regards,<o:p><=
/o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>Chr=
ister<o:p></o:p></p></div></body></html>=

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From albrecht.schwarz@alcatel-lucent.com  Wed Sep 26 16:20:51 2012
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From: "Schwarz, Albrecht (Albrecht)" <albrecht.schwarz@alcatel-lucent.com>
To: "Romascanu, Dan (Dan)" <dromasca@avaya.com>, Qin Wu <bill.wu@huawei.com>,  "xrblock@ietf.org" <xrblock@ietf.org>
Date: Thu, 27 Sep 2012 01:20:30 +0200
Thread-Topic: [xrblock] FW: I-DAction:draft-alvestrand-rtcweb-stats-registry-00.txt
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] [xrblock] FW:	I-DAction:draft-alvestrand-rtcweb-stats-registry-00.txt
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> I am happy to see my comments being forwarded to rtcweb@ietf.org if
they
> are interested and want to more input from XRBLOCK WG. However I am
not
> sure how much of their work is related to XRBLOCK work? Do they only
> look for some basic metrics obtained from SR/RR. But I agree with you
> consistency between the metrics used by RTCP XR and RTCWEB is a good
> thing.

The work on "rtcweb-stats" is related to XRBLOCK in my opinon.
At least the "basic metrics from SR/RR" with respect to packet loss are con=
troversial and could be misleading, and should be replaced by correspondent=
 XR performance metric types.
The deficiency of some of these basic metrics was one reason to start work =
on extension reports (XR, and former HR).

Albrecht

________________________________________
Von: xrblock-bounces@ietf.org [xrblock-bounces@ietf.org] im Auftrag von Rom=
ascanu, Dan (Dan) [dromasca@avaya.com]
Gesendet: Mittwoch, 26. September 2012 11:57
An: Qin Wu; xrblock@ietf.org
Betreff: Re: [xrblock] FW:      I-DAction:draft-alvestrand-rtcweb-stats-reg=
istry-00.txt

Hi Qin,

If you subscribed to the rtcweb list it would be best that you forward
directly your comments there - does this seem OK?

Dan




> -----Original Message-----
> From: Qin Wu [mailto:bill.wu@huawei.com]
> Sent: Wednesday, September 26, 2012 3:45 AM
> To: Romascanu, Dan (Dan); xrblock@ietf.org
> Subject: Re: [xrblock] FW: I-DAction:draft-alvestrand-rtcweb-stats-
> registry-00.txt
>
> Hi, Dan:
> I am happy to see my comments being forwarded to rtcweb@ietf.org if
they
> are interested and want to more input from XRBLOCK WG. However I am
not
> sure how much of their work is related to XRBLOCK work? Do they only
> look for some basic metrics obtained from SR/RR. But I agree with you
> consistency between the metrics used by RTCP XR and RTCWEB is a good
> thing.
> It will be great to see metrics used by RTCP XR can be used in RTCWEB
as
> well.
>
> BTW: I have subscribed to rtcweb list and am interest to join the
> discussion as I said and expected.
>
> Regards!
> -Qin
> ----- Original Message -----
> From: "Romascanu, Dan (Dan)" <dromasca@avaya.com>
> To: "Qin Wu" <bill.wu@huawei.com>; <xrblock@ietf.org>
> Sent: Tuesday, September 25, 2012 5:45 PM
> Subject: Re: [xrblock] FW: I-DAction:draft-alvestrand-rtcweb-stats-
> registry-00.txt
>
>
> > Hi Qin,
> >
> > Would you like me to forward your comments to rtcweb@ietf.org? You
can
> > also subscribe to the list to follow the responses and participate
in
> > the discussions, but I should warn you that performance statistics
are
> a
> > fraction of the many problems this very active WG have on their
hands.
> >
> > Dan
> >
> >
> >
> >> -----Original Message-----
> >> From: Qin Wu [mailto:bill.wu@huawei.com]
> >> Sent: Tuesday, September 25, 2012 4:17 AM
> >> To: Romascanu, Dan (Dan); xrblock@ietf.org
> >> Subject: Re: [xrblock] FW: I-D
Action:draft-alvestrand-rtcweb-stats-
> >> registry-00.txt
> >>
> >> Interesting -00 draft.
> >> Have a quick look at this draft, I have the folowing comments:
> >> 1. Jitter should also refers to PDV draft since PDV draft provide
> >> another two measurement methods with different measurement
precision
> >> which are different from "inter-arrival jitter" described in
RFC3550.
> >> In the early version of PDV draft, we also accomodate
"inter-arrival
> >> jitter" and support reporting inter-arrival jitter. However we
remove
> >> inter-arrival jitter from this draft based on WG review since
inter-
> >> arrival jitter use different measurement unit and has wider range.
> >> 2.ReceivedPacketCount
> >> Loss RLE report block can be used to report the packet received. It
> is
> >> more staightfoward to calculate ReceivedPacket Count based on  Loss
> > RLE
> >> report block defined in RFC3611 rather than based on MIB
information
> >> defined in RFC2959. Also in some case, WebClient may not support
> SNMP.
> >> So I suggest to refer to section 4.1, RFC3611.
> >> 3. RecevedOctetCount
> >> Same as RecevedPacketCount. I think Loss RLE report block also can
be
> >> used to calculate RecevedOctetCount.
> >> It is better to refer to section 4.1, RFC3611.
> >> 4. Section 4.2 Transport variable
> >> It is not clear how Transport variable is relate to Variables from
> > basic
> >> RTCP SR/RR defined in section 4.1, how transport vaiable is
> > calculated,
> >> e.g., what's the relationship between ReceivedPackets and
> >> ReceivedPacketcount? Are they the same variable?
> >> Is 4 transport variable defined in section 4.2 sufficient to
measure
> > the
> >> performance of media stream? Why not refer to some variable defined
> in
> >> XRBLOCK WG?
> >>
> >> Regards!
> >> -Qin
> >>
> >>  ----- Original Message -----
> >> From: "Romascanu, Dan (Dan)" <dromasca@avaya.com>
> >> To: <xrblock@ietf.org>
> >> Sent: Monday, September 24, 2012 7:48 PM
> >> Subject: [xrblock] FW: I-D Action:draft-alvestrand-rtcweb-stats-
> >> registry-00.txt
> >>
> >>
> >> >I believe that this I-D (individual submission targeting the
RTCWEB
> > WG)
> >> > is of interest for some of the folks who participate in XRBLOCK.
> >> > Reaching consistency between the metrics used by RTCP XR and
RTCWEB
> >> > (when relevant) seems to be desirable.
> >> >
> >> > Regards,
> >> >
> >> > Dan
> >> >
> >> >
> >> >
> >> >
> >> > -----Original Message-----
> >> > From: i-d-announce-bounces@ietf.org
> >> > [mailto:i-d-announce-bounces@ietf.org] On Behalf Of
> >> > internet-drafts@ietf.org
> >> > Sent: Monday, September 24, 2012 12:17 PM
> >> > To: i-d-announce@ietf.org
> >> > Subject: I-D Action:
draft-alvestrand-rtcweb-stats-registry-00.txt
> >> >
> >> >
> >> > A New Internet-Draft is available from the on-line
Internet-Drafts
> >> > directories.
> >> >
> >> >
> >> > Title           : A Registry for WebRTC statistics identifiers
> >> > Author(s)       : Harald T. Alvestrand
> >> > Filename        : draft-alvestrand-rtcweb-stats-registry-00.txt
> >> > Pages           : 8
> >> > Date            : 2012-09-24
> >> >
> >> > Abstract:
> >> >   This memo describes a registration procedure for statistics
> >> >   identifiers used in the WebRTC Javascript API to access
> > statistical
> >> >   information about a PeerConnection.
> >> >
> >> >   It also gives some identifiers that will, when approved, form
the
> >> >   initial content of this registry.
> >> >
> >> >
> >> >
> >> > The IETF datatracker status page for this draft is:
> >> > https://datatracker.ietf.org/doc/draft-alvestrand-rtcweb-stats-
> >> registry
> >> >
> >> > There's also a htmlized version available at:
> >> >
http://tools.ietf.org/html/draft-alvestrand-rtcweb-stats-registry-
> 00
> >> >
> >> >
> >> > Internet-Drafts are also available by anonymous FTP at:
> >> > ftp://ftp.ietf.org/internet-drafts/
> >> >
> >> > _______________________________________________
> >> > 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
> >> > _______________________________________________
> >> > xrblock mailing list
> >> > xrblock@ietf.org
> >> > https://www.ietf.org/mailman/listinfo/xrblock
> > _______________________________________________
> > xrblock mailing list
> > xrblock@ietf.org
> > https://www.ietf.org/mailman/listinfo/xrblock
_______________________________________________
xrblock mailing list
xrblock@ietf.org
https://www.ietf.org/mailman/listinfo/xrblock

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From: Qin Wu <bill.wu@huawei.com>
To: "Schwarz, Albrecht (Albrecht)" <albrecht.schwarz@alcatel-lucent.com>, "Romascanu, Dan (Dan)" <dromasca@avaya.com>, <xrblock@ietf.org>
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From harald@alvestrand.no  Fri Sep 28 04:31:55 2012
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Subject: Re: [rtcweb] [xrblock] FW:	I-DAction:draft-alvestrand-rtcweb-stats-registry-00.txt
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On 09/27/2012 01:20 AM, Schwarz, Albrecht (Albrecht) wrote:
>> I am happy to see my comments being forwarded to rtcweb@ietf.org if
> they
>> are interested and want to more input from XRBLOCK WG. However I am
> not
>> sure how much of their work is related to XRBLOCK work? Do they only
>> look for some basic metrics obtained from SR/RR. But I agree with you
>> consistency between the metrics used by RTCP XR and RTCWEB is a good
>> thing.
> The work on "rtcweb-stats" is related to XRBLOCK in my opinon.
> At least the "basic metrics from SR/RR" with respect to packet loss are controversial and could be misleading, and should be replaced by correspondent XR performance metric types.
> The deficiency of some of these basic metrics was one reason to start work on extension reports (XR, and former HR).
Indeed. The reason I started the document with packet counts and packet 
loss from SR/RR was that I thought these would be uncontroversial and 
unambiguous; I'm seeking guidance from XR people on what other metrics 
are well known enough - and implemented widely enough - that it makes 
sense to include them.

(For reference, the WebRTC codebase seems to decode block type 7 and no 
other block type....)


From albrecht.schwarz@alcatel-lucent.com  Fri Sep 28 16:51:13 2012
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From: "Schwarz, Albrecht (Albrecht)" <albrecht.schwarz@alcatel-lucent.com>
To: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Sat, 29 Sep 2012 01:51:05 +0200
Thread-Topic: [rtcweb] [xrblock] FW: I-DAction:draft-alvestrand-rtcweb-stats-registry-00.txt
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Subject: Re: [rtcweb] [xrblock] FW: I-DAction:draft-alvestrand-rtcweb-stats-registry-00.txt
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Harald,

just to double check:

> (For reference, the WebRTC codebase seems to decode block type 7 and no=20
other block type....)

BT=3D7 would be the "VoIP Metrics Report Block" (RFC 3611).

Whereas BT=3D6 would be the "Statistics Summary Report Block", related also=
 to some improved SR/BR metric.

Regards,
Albrecht

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Harald Alvestrand
Sent: Freitag, 28. September 2012 13:32
To: rtcweb@ietf.org
Subject: Re: [rtcweb] [xrblock] FW: I-DAction:draft-alvestrand-rtcweb-stats=
-registry-00.txt

On 09/27/2012 01:20 AM, Schwarz, Albrecht (Albrecht) wrote:
>> I am happy to see my comments being forwarded to rtcweb@ietf.org if
> they
>> are interested and want to more input from XRBLOCK WG. However I am
> not
>> sure how much of their work is related to XRBLOCK work? Do they only
>> look for some basic metrics obtained from SR/RR. But I agree with you
>> consistency between the metrics used by RTCP XR and RTCWEB is a good
>> thing.
> The work on "rtcweb-stats" is related to XRBLOCK in my opinon.
> At least the "basic metrics from SR/RR" with respect to packet loss are c=
ontroversial and could be misleading, and should be replaced by corresponde=
nt XR performance metric types.
> The deficiency of some of these basic metrics was one reason to start wor=
k on extension reports (XR, and former HR).
Indeed. The reason I started the document with packet counts and packet=20
loss from SR/RR was that I thought these would be uncontroversial and=20
unambiguous; I'm seeking guidance from XR people on what other metrics=20
are well known enough - and implemented widely enough - that it makes=20
sense to include them.

(For reference, the WebRTC codebase seems to decode block type 7 and no=20
other block type....)

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

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References: <EDC652A26FB23C4EB6384A4584434A04081AB2D9@307622ANEX5.global.avaya.com><3C8D9E30D9654AD0A3E1F21668D1CBFB@china.huawei.com><EDC652A26FB23C4EB6384A4584434A04081AB4AD@307622ANEX5.global.avaya.com><20E112E82D30472AA48E653041BC0A78@china.huawei.com>, <EDC652A26FB23C4EB6384A4584434A04081AB5BE@307622ANEX5.global.avaya.com><5F7BCCF5541B7444830A2288ABBEBC96218CFC7599@FRMRSSXCHMBSD2.dc-m.alcatel-lucent.com> <50658AA6.5090200@alvestrand.no>
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From harald@alvestrand.no  Sat Sep 29 02:34:14 2012
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Subject: Re: [rtcweb] [xrblock] FW: I-DAction:draft-alvestrand-rtcweb-stats-registry-00.txt
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On 09/29/2012 01:51 AM, Schwarz, Albrecht (Albrecht) wrote:
> Harald,
>
> just to double check:
>
>> (For reference, the WebRTC codebase seems to decode block type 7 and no
> other block type....)
>
> BT=7 would be the "VoIP Metrics Report Block" (RFC 3611).
>
> Whereas BT=6 would be the "Statistics Summary Report Block", related also to some improved SR/BR metric.
Yep. I looked in the source code and found code that special cased the 
number 7. (This is part of the open source WebRTC release).

I guess this means there has been customer demand for the VoIP Metrics, 
but no customer demand for the Statistics Summary - which led to my 
query about which XR metrics are widely supported in others' equipment.

                     Harald

